Re: [asterisk-users] Problem in making outbound calls in PRI

Hey everyone ! This config worked ! ; zapata.conf [channels] language=en context=from-pstn switchtype=euroisdn pridialplan=local signalling=pri_cpe usecallerid=yes hidecallerid=no usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes

[asterisk-users] Hardware TDM Switching

Hello, Do anybody know, if there is a way to connect 2 zap-channels with Hardware TDM Switching? Thanks Nico ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:

Re: [asterisk-users] Stress test

http://sipp.sourceforge.net/ SIPp is a free Open Source test tool / traffic generator for the SIP protocol On 12/13/06, Andre Luiz Martins Rodrigues [EMAIL PROTECTED] wrote: Hello peoples, I need to do a test of urgent stress. It know as much as connections simultaneous my equipment is

Re: [asterisk-users] MeetMe Conferencing and Marked Mode

Savoy, Kevin - Williston, ND schrieb: I'll give this a try but seems silly to require 2 different extensions for one conference room. Thanks for the input. Actually you don't need 2 different extension, but two different parameter-sets for the meetme-App. So, you have to implement some logic

[asterisk-users] IAX trunk problem

I wonder if anyone can help me with this. I have 4 sites running Asterisk and these are linked via IAX trunks and ADSL lines. Calls coming into any of these sites are received locally and forwarded to a central operator. E.g. Call comes in on site A and is forwarded to the operator on

Re: [asterisk-users] How to temporarily unload modules.

Thanks Tzafrir and Marco for the info. If I want to unload modules during start-up, I have to edit my /etc/asterisk/mudules.conf and add something like; noload = app_test.so or I can unload them immediately at CLI using Mr. Cohen suggestion. Regards. /etc/asterisk/modules.conf Marco

Re: [asterisk-users] MeetMe Conferencing and Marked Mode

On 12/14/06, Tobias Wolf [EMAIL PROTECTED] wrote: Actually you don't need 2 different extension, but two different parameter-sets for the meetme-App. So, you have to implement some logic that detects, if the calling user has to be marked or not. It's your choice if you do this by dialplan logic

[asterisk-users] RE: Extending Avaya IP Office ISDN30e with Asterisk

Has anyone hooked up * as an extension/trunk of an Avaya system that has around 2 ISDN30e's. I'm currently running an Asterisk box between my ISDN30 PRI and my Argent Office (pre Avaya takeover of Network Alchemy but still the same box as the Avaya IP Office). All it took was a two PRI digium

[asterisk-users] WRAP+astlinux g729

Hello How many simultaneous conversations g.729a should one expect with a WRAP board running Asterisk? Has anybody tried this? Kind Regards Jon Leren Schøpzinsky ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list

[asterisk-users] AOC-D or similar

hi all, I'm trying to send text messages to Snom 300 to show the credit remaining during the call... Sending a MESSAGE directly to the phone via udp i'm able to update the text on the display... but not during the conversation. I read about AOC, but i can't find any documentation about

Re: [asterisk-users] MeetMe Conferencing and Marked Mode

RR schrieb: And what would someone have to do to sweet-talk you into sharing this AGI ;) Hmmm, there is really not much to share. Most of the code handles Authentication or other stuff, like informing another server that a new user has entered an conf-room, or updating databases. Mostly I

Re: [asterisk-users] send fax by Iaxmodem ?

Hi all, First let me say thank you for Lee Howard, you definitely found my problem on sending faxes! I'm using hy-email2fax to send faxes, and i notice that is there the problem is starting, as the subject of my .eml file contains only the phone number but then some how hy-email2fax is not

Re: [asterisk-users] Measuring VoIP latency and packet loss

Mochamad Susantok wrote: I have already use smokeping, and great for measure latency and packet loss, but not voip packet especialy, or you has been modified smokeping ? I have not modified it. I take it that if the network has considerable latency, so will VOIP. It has been my experience

Re: [Asterisk-Users] Siemens Gigaset SL75

Do you know if it has 802.1x authentication as it is defined in EDUroam ( http://www.eduroam.org/ ) ? I never found a WiFi phone working with 802.1x I tested ZyXel Prestige 2000 but the sound was bad and it doesnt support 802.1x :( Thanks Joao Pereira [EMAIL PROTECTED] wrote: No, the

[asterisk-users] matching the beginning of an EXTEN

Hello how can I distinguish all the calls that arrive to my Asterisk starting with: 351217588XXX ? I want match the first 9 digits does Asterisk has any function for this? Thanks Regards Joao Pereira ___ --Bandwidth and Colocation provided by

Re: [asterisk-users] matching the beginning of an EXTEN

Use ${EXTEN:0:9} Regards, Ove Joao Pereira wrote: Hello how can I distinguish all the calls that arrive to my Asterisk starting with: 351217588XXX ? I want match the first 9 digits does Asterisk has any function for this? Thanks Regards Joao Pereira

Re: [asterisk-users] PRI to SIP

For PRI you have 3 main solutions. This is the order of stability (and pricing): 1. Digium or Sangoma cards use the computer processor and that could be bad if you have huge traffic through the PRI 2. Eicon Diva cards have their own processor, which releases the PC processor and gives more

Re: [Asterisk-Users] Siemens Gigaset SL75

I think, Nokia E60/61/70 currently supports 802.1x Joao Pereira wrote: Do you know if it has 802.1x authentication as it is defined in EDUroam ( http://www.eduroam.org/ ) ? I never found a WiFi phone working with 802.1x I tested ZyXel Prestige 2000 but the sound was bad and it doesnt

RE: [asterisk-users] matching the beginning of an EXTEN

Try Exten = _351217588XXX, 1, Dial ( ... ) Thanks and Regards --Sandeep Kalra Ph: +91-120-4342000-X-2966 : +91-120-4342966 (direct) M- 9810683168 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joao Pereira Sent:

Re: [asterisk-users] MeetMe Conferencing and Marked Mode

On 12/14/06, Tobias Wolf [EMAIL PROTECTED] wrote: Hmmm, there is really not much to share. Most of the code handles Authentication or other stuff, like informing another server that a new user has entered an conf-room, or updating databases. Mostly I look an the CallerId to decide if this

[asterisk-users] Zaptel under FC6

Hi, all I am building a new server. Have installed FC 6 and put in TDM400 card. Checked out latest asteriusk code, run make install in zaptel directory. So far all is fine. Now I am trying to install the drivers. # modprobe zaptel FATAL: Module zaptel not found. Fair enough, no zaptel driver

Re: [asterisk-users] TDM400P won't ring GM phone of mere 0.1B

Yuan LIU wrote: A configuration string boostringer was mentioned in several messages, including one concerning TDM400P, all without indicating the applicable configuration file. This has no apparent effect on TDM400P wherever I tried. That would go in your /etc/modprobe.conf which controls

[asterisk-users] (no subject)

Hello everyone! I'm planning on setting up a new system shortly and can't pick the right card... We will have 2 or 3 lines coming in and 7 extensions (GXP2k's). Should I just get 2 or 3 X100P cards? Or do I need the Sangoma A20200 or even the A20200D (Echo cancelation)... I was

[asterisk-users] agi scripts running slowly

Hi all, I recently installed asterisk 1.2.4 on a HP DL140 G2 server and co-located it. My only problem with the box is that there is a noticeable delay in the processing of agi scripts compared to any other install of asterisk I have. Has anyone got any ideas why this is happening and any

[asterisk-users] Ssh access over a zap channel...

My need to do this through asterisk is simply the ability to provide me access with no additional cost to my customer. It seems like a nice thing to include as long as authentication is done well. I have worked on a dozen or more types of switches and all of them have supported this or had the

Re: [asterisk-users] TDM400P won't ring GM phone of mere 0.1B

On Thu, Dec 14, 2006 at 07:15:30AM -0600, Russ Price wrote: Yuan LIU wrote: A configuration string boostringer was mentioned in several messages, including one concerning TDM400P, all without indicating the applicable configuration file. This has no apparent effect on TDM400P wherever I

Re: [asterisk-users] PRI to SIP

Or any of a number of gateways that do this. Off the top of my head you can get one from CarrierAccess, Vega, Audiocodes, Mediatrix, Adtran, and others. Just try to be very careful as they all have their strengths and weaknesses and you need to evaluate how they would fit your needs. Best

Re: [asterisk-users] Ssh access over a zap channel...

On 14 Dec 2006, at 13:32, Jordan Novak wrote: My need to do this through asterisk is simply the ability to provide me access with no additional cost to my customer. It seems like a nice thing to include as long as authentication is done well. I have worked on a dozen or more types of

Re: [asterisk-users] PRI to SIP

did you use T38 with you patton smart node 2400 ? why Patton are very good GW and fax must work. you must also check that the clock source is the Primary and not the internal clock... 2006/12/14, Jerry Jones [EMAIL PROTECTED]: Or any of a number of gateways that do this. Off the top of my

Re: [asterisk-users] (no subject)

Todd- Asterisk wrote: Hello everyone! I'm planning on setting up a new system shortly and can't pick the right card... We will have 2 or 3 lines coming in and 7 extensions (GXP2k's). Should I just get 2 or 3 X100P cards? Or do I need the Sangoma A20200 or even the A20200D (Echo

Re: [asterisk-users] matching the beginning of an EXTEN

perfect!!! its now working this way: exten = _.,4,GotoIf($[ ${EXTEN:0:9} = 351217588] ? 20:10) Thanks a lot Joao Pereira Ove Aursand wrote: Use ${EXTEN:0:9} Regards, Ove Joao Pereira wrote: Hello how can I distinguish all the calls that arrive to my Asterisk starting with: 351217588XXX ?

Re: [asterisk-users] Pickup application

On 12/13/06, Aaron Daniel [EMAIL PROTECTED] wrote: Does anyone have the pickup application working? I'm attempting to get I did have it working. The problem I'm having is in the fact that my phones register with mac addresses instead of extensions, so I'm unsure as to what to put in the

Re: [asterisk-users] how to define a secure trunk

Can I do the encrypted trunk in SIP? Does Asterisk supports it? Thanks Joao Pereira Pavel Jezek wrote: http://www.voip-info.org/wiki/view/IAX+encryption Joao Pereira wrote: Hello I would like to define a trunk from my Asterisk to a VoIP provider, but I want to make it secure, because its

Re: [asterisk-users] how to define a secure trunk

as I know, only preliminary support: 0005413: [patch] Secure RTP (SRTP) http://bugs.digium.com/view.php?id=5413 Joao Pereira wrote: Can I do the encrypted trunk in SIP? Does Asterisk supports it? Thanks Joao Pereira Pavel Jezek wrote: http://www.voip-info.org/wiki/view/IAX+encryption

[asterisk-users] IBM Server / USB Ports

Hi, I have an IBM xSeries server... and the digium card is sharing IRQ with USB and giving me crackling audio. cat /proc/interrupts It brings up these results: 0: 10566547IO-APIC-edge timer 1: 9IO-APIC-edge i8042 2: 0 XT-PIC cascade 8:

RE: [asterisk-users] IBM Server / USB Ports

compile kernel without usb support or unload usb modules turby ps your tdm card don't share the irq, your network card share the irq... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Sent: Thursday, December 14, 2006 5:13 PM To: Asterisk Users

RE: [asterisk-users] how to define a secure trunk

joao, you can use ssh tunel, pptp or vpn for any sip/iax trunks or users. turby -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Pavel Jezek Sent: Thursday, December 14, 2006 4:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject:

Re: [asterisk-users] TDM400P won't ring GM phone of mere 0.1B

From: Tzafrir Cohen [EMAIL PROTECTED] Have you tried toe boost ring voltage option then recompile Zaptel? It is normally set to a fairly low voltage John Novack Thank you so much! I googled a bit about how to change ring voltage and only found an old and suspended feature request from

Re: [asterisk-users] AOC-D or similar

Ale wrote: hi all, I'm trying to send text messages to Snom 300 to show the credit remaining during the call... Sending a MESSAGE directly to the phone via udp i'm able to update the text on the display... but not during the conversation. I read about AOC, but i can't find any

Re: [asterisk-users] how to define a secure trunk

tunneling small rtp packets through vpn has big overhead, better to use application level encryption - encrypted iax or srtp. PJ [EMAIL PROTECTED] wrote: joao, you can use ssh tunel, pptp or vpn for any sip/iax trunks or users. turby -Original Message- From: [EMAIL PROTECTED]

Re: [asterisk-users] IBM Server / USB Ports

I see that the digium card doesn't share the IRQ however Digium has recommended diabled USB still... additionally the Digium card is on 169 which isn't a valid IRQ.. how can I find out what it is sharing with? On 12/14/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: compile kernel without

[asterisk-users] Web-MeetMe ready for prime time?

What kind of luck are people having with the Web-MeetMe control? The condition of the page on the voip-info wiki makes me a bit nervous about putting Web-MeetMe into a production environment. Use of MeetMe has really taken off here since installation and I need a scheduling and provisioning

Re: [asterisk-users] TDM400P won't ring GM phone of mere 0.1B

Yuan LIU wrote: The feature request # is 4542, but I don't know any associated bug number, nor with what phones other people had to tweak. My phone is a GE 27935GE3-B. (Don't know what possessed me to say GM:-) Yuan Liu Just gotta ask - you did plug in the power supply connection on the

RE: [asterisk-users] how to define a secure trunk

right, but who have production and tested code of application level encryption for SIP and IAX for SECURE(!) trunks? turby -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Pavel Jezek Sent: Thursday, December 14, 2006 6:15 PM To: Asterisk Users Mailing

Re: [asterisk-users] asterisk billing software

Sorry to bring back a post from the grave, but what do you feel is the worst problem you are having with AgileVoice? Is their support easy to reach? On 11/16/06, Chris Mazuc [EMAIL PROTECTED] wrote: Andrew Joakimsen wrote: Chris: We were evaluating AgileVoice currently, could you please

Re: [asterisk-users] Web-MeetMe ready for prime time?

You might check out http://sourceforge.net/projects/web-meetme and version 2.1.0, I had to tweak it a little, but it has worked well for people to schedule their own meetme conferences. On 12/14/06, Porier, Jeremy M. [EMAIL PROTECTED] wrote: What kind of luck are people having with the

RE: [asterisk-users] Web-MeetMe ready for prime time?

Jeremy wrote: What kind of luck are people having with the Web-MeetMe control? The condition of the page on the voip-info wiki makes me a bit nervous about putting Web-MeetMe into a production environment. Use of MeetMe has really taken off here since installation and I need a scheduling and

[asterisk-users] Show agent queue status on the phone?

Title: Message Hi All, Is it possible to show an agent's queue status on the phone? For example, in our currentnon-asterisk PBX, if a member of a call queue does not answer the phone when a queue call is sent to them, they go to a 'not ready' status, and this is indicated on their phone. So

Re: [Asterisk-Users] Siemens Gigaset SL75

At 03:40 AM 12/14/2006, you wrote: I tested ZyXel Prestige 2000 but the sound was bad and it doesnt support 802.1x :( Wow, I've always been impressed with the sound from my Zyxel 2000W. Ira ___ --Bandwidth and Colocation provided by Easynews.com

Re: [asterisk-users] (no subject)

At 05:23 AM 12/14/2006, you wrote: Should I just get 2 or 3 X100P cards? Or do I need the Sangoma A20200 or even the A20200D (Echo cancelation)... When I started down this path I choose the TDM04 and have always had occasional echo issues, not bad and not often, but it annoys the wife and

Re: [asterisk-users] TDM400P won't ring GM phone of mere 0.1B

From: Steve Prior [EMAIL PROTECTED] The feature request # is 4542, but I don't know any associated bug number, nor with what phones other people had to tweak. My phone is a GE 27935GE3-B. (Don't know what possessed me to say GM:-) Yuan Liu Just gotta ask - you did plug in the power supply

[asterisk-users] Broadvoice registration problems

Hello, I have two broadvoice accounts. Lately, very often my broadvoice accounts are in unregistered state. When I log into asterisk I see: voip*CLI sip show registry Host Username Refresh State sip.broadvoice.com:5060 [EMAIL PROTECTED] 120 Request Sent sip.broadvoice.com:5060 [EMAIL PROTECTED]

[asterisk-users] On-Hold

Hello, When in conversation, how can I put somebody on hold? thx ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:

Re: [asterisk-users] IBM Server / USB Ports

On Thu, Dec 14, 2006 at 11:13:29AM -0500, Matt wrote: Hi, I have an IBM xSeries server... Which model, exactly? With which customizations? and the digium card is sharing IRQ with USB and giving me crackling audio. Do you actually get any interrupts from the USB? cat /proc/interrupts

RE: [asterisk-users] Zaptel under FC6

From: Rudolf Ladyzhenskii [EMAIL PROTECTED] Now I am trying to install the drivers. # modprobe zaptel FATAL: Module zaptel not found. Fair enough, no zaptel driver is found on the system. Is there are any known problems with FC6? I did not have much trouble running on FC3 before. I'm not

[asterisk-users] Re: Vonage SIP access via asterisk?

This may not be vonage related as it appears that I can not register with any sip servers. I tried FWD and also get a black sip show registry Could it be a firewall issue? I am running IP tables on the computer which is on the internet with no NAT. Asterisk 1.2.13 I have allow outbound all.

[asterisk-users] Re: Vonage SIP access via asterisk?

I can dial out via FWD because the login is in the session. Vonage requires a register even for outbound. My understanding is that they log the register and then any call from that IP is from that user. This is why I can't dial out vonage. The root cause is that sip is not registering at all.

Re: [asterisk-users] Hardware TDM Switching

[EMAIL PROTECTED] wrote: Do anybody know, if there is a way to connect 2 zap-channels with Hardware TDM Switching? It's called DACS. See the /etc/zapata.conf config file sample. ___ --Bandwidth and Colocation provided by Easynews.com --

Re: [asterisk-users] matching the beginning of an EXTEN

Joao Pereira wrote: Hello how can I distinguish all the calls that arrive to my Asterisk starting with: 351217588XXX ? I want match the first 9 digits does Asterisk has any function for this? exten = _51217588XXX,1,Whatever ___ --Bandwidth and

[asterisk-users] Voicemail Live

Hi, Philipp von Klitzing posted this solution in Dec. 2005 Answering machine mimic: Listen while caller is leaving voicemail for you; with pick-up option Is there any other way to listen while caller is leaving a voicemail for you? Thanks Fernando ___

Re: [asterisk-users] IBM Server / USB Ports

Matt wrote: Hi, I have an IBM xSeries server... and the digium card is sharing IRQ with USB and giving me crackling audio. cat /proc/interrupts It brings up these results: 0: 10566547IO-APIC-edge timer 1: 9IO-APIC-edge i8042 2: 0 XT-PIC cascade

Re: [asterisk-users] Ssh access over a zap channel...

Jordan Novak wrote: My need to do this through asterisk is simply the ability to provide me access with no additional cost to my customer. It seems like a nice thing to include as long as authentication is done well. I have worked on a dozen or more types of switches and all of them have

[asterisk-users] Console latency

Another bizarry: If I run the Echo application from the console, I can hear a very long delay (upward to 1,000 ms). I can run the same application from a GrandStream phone (on the same LAN) and hear little delay. What could possibly be wrong? If it were interrupt overload, I'd hear lots of

[asterisk-users] Fast Busy

We currently have a pri coming into our asterisk system. Most of the time, the did numbers that we call into it work great. However, occationally, we get fast busies, but we noticed those busies were not due to anyone being on the line, etc... Any ideas what could cause this? Is this a congestion

[asterisk-users] StripXXX apps missing from asterisk-1.2.13?

All of StripMSD, StripLSD, etc., are missing when I downloaded asterisk-1.2-current.tar.gz, which explodes into 1.2.13. Are the strip club deprecated? What replacement functions should I use? Yuan Liu ___ --Bandwidth and Colocation provided by

Re: [asterisk-users] Console latency

Yuan LIU wrote: Another bizarry: If I run the Echo application from the console, I can hear a very long delay (upward to 1,000 ms). I can run the same application from a GrandStream phone (on the same LAN) and hear little delay. What could possibly be wrong? If it were interrupt overload,

Re: [asterisk-users] StripXXX apps missing from asterisk-1.2.13?

StripLSD is obsolete: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+StripLSD StripMSD is being phased out: http://bugs.digium.com/view.php?id=5673 John Beaman Telecom Specialist Voice Telecommunications Services Department. Good Samaritan National Campus 605-362-3331 [EMAIL

Re: [asterisk-users] Zaptel under FC6

Thanks for suggestion. Just tried that Was surprised with download size of only 72k. Anyway, command work, but I still have same ptoblem. I tried to run modprobe -- it failed. Tried to run service zaptel start and get: service zaptel start No functioning zap hardware found in /proc/zaptel,

Re: [asterisk-users] Ssh access over a zap channel...

From: Eric \ManxPower\ Wieling [EMAIL PROTECTED] [Goodies skipped] -= Info about application 'ZapRAS' =- [Synopsis] Executes Zaptel ISDN RAS application [Description] ZapRAS(args): Executes a RAS server using pppd on the given channel. The channel must be a clear channel (i.e. PRI source)

Re: [asterisk-users] Fast Busy

Rob Schall wrote: We currently have a pri coming into our asterisk system. Most of the time, the did numbers that we call into it work great. However, occationally, we get fast busies, but we noticed those busies were not due to anyone being on the line, etc... Any ideas what could cause this?

Re: [asterisk-users] StripXXX apps missing from asterisk-1.2.13?

Yuan LIU wrote: All of StripMSD, StripLSD, etc., are missing when I downloaded asterisk-1.2-current.tar.gz, which explodes into 1.2.13. Are the strip club deprecated? What replacement functions should I use? See README.variables in the Asterisk source.

Re: [asterisk-users] Console latency

From: Eric \ManxPower\ Wieling [EMAIL PROTECTED] Yuan LIU wrote: Another bizarry: If I run the Echo application from the console, I can hear a very long delay (upward to 1,000 ms). I can run the same application from a GrandStream phone (on the same LAN) and hear little delay. What could

Re: [asterisk-users] StripXXX apps missing fromasterisk-1.2.13?

From: john beaman [EMAIL PROTECTED] StripLSD is obsolete: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+StripLSD StripMSD is being phased out: http://bugs.digium.com/view.php?id=5673 John Beaman Telecom Specialist Voice Telecommunications Services Department. Good Samaritan

Re: [asterisk-users] Zaptel under FC6

From: Rudolf Ladyzhenskii [EMAIL PROTECTED] I guess, modules are mot there. Running find / -name zaptel* did not find any modules. Be careful here - wildcard expansion takes place locally unless you quote the string: $ find / -name 'zaptel*' Of course search from / is suboptimal as you are

Re: [asterisk-users] Console latency

Yuan LIU wrote: From: Eric \ManxPower\ Wieling [EMAIL PROTECTED] Yuan LIU wrote: Another bizarry: If I run the Echo application from the console, I can hear a very long delay (upward to 1,000 ms). I can run the same application from a GrandStream phone (on the same LAN) and hear little

[asterisk-users] VoipTalk unable to accept calls at present?

I am trying to get asterisks to work with http://www.voiptalk.org 's IAX service. I have configured asterisks as per their instructions and am using the x-lite soft phone. When I get an incoming call the softphone rings but the caller (from pstn) gets a recorded message saying the number is unable

RE: [asterisk-users] Diva Server V-BRI-2 and internal numbers

It looks like 107 is busy ;-) Please increase verbosity, like set verbose 5 capi debug to see what is happening. Hi Armin, Verbose was at 30 :) 107 is not busy since i can call it from 102, which is another internal phone. All internal phones are busy for Asterisk... Here is the log

Re: [asterisk-users] Zaptel under FC6

The Fedora Extras rpm is tiny because it has nothing really of help in it. It's missing the modules. I've had some success on Fedora Core 6 using the ATrpms repository, which has the zaptel-kmdl package for most variations of kernels included in FC6. Simon On 14 Dec 2006, at 22:31,

[asterisk-users] Re: Zaptel under FC6

On Fri, Dec 15, 2006 at 12:05:13AM +1100, Rudolf Ladyzhenskii wrote: Hi, all I am building a new server. Have installed FC 6 and put in TDM400 card. Checked out latest asteriusk code, run make install in zaptel directory. So far all is fine. Now I am trying to install the drivers. #

RE: [asterisk-users] Broadvoice registration problems

Any ideas? Did anyone experience something like that? Thx Yes, unfortunately, all the time. There answer is if it works with a sip softphone client than it's not their problem. It does work with the softphone client. -Original Message- From: Bartosz Wegrzyn - maillists

Re: [asterisk-users] (no subject)

I have been using the sangoma A200 with echo cancelation and I have been real happy. - Original Message - From: Todd- Asterisk [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, December 14, 2006 3:23 PM

Re: [asterisk-users] chan_sip.c:5267 sip_reg_timeout Error

Did you do a reload ? Also when you say you commented it out you mean that you commented out the register statement ? - Original Message - From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Sunday, December 10, 2006 5:02 PM Subject: [asterisk-users]

Re: [asterisk-users] Zaptel under FC6

Howard Lowndes wrote: How old is your mobo? I have that same problem and I think it because the TDM card will only work with PCI 2.2 or later and, although lspci finds the card, udev is not installing the zap devices. Which is why those in the know who don't care to hear Digium's stock

Re: [asterisk-users] IBM Server / USB Ports

Matt wrote: I see that the digium card doesn't share the IRQ however Digium has recommended diabled USB still... additionally the Digium card is on 169 which isn't a valid IRQ.. how can I find out what it is sharing with? the tdm card is not sharing an interrupt with your USB. It's your LAN

Re: [asterisk-users] On-Hold

If using a VOIP phone there should be a button. If using an ATA the instructions should be in the manual of the ATA (you also may be able to look in the web interface of the device). I forgot how to do it if you are using ZAP. Have a look on the wiki. - Original Message - From:

Re: [asterisk-users] Fast Busy

Sounds like you have a disconnect supervision problem. Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada We currently have a pri coming into our asterisk system. Most of the time, the did numbers that we call into it work great. However, occationally, we get fast busies,

Re: [asterisk-users] (no subject)

You might want to take a look at the new 4 port FXO from Grandstream I haven't had one yet to evaluate but assuming it works it is very price competative and off-loads all the analog (TDM) stuff from your PC Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada I have been using

Re: [asterisk-users] IBM Server / USB Ports

So you are saying that the card is on it's own IRQ and is not sharing anything with anything? I realize the eth0 and usb are sharing, but am not too concerned about that. On 12/14/06, Leo Ann Boon [EMAIL PROTECTED] wrote: Matt wrote: I see that the digium card doesn't share the IRQ

[asterisk-users] bridging calls on a samsung pbx from asterisk

I have the following configuration: VoIP Provider Asterisk Samsung PBX --- PSTN ^ Asterisk has a few VoIP extensions connected, but most of the extensions are hanging of the Samsung PBX. Asterisk has 1 NT and 1 TE interface, which are connected to a TE and NT

RE: [asterisk-users] Broadvoice registration problems

I am not sure how far I will go with that, but I did a capture and explained in detail what is the problem, I hope that somebody there will forward it to high level support maybe, who knows, it is so hard to get help when something strange happens, from my experience with broadvoice, everything

Re: [asterisk-users] On-Hold

this is ata , simple flash button works, thx If using a VOIP phone there should be a button. If using an ATA the instructions should be in the manual of the ATA (you also may be able to look in the web interface of the device). I forgot how to do it if you are using ZAP. Have a look on the

Re: [asterisk-users] IBM Server / USB Ports

Matt wrote: So you are saying that the card is on it's own IRQ and is not sharing anything with anything? I realize the eth0 and usb are sharing, but am not too concerned about that. What's your zttest result and did zttool reported any irq misses? If zttest is mostly 99.98%, then the zap

[asterisk-users] Bandwidth.com on asterisk

Does anybody know how to setup bandwidth.com trunk on asterisk. They provide bandwidth services to asterisk.org, but don't know how to setup up asterisk on their system. -- Zeeshan A Zakaria ___ --Bandwidth and Colocation provided by Easynews.com --

Re: [asterisk-users] Bandwidth.com on asterisk

What do they provide you? You normally wouldnt install asterisk on their system unless you are leasing a server from them. On 12/15/06, Zeeshan Zakaria [EMAIL PROTECTED] wrote: Does anybody know how to setup bandwidth.com trunk on asterisk. They provide bandwidth services to asterisk.org, but

[asterisk-users] FYI Panasonic Wireless Phone MWI

Last week I asked about MWI indicators on wireless phones that would work with Asterisk. I sent a message off to Panasonic asking them about it because in their ads they specifically stated that the indicator works with and requires phone company voicemail subscription. The is the model TG5631.

Re: [Asterisk-Users] Siemens Gigaset SL75

Joao Pereira ha scritto: Do you know if it has 802.1x authentication as it is defined in EDUroam ( http://www.eduroam.org/ ) ? I never found a WiFi phone working with 802.1x I tested ZyXel Prestige 2000 but the sound was bad and it doesnt support 802.1x :( Thanks Joao Pereira Well, I

[asterisk-users] fxotune unable to set impedence

My SM56 (Motorola X100P clone) has echo as hight as 38%, according to fxotune -d. But when trying to take action, it fxotune simply says it can't. ./fxotune -i3 - Running with parameters: doset=0 docalibrate=1 dodump=0 startdev=1 stopdev=252

RE: [asterisk-users] Diva Server V-BRI-2 and internal numbers

On Thu, 14 Dec 2006, Gregory Duchatelet wrote: It looks like 107 is busy ;-) Please increase verbosity, like set verbose 5 capi debug to see what is happening. Hi Armin, Verbose was at 30 :) 107 is not busy since i can call it from 102, which is another internal phone. All

RE: [asterisk-users] fxotune unable to set impedence

From: Yuan LIU [EMAIL PROTECTED] How can I fix this? Or does fxotune only tune TDM400? (My TDM400P shows a mere 1.2% echo.) Could it do authentic X100P? I just didn't want to accept fxotune.c's claim about working only with TDM. Several other users indicated that they were not able to