Re: [asterisk-users] Server-to-server BLF

a bit curious which method is better. Regards, Ronald On Fri, Jan 13, 2012 at 3:44 PM, Leandro Dardini ldard...@gmail.com wrote: Me too, an maybe other people on the list are interested in knowing your effort result and maybe appreciate a guide on the topic. Thank you Leandro 2012/1/13

Re: [asterisk-users] Server-to-server BLF

Hi Ishfaq, Thanks for your reply. I've already started trying the XMPP method so I can't help you with the AIS method as of the moment. I'll let you know the result of my test. Regards, Ronald On Fri, Jan 6, 2012 at 5:14 PM, Ishfaq Malik i...@pack-net.co.uk wrote: Hi Ronald I took a bit

[asterisk-users] Asterisk as register server through OpenSIPS

the peer's nat=yes? I appreciate any kind of help. Thanks! Regards, Ronald -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs

Re: [asterisk-users] Server-to-server BLF

is for jabber. IMHO, it means that the XMPP solution can't be used on SIP peers, right? Regards, Ronald On Thu, Nov 17, 2011 at 1:01 AM, Kevin P. Fleming kpflem...@digium.comwrote: On 11/16/2011 04:18 AM, Ronald Cepres wrote: Hi all, Do you have an idea on the best way on how to implement

[asterisk-users] Server-to-server BLF

, Ronald -- _ -- 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

Re: [asterisk-users] Asterisk Realtime Time Dial App

Hi Nick, You mean if it is possible for Asterisk to use realtime dialplan? If it is, AFAIK it is possible using a table format for realtime extensions. Regards, Ronald On Mon, Sep 26, 2011 at 1:33 PM, Sam Govind govoi...@gmail.com wrote: Hmmm..interesting..I haven't came across anything like

[asterisk-users] Asterisk-Radius integration

if this setup is possible. Has anyone achieved this kind of setup? Thanks! Regards, Ronald -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs

Re: [asterisk-users] Asterisk-Radius integration

Hi amit, Thanks for the quick reply. I'll look into this and hopefully get this to work. Thanks again! Regards, Ronald On Wed, Sep 21, 2011 at 2:45 PM, amit anand onewaytoconn...@gmail.comwrote: Hi for this you need to write some agi script that will handle the other feature. Also you

[asterisk-users] Updated: 10 Minutes: Asterisk PBX on Amazon EC2

like to see, please let me know! It's online here: http://ronaldlewis.com/10-minutes-asterisk-pbx-on-amazon-ec2-quickstart-guide http://www.scribd.com/doc/3905321/10-Minutes-Asterisk-PBX-on-Amazon-EC2-A-Quickstart-Guide Thanks for your support! Best, Ronald Lewis Author, 10 Minutes: Asterisk PBX

[asterisk-users] how to improve sound file quality?

We have recorded wav files with 44k, 22k, 16k, 11k and 8k Asterisk does not accept these wav files. I used sox input.wav output.gsm to get them to work. However, the only the 8k file did convert and the quality is poor. How can I improve the quality? bye Ronald

[asterisk-users] Can asterisk work with a dynamic IP?

I know I can setup asterisk without Internet at all and it works as local pbx. Would an asterisk box work with a dynamic IP, with a dyndns name? What must I take care if I try that? bye Ronald ___ -- Bandwidth and Colocation Provided by http

[asterisk-users] Wellgate Asterisk

** DID_1002 No RFC3581 *CLI sip show registry *.133:5060 1002 120 Request Sent bye Ronald ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list

Re: [asterisk-users] [Solved] Wellgate Asterisk

Guillermo Salas M. wrote: El jue, 27-11-2008 a las 21:05 +0800, Ronald Wiplinger (Lists) escribió: I got a Wellgate 3804A and need some hints: Both have public IP *.131=asterisk (1.6.0.1) *.133= Wellgate Wellgate 3804A settings (Line1~Line4): I've one wellgate 3804 (old version

[asterisk-users] [SOLVED] Re: Upgrade 1.4.19 to 1.6 = segementation fault

Ronald Wiplinger (Lists) wrote: During compiling I have not seen an error, however, when I start asterisk again it ends with: app_morsecode.so = (Morse code) == Registered custom function 'SYSINFO' func_sysinfo.so = (System information related functions) Segmentation fault (core dumped

[asterisk-users] Upgrade 1.4.19 to 1.6 = segementation fault

? bye Ronald ___ -- 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] Snom - we are puzzled

we have installed asterisk and snom with PUBLIC IPs (IP/25) on one DSL line we have for our office a different ADSL with one IP shared. Two identical setup snom 360 (except the user name) with two public IP addresses are connected at the hub to the server / DSL line phone A can call B, B cannot

[asterisk-users] IP address on mysql cdr

hi, is it possible to store the IP address of the caller in the CDR? how about the end date/time? thank you. regards, ron ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix,

[asterisk-users] dundi and regcontext

hi, when a user register on my asterisk i can see it adding Noop for that extension, but after awhile i won't see it anymore: what are the reasons for it being removed on the dynamic context? one thing i found when i unregister it's removed. dialplan show myregcontext [ Context

Re: [asterisk-users] iLBC codec

Hi Sir, For this call i did not do anything except just call the extension exten = 100,1,Dial(SIP/100|20|t|M(setmusiconhold,moh-100)) that's how i dial the extension, does musiconhold make asterisk uncompress? but during the call i did not use music on hold. whereelse should i look at?

Re: [asterisk-users] DUNDI Help

],nopartial priv = dundi-static,0,SIP,[EMAIL PROTECTED],nopartial On Wed, Aug 27, 2008 at 3:56 AM, ronald ramos [EMAIL PROTECTED] wrote: Hi Again, Is there a way i can detect whether a user has been added into the regcontext? Currently i'm seeing this and just gives a fast busy. [Aug 27 16:44:46

Re: [asterisk-users] ultramonkey and asterisk

clues why the Authorization Part is not there when i use the Virtual IP? TIA Regards, ronald ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http

Re: [asterisk-users] DUNDI Help

of the registrations and any restarts on the asterisk process it may take some time for phones to re-register. On Tue, Aug 26, 2008 at 2:10 PM, ronald ramos [EMAIL PROTECTED] wrote: Hi Bruce, my apologies, but the error was because of the key. i just run keys init on the CLI and it works, question

[asterisk-users] DUNDI Help

Would like to try setting up dundi with 3-4 asterisk. But for poc, i would like to try setting up dundi on between 2 asterisk. I copied the config from DUNDI enterprise SIP with no password. Only thing i changed is the part where i used regcontext. on both boxes dundi.conf i have [mapping] priv

Re: [asterisk-users] DUNDI Help

PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Tuesday, August 26, 2008, 6:23 PM Ron, What does the peers section in dundi.conf look like? On Tue, Aug 26, 2008 at 3:00 AM, ronald ramos [EMAIL PROTECTED] wrote: Would like to try setting up

[asterisk-users] set callerid with plus sign

Hi, Is it possible to assign a plus sign on the callerid(num) ? currently this is what i do CALLERID(num)=+6523450017 but telco is denying calls, coz they said they are seeing bs523450017 instead of +6523450017. i tried putting it inside double quotes CALLERID(num)=+6523450017 telco says the

Re: [asterisk-users] set callerid with plus sign

Sessions [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] http://www.darrensessions.com _ On Aug 22, 2008, at 1:28 AM, ronald wrote: Hi, Is it possible to assign a plus sign on the callerid(num) ? currently this is what i do CALLERID(num)=+6523450017

Re: [asterisk-users] set callerid with plus sign

Amorsen wrote: ronald [EMAIL PROTECTED] writes: Is it possible to assign a plus sign on the callerid(num) ? Yes. currently this is what i do CALLERID(num)=+6523450017 but telco is denying calls, coz they said they are seeing bs523450017 instead of +6523450017. Which techology? SIP? PRI? POTS

[asterisk-users] ultramonkey and asterisk

hi all, has anyone able to configure ultramonkey for sip (namely asterisk). i tried from this tutorial: http://blog.iclutton.com/2008/01/load-balancing-and-high-availablity.html i have this on my ldirectord.cf: virtual=123.45.67.155:5060 real=123.45.67.130:5060 gate

[asterisk-users] disable auth between two asterisk

Hi, I have setup 2 asterisk talking  a single mysql cluster. I'm also using realtime db. I've setup sip peering between the two asterisk servers. [asterisk-1] insecure=port,invite type=peer host=201.202.203.204 context=from-asterisk-1 [asterisk-2] insecure=port,invite type=peer

[asterisk-users] Maybe a crazy idea, but are there Asterisk hoster outside there?

there? Maybe as a virtual machine? The mini solution does not have all features, but maybe this would still allow me to turn off another machine here. bye Ronald ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September

[asterisk-users] I used to use an Asterisk server, but now it is overkill, ...

support 7. remote gateway support I guess that is all what I would need at home. What is your suggestion for that? bye Ronald ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona

[asterisk-users] how to know what codec is being used

Hi, how would i know what codec is being utilized? currently i have set allow=ilbc disallow=all. i unset all codecs on x-lite except ilbc. i tried to make a call and look at the channel i see these. does this mean it is using ulaw? how about writetranscode? does that mean there is no

[asterisk-users] multiple asterisk approach

Hi, I'm not sure if this is the proper way to approach it but i can't figure out how to setup dundi. what i did is, i try to determine which server a user is registered, by calling an agi to query  the realtime db and capture the regserver of  the user. e.g.  exten =

[asterisk-users] simultaneous dial macro

Hi, Would just like to know if it's possible to be able to call a macro at the same time. i use a macro to dial local extension to another extension. exten = 100,Macro(dial-ext|SIP/100) exten = 101,Macro(dial-ext|SIP/101) but now i would like to use it on a simple ringgroup where it will

Re: [asterisk-users] simultaneous dial macro

/100) Set(DIALGROUP(test,add)=Local/101) Dial(${DIALGROUP(test)}) ronald ramos wrote: Hi, Would just like to know if it's possible to be able to call a macro at the same time. i use a macro to dial local extension to another extension. exten = 100,Macro(dial-ext|SIP/100) exten = 101,Macro

[asterisk-users] need help setting up dundi

Hi, Hope anyone can help me on DUNDi. I got 2 asterisk servers. configs below. tried this on the cli: *CLI dundi lookup [EMAIL PROTECTED] bypass DUNDi lookup returned no results. DUNDi lookup completed in 0 ms *CLI dundi lookup [EMAIL PROTECTED] bypass DUNDi lookup returned no results. DUNDi

[asterisk-users] Asterisk PBX How-to Guide for Amazon EC2

. The official release is set for Wednesday, July 16 and will be available on CloudCrunch. Thanks! Ronald Lewis Denver, Colorado http://ronaldlewis.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25

[asterisk-users] Play Beep if 1 minute remaining on Abosulte timeout

Hi, I have this dialpan to call international: exten =gt; _00.,1,SET(TIMEOUT(absolute)=300) exten =gt; _00.,n,Dial(SIP/[EMAIL PROTECTED]) exten =gt; _00.,n,NoCDR() exten =gt; _00.,n,Hangup Is there a way to check if there is only 1 minute remaining on the absolute timeout? also an additional

[asterisk-users] remote server with Snom 190

Ubuntu 8.04 server. What do I miss? bye Ronald ___ -- 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] trying directrtpsetup

Hi, I recently installed asterisk, i used sterisk-1.4.20.1, i i set directrtpsetup to yes, no whow would i know if the rtp/media is not passing to asterisk. any tool or can u just sniff? regards, ron ___ -- Bandwidth and Colocation

[asterisk-users] install asterisk on linux that uses software raid

hi all, we recently bought a clone box, motherboard with ICH7R raid controller (which i thought was a hardware raid controller). but recently i learned that those things are called FRAID( Fake RAID) which is basically a software raid also. so i decide to just use Software RAID (using CentOS

[asterisk-users] cdr question

Hi, Would just like to ask about cdr, i have an asterisk and i would like to bill only outbound calls not extension to extension, when i'm looking at the CDR, i can't figure out which fields i need to filter all outbound calls only. e.g if i dial 00. or 9XX (for local pstn calls) those

[asterisk-users] ring group question

Hi All, I'm trying to configure a ringgroup, which will ring the extension in the group one by one. this is what i tried on my extension.conf [macro-dial-ringgroup] exten = s,1,Dial(SIP/${ARG1},15) exten = s,n,NoOp( Dial Status: ${DIALSTATUS}) exten = s,n,Goto(s-${DIALSTATUS},1) exten

[asterisk-users] followme scenarios

Hi All, I'm tryng to test different scenarios for followme for different users: [localext] exten = 101,1,Set(FM = ALWAYS); exten = 101,n,Macro(dial-ext|SIP/${EXTEN}|vm-1|moh-101|fm-101); exten = 101,n,Hangup exten = 102,1,Set(FM = NEVER); exten =

[asterisk-users] realtime errors

Hi All, I just started playing around with asterisk realtime, added some extensions and started making test call, sometimes i can call the extension sometimes i can't. below are errors i see on the CLI, has anyone encountered this before? [settings] sippeers = mysql,sipdb,sip_customer sipusers

[asterisk-users] fax detection on sip trunk

Hi, Is it possible for me to detect fax on a sip trunk? my provider has a fax service that can send/receive fax. is it possible that i use a that trunk as a telefax? meaning i will try to detect if it's a fax, if it is i will forward it to an extension that can handle fax if not will forward it

[asterisk-users] audio disappeared after ztdummy install

Hi All, Can't explain what happened, last night i was setting the voicemail configuration, and it worked properly: -- Executing [EMAIL PROTECTED]:3] VoiceMailMain(SIP/1000100-08219db0, @VM-1000) in new stack -- SIP/1000100-08219db0 Playing 'vm-login' (language 'en') i can hear the

Re: [asterisk-users] audio disappeared after ztdummy install

Hi, For now i just turned off acpi. and it works now. just dont know what's the connection of that though :-) i will try to do the things you guys suggested also when i get the chance, thanks for you help! regards, nhadie --- Tzafrir Cohen [EMAIL PROTECTED] wrote: On Sun, Mar 30, 2008 at

[asterisk-users] rxfax does not work (anymore)

)}) exten = s,3,NoOP() exten = s,4,Set(EXTNAME=${DB(${MACRO_EXTEN}/xName)}) exten = s,5,NoOP() exten = s,6,Set(EXTCOMPANY=${DB(${MACRO_EXTEN}/xCompany)}) exten = s,7,rxfax(${FAXFILE}.tif) exten = s,103,SetVar([EMAIL PROTECTED]) exten = s,104,Goto(7) exten = s,105,SetVar(EXTNAME=Ronald) exten

[asterisk-users] Upgrade fails, need system upgrade advice

.x and the above settings. Would you suggest me to install a. OpenSuse 10.x b. Ubuntu desktop c. Ubuntu server Any other hints? to backup directories? or just use a new hard disk. With LVM? bye Ronald ___ -- Bandwidth and Colocation Provided by http

Re: [asterisk-users] dial extension number

Ronald Wiplinger wrote: Can anybody give me a hint, please. I have a Welltech FXO device and from PSTN coming calls will be transfered to the extension number 1001. I want that the caller can reach the extension number by dialing said number. My 1st try was: exten = 1001,1,NoOp

[asterisk-users] Help needed for Fax2Email with Welltech FXO 3804

(MACRO_EXTEN/xName)}) exten = s,5,NoOP() exten = s,6,Set(EXTCOMPANY=${DB(MACRO_EXTEN/xCompany)}) exten = s,7,rxfax(${FAXFILE}.tif) exten = s,103,SetVar([EMAIL PROTECTED]) exten = s,104,Goto(7) exten = s,105,SetVar(EXTNAME=Ronald) exten = s,106,Goto(7) exten = s,107,SetVar(EXTCOMPANY=Boss

[asterisk-users] Multiple fax extensions

as email body. Thanks for your hints. bye Ronald ___ -- 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] I want to record each phone call

1. Instead of using *1 (automon) I need to record each phone call at a certain * box. 2. While already talking about this. I want to autodelete with cron at 2 am in the morning all recordings which are older than 50 hours! How can I do that? bye Ronald

[asterisk-users] SVN update

? 2. Is there anything I have to take care of when updating from such an old version? Thanks! bye Ronald ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http

[asterisk-users] GTalk/Jabber passing audio in 1.4.1!

I've just compiled Asterisk 1.4.1 and I'm happy to report that I've got two-way audio between Google Talk and Asterisk! This IS an exciting moment today in VoIP! This is just GREAT! - Ronald Lewis http://ronaldlewis.com ___ --Bandwidth and Colocation

[asterisk-users] Bandwidth shapping device

I have a link to a building (e.g. 10Mb/s) and want to split up the bandwidth to different users. Each user should get e.g., 512kB/s plus 256kB/s dedicated for VoIP. What kind of device can I use for that ? (managing switch ??? which one?) bye Ronald Wiplinger

[asterisk-users] MRTG with 4 graphs

How can I set-up a MRTG with 4 graphs, whereby: 1 data in 2 data out 3 ONLY voice(/video) data in 4 ONLY voice(/video) data out bye Ronald Wiplinger ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list

[asterisk-users] moving WiFi phone

Can anybody tell me how I can set-up multiple access points with overlapping coverage, so that a moving WiFi phone user can continuesly use the phone. bye Ronald Wiplinger ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users

[asterisk-users] SMS via VoIP and web

Where can I get a starting point for setting up sms via VoIP and via web. I want to send SMS from VoIP or web to VoIP phones and GSM phones. 1. how to set-up? 2. which smsc should I use? (what is the price?) 3. which phones can be used? bye Ronald

[asterisk-users] Asterisk is used in U.S. prisons?

So says The Voice of Asterisk, Allison Smith in this new and informative interview: http://www.ronaldlewis.com/interviews/2007/01/interview-with-allison-smith-north.html (I know this isn't the most appropriate place, but Allison is about as relevant as Mark Spencer and the community)

Re: [asterisk-users] [resolved] asterisk 1,4 and google talk

mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ronald Lewis Producer, Interviews Founder and CTA, Riverscape http://www.ronaldlewis.com/interviews http://www.riverscapecorp.com

Re: [asterisk-users] Soundfiles adding during phone calls

bails wrote: Ronald Wiplinger wrote: Ronald Wiplinger wrote: Tom Lynn wrote: Ron, The guy is trying to help you. Go to the link and read it. There is a feature that you can use to play a recording into the voice channel. Mine is set so when you press #9, the caller hears the lots

Re: [asterisk-users] Soundfiles adding during phone calls

Ronald Wiplinger wrote: Tom Lynn wrote: Ron, The guy is trying to help you. Go to the link and read it. There is a feature that you can use to play a recording into the voice channel. Mine is set so when you press #9, the caller hears the lots of monkeys recording. The best part

Re: [asterisk-users] Soundfiles adding during phone calls

it records the conversion. What do I miss? bye Ronald Wiplinger On 11/11/06, * Ronald Wiplinger* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Andrew Joakimsen wrote: http://www.voip-info.org/wiki-Asterisk+config+features.conf ... and where exactly did you see this feature

Re: [asterisk-users] Soundfiles adding during phone calls

and where should Set(DYNAMIC_FEATURES=hangup#play#testfeature) be and I want that only 601 and 621 can use this feature. bye Ronald Wiplinger The best part of it is that you can hang up and the recording will continue to play to the caller. When it expires, so does the call On 11/11

Re: [asterisk-users] Soundfiles adding during phone calls

Andrew Joakimsen wrote: http://www.voip-info.org/wiki-Asterisk+config+features.conf ... and where exactly did you see this feature bye Ronald Wiplinger On 11/11/06, *Ronald Wiplinger * [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I want to add some sound filed on demand

[asterisk-users] Soundfiles adding during phone calls

each!!! Either the caller-id is not set, is 0 or is a tollfree number. bye Ronald Wiplinger ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman

[asterisk-users] Reg errors? Other anomalies? Check those capacitors!

Three months ago, I was experiencing all sorts of issues with my Asterisk box maintaining a connection to multiple trunks, etc. I also experienced various timing issues as well. In addition, Asterisk would sometimes take almost a minute to fully load and register its SIP and IAX trunks. Puzzled, I

[asterisk-users] OT: (Job) Full-Time Asterisk Opportunity

, but not requiredPlease submit your resume to ron (at) ronaldlewis.com -- I will not respond to inquiries on the list. Regards,Ronald LewisFounder and CTA, Riverscapewww.ronaldlewis.comwww.riverscapecorp.com ___ --Bandwidth and Colocation provided by Easynews.com

Re: [asterisk-users] Re: Real-time and priority n

Brian Capouch wrote: Tony Mountifield wrote: In article [EMAIL PROTECTED], Ronald Wiplinger [EMAIL PROTECTED] wrote: Is it exclusive? Either Realtime or priority n ??? If so, what is the better way? I believe 'n' is just a shorthand way of writing previous line + 1, and gets converted

[asterisk-users] Real-time and priority n

Is it exclusive? Either Realtime or priority n ??? If so, what is the better way? bye Ronald ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com

[asterisk-users] Issues with Monitor in 1.4?

-- Ronald LewisProducer, InterviewsFounder and CTA, Riverscapehttp://www.ronaldlewis.com/interviews http://www.riverscapecorp.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options

Re: [asterisk-users] Issues with Monitor in 1.4?

. On 9/30/06, Doug Lytle [EMAIL PROTECTED] wrote: Ronald Lewis wrote: ever, Asterisk dropped the recording. The same also happened with a friend yesterday. I don't like this, because I RELY on Asterisk to do Sorry, but I've gotta say it.Then you shouldn't be using BETA software in production.Doug

[asterisk-users] Context default incoming ENUM

I want to make the context [default] as an alarm, for not having set-up correct. I am looking for a way to get incoming calls via ENUM or via names (e.g. sip:[EMAIL PROTECTED]) into a defined context. How can I do that? bye Ronald

[asterisk-users] Priority n

, . exten = 617,999,hangup That would greatly help me to throw out the NoOp statements I have inserted over the time if I tested some parts, .. bye Ronald ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list

[asterisk-users] WARNING: chan_sip.c add_realm_authentication: ???

add_realm_authentication: Format for authentication entry is user[:[EMAIL PROTECTED] at line 872 [Sep 27 11:46:11] == Parsing '/etc/asterisk/sip_notify.conf': [Sep 27 11:46:11] Found What does it mean? Should I care? bye Ronald ___ --Bandwidth and Colocation

[asterisk-users] Accounting and re-invite

a rtpproxy or mediaproxy help? If how and why? bye Ronald ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] University switches to Asterisk

on its large-scale deployments.-- Ronald Lewis Producer, InterviewsFounder and CTA, Riverscapehttp://www.ronaldlewis.com/interviewshttp://www.riverscapecorp.com On 9/13/06, Doug Lytle [EMAIL PROTECTED] wrote: Interesting article I found linked from Groklaw:Sam Houston State University replaces Cisco

[asterisk-users] pickupgroup 1

context=ELMIT username=hotline secret=shhshh canreinvite=no host=dynamic ;defaultip=61.220.121.19 dtmfmode=rfc2833 [EMAIL PROTECTED] nat=yes callgroup=1 pickupgroup=1 callerid=Ronald Hotline,601 qualify=1000 and and extension 621: CREATE TABLE `sip_buddies` ( `id` int(11) NOT NULL auto_increment

[asterisk-users] Help spread the word about Asterisk!

to the top of digg.com by digging the URL below:http://digg.com/tech_news/University_Dumps_Cisco_VoIP_Moving_6_000_Students_to_Asterisk -- Ronald LewisProducer, InterviewsFounder and CTA, Riverscapehttp://www.ronaldlewis.com/interviewshttp://www.riverscapecorp.com

[asterisk-users] ASTCC: change from no pin to pin request?

Ronald ___ --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] Makefile.moddir_rules: No such file or directory

-versions/asterisk' make: *** [update] Error 2 Why is Makefile.moddir_rules missing, or what have I forgotten to do? bye Ronald ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options

[asterisk-users] ast_parse_allow_disallow: Cannot allow unknown format 'h264'

, if I use SER, would this be solved? bye Ronald ___ --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] svn trunk or branches ???

My last update was a while back and as I remember svn trunk did not compile and I was advised to use branches 1.2 till further notice. Have I missed the further notice and can we use now svn trunk or is the advice still to use branches 1.2 ??? bye Ronald

[asterisk-users] How to check which rtp ports my firewall let through?

Ronald ___ --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] Need somebody for video phone testing

I need somebody who can test with me video phone settings. I use Eyebeam! Please contact me via MSN first: [EMAIL PROTECTED] bye Ronald ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update

Re: [asterisk-users] Blind transfer 3/4 digits

Koopmann, Jan-Peter wrote: On Sunday, September 03, 2006 3:40 AM Ronald Wiplinger wrote: try that way. However, I have doubts as well. If you are right, than why snom phone does not have this problem? Would not here also the first match count? Because the transfer button

Re: [asterisk-users] Blind transfer 3/4 digits

David Gagnon wrote: Ronald, Like someone already told you, you should explain more clearly the way you try to transfer, we need more details on the procedure, using which button on which phone. We need every detail to help you. This as nothing to do with the way the dial plan is loaded

Re: [asterisk-users] Blind transfer 3/4 digits

, bye Ronald HTH routerguy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ronald Wiplinger Sent: Monday, September 04, 2006 5:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Blind transfer 3/4 digits

Re: [asterisk-users] Blind transfer 3/4 digits

have different length and overlapping, bye Ronald CP On 1-Sep-06, at 10:37 PM, Ronald Wiplinger wrote: I found a problem in blind transfer: I have an extension number 601 and I have an extension 6014 If I get a call on 615 (snom) and transfer to 6014 it works, since snom requires

Re: [asterisk-users] Blind transfer 3/4 digits

David Gagnon wrote: Ronald, You seem to be a little bit angry about VoIP. If so, I could give you my old Nortel system. Does this would make you happy? David David, I am not angry about VoIP, but please send my your old Nortel system ! I just do not understand why I can

Re: [asterisk-users] Keys pressed not registering ...

, the latest trixbox v1.1 .. FreePBX 2.1.1. Everything else works just fine. I’m using VoIPDiscount for outgoing and Stana-in/Stanaphone to receive calls. Any help is appreciated.. Have a look at the dtmfmode settings, inband, rfc2833, ... and try different settings. bye Ronald Regards

Re: [asterisk-users] Keys pressed not registering ...

Lenny wrote: Hello Ronald .. This is what I'm trying to learn of now .. Where in freepbx do I place these settings? sip.conf ;-) that was easy, ... do you have another question? bye Ronald Trunk settings? If I could just get that bit of info.. Thanks LB -Original Message

Re: [asterisk-users] Blind transfer 3/4 digits

point and I will contact the manufacturer of these no-name phones. My guess the problem lies with the Phones, not Asterisk form the information you provided. I disagree with that! Why Asterisk treats dialing and transfer different. That makes not really sense, does it? bye Ronald Kevin

Re: [asterisk-users] Blind transfer 3/4 digits

count? bye Ronald -Tim On September 2, 2006 20:12, Ronald Wiplinger wrote: Kevin Smith wrote: Dialing a number and transferring a number are two different things. And no offense, you are not really providing a lot of details along with your problem. So you can dial the numbers

[asterisk-users] Blind transfer 3/4 digits

as usually with transfer. Howerver the caller get the announcements: I could not get that, What could be the problem ? bye Ronald ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update

[asterisk-users] GIZMO and Asterisk, Failed to authenticate

? bye Ronald ___ --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] Wellgate 3804a: Got SIP response 486 Busy Here

I cannot explain why I get all the time: Got SIP response 486 Busy Here back from 192.168.250.244 I have a Wellgate 3804a there. How can I solve it? bye Ronald ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list

[asterisk-users] Am I looking for automon?

I want to record a call, either it is an incoming call or an outgoing call. I have in features.conf: automon = *1 However, I am not sure if that is what I need, and how to use it. bye Ronald ___ --Bandwidth and Colocation provided by Easynews.com

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