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 list). 3. Stop sending (multiple) test e-mails to mailing lists. -- Herbert -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] test please ignore
my last few emails to this list haven't appeared so I'm just testing -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Test
TestSent from my Galaxy-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Test from Digium address
Testing again. -- Matthew Fredrickson Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] test delivery for lists.digium.com
Testing the new lists.digium.com server. Apologies for the email noise. Digium IT -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Test
Testing again :-) -- Matthew Fredrickson Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Test
Yes it is working -Ursprungligt meddelande- Från: asterisk-users-boun...@lists.digium.com <asterisk-users-boun...@lists.digium.com> För Matt Fredrickson Skickat: den 21 mars 2018 03:46 Till: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Ämne: [asterisk-users] Test Testing, 1, 2, 3. -- Matthew Fredrickson Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Test
Testing, 1, 2, 3. -- Matthew Fredrickson Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Test
This is a test. -- Matthew Fredrickson Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Test
Just a test, my apologies. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Test
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 for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 on unexpected message for Call-Id '19-12768@12... What I see at logs: 2011-06-28 14:32:57:6241309289577.624809: Aborting call on unexpected message for Call-Id '1-12768@127.0.0.1': while expecting '100' (index 1), received 'SIP/2.0 488 Not acceptable here^M Via: SIP/2.0/UDP 127.0.0.1:5061 ;branch=z9hG4bK-12768-1-0;received=192.168.25.253^M From: sipp sip:sipp@127.0.0.1:5061;tag=12768SIPpTag091^M To: sut sip:2005@192.168.1.18:5060;tag=as3614adc3^M Call-ID: 1-12768@127.0.0.1^M CSeq: 1 INVITE^M Server: Asterisk PBX 1.8.4.1^M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH^M Supported: replaces, timer^M Content-Length: 0^M This is my asterisk 1.8's configuration: *sip.conf* [sipp] type=friend context=sipp host=dynamic port=6000 user=sipp canreinvite=no disallow=all allow=ulaw * * *extensions.conf:* [sipp] exten = 2005,1,Answer same=n,Dial(SIP/intern,30) same=n,Hangup() exten = 2006,1,Answer() same= n,WaitMusicOnHold(4) same= n,Hangup() I'm using sipp.3.1.src.tar.gz and I have installed it this way: ..sip.svn# make pcapplay Thanks in advance. Elder On Thu, May 12, 2011 at 2:51 PM, Steve Totaro stot...@asteriskhelpdesk.comwrote: http://tinyurl.com/3hx5652 On Thu, May 12, 2011 at 11:52 AM, Daniel - Asterisk earohua...@gmail.comwrote: 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 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 www.safisystems.com On Sat, Sep 27, 2008 at 5:58 AM, Alex Balashov abalas...@evaristesys.com wrote: 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: 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com-- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hey Sam, I've been looking for such a tool also. I can't seem to find a tool that does those things. If nothing comes up in the next couple of weeks I'm going to code something up, I wouldn't mind letting you and anyone else who might be interested have the source once its done. Let me know if you find anything thats already out there in the meantime, might just save me a few hours of work. Regards, -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --
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 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 www.safisystems.com On Sat, Sep 27, 2008 at 5:58 AM, Alex Balashov abalas...@evaristesys.comwrote: 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: 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hey Sam, I've been looking for such a tool also. I can't seem to find a tool that does those things. If nothing comes up in the next couple of weeks I'm going to code something up, I wouldn't mind letting you and anyone else who might be interested have the source once its done. Let me know if you find anything thats already out there in the meantime, might just save me a few hours of work. Regards, -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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, Elder Arohuanca Lima - Peru On Tue, Sep 30, 2008 at 12:59 PM, zac wolfe zac.wo...@gmail.com mailto:zac.wo...@gmail.com wrote: 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 www.safisystems.com http://www.safisystems.com On Sat, Sep 27, 2008 at 5:58 AM, Alex Balashov abalas...@evaristesys.com mailto:abalas...@evaristesys.com wrote: 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: 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hey Sam, I've been looking for such a tool also. I can't seem to find a tool that does those things. If nothing comes up in the next couple of weeks I'm going to code something up, I wouldn't mind letting you and anyone else who might be interested have the source once its done. Let me know if you find anything thats already out there in the meantime, might just save me a few hours of work. Regards, -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- SJREIA South Jersey Real Estate Investors Association Want to invest in Real Estate? come out and join over 450 real estate investors http://www.SJREIA.org Licensed NJ Real Estate Agent Buy This House REALTORs david.cant...@ibsonecall.com Mobile (856)813-7098 Office (856)324-4488 Pers Fax (646)827-7108 Ofc Fax (888)487-7711 Interlocking Business Solutions, LLC david.cant...@ibsonecall.com (856)581-8971 Home of the Videophone2009.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit:
[asterisk-users] Test numbers Worldwide
Hi, We are searching for a pool of test numbers to call from Asterisk, record voice and test it with our non-intrusive voice quality testing software (NIQA). The problem is that we could find some test numbers, but our customer would like to have a global pool of test numbers, so that we can call them and test voice quality. Greatly appreciate any help! Thank you! Sevana Oy, Finland http://www.sevana.fi-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] test message
Looking to see if it shows up - thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] test
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 _ Hotmail: Free, trusted and rich email service. https://signup.live.com/signup.aspx?id=60969-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 -- Hotmail: Free, trusted and rich email service. Get it now.https://signup.live.com/signup.aspx?id=60969 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] test
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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: http://lists.digium.com/mailman/listinfo/asterisk-users
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 as client is the first he hears a background and then
if he inputs any number, asterisk executes system command like wget
example.org/?p=input number and call terminated.
I'm reading asteriskbook but can't connect all together right now.
I think you'll have to use the Local channel as your queue member, like
this (in queues.conf):
member = Local/s...@systemcommand
And then in your dialplan (extensions.conf) you'd have something like
this:
[systemcommand]
exten = s,1,Background(press-a-key)
exten = s,n,Read(INPUT_NUMBER)||1)
exten = s,n,System(wget http://example.org/?p=${INPUT_NUMBER})
exten = s,n,Playback(goodbye)
Please note, these are only examples to get you started, and they
probably won't work without some tuning.
A good resource to learn more about applications is
http://voip-info.org/. Here are a few links:
http://www.voip-info.org/wiki/view/Asterisk+-+documentation+of+application+commands
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+System
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Read
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Curl
You can also use 'core show application System' and such on the Asterisk
CLI.
GLHF!
--
C. Chad Wallace, B.Sc.
The Lodging Company
http://www.lodgingcompany.com/
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Re: [asterisk-users] test case with queues and system()
On Wed, Jan 20, 2010 at 10:18 PM, C. Chad Wallace
cwall...@lodgingcompany.com wrote:
At 5:59 PM on 19 Jan 2010, __ wrote:
Test case:
We have e1 trunk and multi-channel sip line. Clients waiting in the
queue, which can handle 30 clients. They listen mellody and their
position, while waiting. The system can handle only 5 clients at the
moment. As soon as client is the first he hears a background and then
if he inputs any number, asterisk executes system command like wget
example.org/?p=input number and call terminated.
I'm reading asteriskbook but can't connect all together right now.
I think you'll have to use the Local channel as your queue member, like
this (in queues.conf):
member = Local/s...@systemcommand
And then in your dialplan (extensions.conf) you'd have something like
this:
[systemcommand]
exten = s,1,Background(press-a-key)
exten = s,n,Read(INPUT_NUMBER)||1)
exten = s,n,System(wget http://example.org/?p=${INPUT_NUMBER})
exten = s,n,Playback(goodbye)
Please note, these are only examples to get you started, and they
probably won't work without some tuning.
Thank you, it helped a lot.
Now i have only one thing - how can i tell asterisk to work with 5
clients? I have to make 5 members?
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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 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 as client is the first he hears a background
and then if he inputs any number, asterisk executes system command
like wget example.org/?p=input number and call terminated.
I'm reading asteriskbook but can't connect all together right now.
I think you'll have to use the Local channel as your queue member,
like this (in queues.conf):
member = Local/s...@systemcommand
And then in your dialplan (extensions.conf) you'd have something
like this:
[systemcommand]
exten = s,1,Background(press-a-key)
exten = s,n,Read(INPUT_NUMBER)||1)
exten = s,n,System(wget http://example.org/?p=${INPUT_NUMBER})
exten = s,n,Playback(goodbye)
Please note, these are only examples to get you started, and they
probably won't work without some tuning.
Thank you, it helped a lot.
Now i have only one thing - how can i tell asterisk to work with 5
clients? I have to make 5 members?
Maybe... But I think the Local channel queue member will accept
multiple callers at the same time, so you could use GROUP_COUNT in your
dialplan to limit it:
[systemcommand]
exten = s,1,GotoIf($[${GROUP_COUNT(systemcommand)} 5]?continue)
exten = s,n,Busy()
exten = s,n(continue),Set(GROUP()=systemcommand)
exten = s,n,Background(press-a-key)
exten = s,n,Read(INPUT_NUMBER)||1)
exten = s,n,System(wget http://example.org/?p=${INPUT_NUMBER})
exten = s,n,Playback(goodbye)
It returns Busy if there are already 5 calls being serviced.
Also, you could replace the 5 above with a variable, and set that
variable in your globals, so it's easier to maintain later.
--
C. Chad Wallace, B.Sc.
The Lodging Company
http://www.lodgingcompany.com/
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[asterisk-users] test case with queues and system()
Hello, list. First of all i want to say sorry for my english. Long story short, on my future work i'll deal with asterisk and now i have a test case. But i'm very young to asterisk and don't have a lot of time so any help is appreciated. 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 as client is the first he hears a background and then if he inputs any number, asterisk executes system command like wget example.org/?p=input number and call terminated. I'm reading asteriskbook but can't connect all together right now. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] test request for new event Pickup when a call is picked up from an other phone
A new patch has been made for an extra Manager Event when a call-pickup has occurred. There are two possible situations 1) by using *8 2) by using *8123 (to pickup extension 123 when it is ringing) The manager event looks like: Event: Pickup Privilege: call,all Channel: SIP/ast163-000c UniqueID: astium-21-1261065321.12 TargetChannel: SIP/ast165-000b TargetUniqueID: astium-21-1261065314.11 the patch is described in: https://issues.asterisk.org/view.php?id=16431 Ideally, if you can get someone from the asterisk-dev and/or asterisk-users mailing lists to test this out and report back here, that would be fantastic! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 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 only children read books with only pictures in them. Language, be it English or something else, is the only tool flexible enough to accomplish a sufficiently broad range of tasks. -- Bill Garrett ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] test
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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 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 command work? Thanks, Elliot On Thu, Jul 23, 2009 at 5:08 PM, Ishfaq Maliki...@pack-net.co.uk wrote: 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 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 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 -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 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? Yes it does work, at least with rtcachefriends=yes in sip.conf. Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Test Function if SIP Device is Still Alive
Hello! I am looking for a way to test if a SIP device is still alive or not. 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. Thank you, Elliot ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 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 -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 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 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 -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 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 Kempgenphilipp.kemp...@amooma.de wrote: 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 -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 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 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 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 -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] test
Had an inbound email server issue, just double checking it is working again. James Shigley Monroe Telephone Answering Service 409-981-9213 Infinity 5.5,UC 4.02.3803, Blink 3.0.104 Ecreator:2.21, eResponse 1.1.7 Webportal,WebApps, CONFIDENTIALITY NOTICE: This email, including any attachments, contains information which may be confidential or privileged. The information is intended to be for the use of the individual or entity named above. If you are not the intended recipient, be aware that any disclosure, copying, distribution or use of the contents of this information is prohibited. If you have received this email in error, please notify the sender immediately by reply to sender only message and destroy all electronic and hard copies of the communication, including attachments. Common sense is the collection of prejudices acquired by age eighteen. -- Albert Einstein Once you can accept the universe as matter expanding into nothing that is something,wearing stripes with plaid comes easy. -- Albert Einstein I know a little of everything, but a lot of nothing ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 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 5.5,UC 4.02.3803, Blink 3.0.104 Ecreator:2.21, eResponse 1.1.7 Webportal,WebApps, CONFIDENTIALITY NOTICE: This email, including any attachments, contains information which may be confidential or privileged. The information is intended to be for the use of the individual or entity named above. If you are not the intended recipient, be aware that any disclosure, copying, distribution or use of the contents of this information is prohibited. If you have received this email in error, please notify the sender immediately by reply to sender only message and destroy all electronic and hard copies of the communication, including attachments. Common sense is the collection of prejudices acquired by age eighteen. -- Albert Einstein Once you can accept the universe as matter expanding into nothing that is something,wearing stripes with plaid comes easy. -- Albert Einstein I know a little of everything, but a lot of nothing ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Test asterisk from behind my firewall
I have an asterisk server at home. I'd like to test one just installed elsewhere. Both servers are behind firewalls. I can see the session start in CLI, my congratulations is apparently playing and RTP is being sent. Hearing no audio. Can send key presses and see audio playing changed. Peer audio RTP is at port 198.145.28.177:10180, but that never shows at the client side, behind a linksys wrt54g, ver 1. w/ latest firmware update. My belief is this should be possible, as the SIP phone is registered to my asterisk box inside my home network, asterisk should stay in the middle and forward the RTP packets to my laptop... am I totally off base? If so, what are some key elements to make that happen? I'll stop now, before I get ignored for being too verbose. '-) Cheers, -- |\ /|| | ~ ~ | \/ ||---| `|` ? ||ichael | |iggins\^ / michael.higgins[at]evolone[dot]org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 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! -- |\ /|| | ~ ~ | \/ ||---| `|` ? ||ichael | |iggins\^ / michael.higgins[at]evolone[dot]org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] test
test ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 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 ec_write methond to support other HFC BRI cards? regards! zhu My basic work in progress is here: http://bugs.digium.com/view.php?id=13897 Please submit your patches. Please also use latest svn (or 2.1.0-rc4) as it seems to include a number of other fixes (in the D-channel handling) BTW: keeping to one thread can help others follow this. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] test OpenVox B400P and junghans card for dahdi BRI wcb4xxp
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 ec_write methond to support other HFC BRI cards? regards! zhu ___ 好玩贺卡等你发，邮箱贺卡全新上线！ http://card.mail.cn.yahoo.com/___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 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 printk: 13708 messages suppressed. wcb4xxp :02:02.0: ec_write: Wrote 0x20 to register 0x1ab of VPM 0 but got back 0x01 printk: 13708 messages suppressed. 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 printk: 13708 messages suppressed. wcb4xxp :02:02.0: ec_write: Wrote 0x20 to register 0x1ab of VPM 0 but got back 0x01 printk: 13705 messages suppressed. wcb4xxp :02:02.0: ec_write: Wrote 0x64 to register 0x1ab of VPM 0 but got back 0x01 I noticed that this is called (1000 times per second) even with vpm_support=0 . system.conf: loadzone=us defaultzone=us span=1,1,3,ccs,ami span=2,2,3,ccs,ami span=3,3,3,ccs,ami span=4,4,3,ccs,ami bchan=1,2 dchan=3 bchan=4,5 dchan=6 bchan=7,8 dchan=9 bchan=10,11 dchan=12 It's not dchan . It's 'hardhdlc' . See the sample system.conf: http://docs.tzafrir.org.il/dahdi-tools/#_channel_configuration You can also set: bri_hardhdlc yes in /etc/dahdi/genconf_parameters (which instructs dahdi_genconf to write there 'hardhdlc'). I figure it would become the default at some point. == chan_dahdi.conf: [channels] ; ; Default language ; ;language=en ; ; Default context ; ; switchtype = euroisdn ; p2mp TE mode (for connecting ISDN lines in point-to-multipoint mode) signalling = bri_cpe_ptmp ; p2p TE mode (for connecting ISDN lines in point-to-point mode) ;signalling = bri_cpe ; p2mp NT mode (for connecting ISDN phones in point-to-multipoint mode) ;signalling = bri_net_ptmp ; p2p NT mode (for connecting an ISDN pbx in point-to-point mode) ;signalling = bri_net pridialplan = local prilocaldialplan = dynamic nationalprefix = 0 internationalprefix = 00 priindication = passthrough echocancel = yes context=demo group = 1 ; S/T port 1 channel = 1-2 group = 2 ; S/T port 2 channel = 4-5 group = 3 ; S/T port 3 channel = 7-8 group = 4 ; S/T port 4 channel = 10-11 anyone knows that? no incoming calls and only three leds are on. i will make further study on that. thanks! james.zhu 'pri debug' ? 'pri intense debug' -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] test OpenVox B400P and junghans card for dahdi BRI wcb4xxp
hello:
thanks for Tzafrir Cohen for dahdi testing.
I installed dahdi-2.1-r3c svn code and asterisk1-6
for testing OpenVox B400P and junghans card. i fund that there is bug (i think)
to dectect NT or TE mode. actually on the board,
i set it as TE mode, but after start wcb4xxp, but
it show the port is NT mode. to detect the TE mode, I modefy the code in base.c
static void hfc_init_all_st(struct b4xxp *b4)
1386 {
1387 int i, gpio, nt;
1388 struct b4xxp_span *s;
1389
1390 gpio = b4xxp_getreg8(b4, R_GPI_IN3);
1391
1392 for (i=0; i 4; i++) {
1393 s = b4-spans[i];
1394 s-parent = b4;
1395 s-port = i;
1396
1397 nt = ((gpio (1 (i + 4))) != 0); /*
GPIO=0 = NT mode change ==0 to !=0 */
1398 s-te_mode = !nt;
1399
1400 dev_info(b4-dev, Port %d: %s mode\n, i + 1, (nt ?
NT : TE));
1401
1402 hfc_reset_st(s);
1403 hfc_start_st(s);
1404 }
1405 }
beside that, i stil can not make calls. the driver starts up and loaded into
asterisk, run the command: misdn show status:
*CLI dahdi show status
Description Alarms IRQ bpviol CRC4 Fra Codi
Options LBO
B4XXP (PCI) Card 0 Span 1 RED 0 0 0 CCS AMI
YEL 399-533 feet (DSX-1)
B4XXP (PCI) Card 0 Span 2 RED 0 0 0 CCS AMI
YEL 399-533 feet (DSX-1)
B4XXP (PCI) Card 0 Span 3 OK 0 0 0 CCS AMI
YEL 399-533 feet (DSX-1)
B4XXP (PCI) Card 0 Span 4 RED 0 0 0 CCS AMI
YEL 399-533 feet (DSX-1)
*CLI
==
i add more PCI for wcb4xxp:
==
static struct pci_device_id b4xx_ids[] __devinitdata =
2625 {
2626 { 0xd161, 0xb410, PCI_ANY_ID, PCI_ANY_ID, 0, 0, (unsigned
long)wcb4xxp },
2627 {0x1397, 0x08b4, PCI_ANY_ID,PCI_ANY_ID,0,0,(unsigned
long)wcb4xxp},
2628 {0x1397, 0xe888, PCI_ANY_ID,PCI_ANY_ID,0,0,(unsigned
long)wcb4xxp},
2629
2630 { 0, }
2631 };
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
printk: 13708 messages suppressed.
wcb4xxp :02:02.0: ec_write: Wrote 0x20 to register 0x1ab of VPM 0 but got
back 0x01
printk: 13708 messages suppressed.
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
printk: 13708 messages suppressed.
wcb4xxp :02:02.0: ec_write: Wrote 0x20 to register 0x1ab of VPM 0 but got
back 0x01
printk: 13705 messages suppressed.
wcb4xxp :02:02.0: ec_write: Wrote 0x64 to register 0x1ab of VPM 0 but got
back 0x01
system.conf:
loadzone=us
defaultzone=us
span=1,1,3,ccs,ami
span=2,2,3,ccs,ami
span=3,3,3,ccs,ami
span=4,4,3,ccs,ami
bchan=1,2
dchan=3
bchan=4,5
dchan=6
bchan=7,8
dchan=9
bchan=10,11
dchan=12
==
chan_dahdi.conf:
[channels]
;
; Default language
;
;language=en
;
; Default context
;
;
switchtype = euroisdn
; p2mp TE mode (for connecting ISDN lines in point-to-multipoint mode)
signalling = bri_cpe_ptmp
; p2p TE mode (for connecting ISDN lines in point-to-point mode)
;signalling = bri_cpe
; p2mp NT mode (for connecting ISDN phones in point-to-multipoint mode)
;signalling = bri_net_ptmp
; p2p NT mode (for connecting an ISDN pbx in point-to-point mode)
;signalling = bri_net
pridialplan = local
prilocaldialplan = dynamic
nationalprefix = 0
internationalprefix = 00
priindication = passthrough
echocancel = yes
context=demo
group = 1
; S/T port 1
channel = 1-2
group = 2
; S/T port 2
channel = 4-5
group = 3
; S/T port 3
channel = 7-8
group = 4
; S/T port 4
channel = 10-11
anyone knows that? no incoming calls and only three leds are on. i will make
further study on that.
thanks!
james.zhu
___
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[asterisk-users] test OpenVox B400P and junghans card for dahdi BRI wcb4xxp
hello, users: I tried to change to hardhdlc in system. but i still can not make calls. the port 4 led still can be be on. =system.conf== # Autogenerated by ./dahdi_genconf on Wed Nov 12 19:22:36 2008 -- do not hand edit # Dahdi Configuration File # # This file is parsed by the Dahdi Configurator, dahdi_cfg # # Global data loadzone = us defaultzone = us span=1,1,3,ccs,ami span=2,2,3,ccs,ami span=3,3,3,ccs,ami span=4,4,3,ccs,ami bchan=1,2 hardhdlc=3 bchan=4,5 hardhdlc=6 bchan=7,8 hardhdlc=9 bchan=10,11 hardhdlc=12 =chan_dahdi.conf=== [channels] ; ; Default language ; ;language=en ; ; Default context ; ; switchtype = euroisdn ; p2mp TE mode (for connecting ISDN lines in point-to-multipoint mode) signalling = bri_cpe_ptmp ; p2p TE mode (for connecting ISDN lines in point-to-point mode) ;signalling = bri_cpe ; p2mp NT mode (for connecting ISDN phones in point-to-multipoint mode) ;signalling = bri_net_ptmp ; p2p NT mode (for connecting an ISDN pbx in point-to-point mode) ;signalling = bri_net pridialplan = local prilocaldialplan = dynamic nationalprefix = 0 internationalprefix = 00 priindication = passthrough echocancel = yes context=demo group = 1 ; S/T port 1 channel = 1-2 group = 2 ; S/T port 2 channel = 4-5 group = 3 ; S/T port 3 channel = 7-8 group = 4 ; S/T port 4 channel = 10-11 dmesg ACPI: PCI Interrupt :02:02.0[A] - GSI 22 (level, low) - IRQ 217 wcb4xxp :02:02.0: Identified Wildcard B410P (controller rev 1) at 0001a000, IRQ 217 wcb4xxp :02:02.0: VPM 0/1 init: chip ver 01 printk: 5 messages suppressed. wcb4xxp :02:02.0: ec_write: Wrote 0x00 to register 0x1a8 of VPM 0 but got back 0x01 wcb4xxp :02:02.0: ec_write: Wrote 0x00 to register 0x1ac of VPM 0 but got back 0x01 wcb4xxp :02:02.0: ec_write: Wrote 0x00 to register 0x1b0 of VPM 0 but got back 0x01 wcb4xxp :02:02.0: ec_write: Wrote 0x00 to register 0x1a9 of VPM 0 but got back 0x01 wcb4xxp :02:02.0: VPM 1/1 init: chip ver 01 wcb4xxp :02:02.0: NOTE: hardware echo cancellation has been disabled wcb4xxp :02:02.0: Port 1: TE mode wcb4xxp :02:02.0: Port 2: TE mode wcb4xxp :02:02.0: Port 3: TE mode wcb4xxp :02:02.0: Port 4: TE mode wcb4xxp :02:02.0: Did not do the highestorder stuff wcb4xxp :02:02.0: Reconfigured channel 1 (B4/0/1/1) sigtype 0080 wcb4xxp :02:02.0: Reconfigured channel 2 (B4/0/1/2) sigtype 0080 wcb4xxp :02:02.0: Reconfigured channel 3 (B4/0/1/3) sigtype 00080080 wcb4xxp :02:02.0: Configuring hardware HDLC on B4/0/1/3 wcb4xxp :02:02.0: Reconfigured channel 4 (B4/0/2/1) sigtype 0080 wcb4xxp :02:02.0: Reconfigured channel 5 (B4/0/2/2) sigtype 0080 wcb4xxp :02:02.0: Reconfigured channel 6 (B4/0/2/3) sigtype 00080080 wcb4xxp :02:02.0: Configuring hardware HDLC on B4/0/2/3 wcb4xxp :02:02.0: Reconfigured channel 7 (B4/0/3/1) sigtype 0080 wcb4xxp :02:02.0: Reconfigured channel 8 (B4/0/3/2) sigtype 0080 wcb4xxp :02:02.0: Reconfigured channel 9 (B4/0/3/3) sigtype 00080080 wcb4xxp :02:02.0: Configuring hardware HDLC on B4/0/3/3 wcb4xxp :02:02.0: Reconfigured channel 10 (B4/0/4/1) sigtype 0080 wcb4xxp :02:02.0: Reconfigured channel 11 (B4/0/4/2) sigtype 0080 wcb4xxp :02:02.0: Reconfigured channel 12 (B4/0/4/3) sigtype 00080080 wcb4xxp :02:02.0: Configuring hardware HDLC on B4/0/4/3 dahdi: Registered tone zone 0 (United States / North America) wcb4xxp :02:02.0: open() on chan B4/0/1/1 (1/1) wcb4xxp :02:02.0: close() on chan B4/0/1/1 (1/1) wcb4xxp :02:02.0: open() on chan B4/0/1/2 (2/2) wcb4xxp :02:02.0: close() on chan B4/0/1/2 (2/2) wcb4xxp :02:02.0: open() on chan B4/0/1/3 (3/3) wcb4xxp :02:02.0: close() on chan B4/0/1/3 (3/3) wcb4xxp :02:02.0: open() on chan B4/0/2/1 (4/1) wcb4xxp :02:02.0: close() on chan B4/0/2/1 (4/1) wcb4xxp :02:02.0: open() on chan B4/0/2/2 (5/2) wcb4xxp :02:02.0: close() on chan B4/0/2/2 (5/2) wcb4xxp :02:02.0: open() on chan B4/0/2/3 (6/3) wcb4xxp :02:02.0: close() on chan B4/0/2/3 (6/3) wcb4xxp :02:02.0: open() on chan B4/0/3/1 (7/1) wcb4xxp :02:02.0: close() on chan B4/0/3/1 (7/1) any idea for that? thanks! james ___ 雅虎邮箱，您的终生邮箱！ http://cn.mail.yahoo.com/___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 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 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 provide you with all sorts of statistics. I suspect the Empirix Hammer products would be able to take care of any load, monitoring or analysis scenarios you have including signalling and media. The price is going to be the issue. When we looked at the solution the Call Analyzer software was 5 figures and the XMS solution was 6. Regards, Greyman. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG - http://www.avg.com Version: 8.0.169 / Virus Database: 270.7.3/1694 - Release Date: 26/09/2008 06:55 p.m. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks, Steve Totaro 1.888.777.1888 1.240.938.1212 (cell) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] test call generator
I looked at the hammer thing. It is quite complicate and quite useless too All I want is something that will dial a list of number in schedule per hr or per 3 hours Collect the PDD, ASR and comparing it with other route and determine which is the best. If the call does not pass through then alert 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 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 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 provide you with all sorts of statistics. I suspect the Empirix Hammer products would be able to take care of any load, monitoring or analysis scenarios you have including signalling and media. The price is going to be the issue. When we looked at the solution the Call Analyzer software was 5 figures and the XMS solution was 6. Regards, Greyman. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG - http://www.avg.com Version: 8.0.169 / Virus Database: 270.7.3/1694 - Release Date: 26/09/2008 06:55 p.m. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks, Steve Totaro 1.888.777.1888 1.240.938.1212 (cell) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] test call generator
You can't touch this. Anyways, I am sure I could do it with Hammer, but a tool is just that, use a screwdriver if you feel a Hammer is too complicated for you. On Tue, Sep 30, 2008 at 5:25 AM, Sam Tam [EMAIL PROTECTED] wrote: I looked at the hammer thing. It is quite complicate and quite useless too All I want is something that will dial a list of number in schedule per hr or per 3 hours Collect the PDD, ASR and comparing it with other route and determine which is the best. If the call does not pass through then alert 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 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 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 provide you with all sorts of statistics. I suspect the Empirix Hammer products would be able to take care of any load, monitoring or analysis scenarios you have including signalling and media. The price is going to be the issue. When we looked at the solution the Call Analyzer software was 5 figures and the XMS solution was 6. Regards, Greyman. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG - http://www.avg.com Version: 8.0.169 / Virus Database: 270.7.3/1694 - Release Date: 26/09/2008 06:55 p.m. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks, Steve Totaro 1.888.777.1888 1.240.938.1212 (cell) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks, Steve Totaro 1.888.777.1888 1.240.938.1212 (cell) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 www.safisystems.com On Sat, Sep 27, 2008 at 5:58 AM, Alex Balashov [EMAIL PROTECTED]wrote: 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: 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hey Sam, I've been looking for such a tool also. I can't seem to find a tool that does those things. If nothing comes up in the next couple of weeks I'm going to code something up, I wouldn't mind letting you and anyone else who might be interested have the source once its done. Let me know if you find anything thats already out there in the meantime, might just save me a few hours of work. Regards, -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 provide you with all sorts of statistics. I suspect the Empirix Hammer products would be able to take care of any load, monitoring or analysis scenarios you have including signalling and media. The price is going to be the issue. When we looked at the solution the Call Analyzer software was 5 figures and the XMS solution was 6. Regards, Greyman. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG - http://www.avg.com Version: 8.0.169 / Virus Database: 270.7.3/1694 - Release Date: 26/09/2008 06:55 p.m. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] test call generator
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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 route. As well as do test at every interval too If you know something like this please enlighten me. Sam -- Sent from Gmail for mobile | mobile.google.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hey Sam, I've been looking for such a tool also. I can't seem to find a tool that does those things. If nothing comes up in the next couple of weeks I'm going to code something up, I wouldn't mind letting you and anyone else who might be interested have the source once its done. Let me know if you find anything thats already out there in the meantime, might just save me a few hours of work. Regards, -- Igor Hernandez Escape Communications http://www.escapetel.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 you know something like this please enlighten me. Sam Hey Sam, I've been looking for such a tool also. I can't seem to find a tool that does those things. If nothing comes up in the next couple of weeks I'm going to code something up, I wouldn't mind letting you and anyone else who might be interested have the source once its done. Let me know if you find anything thats already out there in the meantime, might just save me a few hours of work. Regards, -- Igor Hernandez Escape Communications http://www.escapetel.com It's not free but if you want some good ideas for features, or don't mind paying, there is the Empirix Hammer. http://www.empirix.com/ Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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: 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hey Sam, I've been looking for such a tool also. I can't seem to find a tool that does those things. If nothing comes up in the next couple of weeks I'm going to code something up, I wouldn't mind letting you and anyone else who might be interested have the source once its done. Let me know if you find anything thats already out there in the meantime, might just save me a few hours of work. Regards, -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 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 you know something like this please enlighten me. Sam Hey Sam, I've been looking for such a tool also. I can't seem to find a tool that does those things. If nothing comes up in the next couple of weeks I'm going to code something up, I wouldn't mind letting you and anyone else who might be interested have the source once its done. Let me know if you find anything thats already out there in the meantime, might just save me a few hours of work. Regards, -- Igor Hernandez Escape Communications http://www.escapetel.com It's not free but if you want some good ideas for features, or don't mind paying, there is the Empirix Hammer. http://www.empirix.com/ Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 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 route. As well as do test at every interval too If you know something like this please enlighten me. Sam -- Sent from Gmail for mobile | mobile.google.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] test call generator
There is no reason that outbound cannot also be inbound.. mind wandering to the mobius strip I am not using it but I do have plans to shortly. I think if you want any kind of real testing and validation, then a product like this is almost required. As Alex noted, you could use SIPp, you could also use originate, .call files, and other methods, but do you get anything useful except some info from top and maybe a self monitored call or two? Thanks, Steve Totaro On Sat, Sep 27, 2008 at 12:39 PM, Sam Tam [EMAIL PROTECTED] wrote: 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 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 you know something like this please enlighten me. Sam Hey Sam, I've been looking for such a tool also. I can't seem to find a tool that does those things. If nothing comes up in the next couple of weeks I'm going to code something up, I wouldn't mind letting you and anyone else who might be interested have the source once its done. Let me know if you find anything thats already out there in the meantime, might just save me a few hours of work. Regards, -- Igor Hernandez Escape Communications http://www.escapetel.com It's not free but if you want some good ideas for features, or don't mind paying, there is the Empirix Hammer. http://www.empirix.com/ Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 analysis scenarios you have including signalling and media. The price is going to be the issue. When we looked at the solution the Call Analyzer software was 5 figures and the XMS solution was 6. Regards, Greyman. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] test message please do not reply and clog up the list
___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TEST MAIL
sorry just a testmail to the list, becausemy last mail does not show up on the list. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Test
Just a test, please discard Looks like something is eating my messages on their way :-( Martin -- http://mujblog.atlas.cz/___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] test please ignore
___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Test - Please Ignore
My posts were not going thru, so I testing and debugging why. Please ignore ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] test
checking wheather my mail goes to asterisk users mailling list or not ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Test
Test Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 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 UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I as well have been having rotten luck lately with the mailing list. Replies to questions get axed. New questions get axed. I mailed the list admin, and never got a reply... for all I know, the mail I sent got axed. N. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] test please ignore
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 UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] test
Sorry, but it seems that I have banned from list. I can reciveve, but can not send posts. Hi! When I use Dial(type/identifier, timeout, A(some_file)) CDR billsec starts when announcement ends. But I have to bill from when called party answers to phone. How can I solve my problem? -- Suich ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 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 wrote: Sorry, but it seems that I have banned from list. I can reciveve, but can not send posts. Hi! When I use Dial(type/identifier, timeout, A(some_file)) CDR billsec starts when announcement ends. But I have to bill from when called party answers to phone. How can I solve my problem? -- Suich ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- # Jesse Molina # Mail = [EMAIL PROTECTED] # Page = [EMAIL PROTECTED] # Cell = 1.602.323.7608 # Web = http://www.opendreams.net/jesse/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 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. I've experienced the same prolem. I was trying to send mail to asterisk-dev some 10 times (from address i'm subscribed, using GMail's SMTP), but without any bounces and any success.. Then i tried sending from GMail's web interface - and it worked. Regards, Atis On Wed, Nov 28, 2007 at 03:02:11PM +0100, Suity Zsolt wrote: Sorry, but it seems that I have banned from list. I can reciveve, but can not send posts. Hi! When I use Dial(type/identifier, timeout, A(some_file)) CDR billsec starts when announcement ends. But I have to bill from when called party answers to phone. How can I solve my problem? -- Suich ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- # Jesse Molina # Mail = [EMAIL PROTECTED] # Page = [EMAIL PROTECTED] # Cell = 1.602.323.7608 # Web = http://www.opendreams.net/jesse/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Test - pls ignore
Test ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] test list
test list not working ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 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 well. Shame. Seriously though I posted a new thread right after I posted that He got the joke. -Stephen- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 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 reply to an existing thread... -- Does your Canadian VoIP service need CRTC-compliant 9-1-1 services? Please visit http://www.digitalcon.ca/voip9-1-1/ to find out more! ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 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 try posting it to the -dev list since it is a suggestion for asterisk documentation tweaks in regards to the HPEC. -- Does your Canadian VoIP service need CRTC-compliant 9-1-1 services? Please visit http://www.digitalcon.ca/voip9-1-1/ to find out more! ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 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 to be MIA. No bounce or anything (and I have no filtering on this account). Weird... Maybe I'll try posting it to the -dev list since it is a suggestion for asterisk documentation tweaks in regards to the HPEC. -- Does your Canadian VoIP service need CRTC-compliant 9-1-1 services? Please visit http://www.digitalcon.ca/voip9-1-1/ to find out more! ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 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 He got the joke. -Stephen- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 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? Please visit http://www.digitalcon.ca/voip9-1-1/ to find out more! ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] test the email-list
test only. good luck! james.zhu ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Test
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: http://lists.digium.com/mailman/listinfo/asterisk-users
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: http://lists.digium.com/mailman/listinfo/asterisk-users The list was unreachable to me for hours yesterday. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Test Message
Sorry to clutter up the mailiing list, but I've been unable to post to this list for the past 2 WEEKS! My ISP's blocking SMPT from other than his own servers. I think I've worked around it. - But if I see this message in the digest then I know I'm okay. Again. - Sorry for any inconvenience. Gary Guthary ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 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 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 this solution: I mean is there a tool that enables me to test the capacity of this SIP server in terms of simultaneous calls that could be treated, the comsuption of bandwidth.. or any thing like this? I am in urgent need to such a tool, If anyone could help, I would be geatful. -- Appelez vos amis de PC à PC -- C'EST GRATUIT Essayez-le maintenant ! -- ___ --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] test tools of Asterisk server
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 this solution: I mean is there a tool that enables me to test the capacity of this SIP server in terms of simultaneous calls that could be treated, the comsuption of bandwidth.. or any thing like this? I am in urgent need to such a tool, If anyone could help, I would be geatful. _ Appelez vos amis de PC à PC -- C'EST GRATUIT http://get.live.com/messenger/overview___ --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 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 this solution: I mean is there a tool that enables me to test the capacity of this SIP server in terms of simultaneous calls that could be treated, the comsuption of bandwidth.. or any thing like this? I am in urgent need to such a tool, If anyone could help, I would be geatful. Appelez vos amis de PC à PC -- C'EST GRATUIT Essayez-le maintenant ! ___ --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] 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 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 this solution: I mean is there a tool that enables me to test the capacity of this SIP server in terms of simultaneous calls that could be treated, the comsuption of bandwidth.. or any thing like this? I am in urgent need to such a tool, If anyone could help, I would be geatful. This will help you for 99.9% of your problems: echo '16i[q]sa[ln0=aln100%Pln100/snlbx]sbA0D4D465452snlbxq' | dc ___ --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
where are the out of office replies when they're needed? On 4/30/07, Dovid B [EMAIL PROTECTED] wrote: 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: ___ --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 visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 are the out of office replies when they're needed? On 4/30/07, Dovid B [EMAIL PROTECTED] wrote: 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: ___ --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 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] 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: ___ --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] 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 ___ --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] Test
___ --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] test
ggcc___ --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: [asterisk-users] test ggcc ___ --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
Ha This does not directly relate, but I have NO respect for people who use braindumps. Learn the material, do not be a paper certification name here. Just my 2 cents, sorry, had to get that out. :P Cheers, Bkruse - Original Message - From: Steve Totaro [EMAIL PROTECTED] To: Asterisk 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-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] test
I do not even consider certs when evaluating someone's ability. If you want certs, I have no problem with brain dumps since the material may or may not be the knowledge needed in the field. Experience and a hypothetical, how would you implement this? usually tells me all I need to know. Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Brandon Kruse Sent: Wednesday, April 25, 2007 12:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] test Ha This does not directly relate, but I have NO respect for people who use braindumps. Learn the material, do not be a paper certification name here. Just my 2 cents, sorry, had to get that out. :P Cheers, Bkruse - Original Message - From: Steve 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 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
