[asterisk-users] Fwd: [CFP] FOSDEM 2020, RTC devroom, speakers, volunteers neeeded
FOSDEM - Real Time Communications devroom CfP = Overview [FOSDEM](https://fosdem.org) is one of the world's premier meetings of free software developers, with over five thousand people attending each year. FOSDEM 2020 takes place 1-2 February 2020 in Brussels, Belgium. This document contains information about: - Real-Time Communications developer room (devroom) and lounge - speaking opportunities - volunteering in the devroom and lounge - social events (the legendary FOSDEM Beer Night and Saturday night dinners provide endless networking opportunities) - the Planet aggregation sites for RTC blogs **NEW:** Save yourself entering Free-RTC events and CFP deadlines into your calendar and task list, follow our iCalendar feed: https://freertc.org/events.ics Call for participation - Real Time Communications (RTC) --- The Real-Time devroom and Real-Time lounge are about all things involving real-time communication, including: XMPP, SIP, WebRTC, telephony, mobile VoIP, codecs, peer-to-peer, privacy and encryption. **We are looking for speakers for the devroom and volunteers and participants for the tables in the Real-Time lounge.** The devroom is only on Sunday, 2nd of February 2020. The lounge will be present for both days. To discuss the devroom and lounge, please join the [Free-RTC mailing list](http://lists.freertc.org/mailman/listinfo/discuss). ### Speaking opportunities Note: if you used FOSDEM Pentabarf before, please use the same account/username Real-Time Communications devroom: deadline 23:59 UTC on 15th of December. Please use the [Pentabarf](https://penta.fosdem.org/submission/FOSDEM20/) system to submit a talk proposal for the devroom. On the "General" tab, please look for the "Track" option and choose "Real Time Communications devroom". Other devrooms and lightning talks: some speakers may find their topic is in the scope of more than one devroom. It is encouraged to apply to more than one devroom and also consider proposing a lightning talk, but please be kind enough to tell us if you do this by filling out the notes in the form. Here you can find the [full list of devrooms](https://www.fosdem.org/2020/schedule/tracks/) and here you can apply for a [lightning talk](https://fosdem.org/submit). ### First-time speaking? FOSDEM devrooms are a welcoming environment for people who have never given a talk before. Please feel free to contact the devroom administrators personally if you would like to ask any questions about it. ### Submission guidelines The Pentabarf system will ask for many of the essential details. Please remember to re-use your account from previous years if you have one. In the "Submission notes", please tell us about: - The purpose of your talk - Any other talk applications (devrooms, lightning talks, main track) - Availability constraints and special needs You can use HTML and links in your bio, abstract and description. If you maintain a blog, please consider providing us with the URL of a feed with posts tagged for your RTC-related work. We will be looking for relevance to the conference and devroom themes, presentations aimed at developers of free and open source software about RTC-related topics. Please feel free to suggest a duration between 20 minutes and 55 minutes but note that the final decision on talk durations will be made by the devroom administrators based on the number of received proposals. As the two previous devrooms have been combined into one, we may decide to give shorter slots than in previous years so that more speakers can participate. Please note FOSDEM aims to record and live-stream all talks. The CC-BY license is used. Volunteers needed - To make the devroom and lounge run successfully, we are looking for volunteers: - FOSDEM provides video recording equipment and live streaming, volunteers are needed to assist in this - Organizing one or more restaurant bookings (dependending upon number of participants) for the evening of Saturday, 1 February - Participation in the Real-Time lounge - Circulating this Call for Participation to other mailing lists Social events and dinners - The traditional FOSDEM beer night occurs on Friday, 31st of January. On Saturday night, there are usually dinners associated with each of the devrooms. Most restaurants in Brussels are not so large so these dinners have space constraints and reservations are essential. Please subscribe to the [Free-RTC mailing list](http://lists.freertc.org/mailman/listinfo/discuss) for further details about the Saturday night dinner options and how you can register for a seat. Related events around FOSDEM As per usual, the [XMPP Summit](https://wiki.xmpp.org/web/Conferences/Summit_24) is happening ahead of FOSDEM. This time it will take place on the 30th and 31st of
[asterisk-users] problems with blind transfer on GXP-2000 - Multi tenant asterisk !!
Hello Users, We have Thirdlane Multi tenant PBX system in production. Asterisk version is 1.6.2.15. Attendant transfer is working, but blind transfer is not working with Grandstream (gxp-2000) phone. We have read from google that it is a bug in Asterisk 1.6.2.15. We saw the below links: http://www.odesk.com/leaving_odesk.php?ref=http%253A%252F%252Fwww.freepbx.o rg%252Fforum%252Ffreepbx%252Fusers%252Ftransfer-bug-on-asterisk-1-4-38-1-6-2 -15-and-1-8-0-1 http://www.freepbx.org/forum/freepbx/users/transfer-bug-on-asterisk-1-4-38-1 -6-2-15-and-1-8-0-1 http://www.odesk.com/leaving_odesk.php?ref=http%253A%252F%252Fasteriskfaqs. org%252F2010%252F10%252F21%252Fasterisk-users%252Fblind-transfer-failed-sip- refer-method.html http://asteriskfaqs.org/2010/10/21/asterisk-users/blind-transfer-failed-sip- refer-method.html http://www.odesk.com/leaving_odesk.php?ref=https%253A%252F%252Fissues.aster isk.org%252Fview.php%253Fid%253D18185 https://issues.asterisk.org/view.php?id=18185 http://www.odesk.com/leaving_odesk.php?ref=https%253A%252F%252Fissues.aster isk.org%252Fview.php%253Fid%253D18185%2523c128539 https://issues.asterisk.org/view.php?id=18185#c128539 How to do this: We make a call from number to DDI, assigned to a Tenant in PBX. Incoming call is routed to an extension 100. We press transfer button and press the extensions. Call is dropped immediately. We couldn't see any error in the SIP log. I have attached the sip log, where call is made to ddi 016700202 and 100-solitaire rang. blindtransfer was made to 101-solitaire. If anyone is interested to fix this for us, it will be good. My yahoo id is sreedha...@yahoo.com Myskypeid is kvss2010 Note: I can provide more information upon request. Regs Sreedhar. -- _ -- 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] Drop Call on ICMP Port Unreachable?
One of our users recently had a powerfail while connected to our meetme gateway. (Asterisk 1.4.17 on debian 4.0) Through the course of it, asterisk never hung up. His system came back up, and started sending ICMP port unreachables, but the stream went on, flooding him with silence media stream packets (there was nobody else in the conference). Is asterisk aware of ICMP unreachables? Is there a tunable I can set to make it be? I found a thread here that discusses it briefly: http://lists.digium.com/pipermail/asterisk-users/2005-March/086626.html However, there's no real resolution there. If it's not aware of it, how difficult would it be to add? -Dan Mahoney -- Dan Mahoney Techie, Sysadmin, WebGeek Gushi on efnet/undernet IRC ICQ: 13735144 AIM: LarpGM Site: http://www.gushi.org --- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Voicemail Caller ID
Hello, I'm having an issue with caller ID in voicemail that I'd appreciate any input on. I have two sip peers defined as extension 100 and 101 each with separate voicemail accounts. Each sip peer has its own DID number, which is established via cid_number = 6021231234. When a call is placed from SIP peer #100 to SIP peer #101, and SIP peer #101 wants to reply to #100's voicemail via the option #3 advanced options menu, * responds with an error that #100 does not have a voicemailbox. Of course * thinks the voicemailbox = the caller ID/DID of #100, which in this case would be 6021231234 rather than 100, which is the defined voicemailbox for 100. My question is, do I need to rewrite callerid for internal extension-to-extension calls or is there something simple I'm missing? Thank you for any advice you can provide. ___ -- 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] Message
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Re: [asterisk-users] Message 0841984
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Re: [asterisk-users] MedHelp 34189
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[asterisk-users] ASTCC installation error install: invalid user `apache'
I am attempting a fresh install of ASTCC on Ubuntu. Getting install
invalid user as bellow. Has any one seen this? Can some one help out?
/usr/src/astcc# make install
mkdir -p /var/www
mkdir -p /var/www/html/_astcc
mkdir -p /var/www/cgi-bin/astcc-admin
chmod 755 ./astcc.agi
chmod 755 ./astcc-admin.cgi
echo | ./astcc.agi /dev/null
Detected dry run!
./astcc-admin.cgi /dev/null
DBI connect('database=;host=','',...) failed: Access denied for user
'root'@'localhost' (using password: NO) at ./astcc-admin.cgi line 69
install -m 755 -o apache -g root astcc-admin.cgi
/var/www/cgi-bin/astcc-admin/astcc-admin.cgi
install: invalid user `apache'
make: *** [install] Error 1
Thanks,
___
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Re: [asterisk-users] ASTCC installation error install: invalid user `apache'
Running this on Ubuntu 7/0 and appache2.
Googled around and saw possible solution
edit /etc/apache2/sites-available/default file and added the following
Directory /usr/lib/cgi-bin
AllowOverride None
Options +ExecCGI -MultiViews +SymLinksIfOwnerMatch
Order allow,deny
Allow from all
/Directory
Then restarted apache
/etc/apache2/sites-available# /etc/init.d/apache2 restart
* Restarting web server apache2
apache2: Could not reliably
determine the server's fully qualified domain name, using 127.0.1.1 for
ServerName
apache2: Could not reliably determine the server's fully qualified
domain name, using 127.0.1.1 for ServerName
[ OK ]
BUT I STILL AM NOT ABLE to run install on MTCC I get the same error
install: invalid user `apache' make: *** [install] Error 1
appreciate any support on this
Original Message
Subject: ASTCC installation error install: invalid user `apache'
From: [EMAIL PROTECTED]
Date: Sat, March 01, 2008 7:12 am
To: asterisk-users@lists.digium.com
I am attempting a fresh install of ASTCC on Ubuntu. Getting install
invalid user as bellow. Has any one seen this? Can some one help out?
/usr/src/astcc# make install
mkdir -p /var/www
mkdir -p /var/www/html/_astcc
mkdir -p /var/www/cgi-bin/astcc-admin
chmod 755 ./astcc.agi
chmod 755 ./astcc-admin.cgi
echo | ./astcc.agi /dev/null
Detected dry run!
./astcc-admin.cgi /dev/null
DBI connect('database=;host=','',...) failed: Access denied for user
'root'@'localhost' (using password: NO) at ./astcc-admin.cgi line 69
install -m 755 -o apache -g root astcc-admin.cgi
/var/www/cgi-bin/astcc-admin/astcc-admin.cgi
install: invalid user `apache'
make: *** [install] Error 1
Thanks,
___
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Re: [asterisk-users] ASTCC installation error install: invalid user `apache'
The error I see in the log...
/var/log/apache2# grep cgi-bin error.log
[error] [client 127.0.0.1] attempt to invoke directory as script:
/usr/lib/cgi-bin/, referer: http://localhost/
Original Message
Subject: RE: ASTCC installation error install: invalid user `apache'
From: [EMAIL PROTECTED]
Date: Sat, March 01, 2008 2:21 pm
To: [EMAIL PROTECTED]
Cc: asterisk-users@lists.digium.com
Running this on Ubuntu 7/0 and appache2.
Googled around and saw possible solution
edit /etc/apache2/sites-available/default file and added the following
Directory /usr/lib/cgi-bin
AllowOverride None
Options +ExecCGI -MultiViews +SymLinksIfOwnerMatch
Order allow,deny
Allow from all
/Directory
Then restarted apache
/etc/apache2/sites-available# /etc/init.d/apache2 restart
* Restarting web server apache2
apache2: Could not reliably
determine the server's fully qualified domain name, using 127.0.1.1 for
ServerName
apache2: Could not reliably determine the server's fully qualified
domain name, using 127.0.1.1 for ServerName
[ OK ]
BUT I STILL AM NOT ABLE to run install on MTCC I get the same error
install: invalid user `apache' make: *** [install] Error 1
appreciate any support on this
Original Message
Subject: ASTCC installation error install: invalid user `apache'
From: [EMAIL PROTECTED]
Date: Sat, March 01, 2008 7:12 am
To: asterisk-users@lists.digium.com
I am attempting a fresh install of ASTCC on Ubuntu. Getting install
invalid user as bellow. Has any one seen this? Can some one help out?
/usr/src/astcc# make install
mkdir -p /var/www
mkdir -p /var/www/html/_astcc
mkdir -p /var/www/cgi-bin/astcc-admin
chmod 755 ./astcc.agi
chmod 755 ./astcc-admin.cgi
echo | ./astcc.agi /dev/null
Detected dry run!
./astcc-admin.cgi /dev/null
DBI connect('database=;host=','',...) failed: Access denied for user
'root'@'localhost' (using password: NO) at ./astcc-admin.cgi line 69
install -m 755 -o apache -g root astcc-admin.cgi
/var/www/cgi-bin/astcc-admin/astcc-admin.cgi
install: invalid user `apache'
make: *** [install] Error 1
Thanks,
___
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[asterisk-users] sip reload causes unreachable
Hi I have a asterisk with many phones (type=friend) When I issue the command sip reload some of the phones become unreachable and they come back just after. I guess that the sip.conf file is too big and asterisk takes too much time reloading the entire file. Is there a way to avoid this probleme or another way to add/remove sip phones dynamically ? Patrick ___ --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] New to Asterisk
Hi everyone, I'm completely new to Asterisk and before I buy any card, I would like to ask for some information. 1. We'll be using analog PSTN phone lines. Is there anything that I should ask the telecom company before I buy the card? What I mean is whether the card will be compatible with the line? 2. What about the hardware on the PC? I will be using at least a Pentium 3 with a 600 or 700 MHz processor with at least 256 MB. Is there a way to know how much traffic or calls it can handle? 3. Suppose I buy a TDM04B card. It has 4 FXO modules and 0 FXS module. Later I decide that I need a TDM13B configuration. Can I just buy 1 FXS S110M module and replace one existing FXO module myself and reconfigure Asterisk? 4. Does fax work fine with Asterisk? Should I use one FXS module for each fax machine? 5. Is the power connector on the card identical to the power connectors inside PCs? Thank you for any help. I choose Polesoft Lockspam to fight spam, and you? http://www.polesoft.com/refer.html ___ --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] New to Asterisk
Hi everyone, I'm completely new to Asterisk and before I buy any card, I would like to ask for some information. 1. We'll be using analog PSTN phone lines. Is there anything that I should ask the telecom company before I buy the card? What I mean is whether the card will be compatible with the line? 2. What about the hardware on the PC? I will be using at least a Pentium 3 with a 600 or 700 MHz processor with at least 256 MB. Is there a way to know how much traffic or calls it can handle? 3. Suppose I buy a TDM04B card. It has 4 FXO modules and 0 FXS module. Later I decide that I need a TDM13B configuration. Can I just buy 1 FXS S110M module and replace one existing FXO module myself and reconfigure Asterisk? 4. Does fax work fine with Asterisk? Should I use one FXS module for each fax machine? 5. Is the power connector on the card identical to the power connectors inside PCs? Thank you for any help. I choose Polesoft Lockspam to fight spam, and you? http://www.polesoft.com/refer.html ___ --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] Cisco 7970 SIP upgrade issues
Dear Asterisk People, I am having problems putting a SIP image on a 7970. I was wondering if anyone can help? First problem is the phone is running version Load IDJar70.2-5-47-17.sbn Boot Load ID7970_64054100.bin Version5.0(0.6S) So I did read that you couldn't simply put the latest SIP image on such an old phone and a newer firmware version should be used. I got cmterm-7970_7971-sccp.7-0-2SR1 However I can't figure out how to update the firmware without a Callmanager. Can anyone enlighten me? If I do that I can then put the latest SIP image on I think Best Regards___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Outbound CallerID Teliax
Is anyone successfully passing Outbound CallerID to Teliax? If so can you please tell me how. Thanks in advance! Dan ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] wholesale DID's?
We would like to use Asterisk to deploy VoIP to our broadband internet access customers. Which VoIP providers (that are reliable stable) provide wholesale DID's? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk, RH9, minimal install
I have a copy of RH9 and would like to build a Asterisk box for my office but I do not want to load any unnecessary software. Can someone provide me a list of required items above a minimal install. Thanks in advance. Dan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Monday, July 04, 2005 4:29 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Proper way to start * and load modules on aRedHatbox On Mon, Jul 04, 2005 at 09:55:46PM +0200, Roland Zagler wrote: i experienced that on some configs the service asterisk restart does not work correctly, so go to /etc/rc.d/init.d and edit the file asterisk and insert a sleep 5 between stop and start in restart. Why is that? 'restart' on debian is simply: $0 stop $0 start 'stop' has some logic to stop asterisk: first asterisk -rx 'stop now' in the background, so it won't hang, after two seconds sigterm to the asterisk processes and after 5 sends sigkill. Seems to work well. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk, RH9, minimal install
Thanks Carlos this is the link I was looking for. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carlos Alperin Sent: Monday, July 04, 2005 5:28 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Asterisk, RH9, minimal install Importance: High Dan, I believe that this can help you on your question. http://www.automated.it/guidetoasterisk.htm#_Toc49248757 Regards, Carlos Alperin Senior System Engineer Seneca Communications, LLC [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Plexicomm Admin Sent: Monday, July 04, 2005 5:06 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Asterisk, RH9, minimal install I have a copy of RH9 and would like to build a Asterisk box for my office but I do not want to load any unnecessary software. Can someone provide me a list of required items above a minimal install. Thanks in advance. Dan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Monday, July 04, 2005 4:29 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Proper way to start * and load modules on aRedHatbox On Mon, Jul 04, 2005 at 09:55:46PM +0200, Roland Zagler wrote: i experienced that on some configs the service asterisk restart does not work correctly, so go to /etc/rc.d/init.d and edit the file asterisk and insert a sleep 5 between stop and start in restart. Why is that? 'restart' on debian is simply: $0 stop $0 start 'stop' has some logic to stop asterisk: first asterisk -rx 'stop now' in the background, so it won't hang, after two seconds sigterm to the asterisk processes and after 5 sends sigkill. Seems to work well. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Colocation/Telehousing
We do colocation, but we are on the east coast. What are your specific needs? -Dan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sahil Gupta Sent: Monday, July 04, 2005 10:10 PM To: asterisk-users@lists.digium.com Cc: asterisk-biz@lists.digium.com Subject: [Asterisk-Users] Colocation/Telehousing Hi, Is there anybody on the list that recommends anyone for colocation/telehousing in the US? I'm after 2 Servers to be hosted in the US, preferably on the west coast. Regards, Sahil Gupta VoiceValley ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What does Asterisk need in the way of a GUI?
Here are a couple of items I hear people asking for regularly. - Multi-tenant functionality - Allow users to change their own preferences via web (call forwarding, MoH, etc...) We are two programmers who are passionate for Asterisk and we will be dedicating the next three months towards programming for Asterisk and would like to get some input from everyone on what they feel Asterisk is lacking or needs based on what is not currently a part of it or available through third parties. Hopefully, by asking up front we won't be wasting our time on something nobody wants or needs. Specifically I am asking in the way of GUI's (web-based or not), not in backend programming as Mark and others have that well under control! Thank you for your suggestions, Mitchel Tom ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What does Asterisk need in the way of a GUI?
Here are a couple of items I hear people asking for regularly. - Multi-tenant functionality - Allow users to change their own preferences via web (call forwarding, MoH, etc...) We are two programmers who are passionate for Asterisk and we will be dedicating the next three months towards programming for Asterisk and would like to get some input from everyone on what they feel Asterisk is lacking or needs based on what is not currently a part of it or available through third parties. Hopefully, by asking up front we won't be wasting our time on something nobody wants or needs. Specifically I am asking in the way of GUI's (web-based or not), not in backend programming as Mark and others have that well under control! Thank you for your suggestions, Mitchel Tom ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MusicOnHold probelms
Do you have mpg123 installed? Is there a .mp3 file available to play in your /var/lib/asterisk/mohmp3 directory? -daryl -Original Message- From: chawki hammoud [EMAIL PROTECTED] To: Asterisk-Users@lists.digium.com Cc: Date: Thu, 19 May 2005 06:03:55 -0700 (PDT) Subject: [Asterisk-Users] MusicOnHold probelms This is my second attempt trying to get help and I am hoping someone can. When the musiconhold extension is matched, Asterisk attempts to execute musiconhold and stops right away, this is what I gets: Executing MusicOnHold(OSS/dsp, ) in new stack -- Started music on hold, class 'default', on OSS/dsp -- Stopped music on hold on OSS/dsp Is there a file that musiconhold try to play and can't find. Please help withy any suggestions. Discover Yahoo! Stay in touch with email, IM, photo sharing and more. Check it out! http://discover.yahoo.com/stayintouch.html ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] no music on hold
I am using a UTStarcom ATA. I am having the same experience. I can hear the music via firefly softphone if I call a test MoH extension directly, but neither the hold button on the softphone, or the hold/flash buttons on a regular phone (connected through the ATA) seem to work. When I call a test MoH extension through the UTStarcom I sounds like it is trying to play with random spurts of sound. Asterisk CLI shows the MoH happening as normal, just no sound when using the hold buttons. I am using HEAD version of *. Does anyone have * HEAD running MoH without problems? -Original Message- From: Bartosz Jozwiak [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Thu, 19 May 2005 20:11:31 -0300 Subject: Re: [Asterisk-Users] no music on hold I am having problems with music on hold on grandstream phones. When I press Hold button on grandstream phone this is the debug of sip. But nothing happens, no music. Is it problem of asterisk or grandstream budget phone? Do you have mpg123 installed? Do you have any mp3's (CBR preferably) in your moh directory? Have you defined any moh classes in musiconhold.conf? Yes mpg123 is installed and it plays music, put it does not play when u pres button HOLD on grandstram. I just down grade asterisk and now eveything works ok Bartosz ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] HELP... SER + Asterisk as feature server
Can anyone here help me understand what I missing with this setup. I want to use Asterisk as a feature server only, speaking only SIP (no IAX), and use SER for registration to minimize necessary bandwidth. SIP-phone --SER -- * -- PSTN Provider -- Regular-phoneRegular-phone -- PSTN Provider -- SER -- * -- SIP-phone I want to allow SIP users to transfer calls to other users, either on the system or on the PSTN. I'm not sure how to make this work with *. From what I understand, once a call is setup by SER the caller has no access to * because * is not in the media path. If so, * would not be able to catch the DTMF tones and transfer the call. Is this correct? Any help would be greatly appreciated! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] debugging trunks between two asterisk boxes at two different locations
objective: users connected to box A can dial the extension number of users connected to box B boxA at location 1: works fine for internal lan users using the firefly softphone boxB at location 2: works fine for internal lan users using the firefly softphone Both the boxes have a IAX trunk defined following the instructions on: https://sourceforge.net/docman/display_doc.php?docid=26418group_id=121515 and guess what .. it doesnt work how do i go about debugging this thing ? t ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] why even use SIP
after two days of experiments finally decided to go with sipura 2001. I was wondering to support a 50 people call center do i need 25 sipura 2001 or 50 of these ? t On Thu, 24 Mar 2005 16:39:25 -0500, Dana Olson [EMAIL PROTECTED] wrote: On Thu, 24 Mar 2005 13:03:46 -0700, JD Austin [EMAIL PROTECTED] wrote: xlite doesn't seem to have this problem. X-Lite doesn't support IAX. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] why even use SIP
each of the desks already has two rj45 network ports so it makes sense for me to put a sipura 2001 at each of the deks rather then getting a channel bank and then having to do new cabling, will i be ok with ordering 25 of the sipura 2001 since each one of them have 2 FXS ports. Or are there firmware/voice quality/asterisk integration issues to use both FXS ports on a sipura 2001 simultaneously. t On Fri, 25 Mar 2005 14:52:48 -0600, Michael Graves [EMAIL PROTECTED] wrote: On Fri, 25 Mar 2005 12:46:41 -0800, Sys Admin wrote: after two days of experiments finally decided to go with sipura 2001. I was wondering to support a 50 people call center do i need 25 sipura 2001 or 50 of these ? t For such a large installation you'd be far better of with a channel bank to provide FXS ports. Much less network cabling and hassle. You'd use a t-1 connection to bridge between * and the channel bank. The wiki has lots of detail on this stuff. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk compare with Skype
if some one was to create a open source IAX client as good/better then skype, even then a asterisk IAX based network will not be able to compete with skype. Since asterix is a centralized server regitration network it can not grow as big as a skype P2P network can grow, t On Fri, 25 Mar 2005 12:49:24 -0800, Kerry Garrison [EMAIL PROTECTED] wrote: Why would this ever change or need to change? Many many people are quite happy with how Skype works and would never need anything else. Its like saying why would anyone use ICQ/AIM/Yahoo/MSN when you can run your own email server. The application is similar but still very very different. Kerry http://www.geekgazette.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of snacktime Sent: Friday, March 25, 2005 12:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk compare with Skype On Fri, 25 Mar 2005 15:19:55 -0500, Brian Capouch [EMAIL PROTECTED] wrote: snacktime wrote: I guess I'm a bit frustrated because it is very difficult to find reasons why people who use skype should switch to something else. For the average person, there really aren't any compelling reasons I can find when you take everything into account. Your points are very well taken. I never said the Skype folks weren't brilliant, or that they aren't efficiently fulfilling a market need. All of us dedicated Asterholics hope, I'm sure, that someday this imbalance will be rectified. I think it will be sooner than later actually, maybe I'm just impatient:) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] why even use SIP
exactly u bring me to one more anamoly in this industry: (the first one was IAX being beter then SIP but not yet ready for prime time) with a sipura 2001 FXS port the company I am consulting with can plug in very high end quality analog phones each say for $50 offering speaker phone / cordless phone / answering machine integrated with the phone which they can hear before they decide to pick up or not etc etc... So each workstation ends up costing say $90 ($50 for the phone and $40 for the one FXS port on a sipura 2001) Now this same feature set if I was to look for in a IP phone, the cost would be more then $300, whats up with that ? t On Fri, 25 Mar 2005 15:21:16 -0600, Michael Graves [EMAIL PROTECTED] wrote: On Fri, 25 Mar 2005 13:14:12 -0800, Sys Admin wrote: each of the desks already has two rj45 network ports so it makes sense for me to put a sipura 2001 at each of the deks rather then getting a channel bank and then having to do new cabling, will i be ok with ordering 25 of the sipura 2001 since each one of them have 2 FXS ports. Or are there firmware/voice quality/asterisk integration issues to use both FXS ports on a sipura 2001 simultaneously. t On Fri, 25 Mar 2005 14:52:48 -0600, Michael Graves [EMAIL PROTECTED] wrote: On Fri, 25 Mar 2005 12:46:41 -0800, Sys Admin wrote: after two days of experiments finally decided to go with sipura 2001. I was wondering to support a 50 people call center do i need 25 sipura 2001 or 50 of these ? t For such a large installation you'd be far better of with a channel bank to provide FXS ports. Much less network cabling and hassle. You'd use a t-1 connection to bridge between * and the channel bank. The wiki has lots of detail on this stuff. If you already have all that Cat5 then why not consider low end IP phones? I shudder to think about having to configure and support all those SPAs. All those wall-wart PSUs. Seems less than elegant. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk compare with Skype
You think way to small. :( On Fri, 25 Mar 2005 15:55:54 -0600, Steven Critchfield [EMAIL PROTECTED] wrote: On Fri, 2005-03-25 at 13:18 -0800, Sys Admin wrote: if some one was to create a open source IAX client as good/better then skype, even then a asterisk IAX based network will not be able to compete with skype. Since asterix is a centralized server regitration network it can not grow as big as a skype P2P network can grow, You think way to small. You don't have to be centralized with asterisk. Administration of a secure network is easier being centralized, but you could easily run several asterisk machines in a network much like IRC is. A few or even several asterisk boxes peered to each other to know who is where and the users spread over the many asterisk machines. With IAX attempting to handoff calls if possible, it is possible to create true P2P calls as well as routed calls to handle NAT or anything else. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] why even use SIP
just called digium using firefly softphone connected to a asterisk server using IAX2 they said that the IAXy device is not in stock and the earliest expected arrival is after a month. On a dell insipiron 600m laptop with 512 MB RAM each time i maximize or minimize even a small application like putty the firefly softphone looses sound for 1/2 a second. Why is the softphone application so bad that it can not even handle another application being maximized and minimized. This really throws me off !! t On Thu, 24 Mar 2005 12:07:24 -0500, Dana Olson [EMAIL PROTECTED] wrote: On Thu, 24 Mar 2005 15:34:26 -, Giles Coochey [EMAIL PROTECTED] wrote: How about scanning for it's mac address? http://ipscan.sf.net/ipscan.exe -- http://www.umich2.com Digium doesn't label the MAC address on the device, unless it's such a fine print that no one can read it. I believe this has been said a few times in the conversation. Connect it with a cross-over ethernet cable to a Linux box and run tcpdump on the Linux box, before long you'll see the IP address come up on the tcpdump logs. Don't power it off, you want it to have an existing DHCP lease. If you don't see any traffic, try making a call. Once you have the IP you can put it back on the normal network and configure it. I know how to work around these limitations already. My point is that this is not an enterprise-ready solution. If I order 1000 of these for our IT staff, I have to go through each and every one with a crossover cable just to find the IP? Why would we bother when there so many other devices that don't have any of the flaws of the IAXy? Of course they are SIP-only, so that's the answer to the question of why use SIP at all. Because there is no good solution for IAX yet. With a little work, the IAXy can become a product not only for hobbyists but for the corporate world as well. Until then, we will need to rely on Sipura, Grandstream, and the like for devices that can be much easier provisioned, either by keypad entry on the device itself, TFTP config files, or an HTTP interface, that support DNS name resolution, G729/iLBC/GSM codecs, have their MAC addresses labeled on them, etc. This is for my company only. Perhaps yours isn't so large and you have the time and desire to go through this process for every device in your organization, but we don't. Yes, for home users who run Asterisk, it's fine, except if they want to take the IAXy on the road with them and they don't have a static IP address. For internal use in a small company, yeah, the IAXy may be a fine solution. But when you're looking at purchasing hundreds of devices at a time, I don't think this is a good product at this time. All of that said, I like the IAXy, and I will gladly recommend buying it if you're not in my position, or if Digium develops it further to address these issues. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] why even use SIP
well i even pressed ctrl+alt+del went into the process monitor and gave the firefly process high priority. Still it looses half a second of sound each time i maximize or minimize a app like putty, whats the word for this . sucks . why doesnt skype have this problem ? t On Thu, 24 Mar 2005 10:39:54 -0800 (PST), Robert Hajime Lanning [EMAIL PROTECTED] wrote: Because the video driver is a kernel thread and not allowed to lag. That would cause framerate issues with games. :) oh winderz... quote who=Sys Admin On a dell insipiron 600m laptop with 512 MB RAM each time i maximize or minimize even a small application like putty the firefly softphone looses sound for 1/2 a second. Why is the softphone application so bad that it can not even handle another application being maximized and minimized. This really throws me off !! -- END OF LINE -MCP ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] why even use SIP
so seems like the verdict is go IAXy with a IAX only network ? Most of the problems of the IAXy device seems like will be fixed with firmware updates and wont require a hardware update.. this way we get the advantage of a Hardphone (human factor, just feel good to talk on a real phone) with all the goodies of the IAX protocol. t On Wed, 23 Mar 2005 14:23:17 -0500, Dana Olson [EMAIL PROTECTED] wrote: On Wed, 23 Mar 2005 13:59:07 -0500, Time Bandit [EMAIL PROTECTED] wrote: There's another feature request. Let me dial ### or something to find my IP... That's not something to do with the IAXy, you can make an AGI script that will tell you your IP. I had this script somewhere but I can't find it at the moment. This would not only be valid for the IAXy, but for any phone connected to asterisk (well, except analog phones) If I find it, I'll let you know. But I'm confident that somebody on this list has something like this. You dial some extension that call this script and it tells you your IP using SayDigits. hth So how am I going to provision the device in the first place, to be able to dial this extension, if I don't even know the IP? ./iaxyprov Usage: provision ip file ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Reg Asterisk
couldnt agree with u more !! On Wed, 23 Mar 2005 11:15:55 -0800, Robert Goodyear [EMAIL PROTECTED] wrote: Confidentiality Notice The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain confidential or privileged information. If you are not the intended recipient, please notify the sender at Wipro or [EMAIL PROTECTED] immediately and destroy all copies of this message and any attachments. I'm not one of the addressees' unless that since I'm subscribed I am. However these messages get archived and are available for everybody to see at lists.digium.com I believe this makes your posting messages to this list a paradox with the disclaimer you have. CANT YOU JUST GET RID OF THIS RIDICULES DISCLAIMER? at least when posting to lists. Moreover than just an annoyance, all the pointless (and off topic!) words in these footers create bogus indexes when Googling. Which further diminishes the power of this list and its searchability/value/self-sustainment. I wish the subscribe info contained some key points such as: 1. SEND NO FOOTERS 2. Use a REAL subject line... you might as well NEVER say: a. Noob question b. Need help please c. Weird problem ... because all of the above are completely self-evident by the mere submission of an email to the list. /rg ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] why even use SIP
is some one from digium reading this thread. !! Looks like they have a ready and a big market for this device. And all they need to do is invest say 6 man months of development effort :) come on digium do it !! How about making the firmware open source so we can hack on it ... t On Wed, 23 Mar 2005 16:37:02 -0500, Dana Olson [EMAIL PROTECTED] wrote: On Wed, 23 Mar 2005 16:13:12 -0500, Time Bandit [EMAIL PROTECTED] wrote: So how am I going to provision the device in the first place, to be able to dial this extension, if I don't even know the IP? Oups, sorry, didn't think about this one. Check winiaxyprov, the version 1.01 can scan your network to find IAXy. Now the only thing we need is for Digium to write the MAC address on the device before sending it in the open world. Because if you have more than one on your network, you can't really know which one you need to provision. hth Yeah, I found that app earlier in the thread and thanked whoever it was (maybe you, can't remember) for linking to it. It's handy, as I had no way to determine the MAC or IP address prior to this, my IAXys kinda sat on the shelf collecting dust. (I did bring one home and plugged it into my Linksys router, but that's hardly an option in a large IT organization with many IAXys.) My company has thousands of entries in the DHCP server, and it would take forever to go through each and every one of them. Not to mention that I, being in the telecom division, do not have access to the DHCP servers. Luckily I actually have a Windows desktop here at work. I'd like a scanner like that for Linux though. Maybe it's possible with some other kind of application? Anyhow, I still think it wouldn't kill them to add an IP address feature or something (an alternative would be to allow the iaxyprov tool to provision by MAC or IP, and yes, start labeling the devices with their MACs). To me, it just doesn't seem like a product that was really ready for release yet. I think it could be really great after a bit of development though, and wouldn't discourage Digium from doing so, but for now, our company can't really use these for many applications. -- Dana ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] why even use SIP
I am setting up a new asterisk based call center. I just read: http://www.voip-info.org/wiki-IAX+versus+SIP After reading this and other google results for IAX vs SIP is there any reason why i should use SIP anywhere !! t ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: asterisk-1.0.7 make install on fedora corre 3 give errors
an update: since it might help others
I did the same make on another machine and it worked fine. So it seems
to be a problem with my tool-chain
t
On Sun, 20 Mar 2005 22:18:50 -0800, Sys Admin [EMAIL PROTECTED] wrote:
I am trying to install asterisk on fedora core 3 these are the steps i took:
1. download asterisk-1.0.7.tar.gz
2. make clean and make install and then it gives me these errors:
{standard input}:9975: Error: symbol `i' is already defined
{standard input}:9978: Error: symbol `__result' is already defined
{standard input}:9979: Error: symbol `__result' is already defined
{standard input}:9981: Error: symbol `__result' is already defined
{standard input}:9982: Error: symbol `__result' is already defined
{standard input}:9984: Error: symbol `__result' is already defined
{standard input}:9985: Error: symbol `__result' is already defined
{standard input}:9987: Error: symbol `__result' is already defined
{standard input}:9988: Error: symbol `__result' is already defined
{standard input}:9990: Error: symbol `__result' is already defined
{standard input}:9991: Error: symbol `__result' is already defined
{standard input}:9993: Error: symbol `__result' is already defined
{standard input}:9994: Error: symbol `__result' is already defined
{standard input}:9996: Error: symbol `__result' is already defined
{standard input}:9997: Error: symbol `__result' is already defined
{standard input}:: Error: symbol `__result' is already defined
{standard input}:1: Error: symbol `__result' is already defined
{standard input}:10002: Error: symbol `__result' is already defined
{standard input}:10003: Error: symbol `__result' is already defined
{standard input}:10005: Error: symbol `__result' is already defined
{standard input}:10006: Error: symbol `__result' is already defined
{standard input}:10008: Error: symbol `__result' is already defined
{standard input}:10009: Error: symbol `__result' is already defined
{standard input}:10011: Error: symbol `__result' is already defined
{standard input}:10012: Error: symbol `__result' is already defined
{standard input}:10014: Error: symbol `__result' is already defined
{standard input}:10015: Error: symbol `__result' is already defined
{standard input}:10017: Error: symbol `__result' is already defined
{standard input}:10018: Error: symbol `__result' is already defined
{standard input}:10020: Error: symbol `__result' is already defined
{standard input}:10021: Error: symbol `__result' is already defined
{standard input}:10023: Error: symbol `__result' is already defined
{standard input}:10024: Error: symbol `__result' is already defined
{standard input}:10030: Error: symbol `i' is already defined
{standard input}:10042: Error: symbol `i' is already defined
{standard input}:10048: Error: symbol `arg_cols' is already defined
{standard input}:10054: Error: symbol `i' is already defined
{standard input}:10060: Error: symbol `arg_rows' is already defined
make[1]: *** [editline.o_a] Error 1
make[1]: Leaving directory `/root/download/asterisk/asterisk-1.0.7/editline'
make: *** [editline/libedit.a] Error 2
any help would be appreciated,
sysadmin
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Re: [Asterisk-Users] why even use SIP
2 reasons for not using IAX: A. CDR as part of the media B. Not good hardphone / softphone Problem A (CDR as part of the media) I am not worried about too much, as long as the data is there some parsing will allow me to extract it and then do what i want to do with it. Problem B (Not good hardphone / softphone) what would be the best hardphone/softphone to make it work ? t On Mon, 21 Mar 2005 13:15:22 -0500, Andrew Kohlsmith [EMAIL PROTECTED] wrote: On March 21, 2005 01:07 pm, Kevin P. Fleming wrote: Well, let's see.. 99.99% of the available VOIP hardware only support SIP, MGCP and H.323, but not IAX2. Is that a good reason? No. 95% of the marketplaces uses Windows. Drive the marketplace to use better protocols. IAX2 calls between servers carry the signaling and media in the same connection, which is good for NAT issues, but bad for CDR and traffic control issues. SIP handles them separately, so you can keep complete CDR without forcing the media to follow the same path. Is that a good reason? Yes, but the dynamic port allocation and ABSOLUTELY INSANE WORDINESS and WASTAGE of the SIP control protocol is reason enough for me to never want to support it. While perhaps not worth much on my own, I am voting with my wallet and my feet. I will not support SIP, nor will I purchase products or services which require it. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] why even use SIP
btw if there is no good softphone for IAX which does G729, i could possibly get my company to buy consulting time from say DIAX and make dante develop it further, t On Mon, 21 Mar 2005 11:03:48 -0800, Sys Admin [EMAIL PROTECTED] wrote: 2 reasons for not using IAX: A. CDR as part of the media B. Not good hardphone / softphone Problem A (CDR as part of the media) I am not worried about too much, as long as the data is there some parsing will allow me to extract it and then do what i want to do with it. Problem B (Not good hardphone / softphone) what would be the best hardphone/softphone to make it work ? t On Mon, 21 Mar 2005 13:15:22 -0500, Andrew Kohlsmith [EMAIL PROTECTED] wrote: On March 21, 2005 01:07 pm, Kevin P. Fleming wrote: Well, let's see.. 99.99% of the available VOIP hardware only support SIP, MGCP and H.323, but not IAX2. Is that a good reason? No. 95% of the marketplaces uses Windows. Drive the marketplace to use better protocols. IAX2 calls between servers carry the signaling and media in the same connection, which is good for NAT issues, but bad for CDR and traffic control issues. SIP handles them separately, so you can keep complete CDR without forcing the media to follow the same path. Is that a good reason? Yes, but the dynamic port allocation and ABSOLUTELY INSANE WORDINESS and WASTAGE of the SIP control protocol is reason enough for me to never want to support it. While perhaps not worth much on my own, I am voting with my wallet and my feet. I will not support SIP, nor will I purchase products or services which require it. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] why even use SIP
After 20 posts, in 2005 the ideal setup for a new installtion of a 50 user asterisk is: Option1: IAX2 with softphone firefly Option2: SIP with softphone Option3: IAX2 with hardphones (which brand?) Option4: SIP with hardphones. Seems like we cannot come to a definite conclusion, poll ? so the verdict is ? On Tue, 22 Mar 2005 11:48:00 +0800, Shaun Dwyer [EMAIL PROTECTED] wrote: Hi Scott, Intresting to know, cheers :) Only down side though, is that most people using softphones will be using Windows... If only ALSA was available for windows ;) -Shaun Scott Williamson wrote: That should be the program alsamixer, not amixer. Make sure to press F5 to get all of the playback/capture devices shown. On Mon, 2005-21-03 at 22:02 -0500, Scott Williamson wrote: Ah, Console sound card echo. I found that with my cheep YMFPCI sound card that there is a channel called wave capture that is enabled for recording by default. And this is only visible when one uses ALSA sound drivers. One needs to use an ALSA mixer control program (I use amixer, the text mode one) to disable, or reduce the volume on these sound cards. Once this is done there is no more echo at all! Why this channel does not appear under OSS, and why it appears to be enabled by default in the hardware is beyond me. The GUI mixers all list this channel simply as WAVE, and I have two WAVE channels. So remember amixer. The card is called ICEnsemble ICE1230 by OSS, and Yamaha DS-XG (YMF734) by ALSA On Tue, 2005-22-03 at 10:52 +0800, Shaun Dwyer wrote: Scott Bussinger wrote: We just tried to go entirely with softphones in our office gave up after a month or so of trying. I tried probably 10 different softphones running on 3.0Ghz WinXP machines and none of them were workable. I tried both SIP and IAX2 softphones using headsets plugged into the audio ports, USB headsets, and USB phone interface boxes (www.phoneconnector.com). While it wasn't hard to get them to work and the concept would have been perfect in our environment, the quality was _terrible_! We had many issues snip I found that for the most part, crappy sounds cards caused the bulk of problems with soft phones. I noticed that on, for example, intel D865PERLL motherboards with an onboard realtek AC97 sound device, There was heaps of echo for the remote end (using a softphone or a hardphone). I also found that a stock standard Creative PCI 128 sound card gave great results. Given PCI128s arn't available anymore, I'm sure you could find an alternative, like perhaps even going as far as a SB Live Value.. if you buy these in bulk, im sure you can get em cheap as chips. I've only ever used X-lite as a soft phone and found it to be really very good. Cheers, -Shaun ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Scott J. Williamson [EMAIL PROTECTED] Not every problem someone has with his girlfriend is necessarily due to the capitalist mode of production. -- Herbert Marcuse ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Scott J. Williamson [EMAIL PROTECTED] Humor in the Court: Q: The truth of the matter is that you were not an unbiased, objective witness, isn't it. You too were shot in the fracas? A: No, sir. I was shot midway between the fracas and the naval. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk-1.0.7 make install on fedora corre 3 give errors
I am trying to install asterisk on fedora core 3 these are the steps i took:
1. download asterisk-1.0.7.tar.gz
2. make clean and make install and then it gives me these errors:
{standard input}:9975: Error: symbol `i' is already defined
{standard input}:9978: Error: symbol `__result' is already defined
{standard input}:9979: Error: symbol `__result' is already defined
{standard input}:9981: Error: symbol `__result' is already defined
{standard input}:9982: Error: symbol `__result' is already defined
{standard input}:9984: Error: symbol `__result' is already defined
{standard input}:9985: Error: symbol `__result' is already defined
{standard input}:9987: Error: symbol `__result' is already defined
{standard input}:9988: Error: symbol `__result' is already defined
{standard input}:9990: Error: symbol `__result' is already defined
{standard input}:9991: Error: symbol `__result' is already defined
{standard input}:9993: Error: symbol `__result' is already defined
{standard input}:9994: Error: symbol `__result' is already defined
{standard input}:9996: Error: symbol `__result' is already defined
{standard input}:9997: Error: symbol `__result' is already defined
{standard input}:: Error: symbol `__result' is already defined
{standard input}:1: Error: symbol `__result' is already defined
{standard input}:10002: Error: symbol `__result' is already defined
{standard input}:10003: Error: symbol `__result' is already defined
{standard input}:10005: Error: symbol `__result' is already defined
{standard input}:10006: Error: symbol `__result' is already defined
{standard input}:10008: Error: symbol `__result' is already defined
{standard input}:10009: Error: symbol `__result' is already defined
{standard input}:10011: Error: symbol `__result' is already defined
{standard input}:10012: Error: symbol `__result' is already defined
{standard input}:10014: Error: symbol `__result' is already defined
{standard input}:10015: Error: symbol `__result' is already defined
{standard input}:10017: Error: symbol `__result' is already defined
{standard input}:10018: Error: symbol `__result' is already defined
{standard input}:10020: Error: symbol `__result' is already defined
{standard input}:10021: Error: symbol `__result' is already defined
{standard input}:10023: Error: symbol `__result' is already defined
{standard input}:10024: Error: symbol `__result' is already defined
{standard input}:10030: Error: symbol `i' is already defined
{standard input}:10042: Error: symbol `i' is already defined
{standard input}:10048: Error: symbol `arg_cols' is already defined
{standard input}:10054: Error: symbol `i' is already defined
{standard input}:10060: Error: symbol `arg_rows' is already defined
make[1]: *** [editline.o_a] Error 1
make[1]: Leaving directory `/root/download/asterisk/asterisk-1.0.7/editline'
make: *** [editline/libedit.a] Error 2
any help would be appreciated,
sysadmin
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[Asterisk-Users] Broadvoice Alternatives.
Hello: I am looking for alternatives to Broadvoice. Comparability in price but with much better services (stable) to Asterisk users. Please email off-list. Thanks, Terrell S. Patrick www.dealerbackup.net (800) 651-4484 The best time to respond to a disaster is before it happens. A relatively small investment of time and money now may prevent severe damage and disruption of life and business in the future. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Apply Patch for Broadvoice.
Received errors as follows. [EMAIL PROTECTED] asterisk]# patch channels/chan_sip.c broadvoicesip.txt patching file channels/chan_sip.c Hunk #1 succeeded at 229 (offset 13 lines). Hunk #2 succeeded at 308 (offset -2 lines). Hunk #3 succeeded at 494 (offset 11 lines). Hunk #4 succeeded at 489 (offset -2 lines). Hunk #5 FAILED at 3969. Hunk #6 succeeded at 4059 (offset 66 lines). Hunk #7 succeeded at 4011 (offset -2 lines). Hunk #8 succeeded at 4111 (offset 66 lines). Hunk #9 succeeded at 4084 (offset -3 lines). Hunk #10 FAILED at 4100. Hunk #11 succeeded at 4201 (offset 67 lines). Hunk #12 FAILED at 4216. Hunk #13 succeeded at 4279 (offset 15 lines). Hunk #14 succeeded at 6279 (offset 31 lines). Hunk #15 succeeded at 6387 (offset 15 lines). Hunk #16 FAILED at 6800. Hunk #17 succeeded at 6855 (offset 35 lines). 4 out of 17 hunks FAILED -- saving rejects to file channels/chan_sip.c.rej [EMAIL PROTECTED] asterisk]# Thanks, Terrell S. Patrick www.dealerbackup.net (800) 651-4484 The best time to respond to a disaster is before it happens. A relatively small investment of time and money now may prevent severe damage and disruption of life and business in the future. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Seth Remington Sent: Friday, December 10, 2004 2:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Apply Patch for Broadvoice. On Fri, 2004-12-10 at 14:45, Dealer Backup Info wrote: Hello, I am using Broadvoice for my outgoing calls with my Asterisk box. Broadvoice is requiring my to apply a patch to my Asterisk. Instructions at the following link. http://www.broadvoice.com/support_install_asterisk.html Step 1 is what I need help with, not sure on how to apply patch. I have the rest of the instructions figured out. Copy the file into your /usr/src/asterisk directory. From /usr/src/asterisk run patch channels/chan_sip.c broadvoicesip.txt. As a side note: I have never been able to get outgoing calls to work with the host=proxy.XXX.broadvoice.com setting as they describe. I've always had to set it to host=sip.broadvoice.com. I'd be interested in how it works for you. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Apply Patch for Broadvoice.
Seth: Received an error when applying patch. $ patch channels/chan_sip.c broadvoicesip.txt patching file channels/chan_sip.c Reversed (or previously applied) patch detected! Assume -R? [n] Apply anyway? [n] Skipping patch. 17 out of 17 hunks ignored -- saving rejects to file channels/chan_sip.c.rej Thanks, Terrell S. Patrick www.dealerbackup.net (800) 651-4484 The best time to respond to a disaster is before it happens. A relatively small investment of time and money now may prevent severe damage and disruption of life and business in the future. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Seth Remington Sent: Friday, December 10, 2004 2:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Apply Patch for Broadvoice. On Fri, 2004-12-10 at 14:45, Dealer Backup Info wrote: Hello, I am using Broadvoice for my outgoing calls with my Asterisk box. Broadvoice is requiring my to apply a patch to my Asterisk. Instructions at the following link. http://www.broadvoice.com/support_install_asterisk.html Step 1 is what I need help with, not sure on how to apply patch. I have the rest of the instructions figured out. Copy the file into your /usr/src/asterisk directory. From /usr/src/asterisk run patch channels/chan_sip.c broadvoicesip.txt. As a side note: I have never been able to get outgoing calls to work with the host=proxy.XXX.broadvoice.com setting as they describe. I've always had to set it to host=sip.broadvoice.com. I'd be interested in how it works for you. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Odd PRI Behavior
When using a PRI, after the remote party hangs up, asterisk tries to spawn a call to the h extension. Is this normal behavior for a pri to try to call the h extension to try to clean things up? Call Comes In: -- Executing Dial(Zap/1-1, SIP/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] -- Accepting call from '6315800905' to '16464436000' on channel 0/1, span 1 -- SIP/AST-237.65-ba82 is ringing -- SIP/AST-237.65-ba82 answered Zap/1-1 -- Channel 0/1, span 1 got hangup Call has been hung up. Then this: == Spawn extension (default, 16464436000, 1) exited non-zero on 'Zap/1-1' -- Executing Dial(Zap/1-1, SIP/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] == Spawn extension (default, h, 1) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' -- Got SIP response 404 Not Found back from 65.125.237.65 -- Got SIP response 481 Call Leg Does Not Exist back from 65.125.237.65 If anyone could shed some light on this I'd appreciate it. I haven't played with PRIs enough, but I don't normally see this with pure SIP calls. -Dan Mahoney -- unless is a pr0no book he wont even come close to the bandwidth quota -Racer-X, concerning DanMahoney.com's web hits. Dan Mahoney Techie, Sysadmin, WebGeek Gushi on efnet/undernet IRC ICQ: 13735144 AIM: LarpGM Site: http://www.gushi.org --- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] More on Broadvox
Well, in lieu of dropping us, Broadvox has transferred us to their lab switch (keeping our DID's in the process). Now they're complaining that asterisk is sending a Silence-Suppression OFF request of some sort. There's no way to turn this on in asterisk is there? (Yes, I know it will shoot call quality to shit. Otherwise, does anyone know if SER works with silence suppression? -Dan Mahoney -- Dan Mahoney Techie, Sysadmin, WebGeek Gushi on efnet/undernet IRC ICQ: 13735144 AIM: LarpGM Site: http://www.gushi.org --- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] More on Broadvox
On Thu, 19 Aug 2004, Andrew Kohlsmith wrote: Yes, that's what I've told them, too. Do you know of any software that I can use as a proxy which does support this? -Dan On Thursday 19 August 2004 17:59, Dan Mahoney, System Admin wrote: Now they're complaining that asterisk is sending a Silence-Suppression OFF request of some sort. There's no way to turn this on in asterisk is there? (Yes, I know it will shoot call quality to shit. Silence suppression must be turned OFF for asterisk to function correctly; it times the audio through the received RTP stream; if silence suppression were enabled and silence was detected the RTP stream would stop and so would your audio. -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Your future hasn't been written yet; no one's has. So make it a good one! -Doc Emmet L. Browne, Back to the Future III Dan Mahoney Techie, Sysadmin, WebGeek Gushi on efnet/undernet IRC ICQ: 13735144 AIM: LarpGM Site: http://www.gushi.org --- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] More on Broadvox
On Thu, 19 Aug 2004, Andrew Kohlsmith wrote: SO what do the higher-end products use for timing? -Dan On Thursday 19 August 2004 17:59, Dan Mahoney, System Admin wrote: Now they're complaining that asterisk is sending a Silence-Suppression OFF request of some sort. There's no way to turn this on in asterisk is there? (Yes, I know it will shoot call quality to shit. Silence suppression must be turned OFF for asterisk to function correctly; it times the audio through the received RTP stream; if silence suppression were enabled and silence was detected the RTP stream would stop and so would your audio. -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ca. Tas. Tro. Phy. -John Smedley, March 28th 1998, 3AM Dan Mahoney Techie, Sysadmin, WebGeek Gushi on efnet/undernet IRC ICQ: 13735144 AIM: LarpGM Site: http://www.gushi.org --- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Color in console [SOLVED]
On Wed, 4 Aug 2004, Steven Critchfield wrote: Asterisk apparently decides whether or not to display color to users based on the TERM variable under which asterisk was launched, not under which it's being connected to, so if you start it under screen, but later connect from linux you won't see colors. The fact that the lines in term.c don't consider screen to be vt100-compatible is an interesting thing that I don't consider a bug but merely an oversight. -Dan Mahoney On Wed, 2004-08-04 at 17:56, Dan Mahoney, System Admin wrote: Hey all. I have a color-capable console (color ls works, and I can run any color-smart program like naim and bitchX), but for some reason the color in the console for asterisk, whether started with -c or safe_asterisk, isn't working for me. Any ideas as to why? check your TERM variable. I've seen apps just not like certain term values. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Of course she's gonna be upset! You're dealing with a woman here Dan, what the hell's wrong with you? -S. Kennedy, 11/11/01 Dan Mahoney Techie, Sysadmin, WebGeek Gushi on efnet/undernet IRC ICQ: 13735144 AIM: LarpGM Site: http://www.gushi.org --- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] BroadVOX
Guys, For what it's worth, after months of trying to troubleshoot issues with them, and after paying them around $2500 for setup and a down payment (it's unclear what of that will be refunded, if any) BroadVox -- http://www.broadvox.net/ -- decided to terminate our contract without any valid reason, and the only explanation they could cite was it's because of the software you're using. We use asterisk. We've asked about using other products (such as SER), and they say they don't want to accept *anything* open-source. Apparently they're a cisco shop, and maybe their TAC contract only supports getting calls from other cisco devices (but this is conjecture). I wanted to send this email for two reasons. First, to warn the community to stay away from them in general. More than once we've had issues where we were sure our configuration was dead-on, but Broadvox would be, for example, delivering different inbound DID's with different DTMF encodings, after telling us up and down that everything was right. Their support person, Alex, has an attitude the size of Montana, and is easily offended if you can't understand him (through a thick russian accent) on the phone. He refuses to communicate via email, and if you give him an attitude, he'll simply threaten you. Secondly, the ONLY good thing about Broadvox, was the rates they were offering us. Without getting into a debate here (please feel free to forward me recommendations off-list), does anyone know a good provider who is willing to provide nationwide DIDs at a reasonable cost? Thanks for all the help, -Dan Mahoney -- Dan Mahoney Techie, Sysadmin, WebGeek Gushi on efnet/undernet IRC ICQ: 13735144 AIM: LarpGM Site: http://www.gushi.org --- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SNOM 200 and Asterisk Woes
On Wed, 11 Aug 2004, Olle E. Johansson wrote: Dan Mahoney, System Admin wrote: You start up the phones, they register, all is good. They show up in sip show peers like thus: danm/danm65.125.237.91D N 255.255.255.255 5060 OK (29 ms) We pass a few calls in and out, and asterisk deadlocks (not a true deadlock, see below). The sip show peers list becomes frozen. One of two things will happen: Please explain what you mean with becomes frozen. Inbound SIP calls don't connect. Asterisk's console responds, but hangs up on a stop now and never exits cleanly except with -9. Peers are held in a constant state of whatever-state-they-were-in (ip addresses, etc). Outbound calling has a 50/50 shot of working. If the phones were registered when this happens, they continue to be able to make outbound calls. If registration has failed as a result of this happening, they are unable to re-register. 1) I can power down the phone and it will still show status OKAY. The OK in sip show peers is the result of the qualify= tests. We're sending a SIP packet to the phone and gets a response. It happens with some frequency, so Asterisk will not immediately see that the phone is off line. Turn on SIP debug and wait for the next OPTIONS sent to the phone, you will propably see a couple of retransmits before it becomes not OK. I've waited an hour with the snom phone unplugged. This isn't normal behavior. 2) Or, the other thing I'm seeing is that the phones will forget to re-register. As in, they show up in sip show peers as status UNKNOWN, but under this non-deadlock'ed deadlock, they can still make outbound calls fine. Status UNKNOWN has nothing to do with registration. If the phone hasn't registred, you will not see an IP address to the phone in 'sip show peers'. Status UNKNOWN indicates that our SIP pings is not answered. Do you have NAT between your phone and Asterisk? Seems so since you turn on qualify. I turn on qualify even for phones on the local subnet, for exactly this reason. However, yes, some of these phones were behind nat. But not all of them. In fact the one I pasting above was on the same local subnet. No firewalls. (as stated previously). -Dan Mahoney -- The first annual 5th of July party...have you been invited? It's a Jack Party. Okay, so Long Island's been invited. --Cali and Gushi, 6/23/02 Dan Mahoney Techie, Sysadmin, WebGeek Gushi on efnet/undernet IRC ICQ: 13735144 AIM: LarpGM Site: http://www.gushi.org --- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SNOM 200 and Asterisk Woes
Okay, this one is driving me nuts. I have a fedora core 1 machine running asterisk from CVS. Built last week. I have a couple of snom phones with the latest firmware. Here's the issue, it's a wierd one. You start up the phones, they register, all is good. They show up in sip show peers like thus: danm/danm65.125.237.91D N 255.255.255.255 5060 OK (29 ms) We pass a few calls in and out, and asterisk deadlocks (not a true deadlock, see below). The sip show peers list becomes frozen. One of two things will happen: 1) I can power down the phone and it will still show status OKAY. 2) Or, the other thing I'm seeing is that the phones will forget to re-register. As in, they show up in sip show peers as status UNKNOWN, but under this non-deadlock'ed deadlock, they can still make outbound calls fine. Does anyone have any idea what can cause this? -Dan Mahoney -- Hey Guys, does anyone know what 'poon tang' is? -C.S. Dave, July 8, 2K, about 12:30AM Dan Mahoney Techie, Sysadmin, WebGeek Gushi on efnet/undernet IRC ICQ: 13735144 AIM: LarpGM Site: http://www.gushi.org --- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SNOM 200 and Asterisk Woes
On Tue, 10 Aug 2004, Todd Lieberman wrote: Looks like a firewall issue too me. Some of the snoms are behind NAT. However, my test one was on the same subnet, and exhibited the same problems. The asterisk box has firewalling disabled. A firewall issue, I would think, would not cause a registration to get STUCK even when I've unplugged the snom. Also, I should point out that the classic deadlock symptom still holds: Asterisk doesn't want to exit when this happens. Any ideas? -Dan Mahoney ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Color in console
On Wed, 4 Aug 2004, Steven Critchfield wrote: On Wed, 2004-08-04 at 17:56, Dan Mahoney, System Admin wrote: Hey all. I have a color-capable console (color ls works, and I can run any color-smart program like naim and bitchX), but for some reason the color in the console for asterisk, whether started with -c or safe_asterisk, isn't working for me. Any ideas as to why? check your TERM variable. I've seen apps just not like certain term values. screen in both. That's not it. Although I wonder if installing screen will insert the correct values into the termcap file on the remote system. -Dan -- there is no loyalty in the business, so we stay away from things that piss people off -The Boss, November 12, 2002 Dan Mahoney Techie, Sysadmin, WebGeek Gushi on efnet/undernet IRC ICQ: 13735144 AIM: LarpGM Site: http://www.gushi.org --- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Color in console
Hey all. I have a color-capable console (color ls works, and I can run any color-smart program like naim and bitchX), but for some reason the color in the console for asterisk, whether started with -c or safe_asterisk, isn't working for me. Any ideas as to why? I don't think it's my termcap, although I could post that if y'all really wanted it. Asterisk was built, and is used: From CVS on a fedora core 1 system while logged in via ssh from a machine running FreeBSD 4.9, under Screen version 3.09.13 (FAU) 5-Sep-02 (screen on the BSD machine, not the core 1), and that BSD system is connected to with SecureCRT 3.2 That all SHOULDN'T matter, but I'm posting it because I've known odd things to show up in places (like phpinfo() shows the termcap I compiled php with). This is prolly a low priority, not even something I'd open as a bug, but it makes debugging quite a bit easier, so if anyone has anything, it's appreciated. -Dan -- Of course she's gonna be upset! You're dealing with a woman here Dan, what the hell's wrong with you? -S. Kennedy, 11/11/01 Dan Mahoney Techie, Sysadmin, WebGeek Gushi on efnet/undernet IRC ICQ: 13735144 AIM: LarpGM Site: http://www.gushi.org --- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 5350 One Way Sound
Hello All! I have successfully set up my Cisco 5350 for use with *! Through direct-inward-dial i have all my users dialing my number placed in Asterisk. But I have a problem - one way sound (it IS NOT a codec issue): When I call the 5350, it connects to the Asterisk, and then to the destination. I can hear the other party, but they can't hear me. I tried all codecs possible (G.711a/u, G729). I've done the tests with ATA-186 connectedto my Asterisk, using the same codec as the 5350. Any ideas? Thank you for your time in advance! Greetings, Doichin Dokov NetOne - Silistra ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco AS5350 + Asterisk Configuration
Hello, everybody! I need help connecting my Cisco AS5350 to *. What i want to do is forward all incoming calls coming from the E1 connected to the AS5350, to my * server, using SIP. How could this be done? Greetings, Doichin Dokov NetOne - Silistra ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX Connection-Latency/Bandwidth Question
Ping times (latency) and bandwidth are really not related unless you are filling the pipe. Your ping times are too high. My understanding is that anything over 100ms is not good. Your problem probably lies with too many hops or slow or overburdened router along the path. From a windows box run tracert ip and you will get a better idea where the problem lies. - Original Message - From: Dean Collins [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, March 10, 2004 8:25 PM Subject: RE: [Asterisk-Users] IAX Connection-Latency/Bandwidth Question Ok so Benjamin, can I take it from your comment about beefing up customer sales that you are going to run an off shore call centre and the majority of your calls are going to be coming externally from the pstn into your asterisk then out your dsl connection to the overseas site through their dsl to their ip handsets. Is this correct? Is the dsl service 2mbs up and down? Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Benjamin Hoskins Sent: Thursday, 11 March 2004 12:03 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] IAX Connection-Latency/Bandwidth Question Dean, I'm a newbie butYes, we are going to route all of our call from the remote office through our local office and then to the PSTN. We want to beef up our customer service so they will all be talking at the same time. Ben - Original Message - From: Dean Collins [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, March 10, 2004 4:55 PM Subject: RE: [Asterisk-Users] IAX Connection-Latency/Bandwidth Question Ben, Are you going to be routing all of your calls from the remote office back through your pabx onto the PSTN or are you assuming that they have no need for local calls? If so how many calls do you expect over the network at any one time? Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Benjamin Hoskins Sent: Thursday, 11 March 2004 11:50 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] IAX Connection-Latency/Bandwidth Question Hello! I am new to * here and have a question We are installing * currently in the US. We have a small remote office overseas. We would like to enlarge that office to about 20 people using IP phones. My questions are: 1. I pinged the website of the ISP twice from the dsl line I am on which perhaps is not as fast as our T1 line in our office but the following is the average ping times were 216ms, 229 ms, 133 ms, 142 ms, 329 ms, 251 ms ( that enough? :) ) Is anyone running IAX connection with this type of latency? How is the quality of the connection? 2. Is 2 MBPS DSL fast enough to handle the connection for 20 users? Thank you for your information. Ben ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 3com NBX phones
When an NBX100 is upgraded a .tar file is uploaded and installed on the box. Inside that tar file is the firmware for the phones which is downloaded when the phone boots. If someone can provide the last SIP firmware I will replace the phone firmware in the tar file with the SIP code and see if the phone can take it. I see alot of RMA's go through our office so no loss if it kills the phone and the NBX also retains its previous versions to boot to. - Original Message - From: Derek Bruce [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, March 04, 2004 4:22 PM Subject: Re: [Asterisk-Users] 3com NBX phones The 3com phones can't be flashed... they download their firmware image from the NBX call processor when they power on... However, if a SIP image you can have the phone download the image from a linux box using the bootloader provided by Tim Hogard at http://web.abnormal.com/~thogard/nbx100.shtml - Original Message - From: Steve Totaro [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, March 04, 2004 1:54 PM Subject: Re: [Asterisk-Users] 3com NBX phones They did make a SIP phone and are about to release new SIP phones and a new product line. The old SIP phones look identical to the NBX phones but I am not sure about the guts. Possibly the 2102 could be flashed into a 1002. Here is an ebay auction but these phones are really hard to come by. I would love to hear if a 2102 can be flashed. http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemitem=3082407185category=11909 - Original Message - From: Ejay Hire [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, March 04, 2004 12:34 PM Subject: RE: [Asterisk-Users] 3com NBX phones The original NBX100 phones spoke a proprietary voice-over-l2 ethernet protocol, but would upgrade to ip connectivity with a liscense key on the NBX PBX box. There was an optional software package that would let the NBX talk to an h323 gateway but it ran on nt and was rather klunky. These originally came out in 97 or 98, so sip functionality in the originals is rather unlikely. -e -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rob Fugina Sent: Thursday, March 04, 2004 1:58 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] 3com NBX phones On Thu, Mar 04, 2004 at 02:38:13PM -0500, Tim Sailer wrote: Does anyone know if the 3Com NBX 2102 series phones with with * ? There are a crapload (a very precise measurement) on eBay, but I can't figure out what protocols they talk. I believe they did make a version that spoke SIP, but they're rare. The ones on eBay are most likely 100% proprietary. Rob -- Rob Fugina, Systems Guy [EMAIL PROTECTED] -- http://www.geekthing.com My firewall filters MS Office attachments. 666,000,000 -- The number of the megabeast. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dialing 800 numbers with VOIP
works with voicepulse too - Original Message - From: Jeremy McNamara [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, February 09, 2004 6:39 PM Subject: Re: [Asterisk-Users] Dialing 800 numbers with VOIP Matt Lawson wrote: Hmm. Both Voicepulse and Nufone don't seem to be able to dial out 800 numbers. Are 800 numbers treated differently somehow? Or is there a business reason for disallowing them? It makes the ringing sound but never connects. You can call toll-free numbers via NuFone. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zone Paging
Bogen - Original Message - From: Brian Capouch [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, January 18, 2004 3:35 PM Subject: Re: [Asterisk-Users] Zone Paging Alfred R. Nurnberger wrote: There are a number of paging interfaces available which connect to a regular phone line on one side and to a paging amplifier on the other side. Could you provide a pointer? The search terms pager and telephone together are giving me a heck of a lot of noise. . . Thx. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] New Version of SJPhone
I just installed the new version of SJPhone and it appears that it cannot work with * anymore?
Re: [Asterisk-Users] Mailing list growth
everything is free or the cost of shipping if you think... dont worry, newbs will land at my forums but i still wanna know if i can cut and paste FAQs and the like. I plan on it so sue me, rofl. - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, January 10, 2004 10:55 AM Subject: RE: [Asterisk-Users] Mailing list growth -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philipp von Klitzing Sent: Saturday, January 10, 2004 10:35 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Mailing list growth - asterisk-users: VoIP and Asterisk in general (including newbies) - asterisk-tdm: Use if part of your problem/question involves T1/TDM - asterisk-biz: new topics, not yet really covered on -users Effects: - newbies only need to subscribe and read a lower volume -users - all readers have the same amount of traffic, but get some nice filtering help at least Reasonable, but may need some serious topic policing at first (requiring multiple list admins per list), again due to the fact that people often will not know where their problem lies. Also, just as an example.the VoIP list would have discussions on it like the recent calling card appwell, that doesn't sounds newbieish at all. Has anyone actually taken the time to do a message/category classification and breakdown to see if the proposed split even makes sense? Would we end up with 10 messages a day in -biz, 25 or so in -tdm and 100 in -users? As Robert pointed out LISTSERV has some nice topic features that could help, however the license ist costly (we have two LISTSERVs running). Let me add, though, that besides topic management LISTSERV can also provide super lists that are great to fight cross-postings - super lists group one or more normal lists or super lists. My guess is that there are other MLMs out there that have similar features. LISTSERV is evil, and yes, there are (listserv is evil mostly because of the abhorrent cost of something that is available via open source/free alternatives and a couple of perl/awk/sed scripts). Daryl G. Jurbala BMPC Network Operations Tel: +1 215 825 8401 x235 Fax: +1 508 526 8500 INOC-DBA: 26412*DGJ PGP Key: http://www.introspect.net/pgp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Chagres Technologies, Inc
Just refund the guy his money... - Original Message - From: John Brown (CV) [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, January 10, 2004 2:46 PM Subject: Re: [Asterisk-Users] Chagres Technologies, Inc For the list, Mike received a partial order shipped 15-Dec, SN ending 4CD8. Mike received email replies on 3-Dec and 17-Dec advising him on his order. Mike ack'd those emails. This is the first time we have heard anything (phone calls or email) from Mike since 17-Dec. Our CDR and SMTP logs show no inbound packets or calls. Mike has been sent a private email and has been advised that we will be issuing him a refund on product not received. I can only say that there is a human that answers the phones at Chagres M-F 9-5 MDT (GMT-7). I think I'll change the Auto-Attendent so that it says For a Human press 0, instead of To reach an operator press 0. Most people don't seem to press 0 for order status: orders AT chagres dot net, or call +1 505 830 1200 and please do leave good information (name, phone number, what you ordered) we don't always receive enough info to respond back (missing phone numbers or complete names are common) If you have any issue you can call my direct number at +1 505 998 0567. Thats my desk, ring it. cheers, john On Sat, Jan 10, 2004 at 12:08:14PM -0600, mikeu wrote: My experience has been one of unresponsiveness to my e-mails. I have ordered and received devices from other providers in the time I have been waiting for Chagres. As of now, based on my experiences and those of others that I have heard from I would highly recommend avoiding Chagres and Mr. Brown. All I want now is a refund. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olle E. Johansson Sent: Saturday, January 10, 2004 3:22 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Chagres Technologies, Inc Mail John Brown at Chagres. [EMAIL PROTECTED] He usually responds quickly and I get information about where my products are. Yes, I also have rest orders, but I have acceptable responses on why and when they are expected to arrive in this snowy winterland... /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Chagres Technologies, Inc
I would feel sympathetic to Chagres Technologies but I have read many many posts to the same effect. If you are going to take someone's money then follow through on your service or product in a timely manner. If you cannot, close your business and stop taking people's money. - Original Message - From: John Brown (CV) [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, January 10, 2004 2:56 PM Subject: Re: [Asterisk-Users] Chagres Technologies, Inc Hi List, Matt hasn't contacted us directly about this. I've responded to his previous statement that he hasn't recevied the last 20 units, and never heard back from him. Matt, again, if this is an issue please do contact us. Our CDR and SMTP logs show no such attempt. Our inventory records show 100 Grandstream Serial Numbers have been shipped to you, along with tracking numbers. +1 505 830 1200 Office Number, Auto Attendent answers Pressing 0 takes you to a operator M-F 9-5 MDT (GMT-7) orders at chagres dot net gets email into the order admin which replies within 1 biz day and you should get a auto reply. our email system now auto replys to help verify that your email did reach us. If you don't get an auto reply to the sales or order role accounts then our SMTP box didn't get your email. On Sat, Jan 10, 2004 at 12:57:43AM -0500, mattf wrote: As I've said several times on this list[insert usual apology here], I still haven't received the last 20 of 100 phones I ordered over 2 months ago. If you get a hold of them please let me know MATT--- -Original Message- From: mikeu [mailto:[EMAIL PROTECTED] Sent: Saturday, January 10, 2004 12:41 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Chagres Technologies, Inc Anyone else having problems getting product from Chagres? They took my payment almost two months ago and I still have not seen hardware. They have been horribly unresponsive to my e-mails. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Chagres Technologies, Inc
Sorry, but how can you ID his inbound packets? - Original Message - From: admin [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, January 10, 2004 3:17 PM Subject: Re: [Asterisk-Users] Chagres Technologies, Inc Just refund the guy his money... - Original Message - From: John Brown (CV) [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, January 10, 2004 2:46 PM Subject: Re: [Asterisk-Users] Chagres Technologies, Inc For the list, Mike received a partial order shipped 15-Dec, SN ending 4CD8. Mike received email replies on 3-Dec and 17-Dec advising him on his order. Mike ack'd those emails. This is the first time we have heard anything (phone calls or email) from Mike since 17-Dec. Our CDR and SMTP logs show no inbound packets or calls. Mike has been sent a private email and has been advised that we will be issuing him a refund on product not received. I can only say that there is a human that answers the phones at Chagres M-F 9-5 MDT (GMT-7). I think I'll change the Auto-Attendent so that it says For a Human press 0, instead of To reach an operator press 0. Most people don't seem to press 0 for order status: orders AT chagres dot net, or call +1 505 830 1200 and please do leave good information (name, phone number, what you ordered) we don't always receive enough info to respond back (missing phone numbers or complete names are common) If you have any issue you can call my direct number at +1 505 998 0567. Thats my desk, ring it. cheers, john On Sat, Jan 10, 2004 at 12:08:14PM -0600, mikeu wrote: My experience has been one of unresponsiveness to my e-mails. I have ordered and received devices from other providers in the time I have been waiting for Chagres. As of now, based on my experiences and those of others that I have heard from I would highly recommend avoiding Chagres and Mr. Brown. All I want now is a refund. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olle E. Johansson Sent: Saturday, January 10, 2004 3:22 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Chagres Technologies, Inc Mail John Brown at Chagres. [EMAIL PROTECTED] He usually responds quickly and I get information about where my products are. Yes, I also have rest orders, but I have acceptable responses on why and when they are expected to arrive in this snowy winterland... /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Free Software or not -- that's the question /* New subject */
I work for an interconnect that sells 3com and NEC. When I made this project my own and followed through to show my boss, he said, this is going to ruin our industry If that is the case then so be it. Same with mp3s and the music industry. Had they embraced the technology, everyone could be making a living. Now they have to sue as a last fight on the way out. Really, this is like a car that doesnt run on gas. - Original Message - From: Brian Capouch [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, January 10, 2004 3:03 PM Subject: Re: [Asterisk-Users] Free Software or not -- that's the question /* New subject */ [EMAIL PROTECTED] wrote: And why is this unnecessary cruft included in the source tree? So that Digium can leverage the Free Software community into developing proprietary software for them. Am I way off the mark? I think you're unfairly impugning Digium's motives. And I also think you're--again--salting your post with enough innuendo that a reasonable person might suspect you of flame-baiting. I suscribe to the mailing lists of several OS VoIP solutions, as I'm sure do many others on this list. There is nothing out there like asterisk, in terms of it functionality, or the body of minds that have collected to work on it. I have recently found myself embarking on a mini-career doing fundamental-level VoIP training to network operators, technology freaks, and even some small-telco tech people. I take along a laptop with asterisk on it and do a little song-and-dance that shows off some of its gee-whiz features. It is not much of an exaggeration to say that almost always people's mouths drop open in amazement at what all that asterisk can do. It's comical sometimes how affected people are. So I have all this functionality, and I have all the source code to it, and I can legally keep it forever at this (mostly happy) level of functionality, and if Digium drops off the face of the earth, I can start with what's there (we can start with what's there; I know I won't be alone) and keep going should that happen. So I can look at the same set of facts that you do, but in my mind Digium is not the nefarious would-be crook that you imply in your postings, but rather a brilliant and disruptive force upon the telco world. And they are a *business,* and as many of the people reading this sentence are bound to know, one trick of the Open Source world is to figure out how to keep things open and free and at the same time how to keep bread on the table and enough cashflow to keep up with the technology (VoIP in this case) Joneses. I cannot guess your motives, but I'm pretty sure that I *do* know what Digium's motives are, and they are innocuous and altruistic instead of the way you portray them. Where are you trying to take this? B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fw: snom in Wallstreet report
Great interview with Nicolas-Peter Pohland CEO of SNOM Chief Executive Officer - Original Message - From: Steve Totaro [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, December 06, 2003 1:01 PM Subject: FW: snom in Wallstreet report -Original Message- From: Robert Messer ABP [mailto:[EMAIL PROTECTED] Sent: Sat 12/6/2003 12:37 PM To: Steve Totaro Cc: Subject: snom in Wallstreet report Steve, I thought you might enjoy this report on snom. http://www.wallstreetreporter.com/profiles/SnomTechnology.html Regards and have a great weekend! Robert Messer ABP International, Inc. snom technology America 1-866 330 4ABP (4227) x.220 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
