Re: [asterisk-users] ztdummy does not load properly at server startup
Darryl Dunkin wrote: It's not playing a wav file at all, it is mixing the live audio from all of the callers in that conference room and sending it back out to them. I understand. What I tried to say is that if a wav file can be played at the correct speed, why would a conference application need a special driver to achieve the same? I assume it is needed as part of the hardware driver and that this application happens to use the timing reference part of it. Theo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HPEC audio clipping
Kevin P. Fleming wrote: The code is done and initially tested; it is being reviewed internally and should be available on Friday or Monday. Will this code be available in 1.2 and 1.4 versions alike ? I can testify it's needed in 1.2. Best regards ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ztdummy does not load properly at server startup
Tzafrir Cohen wrote: On Thu, Apr 19, 2007 at 09:23:48PM +0200, Theo Band wrote: Eric "ManxPower" Wieling wrote: In the zaptel source "make config" will install the zaptel init script in /etc/rc.d/init.d for many distros. Thanks. This was the missing configuration step. A manual start/stop seems to work. I will try to reboot the machine tomorrow for the "final" test but I feel this is going to work. Stopping the zaptel drivers does not work properly, but I only need this for rebooting... You don't. You don't need to unload the zaptel modules before rebooting. make config installs both a start and stop entry for me. /etc/init.d/zaptel stop unload the kernel modules. I don't see the point of trying to do that even if you switch runlevel. The error just looks ugly when rebooting, but it's not really an issue of course. [EMAIL PROTECTED] zaptel-1.4.1]# service zaptel stop Unloading zaptel hardware drivers: wcusb wctdm wcfxo wctdm24xxp wcte11xp wct1xxp wct4xxp tor2. Removing zaptel module: ERROR: Module zaptel is in use by ztdummy [FAILED] The unload_module function in the init.d script in the SVN should fix this and get rid of that error. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Polycom IP 501 is displaying wrong time
Hi,
Thank you for your response. As you said, I set it for -5. But, its displaying
wrong time. I don't enter any SNTP Server. Is it must? How can I solve this
problem? Can you tell me?
Thank you.
Regards,
Chandra.
Steve Totaro [EMAIL PROTECTED] wrote:v\:*
{behavior:url(#default#VML);} o\:* {behavior:url(#default#VML);} w\:*
{behavior:url(#default#VML);} .shape {behavior:url(#default#VML);}
st1\:*{behavior:url(#default#ieooui) } You can use the web interface
and set it to -5 gmt. Google for free NTP servers. I used to use
time.nist.gov and got mixed results. I found another one that works almost all
of the time.
Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
KB3OPB
-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Crazy Boy
Sent: Thursday, April 19, 2007 7:17 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Polycom IP 501 is displaying wrong time
Hi,
This is Chandra. I have Polycom IP 501 phone. Its showing wrong time on the
display screen. How can I set the New York time? What value I have to give
to GMT offset value?
Look forward to your response. Thank you.
Regards,
Chandra.
-
Ahhh...imagining that irresistible new car smell?
Check out new cars at Yahoo! Autos.
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Re: [asterisk-users] Polycom IP 501 is displaying wrong time
Hi Noah, Thank you for your response. As you said, I tried to enter -18000 in GMT offset field. But, its not taking input from the phone dial pad or key board. Its giving chance to select the value from -12 to 12. I dont enter any SNTP Server. Is it must? How can I solve this problem? Can you tell me? Thank you. Regards, Chandra. Noah Miller [EMAIL PROTECTED] wrote: Hi Chandra - This is Chandra. I have Polycom IP 501 phone. Its showing wrong time on the display screen. How can I set the New York time? What value I have to give to GMT offset value? The GMT offset value is in seconds. So, for example, the value to use for EST is -18000, because EST is -5 hours from GMT (-5 x 3600 = -18000). - Noah ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - Ahhh...imagining that irresistible new car smell? Check outnew cars at Yahoo! Autos.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cell phone that can be connected to standad phone switch network
On Wed, 18 Apr 2007, Joseph wrote: Are there any cell phone (gadgets) that can be connected to standard switch phone network? (ability to check email would be a plus). Digium adapter S101i can be connected to any network and it allow a standard phone to act as your local extension over the Internet (by registering to asterisk), it works almost perfectly. So it would be handy to have a cellular phone that can be connected to standard switched phone network, are there any toys like this? Dock'n'Talk. Check the archives. Although let's hope you're not in the UK - I eventually managed to get in-touch with the person DT passed me on to who was their UK disty, but after an initial reply they haven't bothered to email me back advising me of stock, prices or avalability. Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] gxp2000 expansion module blf leds not working
On Thu, 19 Apr 2007, Zoilo Gomez wrote: Am I the only one using the GXP2000 expansion module? I hope not ... I'm looking into using one myself for a client soon - it would be nice to know that someone has had some success with one... Or can anyone suggest something similar - I need a console with about 25-30 buttons/lamps, sourced in the UK ... Gordon Thanks, Zoilo. Zoilo Gomez wrote: Today a 56-button expansion module for the GXP2000 came in. When I program the buttons+leds on the expansion module for BLF, then speed-dial works fine: when I press the button the programmed ext number is called properly. However the LEDs are always off: neither green nor red They are not broken, because on reboot the LEDs flash red! On the GXP2000 itself, this function works fine, with LEDs being green when the ext is free, or red whenever it is busy. Does anybody know this problem? Or can anyone confirm that the LEDs on the GXP2000 expansion module should be working properly? Thanks, Z. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Cell phone that can be connected to standadphone switch network
On Wed, 18 Apr 2007, Joseph wrote: Are there any cell phone (gadgets) that can be connected to standard switch phone network? (ability to check email would be a plus). Digium adapter S101i can be connected to any network and it allow a standard phone to act as your local extension over the Internet (by registering to asterisk), it works almost perfectly. So it would be handy to have a cellular phone that can be connected to standard switched phone network, are there any toys like this? Dock'n'Talk. Check the archives. Although let's hope you're not in the UK - I eventually managed to get in-touch with the person DT passed me on to who was their UK disty, but after an initial reply they haven't bothered to email me back advising me of stock, prices or avalability. Gordon --- Some of the new Nokia mobile phones now support SIP over WiFi (802.11g) as well as GSM. Peter This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RE: OT (a little): IPV6 Ramifications Article
Hans Witvliet wrote: On Wed, 2007-04-18 at 17:11 -0400, Dean Collins wrote: Hi guys, I know it’s a little off topic but……Wondering if you can help. My wife has been asked to find a writer to produce a story on “The dramatic ramifications of IPV6 on commercial businesses and how it will change the product designs for ordinary household/commercial use in a 5-10 year time frame” Ordinary household equipment Fridge (sending snmp traps if a dork leaves the door open ;) radio/tv/vcr (obviously) central heating system airco security One interesting thing is that most of these devices have quite long life-times as do the houses they're installed in. The radio/tv/vcr is changing already - the VCR is dead anyway and the tv/radio is slowly becoming an integrated entertainment system. For the rest, a network connection and an IP-address is only useful if the house is up to it. For those device you've mentioned, the network connection is only any good if it's got somewhere to connect, so a sort of intelligent house is virtually a pre-requisite. In the good old days, everybody got a fixed ip by default, and some euro's extra you got four or eight addrresses. Now you are lucky to get one fixed address. There are still some providers who dish out a fixed address, but they're a rapidly dying species. But if you pay for it, you can have almost anything you want. The only obstacles currently, are the ISP's. afaik, all dsl-modems currently can only work with v4. (correct me if i'm wrong) I think my Cisco 836 does IPV6, but otherwiser I think you're right. But that really means it's the DSL modem manufacturers, not the ISPs that are holding things back still. /Per Jessen, Zürich ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RE: OT (a little): IPV6 Ramifications Article
Remco Post wrote: Hans Witvliet wrote: The only obstacles currently, are the ISP's. Any decent ISP (eg. XS4All.nl) will give you an ipv6 address as well as an ipv4 address. Not around here (Zurich, Switzerland) they won't. I think there is one single provider with IPV6 as an option. And the other ones are perfectly decent providers too. Like I said, when the low-cost DSL routers/modems do not yet support IPV6, why should the provider? /Per Jessen, Zürich ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] app_voicemail.c
Hi everyone, I have the project to personalize the voicemail's IVR. During the intro message (when you call the voicemail of someone), Asterisk pronounce the number of the personal extension number by number (like that : 0.1..2.3.) and I would like it pronounce it by couple of numbers (like that : 01.23.). I have read on forums that I have to modify the app_voicemail.c to do it. Is there another solution to modify the voicemail's IVR without modify the app_voicemail.c ? And I want to know if you have ideas about the way I can do this modificaton ? Thank you. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom IP 501 is displaying wrong time
Crazy Boy wrote on 4/19/07 11:41 PM: Thank you for your response. As you said, I set it for -5. But, its displaying wrong time. I don't enter any SNTP Server. Is it must? How can I solve this problem? Can you tell me? Yeah, there's no way to set the clock except by using an NTP server, so you need to set one. -- Dave Miller http://www.justdave.net/ System Administrator, Mozilla Corporation http://www.mozilla.com/ Project Leader, Bugzilla Bug Tracking System http://www.bugzilla.org/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom IP 501 is displaying wrong time
Hi, this code is for italian time is inside the sip.cfg file. SNTP tcpIpApp.sntp.resyncPeriod=86400 tcpIpApp.sntp.address=192.168.0.8 tcpIpApp.sntp.address.overrideDHCP=0 tcpIpApp.sntp.gmtOffset=3600 tcpIpApp.sntp.gmtOffset.overrideDHCP=0 tcpIpApp.sntp.daylightSavings.enable=1 tcpIpApp.sntp.daylightSavings.fixedDayEnable=0 tcpIpApp.sntp.daylightSavings.start.month=3 tcpIpApp.sntp.daylightSavings.start.date=1 tcpIpApp.sntp.daylightSavings.start.time=2 tcpIpApp.sntp.daylightSavings.start.dayOfWeek=1 tcpIpApp.sntp.daylightSavings.start.dayOfWeek.lastInMonth=1 tcpIpApp.sntp.daylightSavings.stop.month=10 tcpIpApp.sntp.daylightSavings.stop.date=1 tcpIpApp.sntp.daylightSavings.stop.time=2 tcpIpApp.sntp.daylightSavings.stop.dayOfWeek=1 tcpIpApp.sntp.daylightSavings.stop.dayOfWeek.lastInMonth=1/ Bruno. Dave Miller wrote: Crazy Boy wrote on 4/19/07 11:41 PM: Thank you for your response. As you said, I set it for -5. But, its displaying wrong time. I don't enter any SNTP Server. Is it must? How can I solve this problem? Can you tell me? Yeah, there's no way to set the clock except by using an NTP server, so you need to set one. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] gxp2000 expansion module blf leds not working
Or can anyone suggest something similar - I need a console with about 25-30 buttons/lamps, sourced in the UK ... I've just had an Aastra 57i w/ LCD sidecar delivered for testing... initial results look very promising. Might be worth looking into those if you want an alternative to the Grandstream with a load of BLF lights on it. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited For full contact details visit http://www.minotaur.it/chris.html This email is made from 100% recycled electrons ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Friday April 20th Asterisk Users Conference at 12:30PM EDT
Hello again, Mark Spencer will be joining us for questions on at least one of these conferences (we've discussed this and he is definitely onboard with the concept), but his schedule is such that we can't *promise* he'll be there this week. I haven't heard back from him about today, but I'll try to call him right before the conf to remind him. Mark's very busy, so thanks for your patience. There are several ways to be involved. You can listen to the live stream, listen to past recorded conference/podcasts or (BEST) phone in via SIP or PSTN. There is an optional Java client to be able to see who is called in and text chat or post links of interest. We'll try to watch IRC #asterisk for comments as well. To participate in these calls please see http://x2z.eu for the information. You need to have a PIN which is available free from Talkshoe. Their signup is simple and risk-free. The rest is a SIP call away. Join us, please. If you're incredibly paranoid, I'll give away a few anon PINs in #asterisk right before. The incentive of the conference is to bring us all together in an informal and independent platform. Talkshoe.com pays a few pennies for each download and all participants in conferences and other shows. Any money generated from our Talktahon.org conferences goes to helping small entrepreneurs in the third world. See http://www.Kiva.org for more on that. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Failed to authenticate on INVITE
I use realtime. Both information and extensions are stored in DB. It is just a simple setting of the user with dial plan Dial([EMAIL PROTECTED]). exten = 9003,1,Dial([EMAIL PROTECTED]) What I found is the following. 9002 --- S1 --- S2 9002 can make request to S1 and S1 forward the request to S2. 9002 --- S1 --- S2 S2 returns the mentioned error message to S1. (What I guess is 9002 only registers in S1 not in S2, so mentioned error message issued by S2). It is what I got from the above case. Do you have such configuration? I have no idea to solve the problem On 4/20/07, dave cantera [EMAIL PROTECTED] wrote: ango, can you provide some sip.conf and extens.conf info? daveC Rilawich Ango wrote: hi, I have 2 asterisks with the following configuration. asterisk server 1 (S1) has an user 9002 asterisk server 2 (S2) has an user 9003 Both users can make call to each other without problem. Now I add both users to both servers, i.e. asterisk server 1 (S1) has users 9002,9003 asterisk server 2 (S2) has users 9002,9003 When 9002 dials 9003, Dial(SIP/[EMAIL PROTECTED]) or visa versa. Both processes failed to make call with the following error. Apr 16 11:55:41 NOTICE[19658]: chan_sip.c:9802 handle_response_invite: Failed to authenticate on INVITE to '9002 sip:[EMAIL PROTECTED];tag=as2ff0c493' Any solution to let them call each others? ango ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Building Strong Relationships w/ Intelligent Customer Service -- Interlocking Business Solutions, LLC 856-380-0894 x5000 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Improve voice quality on Asterisk + chan_capi+ DIVA BRI
Ok, I've made all those changes, called my operator from an outside line and tried alternatively whispering / shouting into the mic, banging the microphone with a metal object and pressing DTMF digits. So far - so good, it seems to work. I've now got an other problem. Clamping DTMF disabled my IVR! Is there any way to enable/disable DTMF clamping on a per-call basis? Or better, disable DTMF only when the call makes it to an operator? -- Thanks, Cosmin Prund -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Armin Schindler Sent: 19 aprilie 2007 14:35 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Improve voice quality on Asterisk + chan_capi+ DIVA BRI On Thu, 19 Apr 2007, Cosmin Prund wrote: Hello everyone! I've got a Eicon Diva Server BRI card into my * box working just fine, but I wander if there's anything I can do to improve voice quality for my operators. I'm thinking something along the lines of auto gain and sudden noise suppression (like when you hit a fax machine or the other party accidently touches the dial pad on the phone). Does one of Asterisk, chan_capi or the Diva driver have support for such functionality? Sure, with the Dialogic (Eicon) DIVA Server card DSPs, you have the following possibilities: 1. Automatic Gain Control and Active Talker Evaluation in conference (by default automatically activated with three or more parties) 2. Recording Stream Automatic Gain Control 3. Manual Control of Signal Level 4. Manual control of the signal pitch and/or bitrate (rate conversion) 5. Suppression of DTMF tones. This feature can be activated using adapter configuration (for all calls) or on per call basis This is always good to activate this feature for operators to protect people from signals or in one gateway to prevent DTMF tones from passing through gateway in band. The DTMF tones are suppressed in the way which will not affect the quality of the voice signal in case voice signal and DTMF tones overlap. 6. Part 68 Voice Signal Limiter (Required in US, by default deactivated in Europe). This protects the ears from clicks and too loud signals. This feature can be activated using the configuration. This is good idea to activate Part 68 voice signal limiter to protect the people. This is the dynamic voice signal limiter in accordance with Part 68 of US requirements. The Part 68 Limiter, Audio Recording Automatic Gain Control (AGC of received signal) and the DTMF Clamping (Suppression of DTMF tones) are can be controlled using adapter configuration and do not require any change in the application (but can be controlled on the per call basis too, which is not implemented in chan-capi yet). Armin ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Improve voice quality on Asterisk + chan_capi+ DIVA BRI
On Fri, 20 Apr 2007, Cosmin Prund wrote: Ok, I've made all those changes, called my operator from an outside line and tried alternatively whispering / shouting into the mic, banging the microphone with a metal object and pressing DTMF digits. So far - so good, it seems to work. I've now got an other problem. Clamping DTMF disabled my IVR! Is there any way to enable/disable DTMF clamping on a per-call basis? Or better, disable DTMF only when the call makes it to an operator? This is possible, but such a command/feature must be implemented into chan-capi first. Anyway, even with DTMF clamping the DTMF detection is activated. So Asterisk should get the DTMF infos. Or is your IVR doing own DTMF detection on voice data? If yes, you should change that. Armin -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Armin Schindler Sent: 19 aprilie 2007 14:35 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Improve voice quality on Asterisk + chan_capi+ DIVA BRI On Thu, 19 Apr 2007, Cosmin Prund wrote: Hello everyone! I've got a Eicon Diva Server BRI card into my * box working just fine, but I wander if there's anything I can do to improve voice quality for my operators. I'm thinking something along the lines of auto gain and sudden noise suppression (like when you hit a fax machine or the other party accidently touches the dial pad on the phone). Does one of Asterisk, chan_capi or the Diva driver have support for such functionality? Sure, with the Dialogic (Eicon) DIVA Server card DSPs, you have the following possibilities: 1. Automatic Gain Control and Active Talker Evaluation in conference (by default automatically activated with three or more parties) 2. Recording Stream Automatic Gain Control 3. Manual Control of Signal Level 4. Manual control of the signal pitch and/or bitrate (rate conversion) 5. Suppression of DTMF tones. This feature can be activated using adapter configuration (for all calls) or on per call basis This is always good to activate this feature for operators to protect people from signals or in one gateway to prevent DTMF tones from passing through gateway in band. The DTMF tones are suppressed in the way which will not affect the quality of the voice signal in case voice signal and DTMF tones overlap. 6. Part 68 Voice Signal Limiter (Required in US, by default deactivated in Europe). This protects the ears from clicks and too loud signals. This feature can be activated using the configuration. This is good idea to activate Part 68 voice signal limiter to protect the people. This is the dynamic voice signal limiter in accordance with Part 68 of US requirements. The Part 68 Limiter, Audio Recording Automatic Gain Control (AGC of received signal) and the DTMF Clamping (Suppression of DTMF tones) are can be controlled using adapter configuration and do not require any change in the application (but can be controlled on the per call basis too, which is not implemented in chan-capi yet). Armin ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Improve voice quality on Asterisk + chan_capi+DIVA BRI
I've implemented my IVR using an FastAGI thing, using the READ application. core show application read shows no information on how the read function gets it's digits, I assume it does it the right way. With DTMF clamping off it works, with DTMF clamping on it no longer works. I've also toggled the softftfm setting in capi.conf, no luck ether way. Is there anything else I can try? Did I miss the obvious (it would not be my first) -- Thanks, Cosmin Prund -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Armin Schindler Sent: 20 aprilie 2007 12:32 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Improve voice quality on Asterisk + chan_capi+DIVA BRI On Fri, 20 Apr 2007, Cosmin Prund wrote: Ok, I've made all those changes, called my operator from an outside line and tried alternatively whispering / shouting into the mic, banging the microphone with a metal object and pressing DTMF digits. So far - so good, it seems to work. I've now got an other problem. Clamping DTMF disabled my IVR! Is there any way to enable/disable DTMF clamping on a per-call basis? Or better, disable DTMF only when the call makes it to an operator? This is possible, but such a command/feature must be implemented into chan-capi first. Anyway, even with DTMF clamping the DTMF detection is activated. So Asterisk should get the DTMF infos. Or is your IVR doing own DTMF detection on voice data? If yes, you should change that. Armin -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk- users- [EMAIL PROTECTED] On Behalf Of Armin Schindler Sent: 19 aprilie 2007 14:35 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Improve voice quality on Asterisk + chan_capi+ DIVA BRI On Thu, 19 Apr 2007, Cosmin Prund wrote: Hello everyone! I've got a Eicon Diva Server BRI card into my * box working just fine, but I wander if there's anything I can do to improve voice quality for my operators. I'm thinking something along the lines of auto gain and sudden noise suppression (like when you hit a fax machine or the other party accidently touches the dial pad on the phone). Does one of Asterisk, chan_capi or the Diva driver have support for such functionality? Sure, with the Dialogic (Eicon) DIVA Server card DSPs, you have the following possibilities: 1. Automatic Gain Control and Active Talker Evaluation in conference (by default automatically activated with three or more parties) 2. Recording Stream Automatic Gain Control 3. Manual Control of Signal Level 4. Manual control of the signal pitch and/or bitrate (rate conversion) 5. Suppression of DTMF tones. This feature can be activated using adapter configuration (for all calls) or on per call basis This is always good to activate this feature for operators to protect people from signals or in one gateway to prevent DTMF tones from passing through gateway in band. The DTMF tones are suppressed in the way which will not affect the quality of the voice signal in case voice signal and DTMF tones overlap. 6. Part 68 Voice Signal Limiter (Required in US, by default deactivated in Europe). This protects the ears from clicks and too loud signals. This feature can be activated using the configuration. This is good idea to activate Part 68 voice signal limiter to protect the people. This is the dynamic voice signal limiter in accordance with Part 68 of US requirements. The Part 68 Limiter, Audio Recording Automatic Gain Control (AGC of received signal) and the DTMF Clamping (Suppression of DTMF tones) are can be controlled using adapter configuration and do not require any change in the application (but can be controlled on the per call basis too, which is not implemented in chan-capi yet). Armin ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options
[asterisk-users] RAD IPmux8
Hi, I'm looking for somebody who has managed to get their IPmux8 or IPmux11 talking to an Asterisk machine. I have it setup properly I think (the two IPmux's are talking to each other, and the zttool says that the PRI is acting okay, but I'm flooded with HDLC aborts and FCS problems. Edwin -- Edwin Groothuis |Personal website: http://www.mavetju.org [EMAIL PROTECTED]| Weblog: http://www.mavetju.org/weblog/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] gxp2000 expansion module blf leds not working
On 08:32, Fri 20 Apr 07, Gordon Henderson wrote: On Thu, 19 Apr 2007, Zoilo Gomez wrote: Am I the only one using the GXP2000 expansion module? I hope not ... I'm looking into using one myself for a client soon - it would be nice to know that someone has had some success with one... Or can anyone suggest something similar - I need a console with about 25-30 buttons/lamps, sourced in the UK ... You can have a look at the Snom phones. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recording and Conferencing
Hi Dovid, Thanks for ur reply. Any problem with using Asterisk 1.2.13? Thanks, rani On 4/20/07, Dovid B [EMAIL PROTECTED] wrote: 1) Why arent you using 1.2.17 ? 2) So you have to use an AGI ? You can use the mix monitor command. - Original Message - *From:* selva rani [EMAIL PROTECTED] *To:* asterisk-users@lists.digium.com *Sent:* Friday, April 13, 2007 8:03 AM *Subject:* [asterisk-users] Recording and Conferencing Hi, I am newbie to asterisk. I would like to use asterisk using VoIP. I don't want to use any hardware. I have installed Asterisk 1.2.13. I would like to record using AGI command RECORD FILE. I would also like to do conferencing and recording in asterisk. How could this be done? Help me out with conf files and also using AGI script. -- ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Passive E1 Pri Tap for Voice Recording
Dear All, Is it possible to install * in front of a Avaya IP 406 system via a T connector E1 tap so it's external to the Avaya system? We would like to record upto 60 channels (2 * ISDN30e). This may increase later. Also, could the calls go into the cdr for retrieval/browsing later? What hardware/server would you recommend? Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Improve voice quality on Asterisk +chan_capi+DIVA BRI
On Fri, 20 Apr 2007, Cosmin Prund wrote: This message includes two snips of CLI output, with DTFM CLAMPING ON (first) and with DTFM CLAMPING OFF (second). You can search for ** to skip to the second CLI Output. In the first case I've enterd 6 DTFM digits (123456), you can see them in a CLI msg at some point. In the second test I enterd way more digits (123456789?) but my IVR didn't react to any of them. Thanks a lot for your time. Both logs don't show any DTMF activity. DMTF detection is not activated at all. Please make sure you DON'T have softdmtf=yes or relaxdtmf=yes in your capi.conf. Armin CLI Output == ISDN1#02: Answering for 206364 CONNECT_RESP ID=001 #0x494d LEN=0040 Controller/PLCI/NCCI= 0x401 Reject = 0x0 BProtocol B1protocol = 0x1 B2protocol = 0x1 B3protocol = 0x0 B1configuration= default B2configuration= default B3configuration= default ConnectedNumber = 00 80206364 ConnectedSubaddress = default LLC = default AdditionalInfo BChannelinformation= default Keypadfacility = default Useruserdata = default Facilitydataarray = default CAPI devicestate requested for ISDN1/206364 -- CAPI/ISDN1/206364-28 Playing '/ram_sounds/intro-activare' (language 'de') CONNECT_ACTIVE_IND ID=001 #0x4954 LEN=0015 Controller/PLCI/NCCI= 0x401 ConnectedNumber = default ConnectedSubaddress = default LLC = default CONNECT_ACTIVE_RESP ID=001 #0x4954 LEN=0012 Controller/PLCI/NCCI= 0x401 CONNECT_B3_IND ID=001 #0x4955 LEN=0013 Controller/PLCI/NCCI= 0x2b0401 NCPI= default CONNECT_B3_RESP ID=001 #0x4955 LEN=0015 Controller/PLCI/NCCI= 0x2b0401 Reject = 0x0 NCPI= default CONNECT_B3_ACTIVE_IND ID=001 #0x4956 LEN=0013 Controller/PLCI/NCCI= 0x2b0401 NCPI= default CONNECT_B3_ACTIVE_RESP ID=001 #0x4956 LEN=0012 Controller/PLCI/NCCI= 0x2b0401 == ISDN1#02: Setting up echo canceller (PLCI=0x401, function=1, options=4, tail=0) FACILITY_REQ ID=001 #0x355c LEN=0024 Controller/PLCI/NCCI= 0x401 FacilitySelector= 0x8 FacilityRequestParameter= 01 00 06 04 00 00 00 00 00 FACILITY_CONF ID=001 #0x355c LEN=0022 Controller/PLCI/NCCI= 0x401 Info= 0x0 FacilitySelector= 0x8 FacilityConfirmationParameter = 01 00 02 00 00 -- ISDN1#02: Echo canceller successfully set up (PLCI=0x401) -- User entered '123456' ) (sample_offset 0)m_sounds/codul-client-nu-este-valid' (escape_digits= INFO_IND ID=001 #0x4a87 LEN=0017 Controller/PLCI/NCCI= 0x401 InfoNumber = 0x1e InfoElement = 82 88 INFO_RESP ID=001 #0x4a87 LEN=0012 Controller/PLCI/NCCI= 0x401 -- ISDN1#02: info element PI 82 88 ISDN1#02: In-band information available INFO_IND ID=001 #0x4a88 LEN=0017 Controller/PLCI/NCCI= 0x401 InfoNumber = 0x1e InfoElement = 82 83 INFO_RESP ID=001 #0x4a88 LEN=0012 Controller/PLCI/NCCI= 0x401 -- ISDN1#02: info element PI 82 83 ISDN1#02: Origination is non ISDN INFO_IND ID=001 #0x4a89 LEN=0017 Controller/PLCI/NCCI= 0x401 InfoNumber = 0x8 InfoElement = 80 90 INFO_RESP ID=001 #0x4a89 LEN=0012 Controller/PLCI/NCCI= 0x401 -- ISDN1#02: info element CAUSE 80 90 INFO_IND ID=001 #0x4a8a LEN=0015 Controller/PLCI/NCCI= 0x401 InfoNumber = 0x8045 InfoElement = default INFO_RESP ID=001 #0x4a8a LEN=0012 Controller/PLCI/NCCI= 0x401 -- ISDN1#02: info element DISCONNECT -- ISDN1#02: Disconnect case 3 -- CAPI queue frame: TYPE: Control (4) SUBCLASS: Hangup (1) ] [ISDN1#02] /CLI Output CLI Output == ISDN1#02: Answering for 206364 CONNECT_RESP ID=001 #0x4a91 LEN=0040 Controller/PLCI/NCCI= 0x301 Reject = 0x0 BProtocol B1protocol = 0x1 B2protocol = 0x1 B3protocol = 0x0 B1configuration= default B2configuration= default B3configuration= default ConnectedNumber = 00 80206364 ConnectedSubaddress = default
Re: [asterisk-users] pci 2.2 - pci-e x16
On 4/20/07, Olivier [EMAIL PROTECTED] wrote: 2007/4/20, asterisk [EMAIL PROTECTED]: Hi, Does anyone know if it is possible to plug a tdm400p pci digium card into an pci-e 16x slot ? np Olivier meant no here as well. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk stops responding to SIP/ZAP
About once a week or so my Asterisk box stops responding to all phones. I can pull up the console, do whatever I want at the CLI but the only way to get things working again is to restart Asterisk altogether. I finally cranked verbose debugging way up (and watched my log files go from 1mb/day to 100mb/day), but below I believe contains my problem. The next line is 1.5 minutes later where I restart Asterisk. SIP/701 is a Grandstream GXP-2000 phone (we have about 30 of them in place here). Zap/3-1 is a Digium TDM400. I can't quite figure out where my problem is, is it the initial exception, is it not getting hung up completely, does it have to do with the call limit on the SIP channel, perhaps 'no provider found' statements? Any help would be appreciated, I have a relatively simple dial-plan, I can send over relevant bits of it if necessary. Thanks, Ken [Apr 19 13:51:13] DEBUG[27722] chan_zap.c: Exception on 12, channel 3 [Apr 19 13:51:13] DEBUG[27722] chan_zap.c: Got event On hook(1) on channel 3 (index 0) [Apr 19 13:51:13] DEBUG[27722] chan_zap.c: disabled echo cancellation on channel 3 [Apr 19 13:51:13] DEBUG[27722] channel.c: Didn't get a frame from channel: Zap/3-1 [Apr 19 13:51:13] DEBUG[27722] channel.c: Bridge stops bridging channels SIP/701-08ee6120 and Zap/3-1 [Apr 19 13:51:13] DEBUG[27722] channel.c: Hanging up channel 'Zap/3-1' [Apr 19 13:51:13] DEBUG[27722] chan_zap.c: zt_hangup(Zap/3-1) [Apr 19 13:51:13] DEBUG[27722] chan_zap.c: Hangup: channel: 3 index = 0, normal = 12, callwait = -1, thirdcall = -1 [Apr 19 13:51:13] DEBUG[27722] chan_zap.c: disabled echo cancellation on channel 3 [Apr 19 13:51:13] DEBUG[27722] chan_zap.c: Set option TDD MODE, value: OFF(0) on Zap/3-1 [Apr 19 13:51:13] DEBUG[27722] chan_zap.c: Updated conferencing on 3, with 0 conference users [Apr 19 13:51:13] VERBOSE[27722] logger.c: -- Hungup 'Zap/3-1' [Apr 19 13:51:13] DEBUG[27722] devicestate.c: Notification of state change to be queued on device/channel Zap/3-1 [Apr 19 13:51:13] DEBUG[27722] pbx.c: Spawn extension (from-internal,201,2) exited non-zero on 'SIP/701-08ee6120' [Apr 19 13:51:13] VERBOSE[27722] logger.c: == Spawn extension (from-internal, 201, 2) exited non-zero on 'SIP/701-08ee6120' [Apr 19 13:51:13] DEBUG[27722] pbx.c: Launching 'Hangup' [Apr 19 13:51:13] VERBOSE[27722] logger.c: -- Executing [EMAIL PROTECTED]:1] Hangup(SIP/701-08ee6120, ) in new stack [Apr 19 13:51:13] DEBUG[27722] pbx.c: Spawn extension (from-internal,h,1) exited non-zero on 'SIP/701-08ee6120' [Apr 19 13:51:13] VERBOSE[27722] logger.c: == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/701-08ee6120' [Apr 19 13:51:13] DEBUG[27722] channel.c: Hanging up channel 'SIP/701-08ee6120' [Apr 19 13:51:13] DEBUG[27722] chan_sip.c: Hangup call SIP/701-08ee6120, SIP callid [EMAIL PROTECTED]) [Apr 19 13:51:13] DEBUG[27722] chan_sip.c: Updating call counter for incoming call [Apr 19 13:51:13] DEBUG[27722] chan_sip.c: Call from peer '701' removed from call limit 6 [Apr 19 13:51:13] DEBUG[27722] devicestate.c: Notification of state change to be queued on device/channel SIP/701 [Apr 19 13:51:13] DEBUG[27722] devicestate.c: Notification of state change to be queued on device/channel SIP/701-08ee6120 [Apr 19 13:51:13] DEBUG[20432] devicestate.c: No provider found, checking channel drivers for Zap - 3 [Apr 19 13:51:13] DEBUG[20432] devicestate.c: Changing state for Zap/3 - state 0 (Unknown) [Apr 19 13:51:13] DEBUG[20432] devicestate.c: No provider found, checking channel drivers for SIP - 701 [Apr 19 13:51:13] DEBUG[20432] chan_sip.c: Checking device state for peer 701 [Apr 19 13:51:13] DEBUG[20432] devicestate.c: Changing state for SIP/701 - state 1 (Not in use) [Apr 19 13:51:13] DEBUG[20432] devicestate.c: No provider found, checking channel drivers for SIP - 701 [Apr 19 13:51:13] DEBUG[20432] chan_sip.c: Checking device state for peer 701 [Apr 19 13:51:13] DEBUG[20432] devicestate.c: No provider found, checking channel drivers for SIP - 701 [Apr 19 13:51:13] DEBUG[20432] chan_sip.c: Checking device state for peer 701 [Apr 19 13:51:13] DEBUG[20432] devicestate.c: Changing state for SIP/701 - state 1 (Not in use) [Apr 19 13:51:13] DEBUG[20432] devicestate.c: No provider found, checking channel drivers for SIP - 701 [Apr 19 13:51:13] DEBUG[20432] chan_sip.c: Checking device state for peer 701 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CallerID Auth
Hi, in my dial plan I've configured two trunks to make outbound calls (one for national calls and other international). I want to allow only 2-3 extension to make use of my international trunk to make outbound calls so I want some kind of auth. based on their callerid . Please guide. thanks arun ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] BSNL caller ID (India)
Yes, As I have mentioned below I tried the link http://bugs.digium.com/view.php?id=6683nbn=24 but was not able to make it work. Regards, Sanjay Rajdev - Original Message - From: Steve Murphy [EMAIL PROTECTED] To: sanjay rajdev [EMAIL PROTECTED] Sent: Thursday, April 19, 2007 4:37:39 AM (GMT+0530) Asia/Calcutta Subject: Re: [asterisk-users] BSNL caller ID (India) Sanjay-- Did you look at bug 6683? Sorry, I haven't reviewed all the messages on this thread. murf On Wed, 2007-04-18 at 01:01 +0530, Sanjay Rajdev wrote: Tzafrir, Can you Please let me know if the zapata.conf below is correct, or do I have to change something. Regards, Sanjay Rajdev - Original Message - From: Sanjay Rajdev [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Cc: tzafrir cohen [EMAIL PROTECTED] Sent: Tuesday, April 17, 2007 3:38:16 AM (GMT+0530) Asia/Calcutta Subject: Re: [asterisk-users] BSNL caller ID (India) Yes below is the zapata.conf [trunkgroups] [channels] context=incoming usecallerid=yes cidsignalling=dtmf cidstart=ring hidecallerid=no callerid=asreceived callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no ;Sangoma A200 [slot:2 bus:4 span:1] group=0 signalling = fxs_ks channel = 1 group=0 signalling = fxs_ks channel = 2 group=0 signalling = fxs_ks channel = 3 group=0 signalling = fxs_ks channel = 4 Regards, Sanjay Rajdev - Original Message - From: Tzafrir Cohen [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, April 17, 2007 2:40:32 AM (GMT+0530) Asia/Calcutta Subject: Re: [asterisk-users] BSNL caller ID (India) On Mon, Apr 16, 2007 at 11:18:46PM +0530, Sanjay Rajdev wrote: Has anyone figured out the way of getting the caller id for BSNL on Asterisk 1.4.2 I have tried following link http://bugs.digium.com/view.php?id=6683nbn=24 but was not able to get it, although did not ge any error too. I always get the caller id as asterisk. Hmmm... are you sure you have configured your system to get callerid from the PSTN? callerid=asrecieved in zapata.conf. -- Steve Murphy [EMAIL PROTECTED] Digium ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OT] OMG Verizon is terrible
Remember the big lie with Verizon is not The tech will be there at noon. It is Your FOC (Firm Order Commit) date is xxx The dates they give are neither Firm or Committed. Just ask them. As long as you remember, they are the Phone Company and you are just the customer. no body will be disapointed. Do I sound a lil grumppy. 5 and half week for fiber buildout quote. Yup, Im grumppy. alabun Lee Jenkins wrote: Steve Totaro wrote: They are all terrible in their own way. Don't you have someone you can delegate the Verizon babysitting responsibility to? I would consider sales calls a little more important than being a babysitter. Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Lee Jenkins Sent: Wednesday, April 18, 2007 6:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] [OT] OMG Verizon is terrible Had an appointment for these schmoes to come out and install another line. Was supposed to be 8-12. Its now 6PM and not even call. Missed 3 sales calls waiting on these jerks. No wonder customers were jumping ship to Vonage. -- Not on that day, I was the only one available unfortunately. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Why duoble digits must be so fast to activate features?
Hi everybody, I'm testing Asterisk 1.2.14(you can say: why such an old version? I say: I'm using Trixbox..). I must be as fast as a flash to press *2 and do an attended transfer. If I wait only a tenth of a second nothing happens. I think it is an issue. I have seen the source code and found nothing bad. Is this a known issue? Many thanks Best regards Mauro _ Don't just search. Find. Check out the new MSN Search! http://search.msn.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Passive E1 Pri Tap for Voice Recording
On 20/04/07, John Treble [EMAIL PROTECTED] wrote: Gavin, Call Endance and ask them about their Lawful Call Intercept solution(s) using their DAG TDM E1 cards on Linux (Endance.com). Thanks, will have a look. Cheers. John Treble Ottawa, Ontario, Canada -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Gavin Henry Sent: April 20, 2007 9:07 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Passive E1 Pri Tap for Voice Recording Dear All, Is it possible to install * in front of a Avaya IP 406 system via a T connector E1 tap so it's external to the Avaya system? We would like to record upto 60 channels (2 * ISDN30e). This may increase later. Also, could the calls go into the cdr for retrieval/browsing later? What hardware/server would you recommend? Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom not picking up phone transferred phone call.
Hi all, I'm having a problem with a polycom 301 not picking up a ZAP call. Below is the CLI output of the call. I have: TDM400 with 2 FXO lines Asterisk 1.2.14 Polycom 301 When I dial the first ZAP line, I choose an extension that rings the polycom, polycom rings and I can pick it up and the call is bridged. When I call my second zap line, the polycom rings, but I cannot pickup the call either by hitting the Answer button or by picking up the handset. The only difference that I can see is that the second line is not sending CID information and I get the 2 NOTICE lines outputted in the CLI output below. Thanks for any help, Lee -- Starting simple switch on 'Zap/2-1' Apr 20 10:22:03 NOTICE[5258]: chan_zap.c:6072 ss_thread: Got event 18 (Ring Begin)... Apr 20 10:22:05 NOTICE[5258]: chan_zap.c:6072 ss_thread: Got event 2 (Ring/Answered)... -- Executing Answer(Zap/2-1, ) in new stack -- Executing Ringing(Zap/2-1, ) in new stack -- Executing SetMusicOnHold(Zap/2-1, default) in new stack -- Executing Wait(Zap/2-1, 1) in new stack -- Executing Goto(Zap/2-1, check_time|s|1) in new stack -- Goto (check_time,s,1) -- Executing Answer(Zap/2-1, ) in new stack -- Executing Set(Zap/2-1, LAST_MENU_REACHED=check_time) in new stack -- Executing Set(Zap/2-1, FAIL_MENU=error_invalid|TIMEOUT_MENU=error_timeout) in new stack -- Executing GotoIfTime(Zap/2-1, 09:00-17:00|mon-fri|*|*|?main_menu|s|1) in new stack -- Goto (main_menu,s,1) -- Executing Answer(Zap/2-1, ) in new stack -- Executing Set(Zap/2-1, LAST_MENU_REACHED=main_menu) in new stack -- Executing Macro(Zap/2-1, ProcessCaller) in new stack -- Executing Set(Zap/2-1, DTCONN=callers) in new stack -- Executing AGI(Zap/2-1, dtfb|GET|CallerLevel|SELECT caller_level FROM callers WHERE caller_phone = '') in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/dtfb -- AGI Script dtfb completed, returning 0 -- Executing AGI(Zap/2-1, dtfb|GET|CallerName|SELECT caller_name FROM callers WHERE caller_phone = '') in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/dtfb -- AGI Script dtfb completed, returning 0 -- Executing NoOp(Zap/2-1, Caller Level is ) in new stack -- Executing NoOp(Zap/2-1, Caller Name is ) in new stack -- Executing GotoIf(Zap/2-1, 0?lee_voice_followme|s|1) in new stack -- Executing BackGround(Zap/2-1, custom/attendant) in new stack -- Playing 'custom/attendant' (language 'en') == CDR updated on Zap/2-1 -- Executing Dial(Zap/2-1, SIP/111|29|tm) in new stack -- Called 111 -- Started music on hold, class 'default', on channel 'Zap/2-1' -- SIP/111-083b2bf8 is ringing -- Stopped music on hold on Zap/2-1 == Spawn extension (main_menu, 111, 1) exited non-zero on 'Zap/2-1' -- Hungup 'Zap/2-1' -- Got SIP response 400 Bad Request back from 192.168.1.105 -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Passive E1 Pri Tap for Voice Recording
Gavin, Call Endance and ask them about their Lawful Call Intercept solution(s) using their DAG TDM E1 cards on Linux (Endance.com). Cheers. John Treble Ottawa, Ontario, Canada -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Gavin Henry Sent: April 20, 2007 9:07 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Passive E1 Pri Tap for Voice Recording Dear All, Is it possible to install * in front of a Avaya IP 406 system via a T connector E1 tap so it's external to the Avaya system? We would like to record upto 60 channels (2 * ISDN30e). This may increase later. Also, could the calls go into the cdr for retrieval/browsing later? What hardware/server would you recommend? Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OT] OMG Verizon is terrible
Steve Totaro wrote: They are all terrible in their own way. Don't you have someone you can delegate the Verizon babysitting responsibility to? I would consider sales calls a little more important than being a babysitter. Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Lee Jenkins Sent: Wednesday, April 18, 2007 6:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] [OT] OMG Verizon is terrible Had an appointment for these schmoes to come out and install another line. Was supposed to be 8-12. Its now 6PM and not even call. Missed 3 sales calls waiting on these jerks. No wonder customers were jumping ship to Vonage. -- Not on that day, I was the only one available unfortunately. -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why duoble digits must be so fast to activate features?
On Fri, 20 Apr 2007, Mauro Zanin wrote: Hi everybody, I'm testing Asterisk 1.2.14(you can say: why such an old version? I say: I'm using Trixbox..). I must be as fast as a flash to press *2 and do an attended transfer. If I wait only a tenth of a second nothing happens. I think it is an issue. I have seen the source code and found nothing bad. Is this a known issue? Change it in features.conf. Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] iaxComm problems
Hi Folks. I have installed two sip phones and two PCs in a network. The later with iaxComm. Calls are made between the sip phones and between a sip phone and a PC. When calling from one PC to the other the iaxComm shows ??? in the status column and the call can't be answered. The same goes if anyone calls from the outside and tries to reach a PC. Any ideas? Regards. -- IEE José Hugo Pérez Casanova Profesor Investigador Departamento de Ingeniería Electrónica Instituto Tecnológico de Veracruz M.A. de Quevedo #2779, colonia Formando Hogar Veracruz, Ver. Mexico Tel/Fax: (52) 229-938-8104 http://electronica.itver.edu.mx ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] adding second TDM400P card causes echo cancellation to fail for all Zap channels
Hi, folks: Yesterday I added a second TDM400P card to a working, echo-free server running HPEC. Today, I'm getting these messages: Apr 20 09:12:12 WARNING[5679]: chan_zap.c:1551 zt_enable_ec: Unable to enable echo cancellation on channel 3 along with complaints of severe echo. The channels have all been tuned using 'fxotune -s' and the echo numbers are all under 2%. I guess that Asterisk can't do echo cancellation on these channels. Why would I be getting this? Thanks, -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Transfer via CTI
Any ideas on this? Closest thing that comes to mind is FOP : http://www.asternic.org/ hth ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OT] OMG Verizon is terrible
Noah Miller wrote: Had an appointment for these schmoes to come out and install another line. Was supposed to be 8-12. Its now 6PM and not even call. Missed 3 sales calls waiting on these jerks. No wonder customers were jumping ship to Vonage. I once had to oversee Verizon install a PRI line in Manhattan. I live 2.5 hours away, but we made the appointment, and I was there, but the Verizon tech never showed. I made another appointment, and it happened again, and again, and again. I don't even remember how many times it finally took, but it was ridiculous. The techs were even lying and saying they came and there was no one there to let them in. They seem to have gotten better in recent years, but they own all the physical lines, and they know it. - Noah On 4/19/07, Steve Totaro [EMAIL PROTECTED] wrote: They are all terrible in their own way. Don't you have someone you can delegate the Verizon babysitting responsibility to? I would consider sales calls a little more important than being a babysitter. Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Lee Jenkins Sent: Wednesday, April 18, 2007 6:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] [OT] OMG Verizon is terrible Had an appointment for these schmoes to come out and install another line. Was supposed to be 8-12. Its now 6PM and not even call. Missed 3 sales calls waiting on these jerks. No wonder customers were jumping ship to Vonage. -- Warm Regards, Lee Thats funny. My last turn up with Verizon was a Multilink with three T1s. Three DIFFERENT techs showed up at about the same time, each with a different work order. Thanks, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OT] OMG Verizon is terrible
I make a habit of just buying hamburgers and stopping by the CO or hut where I see the vans. I tell them that I am buying favors with food and they like it.. Its a lot of work but it helps On 4/20/07, Steve Totaro [EMAIL PROTECTED] wrote: Noah Miller wrote: Had an appointment for these schmoes to come out and install another line. Was supposed to be 8-12. Its now 6PM and not even call. Missed 3 sales calls waiting on these jerks. No wonder customers were jumping ship to Vonage. I once had to oversee Verizon install a PRI line in Manhattan. I live 2.5 hours away, but we made the appointment, and I was there, but the Verizon tech never showed. I made another appointment, and it happened again, and again, and again. I don't even remember how many times it finally took, but it was ridiculous. The techs were even lying and saying they came and there was no one there to let them in. They seem to have gotten better in recent years, but they own all the physical lines, and they know it. - Noah On 4/19/07, Steve Totaro [EMAIL PROTECTED] wrote: They are all terrible in their own way. Don't you have someone you can delegate the Verizon babysitting responsibility to? I would consider sales calls a little more important than being a babysitter. Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Lee Jenkins Sent: Wednesday, April 18, 2007 6:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] [OT] OMG Verizon is terrible Had an appointment for these schmoes to come out and install another line. Was supposed to be 8-12. Its now 6PM and not even call. Missed 3 sales calls waiting on these jerks. No wonder customers were jumping ship to Vonage. -- Warm Regards, Lee Thats funny. My last turn up with Verizon was a Multilink with three T1s. Three DIFFERENT techs showed up at about the same time, each with a different work order. Thanks, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! Hind sight is most always 20/20 or better. --- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unicall/R2 for Asterisk 1.4 Available for TESTING
Thanks a lot for the fix Humberto.
On 4/18/07, Humberto Figuera [EMAIL PROTECTED] wrote:
Hi Moises,
the Asterisk SVN-branch-1.4-r60989 make a change in the
ast_channel_alloc function:
This is a big improvement over the current CDR fixes. It may still
need refinement, but this won't have as many folks bothered.
here the patch for chan_unicall.c ;p
--- chan_unicall.c.orig 2007-04-18 03:32:17.0 -0400
+++ chan_unicall.c 2007-04-18 03:32:26.0 -0400
@@ -2485,7 +2485,7 @@
}
while (x 3);
-if ( ( tmp = ast_channel_alloc(0, state, 0, 0, chan_name) ) == NULL)
+if ( ( tmp = ast_channel_alloc(0, state, 0, 0, i-accountcode,
i-exten, i-context, i-amaflags, chan_name) ) == NULL)
{
ast_log(LOG_WARNING, Unable to allocate channel structure\n);
return NULL;
--
Humberto Figuera - Using Linux 2.6.20
Usuario GNU/Linux 369709
Caracas - Venezuela
GPG Key Fingerprint = 5AAC DF0C 00F4 2834 28BA 37AD 3364 01D1 74CA 0603
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--
Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org;
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Re: [asterisk-users] pci 2.2 - pci-e x16
2007/4/20, asterisk [EMAIL PROTECTED]: Hi, Does anyone know if it is possible to plug a tdm400p pci digium card into an pci-e 16x slot ? np Is there a possibility to work? no I have a sun fire x2100 which doesn't have pci slots. Does Digium make pci-e cards? not yet ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] pci 2.2 - pci-e x16
Hi, Does anyone know if it is possible to plug a tdm400p pci digium card into an pci-e 16x slot ? Is there a possibility to work? I have a sun fire x2100 which doesn't have pci slots. Does Digium make pci-e cards? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Pika boards - anyone are using it?
Hi, I`m looking for boards to use with Asterisk and as I already was used Pika boards few years ago (in a Windows IVR application), I found that they have new options to Asterisk. It will be nice if I can see some opinions from here before go ahead on it. Thanks in advance! Peter (*) I`d put this message in biz list but I realize that it seems to be more related with user list then I resent it here. Sorry about inconvenience. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recording and Conferencing
1) Why arent you using 1.2.17 ? 2) So you have to use an AGI ? You can use the mix monitor command. - Original Message - From: selva rani To: asterisk-users@lists.digium.com Sent: Friday, April 13, 2007 8:03 AM Subject: [asterisk-users] Recording and Conferencing Hi, I am newbie to asterisk. I would like to use asterisk using VoIP. I don't want to use any hardware. I have installed Asterisk 1.2.13. I would like to record using AGI command RECORD FILE. I would also like to do conferencing and recording in asterisk. How could this be done? Help me out with conf files and also using AGI script. -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] G.729 Voicemail
List, I have some cisco phones (7940) and asterisk 1.4 running nicely.. Communication between the phones is G.729, and my sip.conf looks like this: disallow=all; First disallow all codecs allow=g729 ; allow=gsm allow=ulaw allow=alaw However, I cannot call voicemail - I get the following error: [Apr 20 14:58:31] WARNING[87184]: channel.c:2816 set_format: Unable to find a codec translation path from g729 to gsm Shouldn't it switch to gsm automatically? I cannot purchase g729 licenses, as FreeBSD is not yet supported (with asterisk 1.4) Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Passive E1 Pri Tap for Voice Recording
I did this with a Nortel MICS a few years ago. No problem.
The dialplan was something like:
[incoming]
exten = _X.,1,setvar(filename) ;We did something with callerid and call
date and time, but I can't really remember
exten = _X.,2,Monitor(filename)
exten = _X.,3,Dial(Zap/G2/${EXTEN})
[outgoing]
exten = _X.,1,setvar(filename) ; If you want to record outgoing calls
exten = _X.,2,Monitor(filename); use these two lines, otherwise, just skip
them
exten = _X.,3,Dial(Zap/G1/${EXTEN})
Obviously, this isn't production code, but you should get the idea. If
you're in a 2-party area, you probably need to make your employees sign a
disclosure, and play a sound file to your callers to warn them that the call
is/may be recorded. While it will waste space, I recommend starting the
recording before the file is played. That way, if you're ever challenged,
you'd have something to back up your position that the caller knew. Add the
signed disclosure, and you may be OK.
Of course, I am no lawyer. And you probably ought to talk to one before you
do this. We did, and he had some helpful pointers on what to include in the
disclosure.
There are some areas that will require you to play an annoying beep to
callers. We didn't have to do that, so I'm not sure of the best way to go
about it.
Good luck,
David
On 4/20/07, Gavin Henry [EMAIL PROTECTED] wrote:
Dear All,
Is it possible to install * in front of a Avaya IP 406 system via a T
connector E1 tap so it's external to the Avaya system?
We would like to record upto 60 channels (2 * ISDN30e). This may increase
later.
Also, could the calls go into the cdr for retrieval/browsing later?
What hardware/server would you recommend?
Thanks.
___
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[asterisk-users] Agents.conf feature replication using addqueuemember
I (have to) would like to move my agents out of agents.conf in preparation for the deprecation of agentcallback login. Everyone I have spoken to is upset about this but the functionality can be accomplished in the dialplan and that is fine by me. I do have an issue with losing the features contained in the agents.conf though. I have to have the ackcall-yes working. All of my agents login on home phones and cellphones. When a call get presented to them there is a possibilty that their voicemail will answer before the queue timeout is reached. I fear that may connect callers to the voicemail when it answers. We currently get around this using the ackcall-yes which will wait for the caller to press pound, which a voicemail will not do and therefore the call will be put back in the queue. Other features that are important are the recording(which can be done on the dialplan side) and the update cdr. Both are important as the Monitor/mixmonitor will not name the file to be assoicated with the agent and only agent related calls. IE: if you have to look up the recording based on extension you will get personal calls as well as queue calls. Updatecdr is invalueable for the same reasons in the call detail records. Is anyone aware of a way to accomplish these things? Are any efforts being made to replicate the missing features? I can deal with the cdr and recordings, but the ackcall=yes is a show stopper!!! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] gxp2000 expansion module blf leds not working
On 4/19/07, Zoilo Gomez [EMAIL PROTECTED] wrote: Am I the only one using the GXP2000 expansion module? I'm using one but I'm not terribly happy with it. With firmware 1.1.3.1 the phone wouldn't boot, and with 1.1.3.2 having the buttons configured for BLF caused complete lockups on the phone requiring a power cycle. It doesn't reset mind you - the display looks fine until you realize that you haven't received a call for 10 minutes. GS tech support suggested disabling BLFs, so until an firmware fix is available it's a sidecar of 56 speed dials to me. Granted, it doesn't lock up anymore either. I'd love to downgrade to 1.1.2.x, but GS's firmware isn't capable of doing that and the phone shipped with 1.1.3.1, so I'm kind of stuck where I am right now. As to your specific question, I am not sure what would cause this behaviour, especially if on the phone that the sidecar is attached to the BLFs work. I assume that 'show hints' shows state for all your monitored extensions and 'sip show subscriptions' shows that the phone is actually subscribing to the extensions you have assigned to each multifunction key. If your problem isn't there, my gut says bad hardware. I am also looking at the Aastra 57i with the 560 module, but I haven't gotten one in to play with yet. I've got a few 480i CTs and they are performing well, both from a provisioning and usability perspective. The ability to do all changes via provisioning is nice, as right now I can auto-provision changes to the GXP sidecar, but I still have to print up a replacement button label and walk it over the the phone. One concern is whether I have 20 or less most frequently called numbers; if you have more than 20 your operator will be page-shifting a lot (the 560 has 20 soft keys and 3 fixed change page buttons). Hope that helps. -- j. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OT] OMG Verizon is terrible
This could go on forever, I mean take your pick Verizon, Att, Bell South any of them. Same story We are the phone company, who else can you call?. We have time and again seen it take weeks to get the order documents created, not the actual order, just the paperwork to create the order. I personally take great joy in finding anyway not to deal with them. The only way I see them changing their ways is by losing enough customers. I think Verizon is learning that lesson, but their response is not to compete and satisfy the customers, but to put the competition in a strangle hold with patents that are vague and broad. Ultimately I think Verizon will suffer from the court decisions more then anyone else, the true nature of their leadership is not to satisfy the customer. Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Softphone that supports central provisioning?
Has anyone found a softphone that supports pulling it's configuration from a central server via TFTP/FTP/HTTP, much like hard desk phones use? I'm looking for something for a call center that I can provision from a central location by generating config files. If the phone has soft keys (yes, I know they're all soft - but you know what I mean; programmable buttons whose function comes from the provisioning system), even better. I know idefisk Biz says they'll do this, but it's not in the release candidate and will make it's debut in the final version, which is a little too much early adoption for my liking. Other than that, I'm back at X-Lite/eyeBeam, which stores it's configs in binary files, preventing me from I'm open to SIP/IAX, so long as I don't have to jump through hoops to get it talking to *. Thanks for any experience you can share. -- j. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pci 2.2 - pci-e x16
On Fri, 2007-04-20 at 15:37 +0300, asterisk wrote: Hi, Does anyone know if it is possible to plug a tdm400p pci digium card into an pci-e 16x slot ? Is there a possibility to work? I have a sun fire x2100 which doesn't have pci slots. Does Digium make pci-e cards? Can you insert a square peg in a round hole? The only company that has announced PCI-Express products at this moment is Sangoma. Rhino has also stated that they will deliver all their products in a PCI-Ex option. -- Telecomunicaciones Abiertas de Mexico S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] adding second TDM400P card causes echo cancellation to fail for all Zap channels
Stephen Bosch wrote: Hi, folks: Yesterday I added a second TDM400P card to a working, echo-free server running HPEC. Today, I'm getting these messages: Apr 20 09:12:12 WARNING[5679]: chan_zap.c:1551 zt_enable_ec: Unable to enable echo cancellation on channel 3 along with complaints of severe echo. The channels have all been tuned using 'fxotune -s' and the echo numbers are all under 2%. I guess that Asterisk can't do echo cancellation on these channels. Why would I be getting this? Maybe I would be getting this because I didn't have the zaphpec_enable in my system startup :) So -- for future reference, in case anybody else makes this mistake: zaphpec_enable has to run at boot time, after the zap modules are loaded. I'll go back to my corner now. -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call queue problem
Hi, I've been configuring AsteriskNOW from the GUI but could it be that the GUI isn't working properly? because when I make a queue and add a few agents, and when I call the queue none of the phones ring. The queue is also configured at Ringall I checked the queues.conf file and the settings matched. I also noticed that the agents I made in the GUI, that they were not written away in agents.conf file, so I've added them there but still no results... any suggestions? Tim ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ztdummy does not load properly at server startup
Theo, I'm glad my reply was helpful to you. The responses pointed out that it's time for me to update my procedures and documentation, so I'm benefiting as well. My thanks go out to everyone who participated in this thread. Thank you, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OT] OMG Verizon is terrible
Alan Bunch wrote: Remember the big lie with Verizon is not The tech will be there at noon. It is Your FOC (Firm Order Commit) date is xxx The dates they give are neither Firm or Committed. Just ask them. As long as you remember, they are the Phone Company and you are just the customer. no body will be disapointed. Do I sound a lil grumppy. 5 and half week for fiber buildout quote. Yup, Im grumppy. alabun Lee Jenkins wrote: Steve Totaro wrote: They are all terrible in their own way. Don't you have someone you can delegate the Verizon babysitting responsibility to? I would consider sales calls a little more important than being a babysitter. Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Lee Jenkins Sent: Wednesday, April 18, 2007 6:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] [OT] OMG Verizon is terrible Had an appointment for these schmoes to come out and install another line. Was supposed to be 8-12. Its now 6PM and not even call. Missed 3 sales calls waiting on these jerks. No wonder customers were jumping ship to Vonage. -- Not on that day, I was the only one available unfortunately. I just got off the phone with Verizon and had them turn off all POTS lines except one. Between the humming noise they seem to not be able to fix and CID problems with additional lines we just ordered, I think I've had enough of verizon over the last 3 days. I'll keep the one line for when voip lines are down and shed the frustrations... It'd be nice if T1's were more reasonably priced or FIOS was here. -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OT] OMG Verizon is terrible
I once had to oversee Verizon install a PRI line in Manhattan. I live 2.5 hours away, but we made the appointment, and I was there, but the Verizon tech never showed. I made another appointment, and it happened again, and again, and again. I don't even remember how many times it finally took, but it was ridiculous. The techs were even lying and saying they came and there was no one there to let them in. They seem to have gotten better in recent years, but they own all the physical lines, and they know it. Back in the old days, our government de-monoplized monopolys. What happened to the good old days when we could just split apart big companies for fun and better competition? Blasted FTC for approving all these telcom mergers. Comcast is no better than Verizon. Took them three visits to get my new home serviced, and I'm still limping along at 3mb / 256k (they can't tell me why, I'm provisioned for 6mb). Now they're charging me for DVR service I didn't subscribe too. They're just idiots too. The only successful way I've found to deal with large unmanaged companies is to keep calling in trouble tickets, get new techs, and you'll eventually get one that knows what they're doing. I personally hate Verizon so much that I've elimitated all my lines with them, from cell, to home, to business. I refuse to deal with them and recommend alternative (and less expensive) solutions to all my customers. It seems to work great, and the only one loosing is Verizon. They won't have any money left to throw their weight around with if all their customers leave. -Kenneth ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Passive E1 Pri Tap for Voice Recording
On 20/04/07, David Gomillion [EMAIL PROTECTED] wrote:
I did this with a Nortel MICS a few years ago. No problem.
The dialplan was something like:
[incoming]
exten = _X.,1,setvar(filename) ;We did something with callerid and call
date and time, but I can't really remember
exten = _X.,2,Monitor(filename)
exten = _X.,3,Dial(Zap/G2/${EXTEN})
[outgoing]
exten = _X.,1,setvar(filename) ; If you want to record outgoing calls
exten = _X.,2,Monitor(filename); use these two lines, otherwise, just skip
them
exten = _X.,3,Dial(Zap/G1/${EXTEN})
Obviously, this isn't production code, but you should get the idea. If
you're in a 2-party area, you probably need to make your employees sign a
disclosure, and play a sound file to your callers to warn them that the call
is/may be recorded. While it will waste space, I recommend starting the
recording before the file is played. That way, if you're ever challenged,
you'd have something to back up your position that the caller knew. Add the
signed disclosure, and you may be OK.
Of course, I am no lawyer. And you probably ought to talk to one before you
do this. We did, and he had some helpful pointers on what to include in the
disclosure.
There are some areas that will require you to play an annoying beep to
callers. We didn't have to do that, so I'm not sure of the best way to go
about it.
Thanks for this. Given me some ideas. I think our solution has to be
non-evasive, i.e. in case the recording box goes down, the main pbx
works :-)
Good luck,
David
On 4/20/07, Gavin Henry [EMAIL PROTECTED] wrote:
Dear All,
Is it possible to install * in front of a Avaya IP 406 system via a T
connector E1 tap so it's external to the Avaya system?
We would like to record upto 60 channels (2 * ISDN30e). This may increase
later.
Also, could the calls go into the cdr for retrieval/browsing later?
What hardware/server would you recommend?
Thanks.
___
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Re: [asterisk-users] Softphone that supports central provisioning?
2007/4/20, James FitzGibbon [EMAIL PROTECTED]: Other than that, I'm back at X-Lite/eyeBeam, which stores it's configs in binary files, preventing me ... -- j. James, Are you sure eyeBeam config are binary ? I thought it was just the case for XLite. Regards ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ZT_CHANCONFIG failed on channel 1: No suchdeviceor address
lsmod | grep ^zaptel lsmod | grep ^zaptel zaptel183076 2 zttranscode,wctdm Cameron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why duoble digits must be so fast to activate features?
Hi Mauro; Try to add featuredigittimeout = 1500 at features.conf in the [global] section. On 4/20/07, Gordon Henderson [EMAIL PROTECTED] wrote: On Fri, 20 Apr 2007, Mauro Zanin wrote: Hi everybody, I'm testing Asterisk 1.2.14(you can say: why such an old version? I say: I'm using Trixbox..). I must be as fast as a flash to press *2 and do an attended transfer. If I wait only a tenth of a second nothing happens. I think it is an issue. I have seen the source code and found nothing bad. Is this a known issue? Change it in features.conf. Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Softphone that supports central provisioning?
On 4/20/07, Olivier [EMAIL PROTECTED] wrote: Are you sure eyeBeam config are binary ? I thought it was just the case for XLite. Having looked into it further, you're right. For some inexplicable reason it's not putting the files where the manual says they should be - instead of a directory called eyeBeam n.n they're in a folder called 'RegNow Basic', but the .CPS files there are indeed in XML rather than binary format. When I last looked, I suspect I assumed that eyeBeam stored it's configs in the X-Lite directory and was thus looking at the configs for the free version that were no longer being accessed. That does help a little for provisioning, as I can at least generate the configs and then place them someone central, but actually getting them to the phone is still kludgey. Since they are in the Local Settings folder, they can't be made part of a roaming profile. I've tried moving the CounterPath directory from \Documents and Settings\username\Local Settings\Application Data to \Documents and Settings\username\Application Data, but the phone never references the configs held there. Right now (with X-Lite) i'm configuring each phone manually, then zipping up the configs and storing them in a location named for the windows username. On login, the zipfile is fetched and unzipped to the right location. Inelegant to be sure, but it works. XML just saves me having to do the configuration manually. In any case, this is now going down an OT path - I'll take it up with CounterPath on their forums. Thanks for the pointer. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OT] OMG Verizon is terrible
On 20 Apr 2007, at 17:10, Kenneth Padgett wrote: I once had to oversee Verizon install a PRI line in Manhattan. I live 2.5 hours away, but we made the appointment, and I was there, but the Verizon tech never showed. I made another appointment, and it happened again, and again, and again. I don't even remember how many times it finally took, but it was ridiculous. The techs were even lying and saying they came and there was no one there to let them in. They seem to have gotten better in recent years, but they own all the physical lines, and they know it. Back in the old days, our government de-monoplized monopolys. What happened to the good old days when we could just split apart big companies for fun and better competition? Blasted FTC for approving all these telcom mergers. Ah, back in the old days our government privatized the state monopoly (BT) intact (attitudes and all). As one of the conditions they had to deliver within 6 weeks of order. So I ordered a data line to my house (ok a bit obscure in those days, but I needed it). 6 weeks roll past, nothing happens. I call BT and ask why they haven't installed my line within 6 weeks of order. The guy gently explains that it is 6 weeks from order being accepted, and they haven't accepted mine yet! When are you going to accept it ? - About 5 weeks from when we plan to fit it! Hey, at least he was an honest bloke in a twisted system. Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom Phones
Can anyone tell me which config file tells the phone what file to load as bootrom.ld? Or is this hardcoded in the phone? I just got a IP501 but I have a bunch of IP500s... Will the bootrom (2.6.2) work OK with both the IP500 and 501? Thanks! Wiley E. Siler Director of Information Technology 4110 N. Scottsdale Rd. Ste 110 Scottsdale, Arizona 85251 (480) 296.0260 (866) 320.2083 ext. 1003 mailto:[EMAIL PROTECTED] www.education2020.com http://www.education2020.com/ Helping students on a mission. Graduation and beyond. attachment: image001.jpg ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Softphone that supports central provisioning?
On 4/20/07, James FitzGibbon [EMAIL PROTECTED] wrote: On 4/20/07, Olivier [EMAIL PROTECTED] wrote: Are you sure eyeBeam config are binary ? I thought it was just the case for XLite. Having looked into it further, you're right. For some inexplicable reason it's not putting the files where the manual says they should be - instead of a directory called eyeBeam n.n they're in a folder called 'RegNow Basic', but the .CPS files there are indeed in XML rather than binary format. When I last looked, I suspect I assumed that eyeBeam stored it's configs in the X-Lite directory and was thus looking at the configs for the free version that were no longer being accessed. I went around this loop with CounterPath a couple of months back. It seems that their idea of provisioning revolves around customising the software before selling it, so that it is locking the end-user into using your (the seller's) SIP server. They had trouble understanding that the user just paid money for this software, which they want to be provisioned by a server on their own network, and they do not support this. I gave up at this stage, but perhaps if more people apply pressure, it will become possible to extend their current (quite useable) provisioning interface, but have a user-configurable setting to determine where the configuration is fetched from. At present the configuration server setting is fixed at compile-time by CounterPath. Regards, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G.729 Voicemail
On Fri, Apr 20, 2007 at 02:59:28PM +0200, Michael Landin Hostbaek wrote: List, I have some cisco phones (7940) and asterisk 1.4 running nicely.. Communication between the phones is G.729, and my sip.conf looks like this: disallow=all; First disallow all codecs allow=g729 ; allow=gsm allow=ulaw allow=alaw However, I cannot call voicemail - I get the following error: [Apr 20 14:58:31] WARNING[87184]: channel.c:2816 set_format: Unable to find a codec translation path from g729 to gsm Shouldn't it switch to gsm automatically? Cisco 79XX phones only support ulaw, alaw or g729, not gsm. Asterisk only supports g.729 protocol in passthrough mode without the licence (i.e. It can set up a session between two licenced g.729 endpoints to talk to each other, but cannot get into the media path itself.) The voicemail system is presumably trying to transcode from g.729 to gsm and you haven't got the licence for that. (Maybe you can get hold of/convert the sounds in the g729 format for the voicemail system, then it may not have to transcode out of .gsm?) I am not sure what parts of the system are enabled/disabled without the licence. http://www.voip-info.org/wiki-Asterisk+G.729+Licensing I cannot purchase g729 licenses, as FreeBSD is not yet supported (with asterisk 1.4) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G.729 Voicemail
transcode out of .gsm?) I am not sure what parts of the system are enabled/disabled without the licence. This mentions voicemail g729 in pass-thru mode. I'm not sure if it works as I've never tried it, but it may be worth a try... http://www.voip-info.org/wiki/index.php?page=Asterisk+G.729+pass-thru -- Robert Lister - London Internet Exchange- http://www.linx.net/ [EMAIL PROTECTED] - tel: +44 (0)20 7645 3510- RL786-RIPE ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How can I improve call quality?
Hi All, I've a single 1.2.17 Asterisk system. Gradwell here in the UK is used for PSTN calls via IAX2. Our 'net link is a dedicated 2Mb fibre connection (of which we have ever used 50% max bandwidth). We've no E1/T1 links, everything is IP based. My boss complains that many of the calls he holds with others has a bad quality. He also says that its not just him. My iax.conf file has: disallow=all allow=ulaw allow=alaw bandwidth=high jitterbuffer=yes dropcount=2 maxjitterbuffer=1000 maxjitterinterps=10 resyncthreshold=1000 maxexcessbuffer=80 minexcessbuffer=10 jittershrinkrate=1 tos=lowdelay autokill=yes He complains of broken audio, muffled audio, and says compared to Skype its very poor, particularly during conference calls (zaptel meetme). Most of these would be SIP based within our server though, rather than IAX/PSTN based (X-lite/SJphone). Obviously I can't do much about the far end IP connections/Mobiles etc, but what can I do to tweak/improve the call quality on the A*k box itself? The CPU stays at a constant 10% usage, mainly due to a few other monitoring apps on there (with these turned off, its 2%, but still the same issues). Also - are there any useful stats/logs that I can examine to see the quality of calls? Thanks, Adrian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How can I improve call quality?
Hi Also - are there any useful stats/logs that I can examine to see the quality of calls? You didn't mention that you have any QOS on your router, so we can basically guarantee that your problem is the internet connection. Remember that all the research on networking has been how to saturate a single connection and download as fast as possible, so when some spod hits a website and reads a web page then he grabs basically the whole connection for a short space of time. During that time your voip packets tend to loose out and get delayed - the jitter buffer does some stuff to try and compensate, but ultimately it will loose Add some kind of priorisation to the T1 line and your quality should go up dramatically Check first using something like testmyvoip.com to get an idea of your situation (stress the internet by opening up lots of simultaneous downloads during the test) Cheap fix is to get a separate DSL line and run the voice over that... Ed W ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How can I improve call quality?
Our 'net link is a dedicated 2Mb fibre connection (of which we have ever used 50% max bandwidth). Remember in computer terms this means that you used 100% of the connection, 50% of the time Your voice will loose out against the big data packets and spoil the voice quality big time Ed W ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OT] OMG Verizon is terrible
Andrew Latham wrote: I make a habit of just buying hamburgers and stopping by the CO or hut where I see the vans. I tell them that I am buying favors with food and they like it.. Its a lot of work but it helps... This is by far the most effective way of getting something done with a telco. And as for it being a lot of work? No more work than wasting hours, days, or weeks waiting for a problem to get fixed properly or a Remember -- in a lot of ILECs, the technicians are still union. More often than not, they are none to happy with their corporate overlords either. It is sensible and not too difficult to turn them into friends. Cheers, -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How can I improve call quality?
Try turning the jitterbuffer off, I found that often the endpoints can do better on their own. On 20 Apr 2007, at 19:01, Adrian Marsh wrote: Hi All, I've a single 1.2.17 Asterisk system. Gradwell here in the UK is used for PSTN calls via IAX2. Our 'net link is a dedicated 2Mb fibre connection (of which we have ever used 50% max bandwidth). We've no E1/T1 links, everything is IP based. My boss complains that many of the calls he holds with others has a bad quality. He also says that its not just him. My iax.conf file has: disallow=all allow=ulaw allow=alaw bandwidth=high jitterbuffer=yes dropcount=2 maxjitterbuffer=1000 maxjitterinterps=10 resyncthreshold=1000 maxexcessbuffer=80 minexcessbuffer=10 jittershrinkrate=1 tos=lowdelay autokill=yes He complains of broken audio, muffled audio, and says compared to Skype its very poor, particularly during conference calls (zaptel meetme). Most of these would be SIP based within our server though, rather than IAX/PSTN based (X-lite/SJphone). Obviously I can't do much about the far end IP connections/Mobiles etc, but what can I do to tweak/improve the call quality on the A*k box itself? The CPU stays at a constant 10% usage, mainly due to a few other monitoring apps on there (with these turned off, its 2%, but still the same issues). Also - are there any useful stats/logs that I can examine to see the quality of calls? Thanks, Adrian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HPEC audio clipping: IMPORTANT DETAIL
Hi, everybody: Stephen Bosch wrote: Kevin P. Fleming wrote: Eric ManxPower Wieling wrote: Any updates on this? The code is done and initially tested; it is being reviewed internally and should be available on Friday or Monday. Under what circumstances would this clipping be present? Is this patch going to be recommended for anybody using HPEC? Guess what *I* noticed today? We've been using the HPEC for about a month now and hadn't had this clipping problem. Today I added two licences. In the process, I noticed that the HPEC version had been updated to 9.x. We'd been using 8.2. Since I'm going through the process of adding the licence, I thought I'd try updating the HPEC. The moral of that story: if it ain't broke, don't fix it. I've confirmed this: hpec-9.00.002 has the clipping problem. hpec-8.20 definitely doesn't. I've implemented and reverted. The clipping makes the phones unusable. I just count my lucky stars that I kept the old archive, or I'd be up the creek right now. What's the word on the patch? -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RE: OT (a little): IPV6 Ramifications Article
Per Jessen wrote: Remco Post wrote: Hans Witvliet wrote: The only obstacles currently, are the ISP's. Any decent ISP (eg. XS4All.nl) will give you an ipv6 address as well as an ipv4 address. Not around here (Zurich, Switzerland) they won't. I think there is one single provider with IPV6 as an option. And the other ones are perfectly decent providers too. Like I said, when the low-cost DSL routers/modems do not yet support IPV6, why should the provider? Not here, either. The best you can do is a tunnel host. -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OT] OMG Verizon is terrible
Tim Panton wrote: Ah, back in the old days our government privatized the state monopoly (BT) intact (attitudes and all). As one of the conditions they had to deliver within 6 weeks of order. So I ordered a data line to my house (ok a bit obscure in those days, but I needed it). 6 weeks roll past, nothing happens. I call BT and ask why they haven't installed my line within 6 weeks of order. The guy gently explains that it is 6 weeks from order being accepted, and they haven't accepted mine yet! When are you going to accept it ? - About 5 weeks from when we plan to fit it! Hey, at least he was an honest bloke in a twisted system. I find the best approach is to ignore what's in the publicly distributed marketing material. If you have a doubt about something that a company representative is telling you, ask for it to be confirmed in writing. If they waffle, you know they're having you on. Next, assume you are in the jungle, and there is no civilisation. Recruit everybody in the chain as a friend and accomplice. If you have to deal with the provider often, this is worth the initial effort. People tend to like food :) -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with TDM2400 and Polycom 501... Voice Quality Lost...
Arturo Ochoa wrote: Hi List... I have an asterisk box with one TDM2400, it has 8 FXS's and 12 FXO's, and it also has the echo canceller... I'm running Asterisk 1.2.13 and Zaptel 1.2.12 with the Kernel 2.6.9-34.0.2.EL I'm using Polycom's 501 with the SIP 1.6.2.0041 The problem is when someone dials to or from the PSTN through the TDM2400, the voice quality is crappy...Instead of hearing: Hello, this is John You hear.. He o, th s J hn I already tried with the fxotune utility, also using G711 or G729, dealing with the gains... but I can't see the light... This is a bug in the 9.00-002 HPEC echo canceller. I have no idea when a patch will be available. -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] VPM450: Not Present
I've got a system with a TE412P installed under Fedora Core 6 and I continue to see this message in the logs. The card most certainly does have an EC module installed. The system is suffering from echo problems, and I suspect this is no coincidence... I've double checked to ensure the module has been inserted correctly. I've not seen any other complaints on the lists, etc. about this error message, so I'm running out of clues. Same problem under Fedora Core 4. How does one confirm/troubleshoot EC card detection? Chris___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] STUN
Hi guys, I'm trying to implement STUN support in *, is there anyone here which have any experience in something like that? I've got the STUND and I'll try to buld a patch or something for sip. Any ideas or existing implementation would be nice. I know openpbx have it. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom Phones
Hi Wiley - Can anyone tell me which config file tells the phone what file to load as bootrom.ld? Or is this hardcoded in the phone? Yup, it's hardcoded. I believe this is the way it works: If there's a bootrom.ld on your configuration server, and it is newer than the one on the phone, the phone will load it. Otherwise, it will use what is already on the phone. There's no option to change the name or anything. I just got a IP501 but I have a bunch of IP500s… Will the bootrom (2.6.2) work OK with both the IP500 and 501? You bet. - Noah ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with TDM2400 and Polycom 501... Voice Quality Lost...
I have an asterisk box with one TDM2400, it has 8 FXS's and 12 FXO's, and it also has the echo canceller... I'm running Asterisk 1.2.13 and Zaptel 1.2.12 with the Kernel 2.6.9-34.0.2.EL I'm using Polycom's 501 with the SIP 1.6.2.0041 The problem is when someone dials to or from the PSTN through the TDM2400, the voice quality is crappy...Instead of hearing: Hello, this is John You hear.. He o, th s J hn I already tried with the fxotune utility, also using G711 or G729, dealing with the gains... but I can't see the light... This is a bug in the 9.00-002 HPEC echo canceller. I have no idea when a patch will be available. I don't think this is the HPEC issue. I don't think Zaptel 1.2.12 supported HPEC. This must be the hardware echo can on the TDM2400. Arturo, can you post your zapata.conf and sip.conf? Also, I don't think this is your problem, but you may want to consider upgrading to Asterisk 1.2.17 and Zaptel 1.2.16. There have been many bug and security fixes since Asterisk 1.2.13. - Noah ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 7921G running linux
I was just watching the informational video on cisco's web site about the 7921G and they guy mentions that the phone is running Linux. Anyone know if they've released the source code? This page confirms that the phone is running Linux http://www.cisco.com/en/US/products/hw/phones/ps379/products_qanda_item0900aecd80601788.shtml The phone doesn't support sipyet ;) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom SIP Phones On LAN can't register without WAN (Internet) Access
Hi Shawn - We have several Polycom 500/501/601's on both a LAN and at employee homes. The problem we are having is if our internet connection goes down the Local LAN phones loose their connection to the Asterisk Server. I've checked everything I can think of but can't figure out why its happening. I believe since the Asterisk Box is on the LAN and the phones are on the same LAN then it shouldn't need internet to function. The closest I have narrowed this down is to the DNS area. I found that if I block access to our ISP's DNS that the phones aren't able to register with asterisk. This baffles me because the phone has the LAN address for the Asterisk server so I don't know why it's doing DNS lookups. Hmm. Well, you've got me. I don't know why it would be doing that, it certainly shouldn't be. You might try a newer version of the SIP firmware or the 3.2.2 bootrom. If it still happens with the latest bootrom/firmware, you could do a packet trace on the phone. Is it doing DNS queries? If so, I'd call your Polycom reseller and have them take this up with Polycom (support requests are supposed to go through the reseller). Actually, in any case, I'd take it up with your Polycom reseller. - Noah ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] C7960 TFTP [Slightly off-topic]
Hi all, This is slightly off-topic, but I was hoping to be able to receive some insight as I'm sure plenty of experts with c7960's exist on this mailing list. I'm attempting to upgrade from SIP 8.3 - 8.6 on a C7960G that I inherited. I have my TFTP setup and unfiltered. The phone is doing TFTP over the internet as I'm telecommuting today, but I've placed it on the dmz to avoid any firewall headaches. Here's what a packet capture looks like: tcpdump -vv -Xnni eth1 -s 1000 port 69 tcpdump: listening on eth1, link-type EN10MB (Ethernet), capture size 1000 bytes 16:30:25.649429 IP (tos 0x0, ttl 51, id 1612, offset 0, flags [none], proto: UDP (17), length: 56) 1.2.3.4.50770 64.90.184.96.69: [no cksum] 28 RRQ SIP00187330C526.cnf octet 0x: 4500 0038 064c 3311 b2a8 44c1 9145 E..8.L..3...D..E 0x0010: 405a b860 c652 0045 0024 0001 5349 @Z.`.R.E.$SI 0x0020: 5030 3031 3837 3043 3532 362e 636e P00187330C526.cn 0x0030: 6600 6f63 7465 7400 f.octet. 16:30:41.667380 IP (tos 0x0, ttl 51, id 1617, offset 0, flags [none], proto: UDP (17), length: 56) 1.2.3.4.50771 1.2.3.5.69: [no cksum] 28 RRQ SIP00187330C526.cnf octet 0x: 4500 0038 0651 3311 b2a3 44c1 9145 E..8.Q..3...D..E 0x0010: 405a b860 c653 0045 0024 0001 5349 @Z.`.S.E.$SI 0x0020: 5030 3031 3837 3043 3532 362e 636e P00187330C526.cn 0x0030: 6600 6f63 7465 7400 f.octet. 16:31:15.596217 IP (tos 0x0, ttl 51, id 1606, offset 0, flags [none], proto: UDP (17), length: 51) 1.2.3.4.50757 1.2.3.5.69: [no cksum] 23 RRQ SIPDefault.cnf octet 0x: 4500 0033 0646 3311 b2b3 44c1 9145 E..3.F..3...D..E 0x0010: 405a b860 c645 0045 001f 0001 5349 @Z.`.E.E..SI 0x0020: 5044 6566 6175 6c74 2e63 6e66 006f 6374 PDefault.cnf.oct 0x0030: 6574 00 et. 16:31:31.621286 IP (tos 0x0, ttl 51, id 1611, offset 0, flags [none], proto: UDP (17), length: 58) 1.2.3.4.50758 1.2.3.5.69: [no cksum] 30 RRQ ./SIP00187330C526.cnf octet 0x: 4500 003a 064b 3311 b2a7 44c1 9145 E..:.K..3...D..E 0x0010: 405a b860 c646 0045 0026 0001 2e2f @Z.`.F.E../ 0x0020: 5349 5030 3031 3837 3043 3532 362e SIP00187330C526. 0x0030: 636e 6600 6f63 7465 7400 cnf.octet. 16:31:47.652531 IP (tos 0x0, ttl 51, id 1617, offset 0, flags [none], proto: UDP (17), length: 58) 1.2.3.4.50759 1.2.3.5.69: [no cksum] 30 RRQ ./SIP00187330C526.cnf octet 0x: 4500 003a 0651 3311 b2a1 44c1 9145 E..:.Q..3...D..E 0x0010: 405a b860 c647 0045 0026 0001 2e2f @Z.`.G.E../ 0x0020: 5349 5030 3031 3837 3043 3532 362e SIP00187330C526. 0x0030: 636e 6600 6f63 7465 7400 cnf.octet. 16:32:03.679382 IP (tos 0x0, ttl 51, id 1623, offset 0, flags [none], proto: UDP (17), length: 58) 1.2.3.4.50760 1.2.3.5: [no cksum] 30 RRQ ./SIP00187330C526.cnf octet 0x: 4500 003a 0657 3311 b29b 44c1 9145 E..:.W..3...D..E 0x0010: 405a b860 c648 0045 0026 0001 2e2f @Z.`.H.E../ 0x0020: 5349 5030 3031 3837 3043 3532 362e SIP00187330C526. 0x0030: 636e 6600 6f63 7465 7400 cnf.octet. --- The phone also displays TFTP SIP00187330C526.cnf on its LCD, however it does not appear to be retrieving binaries from the tftpserver. tftproot # cat OS79XX.TXT P003-08-6-00 -rwxr-xr-x 1 root root 129824 Dec 12 16:54 P003-08-6-00.bin -rwxr-xr-x 1 root root 130228 Dec 12 17:21 P003-08-6-00.sbn -rwxr-xr-x 1 root root459 Dec 12 17:40 P0S3-08-6-00.loads -rwxr-xr-x 1 root root 753560 Dec 12 17:20 P0S3-08-6-00.sb2 -rw-r--r-- 1 root root 681556 Jan 10 14:55 P0S3-08-6-00.zip -rw-rw-rw- 1 root root779 Apr 20 14:21 SIP00187330C526.cnf -rw-rw-rw- 1 root root 4658 Apr 20 16:29 SIPDefault.cnf -rw-rw-rw- 1 nobody nobody 11675652 Mar 28 16:18 c2600-entbase-mz.123-22.bin -rw-rw-r-- 1 nobody nobody 7735532 Jan 16 15:54 c2600-i-mz.123-21.bin -rw-rw-rw- 1 nobody nobody 11100664 Jan 16 16:24 c2600-ik9s-mz.122-27.bin -rw-rw-r-- 1 nobody nobody 8450865 Feb 27 23:35 c3560-advipservicesk9-mz.122-35.SE1.bin -rw-rw-rw- 1 nobody nobody 2726 Jan 16 16:23 cmeinternetlink-confg -rw-r--r-- 1 root root223 Apr 20 14:13 dialplan.xml -rw-rw-rw- 1 nobody nobody 1413 Jan 16 16:10 helfant-confg -rw-r--r-- 1 root root779 Apr 20 14:11 xmlDefault.CNF.XML Thanks for any suggestions all, - sf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Softphone that supports central provisioning?
From: Steve Davies [EMAIL PROTECTED] Date: Fri, 20 Apr 2007 18:26:57 +0100 On 4/20/07, James FitzGibbon [EMAIL PROTECTED] wrote: On 4/20/07, Olivier [EMAIL PROTECTED] wrote: Are you sure eyeBeam config are binary ? I thought it was just the case for XLite. Having looked into it further, you're right. For some inexplicable reason it's not putting the files where the manual says they should be - instead of a directory called eyeBeam n.n they're in a folder called 'RegNow Basic', but the .CPS files there are indeed in XML rather than binary format. When I last looked, I suspect I assumed that eyeBeam stored it's configs in the X-Lite directory and was thus looking at the configs for the free version that were no longer being accessed. I went around this loop with CounterPath a couple of months back. It seems that their idea of provisioning revolves around customising the software before selling it, so that it is locking the end-user into using your (the seller's) SIP server. They had trouble understanding that the user just paid money for this software, which they want to be provisioned by a server on their own network, and they do not support this. I gave up at this stage, but That's because mainstream service providers only want a branded client that indeed locks users in. Unless a reasonably powerful commercial entity (or even freelance org) exerts pressure, individual users and small companies can't do much. Does a Web deployed client such as JAIN SIP applet count? Yuan Liu perhaps if more people apply pressure, it will become possible to extend their current (quite useable) provisioning interface, but have a user-configurable setting to determine where the configuration is fetched from. At present the configuration server setting is fixed at compile-time by CounterPath. Regards, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk PiX devices
Hi All, Im just getting started in the asterisk world and im wondering if anyone can point me in the right direction towards getting asterisk working from my house to my asterisk server in my colocation facility. Thanks --Don ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FW: Asterisk PiX devices
I forgot to add the hardware. Im using Gentoo Linux a Pix 515 Thanks --Don From: Don E. Wisdom Sent: Friday, April 20, 2007 2:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Asterisk PiX devices Hi All, Im just getting started in the asterisk world and im wondering if anyone can point me in the right direction towards getting asterisk working from my house to my asterisk server in my colocation facility. Thanks --Don ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Big trouble with zap lines
Hi, I recently install asterisk 1.4 over fedora 4, I configured 12 zap channels like this In zaptel.conf fxsks=1-4 fxsks=5-8 fxsks=9-12 loadzone=us defaultzone=us in zapata.conf in channels section context=incoming signalling=fxs_ks channel = 1-4 channel = 5-8 channel = 9-12 when i run ztcfg -vv show 12 channels correctly configured whe i run zap show channels in asterisk console this show 12 channels correctly configured when i call to a zap channel like this Dial(zap/1/somenumber,15,r) in the console appear that asterisk is dialing trought this channel to this somenumber but in the line the call never go out nor in, the same happens when dial from outside, the line is ringing until the normal timeout. the PSTN lines used work normally whit normal hardphones (PSTN) zaptel, asterisk, zttool and ztcfg all never send any error message. What could be the problem?? Could be a damaged wildcard My card is wctdm2400p with 12 fxs ports in 3 modules thanks in advance ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] CallerID Auth
From: Arun Kumar [EMAIL PROTECTED]
Date: Fri, 20 Apr 2007 17:58:10 +0400
Hi,
in my dial plan I've configured two trunks to make outbound calls (one for
national calls and other international). I want to allow only 2-3 extension
to make use of my international trunk to make outbound calls so I want some
kind of auth. based on their callerid . Please guide.
Just detect that a call is international, then branch out. e.g., if 011 is
the prefix required for international,
[outbound]
exten = _011.,1,Dial(Local/${EXTEN}/international)
exten = _X.,1,Dial(ZAP/g1/${EXTEN})
[international]
exten = _X.,1,GotoIf(${DBEXISTS(international/${CALLERID(NUMBER)})}?:deny)
exten = _X.,n,Dial(ZAP/g1/${EXTEN})
exten = _X.,n,Hangup; just in case
exten = _X.,n(deny),Playback(not-a-valid-numbertry-again)
exten = _X.,n,DISA(nopassword,outbound)
This is assuming AstDB contains a family international that includes
extensions/ID's allowed. Hope this helps.
Yuan Liu
thanks
arun
___
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Re: [asterisk-users] Asterisk PiX devices
http://www.voip-info.org/wiki/index.php?page=Asterisk+-+dual+servers Don E. Wisdom wrote: Hi All, Im just getting started in the asterisk world and im wondering if anyone can point me in the right direction towards getting asterisk working from my house to my asterisk server in my colocation facility. Thanks --Don ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Queue problems
Hi, I've been configuring AsteriskNOW from the GUI but could it be that the GUI isn't working properly? because when I make a queue and add a few agents, and when I call the queue none of the phones ring. The queue is also configured at Ringall I checked the queues.conf file and the settings matched. I also noticed that the agents I made in the GUI, that they were not written away in agents.conf file, so I've added them there but still no results... any suggestions? Tim ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Big trouble with zap lines
Do you have fxs modules or fxo modules? PSTN connects to fxo, but the signaling is fxs like you have. On 4/20/07, Ricardo Melendez [EMAIL PROTECTED] wrote: Hi, I recently install asterisk 1.4 over fedora 4, I configured 12 zap channels like this In zaptel.conf fxsks=1-4 fxsks=5-8 fxsks=9-12 loadzone=us defaultzone=us in zapata.conf in channels section context=incoming signalling=fxs_ks channel = 1-4 channel = 5-8 channel = 9-12 when i run ztcfg -vv show 12 channels correctly configured whe i run zap show channels in asterisk console this show 12 channels correctly configured when i call to a zap channel like this Dial(zap/1/somenumber,15,r) in the console appear that asterisk is dialing trought this channel to this somenumber but in the line the call never go out nor in, the same happens when dial from outside, the line is ringing until the normal timeout. the PSTN lines used work normally whit normal hardphones (PSTN) zaptel, asterisk, zttool and ztcfg all never send any error message. What could be the problem?? Could be a damaged wildcard My card is wctdm2400p with 12 fxs ports in 3 modules thanks in advance ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Big trouble with zap lines
Do you have fxs modules or fxo modules? PSTN connects to fxo, but the signaling is fxs like you have. On 4/20/07, Ricardo Melendez [EMAIL PROTECTED] wrote: Hi, I recently install asterisk 1.4 over fedora 4, I configured 12 zap channels like this In zaptel.conf fxsks=1-4 fxsks=5-8 fxsks=9-12 loadzone=us defaultzone=us in zapata.conf in channels section context=incoming signalling=fxs_ks channel = 1-4 channel = 5-8 channel = 9-12 when i run ztcfg -vv show 12 channels correctly configured whe i run zap show channels in asterisk console this show 12 channels correctly configured when i call to a zap channel like this Dial(zap/1/somenumber,15,r) in the console appear that asterisk is dialing trought this channel to this somenumber but in the line the call never go out nor in, the same happens when dial from outside, the line is ringing until the normal timeout. the PSTN lines used work normally whit normal hardphones (PSTN) zaptel, asterisk, zttool and ztcfg all never send any error message. What could be the problem?? Could be a damaged wildcard My card is wctdm2400p with 12 fxs ports in 3 modules thanks in advance ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Developing Marketing materials ...
Hi, I am working on developing a professional Marketing Materials for my systems. I plan on using a very good(expensive) company to do that so splitting the costs with several people would be nice. Let me know if you are interested on taking part in it. robert ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk PiX devices
Sorry I should clarify. I need to pass sip traffic thru the pix to the asterisk server. (from sip phones at my house and wherever else I might be) The pix has 7.2.2 os --Don -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jorge Mendoza Sent: Friday, April 20, 2007 3:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk PiX devices http://www.voip-info.org/wiki/index.php?page=Asterisk+-+dual+servers Don E. Wisdom wrote: Hi All, Im just getting started in the asterisk world and im wondering if anyone can point me in the right direction towards getting asterisk working from my house to my asterisk server in my colocation facility. Thanks --Don ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users !SIG:4629331f169582021920165! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with TDM2400 and Polycom 501... Voice Quality Lost...
Noah Miller wrote: I have an asterisk box with one TDM2400, it has 8 FXS's and 12 FXO's, and it also has the echo canceller... I'm running Asterisk 1.2.13 and Zaptel 1.2.12 with the Kernel 2.6.9-34.0.2.EL I'm using Polycom's 501 with the SIP 1.6.2.0041 The problem is when someone dials to or from the PSTN through the TDM2400, the voice quality is crappy...Instead of hearing: Hello, this is John You hear.. He o, th s J hn I already tried with the fxotune utility, also using G711 or G729, dealing with the gains... but I can't see the light... This is a bug in the 9.00-002 HPEC echo canceller. I have no idea when a patch will be available. I don't think this is the HPEC issue. I don't think Zaptel 1.2.12 supported HPEC. This must be the hardware echo can on the TDM2400. Arturo, can you post your zapata.conf and sip.conf? Also, I don't think this is your problem, but you may want to consider upgrading to Asterisk 1.2.17 and Zaptel 1.2.16. There have been many bug and security fixes since Asterisk 1.2.13. - Noah ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users While trying to fix the problem, yesterday I spoke to Digium, they logged to the computer and they upgraded Zaptel to 1.2.16 with the HPEC support. They enabled the MARK2 software echo cancellation by default. Then we made a few tests but nothing changed. Then I enabled the aggressive mode of this echo canceller, then I ran the fxotune -i, and last I ran the fxotune -s to load the parameters... With this changes, the voice quality seems to be better, at least you can hear almost all the words, but still have low quality... This is the zapata.conf [channels] language=en context=from-zaptel signalling=fxs_ks ;rxwink=300 ; Atlas seems to use long (250ms) winks ; ; Whether or not to do distinctive ring detection on FXO lines ; ;usedistinctiveringdetection=yes usecallerid=yes hidecallerid=no callwaiting=yes ;usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no ;echotraining=800 rxgain=0.0 txgain=0.0 group=0 callgroup=1 pickupgroup=1 immediate=no ;faxdetect=both faxdetect=incoming ;faxdetect=outgoing ;faxdetect=no ;Include genzaptelconf configs #include zapata-auto.conf group=1 -- Ing. Arturo Ochoa N Network Administrator Electrosystems, ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] why do I get this message
set_format: Unable to find a codec translation path from ulaw to g729 Both endpoints are PAP2 set to G711 only I have 1.2.17 on Suse 10.1 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 3rd T1 of quad card won't change signaling
--- Barton Fisher [EMAIL PROTECTED] wrote: Looks like: amaflags=billing switchtype=national is being carry-over from prior PRI.. (All PRI stuff) Try moving below before the first PRI? Thanks all, I tried the: T1 as port 1 and then the PRI as ports 2 and 3 but zap dumped again. I tried to blank the switchtype= , but zap didn't like that. span=3,0,0,d4,ami did not work with the original setup. ztcfg -vv gives no errors at all and shows correct signaling per port. I think this only reads the zaptel.conf and the error is occuring while parsing the zapata.conf. After every change, I remove/readd all zap modules and then ztcfg -vv. I will try the ztcfg -vvf tonight. The customer asked the faxman and said a PRI would be better anyway. Thanks again, I'll let you know. JJ __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] why do I get this message
On Fri, 20 Apr 2007, Bruce Ferrell said something to this effect: set_format: Unable to find a codec translation path from ulaw to g729 Both endpoints are PAP2 set to G711 only Where precisely are they so set? -- Alex Balashov [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue problems
Are your agents logged into the queue? -brandon Tim Verscheure wrote: Hi, I've been configuring AsteriskNOW from the GUI but could it be that the GUI isn't working properly? because when I make a queue and add a few agents, and when I call the queue none of the phones ring. The queue is also configured at Ringall I checked the queues.conf file and the settings matched. I also noticed that the agents I made in the GUI, that they were not written away in agents.conf file, so I've added them there but still no results... any suggestions? Tim ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] why do I get this message
Sip Debug, But I can tell you now that one of them is requesting g729, or, asterisk has g729 set for one of its codecs in sip.conf and needs to translate it. grep -r g729 /etc/asterisk/* Alex Balashov wrote: On Fri, 20 Apr 2007, Bruce Ferrell said something to this effect: set_format: Unable to find a codec translation path from ulaw to g729 Both endpoints are PAP2 set to G711 only Where precisely are they so set? -- Alex Balashov [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
