Re: [asterisk-users] test please ignore

On Wed, Nov 10, 2021 at 09:08:52AM +, Kingsley Tart wrote: > my last few emails to this list haven't appeared so I'm just testing 1. Check the archive: http://lists.digium.com/pipermail/asterisk-users/2021-November/thread.html 2. Check your list settings (e.g: Receive your own posts to the

Re: [asterisk-users] Test

Yes it is working  -Ursprungligt meddelande- Från: asterisk-users-boun...@lists.digium.com För Matt Fredrickson Skickat: den 21 mars 2018 03:46 Till: Asterisk Users Mailing List - Non-Commercial Discussion

Re: [asterisk-users] Test

Fail. On 10/28/2015 04:42 PM, ama...@sevana.fi wrote: Hi, Just checking if my emails reach the list. Thanks, Amanda -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us

Re: [asterisk-users] test call generator

Hi List, I'm trying to get working SIPp with media but something is wrong (it's working well without media), please help: This is the command I send at SIPp server: ./sipp -sn uac_pcap -d 5000 -s 2006 192.168.1.18 -l 20 -trace_err This is the result I see: Last Error: Aborting call

Re: [asterisk-users] test call generator

Hello Everyone, I wonder if someone could share a manual about using SIPp for Asterisk's testing. I'll be gratefull Regards, Elder Arohuanca Lima - Peru On Tue, Sep 30, 2008 at 12:59 PM, zac wolfe zac.wo...@gmail.com wrote: Sipp looks pretty good! I don't know how I missed this one. This

Re: [asterisk-users] test call generator

dan, elder, I have played with scripts to generate calls and track their completion, email me off-list if you have questions. daveC Daniel - Asterisk wrote: Hello Everyone, I wonder if someone could share a manual about using SIPp for Asterisk's testing. I'll be gratefull Regards,

Re: [asterisk-users] test

Test successful On 2010-03-21 9:12 AM, card support asterisk asteriskc...@hotmail.com wrote: hi: only test Best wishes! Asterisk Support group(sangoma, digium...), providing asterisk conf, pri, ss7, elastix, trixbox support. website:www.cnasterisk.com, www.voip88.com voip88

Re: [asterisk-users] test

fail. On Tue, 9 Feb 2010, aster...@opensourcesolution.in wrote: test -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:

Re: [asterisk-users] test case with queues and system()

At 5:59 PM on 19 Jan 2010, __ wrote: Test case: We have e1 trunk and multi-channel sip line. Clients waiting in the queue, which can handle 30 clients. They listen mellody and their position, while waiting. The system can handle only 5 clients at the moment. As soon

Re: [asterisk-users] test case with queues and system()

On Wed, Jan 20, 2010 at 10:18 PM, C. Chad Wallace cwall...@lodgingcompany.com wrote: At 5:59 PM on 19 Jan 2010, __ wrote: Test case: We have e1 trunk and multi-channel sip line. Clients waiting in the queue, which can handle 30 clients. They listen mellody and their

Re: [asterisk-users] test case with queues and system()

At 3:09 AM on 21 Jan 2010, __ wrote: On Wed, Jan 20, 2010 at 10:18 PM, C. Chad Wallace cwall...@lodgingcompany.com wrote: At 5:59 PM on 19 Jan 2010, __ wrote: Test case: We have e1 trunk and multi-channel sip line. Clients waiting in

Re: [asterisk-users] test

On 25/10/2009, Matt mhop...@gmail.com wrote: This is a test... I am being told I am subscribed, but I am not getting messages. Gmail always seems to hide receipt of your own messages to mailing lists... Andrew -- Linux supports the notion of a command line or a shell for the same reason that

Re: [asterisk-users] test

Ping. -- Sent from mobile device On Oct 24, 2009, at 8:33 PM, Matt mhop...@gmail.com wrote: This is a test... I am being told I am subscribed, but I am not getting messages. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com

Re: [asterisk-users] Test Function if SIP Device is Still Alive

Hello! Thank you for the information. Regarding using the sip show peers command, I remember somewhere seeing that it only works for static sip accounts and does not list accounts that are dynamically stored in a database. Most of my accounts are database entries, so would the sip show peers

Re: [asterisk-users] Test Function if SIP Device is Still Alive

Elliot Murdock schrieb: Regarding using the sip show peers command, I remember somewhere seeing that it only works for static sip accounts and does not list accounts that are dynamically stored in a database. Most of my accounts are database entries, so would the sip show peers command work?

Re: [asterisk-users] Test Function if SIP Device is Still Alive

Elliot Murdock schrieb: I am looking for a way to test if a SIP device is still alive or not. What about qualify=yes in sip.conf? I want to add this functionality in an AGI or independent script in order ensure all the SIP phones are properly connected to the system. Philipp Kempgen --

Re: [asterisk-users] Test Function if SIP Device is Still Alive

Hello Philipp, Thank you. I could set that up, but is that status (of qualifying) stored anywhere (besides the log files) that a script could use? Regards, Elliot On Thu, Jul 23, 2009 at 12:47 PM, Philipp Kempgenphilipp.kemp...@amooma.de wrote: Elliot Murdock schrieb: I am looking for a way

Re: [asterisk-users] Test Function if SIP Device is Still Alive

Elliot Murdock schrieb: I could set that up, but is that status (of qualifying) stored anywhere (besides the log files) that a script could use? You could have a script execute asterisk -rx 'sip show peers' and read the status for each peer. On Thu, Jul 23, 2009 at 12:47 PM, Philipp

Re: [asterisk-users] Test Function if SIP Device is Still Alive

Hi You can retrieve it in real time using the AMI from a script http://www.voip-info.org/wiki/view/Asterisk+manager+API Ish Elliot Murdock wrote: Hello Philipp, Thank you. I could set that up, but is that status (of qualifying) stored anywhere (besides the log files) that a script could

Re: [asterisk-users] test

Yes, its working :) Jai Rangi ww.didforsale.com On Thu, Apr 30, 2009 at 12:12 PM, James A. Shigley j...@answeringserv.comwrote: Had an inbound email server issue, just double checking it is working again. James Shigley *Monroe Telephone Answering Service* 409-981-9213** Infinity

Re: [asterisk-users] Test asterisk from behind my firewall [SOLVED]

On Mon, 16 Mar 2009 23:00:32 -0700 Michael Higgins li...@evolone.org wrote: I have an asterisk server at home. I'd like to test one just installed elsewhere. And did succeed just after emailing, of course. :( Sorry for the noise! -- |\ /|| | ~ ~ | \/ |

Re: [asterisk-users] test OpenVox B400P and junghans card for dahdi BRI wcb4xxp

On Tue, Nov 18, 2008 at 02:23:56PM +0800, lizhong zhu wrote: hello, all of users: after dig the code, i found that dahdi wcb4xxp is only for digium B410P. it has VPM hardware based echo cancellation, which  Junghans and openvox bri cards do not have. anyone can tell me how to disable the

Re: [asterisk-users] test OpenVox B400P and junghans card for dahdi BRI wcb4xxp

On Wed, Nov 12, 2008 at 05:55:29PM +0800, lizhong zhu wrote: the dmesg shows: wcb4xxp :02:02.0: ec_write: Wrote 0x20 to register 0x1ab of VPM 0 but got back 0x01 printk: 13709 messages suppressed. wcb4xxp :02:02.0: ec_write: Wrote 0x20 to register 0x1ab of VPM 0 but got back 0x01

Re: [asterisk-users] test call generator

If you have some time, interest and desire, I would like to see how FreeSwitch compares to the 9 calls per second lost SIP message issue. On Sun, Sep 28, 2008 at 2:55 AM, Gnu Devel [EMAIL PROTECTED] wrote: I'm using Sipp to load test, but it lost some SIP message when I increment Call Per

Re: [asterisk-users] test call generator

the admin Obviously hammer can't do that Sam -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Tuesday, September 30, 2008 4:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] test call

Re: [asterisk-users] test call generator

Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Tuesday, September 30, 2008 4:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] test call generator If you have some time, interest and desire, I would

Re: [asterisk-users] test call generator

Sipp looks pretty good! I don't know how I missed this one. This would've saved me tons of time a couple months ago. I plan on using it to load test using 2 Asterisk servers, one to initiate the SIP calls, the other to receive. Thanks for the tip Alex. Zac Wolfe Safi Systems LLC

Re: [asterisk-users] test call generator

I'm using Sipp to load test, but it lost some SIP message when I increment Call Per Second more than 9. Regards Grey Man escribió: I've used both the Hammer Call Analyzer software and als to the Hammer XMS system which is a server that they install in your rack to do the packet captures and

Re: [asterisk-users] test call generator

Are you looking for inbound or outbound. I can get you free inbound test DID. LMK Jai www.didforesale.com On 9/27/08, Sam Tam [EMAIL PROTECTED] wrote: Hello everyone I am trying to look for a free test call generator that will get me some stats like PDD, ASR and call quality etc on each

Re: [asterisk-users] test call generator

Sam Tam wrote: Hello everyone I am trying to look for a free test call generator that will get me some stats like PDD, ASR and call quality etc on each route. As well as do test at every interval too If you know something like this please enlighten me. Sam

Re: [asterisk-users] test call generator

On Sat, Sep 27, 2008 at 4:38 AM, Igor Hernandez [EMAIL PROTECTED] wrote: Sam Tam wrote: Hello everyone I am trying to look for a free test call generator that will get me some stats like PDD, ASR and call quality etc on each route. As well as do test at every interval too If

Re: [asterisk-users] test call generator

What you are looking for is SIPP: http://sipp.sourceforge.net/ It won't intrinsically tell you anything about the data; it's up to you to appropriate the findings. But it accomplishes the generation of traffic (and dummy media!) on a technical level. Igor Hernandez wrote: Sam Tam wrote:

Re: [asterisk-users] test call generator

You actually using that steve? Sam -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Saturday, September 27, 2008 6:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] test call generator

Re: [asterisk-users] test call generator

Unforunately it is outbound -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jai Rangi Sent: Saturday, September 27, 2008 4:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] test call generator Are you looking

Re: [asterisk-users] test call generator

-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Saturday, September 27, 2008 6:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] test call generator On Sat, Sep 27, 2008 at 4:38 AM, Igor Hernandez

Re: [asterisk-users] test call generator

I've used both the Hammer Call Analyzer software and als to the Hammer XMS system which is a server that they install in your rack to do the packet captures and provide you with all sorts of statistics. I suspect the Empirix Hammer products would be able to take care of any load, monitoring or

Re: [asterisk-users] test

On Tue, Feb 26, 2008 at 10:59 AM, Joel Solanki [EMAIL PROTECTED] wrote: checking wheather my mail goes to asterisk users mailling list or not ACK. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list

Re: [asterisk-users] test

Thanks, Joel On Tue, Feb 26, 2008 at 10:47 PM, Erik Anderson [EMAIL PROTECTED] wrote: On Tue, Feb 26, 2008 at 10:59 AM, Joel Solanki [EMAIL PROTECTED] wrote: checking wheather my mail goes to asterisk users mailling list or not ACK. ___ --

Re: [asterisk-users] test please ignore

Ian wrote: Just testing to see if my emails to this mailing list gets through. Tried posting a question, but it failed Thanks Ian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To

Re: [asterisk-users] test

I've been complaining about this problem recently, but nothing has been done about it. I'm guessing some spam filtering software has gone badly wrong. The filtering seems to be based on the content of the message rather than the sender. On Wed, Nov 28, 2007 at 03:02:11PM +0100, Suity Zsolt

Re: [asterisk-users] test

On 11/28/07, Jesse Molina [EMAIL PROTECTED] wrote: I've been complaining about this problem recently, but nothing has been done about it. I'm guessing some spam filtering software has gone badly wrong. The filtering seems to be based on the content of the message rather than the sender.

Re: [asterisk-users] test the email-list OT

Ok, so I was fooled :P On 8/12/07, Stephen Bosch [EMAIL PROTECTED] wrote: C F wrote: OMG, someone thought that it's for real. Wow. I don't think so. Read the sentence carefully: On 8/11/07, Trevor Peirce [EMAIL PROTECTED] wrote: C F wrote: No you cant. This message is being dropped as

Re: [asterisk-users] test the email-list OT

No you cant. This message is being dropped as well. On 8/10/07, Trevor Peirce [EMAIL PROTECTED] wrote: C F wrote: This is the postmaster at the list and I am notifying you that your message failed. Over the past two days my new posts seem to have silently been dropped. I wonder if I can

Re: [asterisk-users] test the email-list OT

C F wrote: No you cant. This message is being dropped as well. Shame. Seriously though I posted a new thread right after I posted that reply. The reply showed up but the new thread still seems to be MIA. No bounce or anything (and I have no filtering on this account). Weird... Maybe I'll

Re: [asterisk-users] test the email-list OT

OMG, someone thought that it's for real. Wow. On 8/11/07, Trevor Peirce [EMAIL PROTECTED] wrote: C F wrote: No you cant. This message is being dropped as well. Shame. Seriously though I posted a new thread right after I posted that reply. The reply showed up but the new thread still seems

Re: [asterisk-users] test the email-list OT

C F wrote: OMG, someone thought that it's for real. Wow. I don't think so. Read the sentence carefully: On 8/11/07, Trevor Peirce [EMAIL PROTECTED] wrote: C F wrote: No you cant. This message is being dropped as well. Shame. Seriously though I posted a new thread right after I posted that

Re: [asterisk-users] test the email-list OT

C F wrote: This is the postmaster at the list and I am notifying you that your message failed. Over the past two days my new posts seem to have silently been dropped. I wonder if I can reply to an existing thread... -- Does your Canadian VoIP service need CRTC-compliant 9-1-1 services?

Re: [asterisk-users] test the email-list

This is the postmaster at the list and I am notifying you that your message failed. On 8/7/07, zhu lizhong [EMAIL PROTECTED] wrote: test only. good luck! james.zhu ___ --Bandwidth and Colocation Provided by http://www.api-digital.com--

Re: [asterisk-users] Test

Alex Roston wrote: Is the list up? I haven't gotten mail in the last 24 hours. Alex ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit:

Re: [asterisk-users] test tools of Asterisk server

I don't know about bandwith consumption but look at sipp (http://sipp.sourceforge.net/) - Original Message - From: khawla khawla To: asterisk-users@lists.digium.com Sent: Saturday, May 26, 2007 10:33 PM Subject: [asterisk-users] test tools of Asterisk server I am using

Re: [asterisk-users] test tools of Asterisk server

HP has a tool that is a free Open Source test tool / traffic generator for the SIP protocol. On 5/26/07, khawla khawla [EMAIL PROTECTED] wrote: I am using Aserisk as a SIP server to interconnect differents PBX in differents sites. I am now looking for a tool that can test the performance of

Re: [asterisk-users] test tools of Asterisk server

HP's tool can be found at sipp.sf.net. Im unshure if you have to use unstable to get rtp support or if they hasve released it as stable. /M Andrew Joakimsen wrote: HP has a tool that is a free Open Source test tool / traffic generator for the SIP protocol. On 5/26/07, khawla khawla [EMAIL

Re: [asterisk-users] Test

, 2007 7:54 PM Subject: Re: [asterisk-users] Test Failed On 4/26/07, gc [EMAIL PROTECTED] wrote: ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http

Re: [asterisk-users] Test

Test emails and out of office emails make my day. - Original Message - From: Wilson Pickett [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, May 01, 2007 5:37 PM Subject: Re: [asterisk-users] Test where

Re: [asterisk-users] Test

I love these :) - Original Message - From: C F [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, April 27, 2007 7:54 PM Subject: Re: [asterisk-users] Test Failed On 4/26/07, gc [EMAIL PROTECTED] wrote

Re: [asterisk-users] Test

Failed On 4/26/07, gc [EMAIL PROTECTED] wrote: ___ --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] test

You failed. Try some brain dumps before attempting again. Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of gc Sent: Wednesday, April 25, 2007 10:10 AM To: asterisk-users@lists.digium.com Subject:

Re: [asterisk-users] test

Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, April 25, 2007 9:57:54 AM (GMT-0600) America/Chicago Subject: RE: [asterisk-users] test ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk

RE: [asterisk-users] test

Totaro [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk- [EMAIL PROTECTED] Sent: Wednesday, April 25, 2007 9:57:54 AM (GMT-0600) America/Chicago Subject: RE: [asterisk-users] test ___ --Bandwidth and Colocation

Re: [asterisk-users] test

ACK 2007/4/12, Razza [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 -- Alberto

Re: [asterisk-users] Test

Hi, I'm the admin on the list your test didin't work you should resend it. Well I am not the admin, just wanted you to realized that you shouldn't annoy thousands of people just

Re: [asterisk-users] Test

I DID receive it. Please don't re-send it. C F wrote: Hi, I'm the admin on the list your test didin't work you should resend it. Well I am not the admin, just wanted you to realized

Re: [asterisk-users] Test to Speech

Someone has worked with any test to speech software with aceptable quality in spanish? Probably in english the text to speech quality will be better. Witch test to speech software gave you the best results in spanish? Hi Andres, Check www.loquendo.com out... They have a nice web front

Re: [asterisk-users] Test to Speech

you gotta check festival site, it works for Spanish too through external language modules, On Feb 5, 2007, at 7:57 AM, voip crazy wrote: Hello all, I am looking for software for text to speech in spanish witch works with asterisk (1.2.13). I have tested festival and the cepstral software,

Re: [asterisk-users] Test Call Script

On Wed, Oct 11, 2006 at 05:56:12PM -0400, John Kane wrote: I am trying to write a script to attempt to make a call on a Zap channel, and if it fails, send an alarm. I can generate the call, but because the Zap channel accepts the call, even though the other end never answers, it sees it as a

Re: [asterisk-users] Test Call Script

on an analog Zap PSTN channel, you have no real way of determining if the remote side answered, because, as you discerned, it IS considered answered as soon as asterisk opens the channel. How about you contact another asterisk server through the PSTN, and dial through to an extension on that

Re: [asterisk-users] Test Call Script

yes.. actualy use 1 did for each proxy to check..then inbound for each use the method he described..On 10/12/06, Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote: on an analog Zap PSTN channel, you have no real way of determining ifthe remote side answered, because, as you discerned, it IS

Re: [asterisk-users] Test Call Script

I can think of a couple of ways to achieve testing of a PSTN line but this would seem to be the easiest. Attempt to call an incoming PSTN/SIP/IAX line from your outgoing PSTN trunk, answer the call at a vmail box and notify you of a message via email. insert a delay of x minutes and do it again.

Re: [asterisk-users] Test E1 channel

On 7/7/06, Moises Silva [EMAIL PROTECTED] wrote: One of the ends must be configured as pri_net and the other aspri_cpe. By the error I think the problem is with your configuration,does zttool says no alarms in spans?Post your configuration files zapata.conf and zaptel.confRegardsOn 7/7/06,

RE: [asterisk-users] Test E1 channel

When you dial directly you are bypassing the zap and just dialing an internal extension. So that is probably why dialing directly works. As far as the cross over cable between ports 1 and 2 I have never attempted something like that before. James Hawks -Original Message-

Re: [asterisk-users] Test E1 channel

I think you are really confused. I dont see a reason why dialing 555666 the call should go to client SIP/test. What you are doing is dialing to Zap channel 1 (whatever it is) the number 5556662, so, what do you have connected at the other end of the Zap/1 channel? On 7/7/06, Ralph Liebessohn

Re: [asterisk-users] Test E1 channel

Oops, i missed the crossover cable part. I have used crossover cable, so it should work, but the DNID must be complete. Wich signaling are you using? Regards On 7/7/06, James Hawks [EMAIL PROTECTED] wrote: When you dial directly you are bypassing the zap and just dialing an internal

Re: [asterisk-users] Test E1 channel

On 7/7/06, James Hawks [EMAIL PROTECTED] wrote: When you dial directly you are bypassing the zap and just dialing an internal extension. So that is probably why dialing directly works. As far as the cross over cable between ports 1 and 2 I have never attempted something like that

Re: [asterisk-users] Test E1 channel

On 7/7/06, Moises Silva [EMAIL PROTECTED] wrote: Oops, i missed the crossover cable part. I have used crossover cable,so it should work, butthe DNID must be complete. Wich signaling areyou using?RegardsHi Moises,I'm signalling=pri_net. I got this error too:app_dial.c: Unable to create channel of

Re: [asterisk-users] Test E1 channel

Newbie guess, Don't you need to set one of the ports NT mode and the other one as TE mode? hope it helps Best regards, PS. give me some feed back if it solved. On 7/7/06, Ralph Liebessohn [EMAIL PROTECTED] wrote: On 7/7/06, James Hawks [EMAIL PROTECTED] wrote: When you dial directly

Re: [asterisk-users] Test E1 channel

by Ports i mean Spans :) On 7/7/06, Marco Mouta [EMAIL PROTECTED] wrote: Newbie guess, Don't you need to set one of the ports NT mode and the other one as TE mode? hope it helps Best regards, PS. give me some feed back if it solved. On 7/7/06, Ralph Liebessohn [EMAIL PROTECTED] wrote: On

Re: [asterisk-users] Test E1 channel

One of the ends must be configured as pri_net and the other as pri_cpe. By the error I think the problem is with your configuration, does zttool says no alarms in spans? Post your configuration files zapata.conf and zaptel.conf Regards On 7/7/06, Marco Mouta [EMAIL PROTECTED] wrote: by Ports

Re: [asterisk-users] test tone

Hi Edwin, I'd check here: http://www.sandman.com/colookup.asp I couldn't find mine listed there, so I had to call the operator a couple of times until someone gave me the number to the local CO. A guy there was able to give me the milliwatt test # and the silence #. Cheers, Barry Edwin

Re: [asterisk-users] test tone

Outdated, but some of the info may still be current: http://www.tek-tips.com/viewthread.cfm?qid=583069 Edwin Lam wrote: hi folks. does anybody know what's the phone number for SBC Nothern California's 102-type milliwatt test line? (specifically in 415 area code) -- Now accepting new

Re: [Asterisk-Users] test numbers in different countries!

On Wednesday 26 April 2006 07:52, Jason Frisch wrote: How about using time announments? I list of these for each country would be great! I have some test numbers on my switch in Latvia: +371 7160201 -- echo +371 7160202 -- music :) +371 7160203 -- time Do you mean something like this?

RE: [Asterisk-Users] test numbers in different countries!

Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] test numbers in different countries! On Wednesday 26 April 2006 07:52, Jason Frisch wrote: How about using time announments? I list of these for each country would be great! I have some test numbers on my switch in Latvia

Re: [Asterisk-Users] test numbers in different countries!

On Wednesday 26 April 2006 11:48, Dmitry Ivanov wrote: On Wednesday 26 April 2006 07:52, Jason Frisch wrote: How about using time announments? I list of these for each country would be great! I have some test numbers on my switch in Latvia: +371 7160201 -- echo +371 7160202 -- music :)

Re: [Asterisk-Users] test numbers in different countries!

How about using time announments? I list of these for each country would be great! Jason Ronald Wiplinger wrote: Hi, --snip-- I would need some testing numbers in different countries. Testing numbers where a tape is or where a long company announcement is. Do you know such numbers?

Re: [Asterisk-Users] test call quality

amaury BOSSE wrote: Is there a free linux tool which can test voip call quality between two Asterisk PBX. It will help me to test the WAN network between them. I have only found commercials ones, so if you know a free one, let me know. For packet loss, rtt etc and a phone call check out:

Re: [Asterisk-Users] test call quality

amaury BOSSE schrieb: Is there a free linux tool which can test voip call quality between two Asterisk PBX. It will help me to test the WAN network between them. I have only found commercials ones, so if you know a free one, let me know. Hi, just some hours ago I published in this list:

Re: [Asterisk-Users] Test my test-branch!

Olle E Johansson wrote: Friends, The developer team for Asterisk not only consists of coders - a very important part are the testers, those that test new code and give feedback. For a few weeks, I've been maintaining a large number of branches with various stuff in them and have gotten

Re: [Asterisk-Users] Test Test

Test Your audio is garbled, but text seems to work fine. ;) ___ --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] Test to see if I'm still on list...

I was having problems too. Mine stopped at 5:19am MST this morning and just picked up a few minutes ago. Isn't the first time it's happened either. -Original Message- From: Francesco Peeters [mailto:[EMAIL PROTECTED] Sent: Monday, January 16, 2006 3:17 AM To:

Re: [Asterisk-Users] test

I am not sure the test modules work yet. PaulH - Original Message - From: RdBSD To: asterisk-users@lists.digium.com Sent: Monday, January 02, 2006 7:41 PM Subject: [Asterisk-Users] test Just Test

Re: [Asterisk-Users] Test numbers for ENUM (e164.arpa, e164.org, etc.)

www.enum-test.at (english translation is comming soon) klaus John Todd wrote: I'm looking to build a decent list of test numbers which have ENUM resolution. The numbers I'm looking for should go to a recording, an echo test, or some other feature which does NOT lead to a human. These

Re: [Asterisk-Users] Test numbers for ENUM (e164.arpa, e164.org, etc.)

On Tue, 22 Nov 2005, John Todd [EMAIL PROTECTED] wrote: I'm looking to build a decent list of test numbers which have ENUM resolution. The numbers I'm looking for should go to a recording, an echo test, or some other feature which does NOT lead to a human. These will be for manual or

Re: [Asterisk-Users] Test environment (Windows Softphone)

Assuming an XP or 2003 box, I use the free xlite client. Create a user for each instance that you want to run. Right click on the shortcut and select run as... enter the username and password of the account, setup the settings for the phone, and repeat the process for each additional

Re: [Asterisk-Users] Test environment for a Predictive Dialer

[EMAIL PROTECTED] wrote: I'm thinking about to set up a test environment for a predictive dialler with two asterisk machines. Each Asterisk should use a Digium TE110P card. One machine should work as predictive dialler; the other box should simulate the PSTN. - Is it in general possible to

Re: [Asterisk-Users] Test environment for a Predictive Dialer

Hello, Yes, we have this kind of test setup in our office using a quad T1 card in each of two servers and cross-connecting each T1 to the other to test VICIDIAL for performance and compatibility. A few notes, if you are going to use a PRI between the two servers make sure you set one to pri_cpe

Re: [Asterisk-Users] Test after Hurricane Wilma

What a creative way to test. GL. On 10/27/05, Waldo Rubinstein [EMAIL PROTECTED] wrote: Hi guys. Please disregard this. I'm testing connectivity after being down due to Hurricane Wilma. Thanks, Waldo ___ --Bandwidth and Colocation sponsored by

RE: [Asterisk-Users] test message - ignore me

Same here... Chris HARIGA -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Hess Sent: Monday, August 01, 2005 3:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] test message - ignore me Haven't seen email

Re: [Asterisk-Users] test message - ignore me

well over 10,000 users getting 80+ emails a day, it was bound to go down. I wonder how this ranks in the size of mailing lists. Other than LKML what other lists would be this size? On 8/1/05, Matt Hess [EMAIL PROTECTED] wrote: Haven't seen email since the 29th.. just testing.

RE: [Asterisk-Users] test message - ignore me

Me neither.. but just started receiving now. WEIRD. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Hess Sent: Monday, August 01, 2005 12:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] test message -

Re: [Asterisk-Users] test message - ignore me

You are duly ignored. Matt Hess wrote: Haven't seen email since the 29th.. just testing. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com

Re: [Asterisk-Users] Test CVS HEAD Voicemail ODBC Storage

Olle E. Johansson wrote: One of the cool new features in CVS head is the ability to store the actual voicemail messages in a database. This is not using ARA, the Asterisk Realtime Architecture, but directly interfaces with ODBC from app_voicemail. It stores both meta data and audio in the

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