Re: [asterisk-users] asterisk on mini-itx
On Sunday 11 March 2007 20:04, Ira wrote: At 01:36 AM 3/11/2007, you wrote: My servers don't run anything more than they need to and don't have packages loaded that they don't need. I could rant on all day about the bloat I see in modern RH/Fedora/SuSe, even my favourite Debian systems, but this isn't the place ... I'd love to have my box running that little, but how do I figure out what's not needed and how to get rid of it? One of my frustrations with the Linux world is the apparent assumption of people that their target audience already knows what they're talking about. My LiveCD wich does not run RH/Fedora/Suse/Debian but Gentoo is about to be publically available. -- Sune Kloppenborg Jeppesen (Jaervosz) ___ --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] Problems with Voice conferencing
How did you install these packages -- make sure you do ./configure and if needed make menuselect in each one of these before the make and make install. This is the only thing I can think of -- check whether there are any built-in modules as well. on Monday 03/12/2007 Asterisk Asterisk([EMAIL PROTECTED]) wrote Hey! Thanks for your interest, i checked the modules and i could not find app_meetme anywhere could you help me Please how to get meetme application and install it to configure voice conference. I have installed asterisk-1.4.1, zaptel-1.4.0 and libpri-1.4.0 Send instant messages to your online friends http://uk.messenger.yahoo.com -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: DTMF not being detected with 1 provider. Works with the other provider...
No one...? This problem is really bugging me... :-/ Regards, Evert Evert wrote: Hi all! Working on the following brain-scratcher. I am setting up a Trixbox system for someone who uses 'provider A'. Everything works fine, except for the IVR: keypresses by callers are not being detected. Just for testing I added my own provider, 'provider B' to their system. And then the IVR works! Is there any possibility that the config on the provider-side is causing this difference? If yes, what could it be, and is there a way for me to fix this? Regards, Evert ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How many outgoing phone line/voip account do I need?
Hi list, I have an application which has to automatically dial and send out a voice message to 50 different phone numbers at the same time. Does it mean that I need to sign up 50 phone lines or voip accounts in order to achieve this purpose? Is there a provider(voip prefer) who offer a special account which is able to handle multiple calls simultaneously? Thanks in advance. Kurt _ Find a local pizza place, movie theater, and more .then map the best route! http://maps.live.com/?icid=hmtag1FORM=MGAC01 ___ --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] How many outgoing phone line/voip account do I need?
On Mon, 2007-03-12 at 09:07 +, Kurt Kuo wrote: Hi list, I have an application which has to automatically dial and send out a voice message to 50 different phone numbers at the same time. Does it mean that I need to sign up 50 phone lines or voip accounts in order to achieve this purpose? Is there a provider(voip prefer) who offer a special account which is able to handle multiple calls simultaneously? Thanks in advance. I love it, a question like this from a _hotmail_ address. Of course he could have a legit reason, but that email address. -- Dave Cotton [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Pickup group
Dears Please can you inform me by how to make a pickup group ?since all users can pick up any line ? Regards Khaled Chehab System Integration Engineer Xplorium Offshore. Sakiet Al Janzir Postal Code: 1102-2080 Tel: (961) 1- 868 686 Fax :(961) 1-808 810 GSM: (961) 3-979 343 * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. *___ --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] Coming events in Europe
Friends, This week I'll be in Lissabon speeking at a Voip Conference on Wednesday. I'm not aware if there's an Asterisk Users group in Lissabon, but if there is maybe there would be a chance to meet. Next week, I'll be at Cebit, in the Digium stand. If you want to meet me, I'll be in the stand between 2 and 4 pm together with Voop on tuesday and wednesday. Regards, /O ___ --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] How many outgoing phone line/voip account do I need?
That sounds like not quite right maths... More importantly, how many calls per day and how long per call. Then you can figure out the other bits. PaulH On Mon, 2007-03-12 at 09:07 +, Kurt Kuo wrote: Hi list, I have an application which has to automatically dial and send out a voice message to 50 different phone numbers at the same time. Does it mean that I need to sign up 50 phone lines or voip accounts in order to achieve this purpose? Is there a provider(voip prefer) who offer a special account which is able to handle multiple calls simultaneously? Thanks in advance. Kurt _ Find a local pizza place, movie theater, and more.then map the best route! http://maps.live.com/?icid=hmtag1FORM=MGAC01 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: DTMF not being detected with 1 provider. Works with the other provider...
Some VOIP providers don't pass DTMF very welland sadly it's pretty common. relaxdtmf=yes (I have never used this function) PaulH On Mon, 2007-03-12 at 09:48 +0100, Evert wrote: No one...? This problem is really bugging me... :-/ Regards, Evert Evert wrote: Hi all! Working on the following brain-scratcher. I am setting up a Trixbox system for someone who uses 'provider A'. Everything works fine, except for the IVR: keypresses by callers are not being detected. Just for testing I added my own provider, 'provider B' to their system. And then the IVR works! Is there any possibility that the config on the provider-side is causing this difference? If yes, what could it be, and is there a way for me to fix this? Regards, Evert ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How many outgoing phone line/voip account do I need?
On Mon, 2007-03-12 at 20:52 +1100, Paul Hales wrote: But top posted That sounds like not quite right maths... What maths was involved? He wants to make 50 simultaneous calls. More importantly, how many calls per day and how long per call. Then you can figure out the other bits. He wants to make 50 simultaneous calls. What difference does the length and frequency make. -- Dave Cotton [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem configuring voice conference
Hey i installed zaptel and when i tried to install asterisk and ran command menuselect it showed me that there are some discrepencies that are not being fullfilled for meetme application, but i have also installed ztdummy when i installed zaptel. I am totally stuck and nowhere to go what should i do. --- Paul Hales [EMAIL PROTECTED] wrote: Sure, but you will probably have to recompile Asterisk to get all the extra bits. Should only take you 10 minutes. later, PaulH On Mon, 2007-03-12 at 06:54 +, Asterisk Asterisk wrote: Hey! Thanks you are absolutely rite could i install ity now after i have compiled and installed asterisk or not. Send instant messages to your online friends http://uk.messenger.yahoo.com Send instant messages to your online friends http://uk.messenger.yahoo.com ___ --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] Re: Problems with Voice conferencing
We would need your exact steps in both installing zaptel and asterisk in order to help, and this is a series of steps which is quite long, so you would have to keep exact logs of what you did to both configure, make and install both zzaptel and asterisk and you would need to tell which zaptel modules you loaded, etc. on Monday 03/12/2007 Asterisk Asterisk([EMAIL PROTECTED]) wrote Thanks for the advicve, but i have done that also but the same error remains, another reply that i recieed said me to check for meetme application in the asterisk modules which i could not find how should i install the meetme application. Thanks --- John covici [EMAIL PROTECTED] wrote: How did you install these packages -- make sure you do ./configure and if needed make menuselect in each one of these before the make and make install. This is the only thing I can think of -- check whether there are any built-in modules as well. on Monday 03/12/2007 Asterisk Asterisk([EMAIL PROTECTED]) wrote Hey! Thanks for your interest, i checked the modules and i could not find app_meetme anywhere could you help me Please how to get meetme application and install it to configure voice conference. I have installed asterisk-1.4.1, zaptel-1.4.0 and libpri-1.4.0 Send instant messages to your online friends http://uk.messenger.yahoo.com -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [EMAIL PROTECTED] Send instant messages to your online friends http://uk.messenger.yahoo.com -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] _ALERT_INFO replacement in 1.4?
Hi All I have just upgraded from Asterisk 1.2 to 1.4 and am having trouble with with one of my ATAs not ringing. Basically, when I execute the Dial command, an error occurs: Got SIP response 400 In alert-info header: Empty value expected Now in 1.2, I just issued the following command to overcome this problem: Set(_ALERT_INFO=). Now in 1.4, _ALERT_INFO is deprecated, so I have to use SIPAddHeader, but I don't know how, or if there is a way to remove the alert-info header. Here is my dialplan snippet: exten = s,9,Playback(my-greeting) exten = s,10,Wait(1) exten = s,11,SIPAddHeader(Alert-Info: info=bellcore-r4) exten = s,12,Dial(SIP/600SIP/602SIP/603,60,tm) exten = s,13,Set(_ALERT_INFO=) exten = s,14,Dial(SIP/604,60,tm) exten = s,15,Voicemail(su600) exten = s,16,Hangup exten = s,115,Voicemail(sb600) exten = s,116,Hangup As you can see, #13 is deprecated, so extension 604 does not ring. Extension 600, 602 and 603 are all hooked up to Sipura ATAs and need the bellcore-r4 ringtone to differentiate from other incoming lines. Any ideas? Thanks Nikhil Jogia ___ --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] Re: Help: CallerID Name not being sent
It has been my experience when working with PRI's that you have very limited options when dealing with outbound CID. Due to restrictions of 911 most Telco's will have to have the PRI split into Trunk Groups for proper CID delivery. This would work for a situation of sharing one asterisk server between 2-3 mid sized businesses and using trunk group 1 channels 1-5 for outgoing company 1, Trunk Group 2 channels 6-10 for outgoing company 2, Trunk Group 3 11-15 Outgoing for Company 3. Channels 16-23 for incoming calls and channel 24 is your switching channel. That is if your provider will alow you to go that far with it. This may not be anywhere whatcha need but it may help someone. Matt ___ --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] Problem with H323
hymy server under heavy traffic give me the folowing error then restarts asterisk: Mar 8 21:35:39 ERROR[514]: ast_h323.cxx:169 void PAssertFunc(const char*): Assertion fail: Operating System error, file tlibthrd.cxx, line 743, Error=24 edit..and again this one... Mar 8 21:40:59 ERROR[21593]: ast_h323.cxx:169 void PAssertFunc(const char*): Assertion fail: Invalid parameter, file ../common/sockets.cxx, line 1354, Error=115 what is the problem? Sebastian BOZIOREANU ___ --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] AMI - DBPut
I'm using AMI on Asterisk 1.2.15 and I'm having problems with DBPut action. If I execute this: Action: DBPut Family: checkin Key: 316 Val: yes Response: Error Message: Missing action in request I don't put anything in Asterisk DB. If I execute this: Action: DBPut Family: checkin Key: 316 Val: yes Response: Success Message: Updated database successfully Then I put data in Asterisk DB, but that data has and . How to enter data in Asterisk DB without this brackets. fc4*CLI database show /checkin/316 : yes /dozvola/148 : yes -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Citel Handset Gateway DST fix - FYI
FYI, If you are using a Citel Handset Gateway, here is a working Time Zone rule to fix the US DST change. rule mar sun GTEQ 2 0200 -0400 nov sun GTEQ 1 0200 -0500 -- -- Steven http://www.glimasoutheast.org ___ --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] Rebooting ALL polycom phones
Hi, I know that if you have Polycom phones properly configured, you can use sip notify polycom-check-cfg SIP_REGISTRATION_ID to have the phones download the new configuration from the provisioning server and reboot. Is there anyway to send the same command to all peers (let's say I had 50 polycom phones that I wanted to reboot)? Thanks, Mike ___ --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] Shoutcast music-on-hold
Hello List I am currently testing, using a shoutcast server as source for MOH. Here is the command im using: /usr/bin/wget -q -O - http://listen.coolfm.dk:80/ | /usr/local/bin/madplay -d -Q -z -o raw:- --mono -R 44100 - | sox -r 44100 -w -s -t raw - -r 8000 -c 1 -t raw - resample vol 0.10 I know that the normal examples, only shows using madplay without sox, but the quality is s bad when I do this, compared to using SoX to do the samplerate conversion. My problem is, that everytime somebody hangs up, and nobody is using the MOH, it seems as though it stops reading data from the shoutcast server. This results in the music re-buffering from the shoutcast server, which skips the music, and in this scenario results in a re-connect to the shoutcast server. Anybody know of a solution for this? Jon ___ --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] Single sign on PC + phone?
Hi all, Does anyone have any experience with creating a Single sign on (SSO) concept where if someone logs in on their PC the phone next to that PC is also automatically assigned to that user? TIA, Patrick ___ --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] Asterisk and Databases
Hi Yes, with AGI you can do all what you need. If you know php, i suggest you phpagi http://phpagi.sourceforge.net/ But take a look at this page, you can interface with AGI with many languages http://www.voip-info.org/wiki-Asterisk+AGI Bye On 3/12/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi my friends ... I need to implement my ivr based on some decisions that would be stored in a database ... like ... to play a sound file if a boolean field in a PostgreSQL database is true ... something like that ... But how do I do that ? Is it through AGI ? If so, can you at least point me through the right direction, because I don't know much about AGI ... Thanks again folks ... Marcelo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Rebooting ALL polycom phones
for i in `seq 100 150` ; do asterisk -rx sip notify polycom-check-cfg $i ; done Julian. On 3/12/07, Mike [EMAIL PROTECTED] wrote: Hi, I know that if you have Polycom phones properly configured, you can use sip notify polycom-check-cfg SIP_REGISTRATION_ID to have the phones download the new configuration from the provisioning server and reboot. Is there anyway to send the same command to all peers (let's say I had 50 polycom phones that I wanted to reboot)? Thanks, Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 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
Re: [asterisk-users] When to use Echo Cancellation cards?
I've been running Digium TDM400P with 2 FXO and now I run a Sangoma A200 with 2 FXO and no hwec, both cards have suffered echo, one of my lines is much worse than the other. I messed about for a year using the software EC in Zaptel and whilst I could remove the echo on one line the other would always take 10-20secs to sort its self out and somtimes would never kill the echo. This situation never ever passed the wife test (that is seriouse business). I was about to bite the bullet and get a Sagoma A200D with HWEC when I discoved the new Digium software EC software called HPEC, it works a treat and at $10 per channel it is much better then spending loads of $$$ on a HWEC. I run it on a P3 650Mhz that seems to have no problem at all keeping up. hth Harvey - Original Message - From: Zeeshan Zakaria To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Friday, March 09, 2007 7:29 AM Subject: [asterisk-users] When to use Echo Cancellation cards? In what scenarios non-Echo Cancellation cards (T1/E1/FXO) should be used ? Don't all good and professional installations need echo cancellation cards? Are there people out there with non-Echo Cancellation cards for T1 or 8 FXO ports and who really don't have any echo issues and they are running serious businesses? -- Zeeshan A Zakaria -- This message has been scanned for viruses and dangerous content by ESVA, and is believed to be clean. -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: queue information into db
nik600 wrote: new update 11/03/2006 - added the module stats - updated the file db.sql with sql instructions for the creation of queue_stats table - added the files view.sql I'm in no position to test your product now. Hopefully I will find some time soon. Please keep group informed about new updates. Bye, -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DST changes for the US
SNTP tcpIpApp.sntp.resyncPeriod=86400 tcpIpApp.sntp.address= tcpIpApp.sntp.gmtOffset= tcpIpApp.sntp.daylightSavings.enable=1 tcpIpApp.sntp.daylightSavings.fixedDayEnable=0 tcpIpApp.sntp.daylightSavings.start.month=3 tcpIpApp.sntp.daylightSavings.start.date=8 tcpIpApp.sntp.daylightSavings.start.time=2 tcpIpApp.sntp.daylightSavings.start.dayOfWeek=1 tcpIpApp.sntp.daylightSavings.start.dayOfWeek.lastInMonth=0 tcpIpApp.sntp.daylightSavings.stop.month=11 tcpIpApp.sntp.daylightSavings.stop.date=1 tcpIpApp.sntp.daylightSavings.stop.time=2 tcpIpApp.sntp.daylightSavings.stop.dayOfWeek=1 tcpIpApp.sntp.daylightSavings.stop.dayOfWeek.lastInMonth=0/ I'm pretty sure this is wrong: tcpIpApp.sntp.daylightSavings.start.date=8 Should be: tcpIpApp.sntp.daylightSavings.start.date=2 which indicates second week of month. ___ --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] Re: Help: CallerID Name not being sent on outbound PRI trunk
C F wrote: Tomislav, really? and how does it show up on my POTS line? It only can be seen if other end is also on Optima provider. Ant it is shown exactly as originator has define it. It's strange when you, for the first time, get the phone call from unknown number and you see his name at your display :)) -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Rebooting ALL polycom phones
Best way I found to do this I wrote a quick bash script that takes an ip address and runs that command. Then if your phones are in an ip range, you can say something like for i in `seq 194 197`; do /usr/sbin/sipReboot 192.168.101.$i; done That will reboot 192.168.194 thru 197. Rob Mike wrote: Hi, I know that if you have Polycom phones properly configured, you can use sip notify polycom-check-cfg SIP_REGISTRATION_ID to have the phones download the new configuration from the provisioning server and reboot. Is there anyway to send the same command to all peers (let's say I had 50 polycom phones that I wanted to reboot)? Thanks, Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DST changes for the US
Hi Peder, I think that CF was correct in his original post. From the Polycom SP IP admin guide: Attribuite: tcpIpApp.sntp.daylightSavings.start.date Values permitted: 1-31 Default: 1 Description: Day of the month to start DST. What the start.date=8 does is tell the phone to start DST on the first start.dayOfWeek it finds after the start.date. So in this case, we're telling it to start DST on the first Sunday (1) after the 8th of March (making it the second Sunday in March). Alex On 3/12/07, Peder @ NetworkOblivion [EMAIL PROTECTED] wrote: SNTP tcpIpApp.sntp.resyncPeriod=86400 tcpIpApp.sntp.address= tcpIpApp.sntp.gmtOffset= tcpIpApp.sntp.daylightSavings.enable=1 tcpIpApp.sntp.daylightSavings.fixedDayEnable=0 tcpIpApp.sntp.daylightSavings.start.month=3 tcpIpApp.sntp.daylightSavings.start.date=8 tcpIpApp.sntp.daylightSavings.start.time=2 tcpIpApp.sntp.daylightSavings.start.dayOfWeek=1 tcpIpApp.sntp.daylightSavings.start.dayOfWeek.lastInMonth=0 tcpIpApp.sntp.daylightSavings.stop.month=11 tcpIpApp.sntp.daylightSavings.stop.date=1 tcpIpApp.sntp.daylightSavings.stop.time=2 tcpIpApp.sntp.daylightSavings.stop.dayOfWeek=1 tcpIpApp.sntp.daylightSavings.stop.dayOfWeek.lastInMonth=0/ I'm pretty sure this is wrong: tcpIpApp.sntp.daylightSavings.start.date=8 Should be: tcpIpApp.sntp.daylightSavings.start.date=2 which indicates second week of month. ___ --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 -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DST changes for the US
Peder @ NetworkOblivion wrote: I'm pretty sure this is wrong: tcpIpApp.sntp.daylightSavings.start.date=8 Should be: tcpIpApp.sntp.daylightSavings.start.date=2 This is what I set it to as well. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --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] DST changes for the US
Peder @ NetworkOblivion wrote: SNTP tcpIpApp.sntp.resyncPeriod=86400 tcpIpApp.sntp.address= tcpIpApp.sntp.gmtOffset= tcpIpApp.sntp.daylightSavings.enable=1 tcpIpApp.sntp.daylightSavings.fixedDayEnable=0 tcpIpApp.sntp.daylightSavings.start.month=3 tcpIpApp.sntp.daylightSavings.start.date=8 tcpIpApp.sntp.daylightSavings.start.time=2 tcpIpApp.sntp.daylightSavings.start.dayOfWeek=1 tcpIpApp.sntp.daylightSavings.start.dayOfWeek.lastInMonth=0 tcpIpApp.sntp.daylightSavings.stop.month=11 tcpIpApp.sntp.daylightSavings.stop.date=1 tcpIpApp.sntp.daylightSavings.stop.time=2 tcpIpApp.sntp.daylightSavings.stop.dayOfWeek=1 tcpIpApp.sntp.daylightSavings.stop.dayOfWeek.lastInMonth=0/ I'm pretty sure this is wrong: tcpIpApp.sntp.daylightSavings.start.date=8 Should be: tcpIpApp.sntp.daylightSavings.start.date=2 which indicates second week of month. Actually, the 8 is correct. That config block is exactly the same as what's in the default sip.cfg for SIP app 2.1.0 in which Polycom provided the new DST rules. It also seems to be working on the 1.6.7 phones I have in service. From the 2.1 admin guide: Day of the month to start DST. Mapping (on or after): 1 = the first occurrence of a given day-of-the-week in a month, 8 = the second occurrence of a given day-of-the-week in a month, 15 = the third occurrence of a given day-of-the-week in a month, 22 = the fourth occurrence of a given day-of-the-week in a month -Dave ___ --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] Rebooting ALL polycom phones
Mike wrote: Hi, I know that if you have Polycom phones properly configured, you can use sip notify polycom-check-cfg SIP_REGISTRATION_ID to have the phones download the new configuration from the provisioning server and reboot. Is there anyway to send the same command to all peers (let's say I had 50 polycom phones that I wanted to reboot)? Thanks, Mike Depending on whether you need them to reboot soon rather than *now* you can set prov.polling.enabled=1 in sip.cfg. I have all my phones check for configuration updates at 3:00am every day. If there are changes then the phone reboots automatically to get the updated configuration. I make the changes on the server, reboot my phone to test it and by the next morning the rest of the phones will be up to date. -Dave ___ --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] DST changes for the US
Alex Robar wrote: Hi Peder, I think that CF was correct in his original post. From the Polycom SP IP admin guide: What the start.date=8 does is tell the phone to start DST on the first start.dayOfWeek it finds after the start.date. So in this case, we're telling it Not according to the web interface. It says that 8 signifies the end of the month. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --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] DST changes for the US
This all depends on the setting before it: tcpIpApp.sntp.daylightSavings.fixedDayEnable=0 Since this isn't a fixed date, it isn't used the same way. It doesn't understand 'second week of the month', so if you use the 8th, it will use the next weekday of tcpIpApp.sntp.daylightSavings.start.dayOfWeek=1. If start date was set to 2, it should change your clock on the 4th. Here are the working defaults from 2.1.0: tcpIpApp.sntp.daylightSavings.enable=1 tcpIpApp.sntp.daylightSavings.fixedDayEnable=0 tcpIpApp.sntp.daylightSavings.start.month=3 tcpIpApp.sntp.daylightSavings.start.date=8 tcpIpApp.sntp.daylightSavings.start.time=2 tcpIpApp.sntp.daylightSavings.start.dayOfWeek=1 tcpIpApp.sntp.daylightSavings.start.dayOfWeek.lastInMonth=0 tcpIpApp.sntp.daylightSavings.stop.month=11 tcpIpApp.sntp.daylightSavings.stop.date=1 tcpIpApp.sntp.daylightSavings.stop.time=2 tcpIpApp.sntp.daylightSavings.stop.dayOfWeek=1 tcpIpApp.sntp.daylightSavings.stop.dayOfWeek.lastInMonth=0 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle Sent: Monday, March 12, 2007 07:12 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DST changes for the US Peder @ NetworkOblivion wrote: I'm pretty sure this is wrong: tcpIpApp.sntp.daylightSavings.start.date=8 Should be: tcpIpApp.sntp.daylightSavings.start.date=2 This is what I set it to as well. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] In Asterisk 1.4.x, Why Digium has two H323 Channels
Now, the H323 Channels is updated and your bugs fixed. But Digium still develop your OOH323 Channel. My question is why? What is the better in Asterisk 1.4.x.? I know that in Asterisk 1.2.x OOH323 is very better than H323 or OH323. Thanks in advanced. Thiago. -- THIAGO MALUF RESENDE Consultor Voip e Programador WEB (Voip Developer and Web Developer) Tel: +55 21 86042100 e-mail: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Rebooting all Aastra phones
Is there a command in Asterisk that will cause all Aastra phones to reboot and/or recheck for new firmware? ___ --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] In Asterisk 1.4.x, Why Digium has two H323 Channels
as I know, ooh323 is external project from objective systems, anyway, for 1.4 I prefer chan_h323 from asterisk tree. Thiago Maluf wrote: Now, the H323 Channels is updated and your bugs fixed. But Digium still develop your OOH323 Channel. My question is why? What is the better in Asterisk 1.4.x.? I know that in Asterisk 1.2.x OOH323 is very better than H323 or OH323. Thanks in advanced. Thiago. -- THIAGO MALUF RESENDE Consultor Voip e Programador WEB (Voip Developer and Web Developer) Tel: +55 21 86042100 e-mail: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] Rebooting all Aastra phones
On Mon, 2007-03-12 at 10:50 -0400, Matt wrote: Is there a command in Asterisk that will cause all Aastra phones to reboot and/or recheck for new firmware? In his answer for the same question on Polycom phones Julian wrote for i in `seq 100 150` ; do asterisk -rx sip notify polycom-check-cfg $i ; done if you substitute aastra for polycom? -- Dave Cotton [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Rebooting all Aastra phones
There does not seem to be an 'aastra' option in Asterisk.. that's why I'm asking if there is another way. On 3/12/07, Dave Cotton [EMAIL PROTECTED] wrote: On Mon, 2007-03-12 at 10:50 -0400, Matt wrote: Is there a command in Asterisk that will cause all Aastra phones to reboot and/or recheck for new firmware? In his answer for the same question on Polycom phones Julian wrote for i in `seq 100 150` ; do asterisk -rx sip notify polycom-check-cfg $i ; done if you substitute aastra for polycom? -- Dave Cotton [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] New to Asterisk
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? Just regular lines... not T1 or PRI? You shouldn't have any issues. 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? This is a little on the low side. Is this a system for work? If so, I'd suggest a GIG of RAM and at least a 1Ghz processor. 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? Absolutely! 4. Does fax work fine with Asterisk? Should I use one FXS module for each fax machine? Fax works ok as long as you aren't going over a high latency VoIP connection. PSTN to FXO should work ok. 5. Is the power connector on the card identical to the power connectors inside PCs? Yes, but you only need this if you hook up FXS modules. 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 ___ --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] DST 2007 Config for Cisco 7970
Anyone have a suitable configuration that takes into account the new DST changes for a Cisco 7970 (XML format) ^gtg ___ --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] Rebooting all Aastra phones
On Mon, 2007-03-12 at 11:29 -0400, Matt wrote: There does not seem to be an 'aastra' option in Asterisk.. that's why I'm asking if there is another way. Then read the excellent Aastra documentation like I and probably many others did and add the required code to sip_notify.conf substituting aastra for polycom -- Dave Cotton [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Filter IDENT(Port 113) on Linksys router puts remote extensions to one way audio
Hi all, I just changed router at my clients office and installed a Linksys router with latest firmware, which gives an option Filter IDENT(Port 113) in its firewall. If it is checked, remote SIP phones do register but audio goes one way. If I uncheck it, everything works fine. What I read on the Internet suggested to keep it turned on due to security reasons, but then how do I keep my remote extensions working? -- Zeeshan A Zakaria ___ --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] ACM question
I can telnet to the ACM on the local machine but I can't get to it from another machine I've been over the information about ACM at voip-info.org and haven't been able to figure out what I'm missing. I've included my manager.conf file and the error I'm getting from the other machine. Can anyone point out my problem? TIA Manager.conf: [general] displaysystemname = yes enabled = yes port = 5038 bindaddr = 0.0.0.0 [myuser] secret=mypass permit=0.0.0.0/0.0.0.0 read = system,call,log,verbose,command,agent,user,config write = system,call,log,verbose,command,agent,user,config Local System that works: [EMAIL PROTECTED] ~]# telnet 192.168.0.160 5038 Trying 192.168.0.160... Connected to 192-168-0-160.safedataisp.net (192.168.0.160). Escape character is '^]'. Asterisk Call Manager/1.0 ^] telnet quit Connection closed. [EMAIL PROTECTED] ~]# Remote system that doesn't work: [EMAIL PROTECTED] ~]# ping 192.168.0.160 PING 192.168.0.160 (192.168.0.160) 56(84) bytes of data. 64 bytes from 192.168.0.160: icmp_seq=1 ttl=64 time=0.298 ms 64 bytes from 192.168.0.160: icmp_seq=2 ttl=64 time=0.278 ms --- 192.168.0.160 ping statistics --- 2 packets transmitted, 2 received, 0% packet loss, time 999ms rtt min/avg/max/mdev = 0.278/0.288/0.298/0.010 ms [EMAIL PROTECTED] ~]# telnet 192.168.0.160 5038 Trying 192.168.0.160... telnet: connect to address 192.168.0.160: No route to host telnet: Unable to connect to remote host: No route to host [EMAIL PROTECTED] ~]# Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP unicode support ?
Hi! Is there unicode support in Asterisk for SIP? E.g. How can I have a displayname with special characters? E.g. if I want to have the Umlaut ä in the display name: callerid=Jeff Gräser 11 AFAIK SIP requires that the ä must be encoded using UTF-8. Thus, the ä must be encoded as 2 bytes: 1111 10100100 But Asterisk only uses one byte for the ä. Is there a way to configure Asterisk to encode the ä as UTF-8? thanks klaus -- Klaus Darilion nic.at ___ --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] Noob Question
On Sun, 2007-03-11 at 22:29 -0800, Yuan LIU wrote:
From: Thomas Patterson [EMAIL PROTECTED]
Date: Mon, 12 Mar 2007 19:03:12 +1300
I have setup my Asterisk server to have 3 outbound routes
1 being for local calls
2 being for toll calls
3 being international call
What I am wanting to do is automaticly setup if you dial a local number it
goes out on the local interface
If you dial a toll call it will go out on the tall provider.
Now for the 3 option I want it to pick up the slack of the other eg if I
have not put the dialing prefix in it will default to this trunk
Just match your local numbering plan. Don't know your country's, but in
North America (NANP), you can do
[general]
NANP = NXXNXX; 10-digit phone number starting with area code
NA_LOCAL = NXX; a North American local #
[outgoing]
exten = _${NA_LOCAL},1,NoOp(Got local number ${EXTEN})
exten = _${NA_LOCAL},n,Dial(Zap/g1/${EXTEN}); G1 is for local
exten = _${NANP),1,Goto(1${EXTEN}); for the lazy people
exten = _1${NANP},1,NoOp(Got toll number ${EXTEN})
exten = _1${NANP},n,Dial(Zap/g2/${EXTEN}); G2 is for toll
exten = _X.,1,NoOp(Likely international number ${EXTEN})
exten = _X.,1,Dial(Zap/g3/${EXTEN}); G3 is for international
Of course the real thing is a bit more complicated, if you want to count for
local toll and toll-free numbers, etc.
Small corrections: [general] should be [globals]; and in the last line,
exten = _X.,1,Dial(Zap/g3/${EXTEN}); G3 is for international
should be
exten = _X.,2,Dial(Zap/g3/${EXTEN}); G3 is for international
or
exten = _X.,n,Dial(Zap/g3/${EXTEN}); G3 is for international
because that last line will be silently dropped because it conflicts in
priority with the line previous to it.
murf
smime.p7s
Description: S/MIME cryptographic signature
___
--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] Re-parking (or transfer) a parked call
Thanks Marc, hadn't seen that one. I'm currently at 1.2.14, looks like this is in .15. Thank you very much! On 3/12/07, Marc Archer [EMAIL PROTECTED] wrote: Barry, Have a look at http://bugs.digium.com/view.php?id=8804 I am assuming that you are trying to transfer using the # key (or whatever is specified in features.conf) to re-park or transfer the parked call. I think this has been fixed in the latest versions of Asterisk 1.2 Marc. *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Barry D. Hassler *Sent:* Monday, 12 March 2007 3:13 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Re-parking (or transfer) a parked call How do you transfer or re-park a call that's been picked up from a parking lot? I don't see any options for specifying the transfer options on the parked call, so that you could transfer or repark it. -- Barry D. Hassler President, HCST http://www.hcst.net/ 937-427-9000 ___ --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 -- Barry D. Hassler President, HCST http://www.hcst.net/ 937-427-9000 ___ --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] ACM question
On Mon, 12 Mar 2007, David Ruggles wrote: I can telnet to the ACM on the local machine but I can't get to it from another machine I've been over the information about ACM at voip-info.org and haven't been able to figure out what I'm missing. I've included my manager.conf file and the error I'm getting from the other machine. Can anyone point out my problem? Maybe iptables is getting in your way? Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --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] _ALERT_INFO replacement in 1.4?
Does SIPAddHeader(Alert-Info:) not do it? On 3/12/07, Nikhil Jogia [EMAIL PROTECTED] wrote: Hi All I have just upgraded from Asterisk 1.2 to 1.4 and am having trouble with with one of my ATAs not ringing. Basically, when I execute the Dial command, an error occurs: Got SIP response 400 In alert-info header: Empty value expected Now in 1.2, I just issued the following command to overcome this problem: Set(_ALERT_INFO=). Now in 1.4, _ALERT_INFO is deprecated, so I have to use SIPAddHeader, but I don't know how, or if there is a way to remove the alert-info header. Here is my dialplan snippet: exten = s,9,Playback(my-greeting) exten = s,10,Wait(1) exten = s,11,SIPAddHeader(Alert-Info: info=bellcore-r4) exten = s,12,Dial(SIP/600SIP/602SIP/603,60,tm) exten = s,13,Set(_ALERT_INFO=) exten = s,14,Dial(SIP/604,60,tm) exten = s,15,Voicemail(su600) exten = s,16,Hangup exten = s,115,Voicemail(sb600) exten = s,116,Hangup As you can see, #13 is deprecated, so extension 604 does not ring. Extension 600, 602 and 603 are all hooked up to Sipura ATAs and need the bellcore-r4 ringtone to differentiate from other incoming lines. Any ideas? Thanks Nikhil Jogia ___ --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 -- Bruce Nortex Networks ___ --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] Rebooting all Aastra phones
At 07:50 AM 3/12/2007, you wrote: Is there a command in Asterisk that will cause all Aastra phones to reboot and/or recheck for new firmware? You can tell the phones to check ever morning at 3:00AM which is what I do or if I want it now, I just unplug the switch for a few seconds. But I only have 3 phones so it's rather a simple problem for me. I believe there is something on the WIKI or the Google Aastra list about how to make Aastra phones reboot. Ira ___ --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] Create meetme conference rooms on the flight.
Hi all, Anyone know how to dynamically create meetme conference rooms on the flight? I remembered a while ago there was a switch that tell meetme to create the conference room is the room is not defined in the meetme.conf. It doen't seem to be working for me anymore. Thnx ___ --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] GXP-2000 DST Change
In case it hasn't been posted before, here's instructions to get the correct time to show up on your Grandstream GXP-2000's: 1. Login to phone 2. Go to Basic Settings tab 3. Change Daylight Savings Time to yes 4. Change Optional Rule to 3,2,7,2,0;11,1,7,2,0;60 (this means change clocks the second sunday of March and back again the first sunday of November - i.e., the new savings times). For additional reference, here's how to configure the 'optional rule' field from the manual at http://support.vocalocity.com/grandstream_guide.pdf: This parameter controls whether the displayed time will be daylight savings time or not. If set to Yes and the Optional Rule is empty, then the displayed time will be 1 hour ahead of normal time. The Automatic Daylight Saving Time Rule shall have thefollowing syntax:start-time;end-time;saving Both start-time and end-time have the same syntax:month,day,weekday,hour,minute month: 1,2,3,..,12 (for Jan, Feb, .., Dec) day: [+|-]1,2,3,..,31 weekday: 1, 2, 3, .., 7 (for Mon, Tue, .., Sun), or 0 which means the daylight saving rule is not based on week days but based on the day of the month. hour: hour (0-23), minute: minute (0-59) If weekday is 0, it means the date to start or end daylight saving is at exactly the given date. In that case, the day value must not be negative. If weekday is not zero and day is positive, then the daylight saving starts on the first dayth iteration of the weekday (1st Sunday, 3rd Tuesday etc). If weekday is not zero and day is negative, then the daylight saving starts on the last dayth iteration of the weekday (last Sunday, 3rd last Tuesday etc). The saving is in the unit of minutes. The saving time may also be preceded by a negative (-) sign if subtraction is desired instead of addition. The default value for Automatic Daylight Saving Time Rule shall be set to 04,01,7,02,00;10,-1,7,02,00;60 which is the rule for US. Examples US/Canada where daylight saving time is applicable: 04,01,7,02,00;10,-1,7,02,00;60 This means the daylight saving time starts from the first Sunday of April at 2AM and ends the last Sunday of October at 2AM. The saving is 60 minutes (1hour). Hope this saves someone a bit of time, Ken ___ --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] Rebooting ALL polycom phones
Thanks Dave, good info! And thanks to those who confirmed I needed to write a script because there were no built in functions, I appreciate that info too. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Fullerton Sent: Monday, March 12, 2007 10:26 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Rebooting ALL polycom phones Mike wrote: Hi, I know that if you have Polycom phones properly configured, you can use sip notify polycom-check-cfg SIP_REGISTRATION_ID to have the phones download the new configuration from the provisioning server and reboot. Is there anyway to send the same command to all peers (let's say I had 50 polycom phones that I wanted to reboot)? Thanks, Mike Depending on whether you need them to reboot soon rather than *now* you can set prov.polling.enabled=1 in sip.cfg. I have all my phones check for configuration updates at 3:00am every day. If there are changes then the phone reboots automatically to get the updated configuration. I make the changes on the server, reboot my phone to test it and by the next morning the rest of the phones will be up to date. -Dave ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Rebooting all Aastra phones
Yup.. got mine set to reboot every morning at 3am and check for updates... just was curious :) On 3/12/07, Ira [EMAIL PROTECTED] wrote: At 07:50 AM 3/12/2007, you wrote: Is there a command in Asterisk that will cause all Aastra phones to reboot and/or recheck for new firmware? You can tell the phones to check ever morning at 3:00AM which is what I do or if I want it now, I just unplug the switch for a few seconds. But I only have 3 phones so it's rather a simple problem for me. I believe there is something on the WIKI or the Google Aastra list about how to make Aastra phones reboot. Ira ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] ACM question
Thanks! That was the problem. Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Edwards Sent: Monday, March 12, 2007 12:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ACM question On Mon, 12 Mar 2007, David Ruggles wrote: I can telnet to the ACM on the local machine but I can't get to it from another machine I've been over the information about ACM at voip-info.org and haven't been able to figure out what I'm missing. I've included my manager.conf file and the error I'm getting from the other machine. Can anyone point out my problem? Maybe iptables is getting in your way? Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re-parking (or transfer) a parked call
Barry D. Hassler wrote: Thanks Marc, hadn't seen that one. I'm currently at 1.2.14, looks like this is in .15. Actually, It didn't start working correctly for me until 1.2.16 Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --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] Rebooting all Aastra phones
There is an aastra option but only for individual extensions which you may already know about. Perhaps if you try the same wildcards as suggested for the polycom it may work? nano /etc/asterisk/sip_notify.conf [aastra-check-cfg] Event=check-sync Content-Length=0 Then from the asterisk console you can type sip notify aastra-check-cfg $i I don't have a system I can check this on right now though. -Original Message- From: Matt [mailto:[EMAIL PROTECTED] Sent: Monday, March 12, 2007 8:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Rebooting all Aastra phones There does not seem to be an 'aastra' option in Asterisk.. that's why I'm asking if there is another way. On 3/12/07, Dave Cotton [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: On Mon, 2007-03-12 at 10:50 -0400, Matt wrote: Is there a command in Asterisk that will cause all Aastra phones to reboot and/or recheck for new firmware? In his answer for the same question on Polycom phones Julian wrote for i in `seq 100 150` ; do asterisk -rx sip notify polycom-check-cfg $i ; done if you substitute aastra for polycom? -- Dave Cotton [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FW: Seamless Multi Office Asterisk Deployment
Hello I have a brief and a long question about a possible Asterisk deployment I am planning. Long Story Short: I have four total offices, one main and three remote. All offices are connected using dedicated network T1 lines creating one unified network across offices. I would like to know if it is possible to set up an Asterisk system with the following capabilities: - Branch Unification (I know this can be done) - Branch Independence (In case of T1 network Failure, PSTN line failover at each branch) - Roaming Extensions (A user can go to any office and log in to a phone - hopefully check voice mail as well) Basically, I am asking if Asterisk can be a system that will seamlessly operate as one big system and handle failovers as well. After spending hours playing with Asterisk, reading voip-info.org, and watching this list, it seems that Asterisk can handle anything. I just would like re-assurance that I am not chasing a lost cause. A simple Yes or No answer is acceptable to me. Below I have a long version of what I am trying to do if anyone is in the mood to give me more pointers :-) Brandon (Long Version Follows) Long Story Version: Here is what I have to work with: - Four Offices (One main and three remote) - Dedicated T1 lines connecting three remote offices to one main office (all connections made through the main office) - Will have a T1 Voice line at the main office - Three PSTN lines at each remote office Essentially what I would like to do is create a system comparable to the ShoreTel ShoreGear product line (if you are familiar with it). This system will seamlessly unite all offices as one and provide failover in the case of line outage. It also allows users to roam from phone to phone across offices seamlessly. It has many more features, but those are two main features I am looking for. About 40 total phones will be deployed. To make it even more difficult, I would like all user extensions to start the same (i.e. across offices all extensions are 5### with no discernable pattern). Progress so far: At this time I have determined that I will need a server at each office as well as a T1 card (TE110P) at the Main office and the four port TDM PSTN cards at each remote office. I plan on using the Polycom IP 430 or 501 (Undecided, 501 if required). I have been using TrixBox to this point, would like to continue if possible. It appears that I will want to use DunDi in some fashion to unite the branches. My main roadblock right now is trying to figure out how to get all the information across the offices at the same time (extensions, voicemail). I have successfully had two boxes communicate, but what I am looking for is much more complex I feel. I have thought of synchronized MySQL databases, but I do not know if that will work the way I wish. If anyone reads this far ;) I am looking for suggestions for routes I might consider or places I should/could look for more information. I am relatively new to Asterisk, but I am not afraid to get my hands dirty. If something I said did not make any sense or if there is more information I could provide, I am happy to help where I can. Thank you for your time and assistance. Brandon Comouche An IT Guy ___ --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] OT: Sipura DST Rules
Since we've had discussion about DST on polycom I thought I'd pass along the rule I used to configure DST on my sipura units as well (This way the date and time passed in caller ID will be correct). Under the admin view go to the regional tab. At the bottom under miscellaneous enter this in Daylight Saving Time Rule: start=3/8/7/2:0:0;end=11/1/7/2:0:0;save=1 This is based off information I found here: http://www.sipura.com/Documents/faq/Section_5.html -Dave ___ --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] Create meetme conference rooms on the flight.
You should take a look at the d or the D option of Meetme application ;) Regards, Tristan Mahé Wai Wu a écrit : Hi all, Anyone know how to dynamically create meetme conference rooms on the flight? I remembered a while ago there was a switch that tell meetme to create the conference room is the room is not defined in the meetme.conf. It doen't seem to be working for me anymore. Thnx ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP unicode support ?
12 mar 2007 kl. 17.05 skrev Klaus Darilion: Hi! Is there unicode support in Asterisk for SIP? E.g. How can I have a displayname with special characters? No. I made a proposal for it a long time ago, without much comments. We need to fix this, since IAX2 now by spec is UTF8 too. In order to do it right we need to be able to convert since PSTN Caller ID Names is *not* UTF8. I started working on something for this in my Astum branch but got lost in libiconv. Må så bra! /O ;-) E.g. if I want to have the Umlaut ä in the display name: callerid=Jeff Gräser 11 AFAIK SIP requires that the ä must be encoded using UTF-8. Thus, the ä must be encoded as 2 bytes: 1111 10100100 But Asterisk only uses one byte for the ä. Is there a way to configure Asterisk to encode the ä as UTF-8? thanks klaus -- Klaus Darilion nic.at ___ --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 --- * Olle E Johansson - [EMAIL PROTECTED] * Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden ___ --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] asterisk on mini-itx
How many simultaneous calls will this device support and with which codecs/transcoding? Do you sell the hardware stand-alone without your software so we can load our own version of Asterisk/Gui? On 3/12/07, Ioan Biris [EMAIL PROTECTED] wrote: Hi , We have done exactly that … fan less , VIA processor , flash card , firewall. http://www.allo.com/products/micropbx.php We sell wholesale. Ioan at allo.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mail Lists Sent: Saturday, March 10, 2007 11:41 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] asterisk on mini-itx Hello, I'm trying to put together a low cost - low powers PBX appliance for several customers. I have purchased a couple of the soekris net4801 boards and have asterisk up and running on them fine but they just don't quite cut it in the processing power department. I've been able to get about 10 simultaneous SIP calls with simple ulaw (no encoding decoding). While this might be OK for a very small business or home I just don't think it leaves a lot of overhead to do anything else. I've had a look around and I think I have settled on one of the VIA EPIA fanless boards. Does anyone have any experience with these running asterisk as far as performance and reliability is concerned? Has anyone run asterisk with any compressed codecs on this setup? I am going to TRY to run the system from flash memory one way or another - I realize the hoops I might have to jump through to prevent a large number of read/write cycles but I'd really like to have the whole thing solid state... Maybe someone has a better idea regarding program storage? Also, I would really like to run this as a router/firewall appliance as well so that that the box can sit on a public IP if the client only has one. For this reason I kind of have my heart set on openbsd. The routing and firewall utilities on openbsd are very simple to configure and easy to use. Does anyone know what limitations asterisk might have on openbsd (besides lack of zaptel.. ) ? I have run asterisk 1.2.? on openbsd before and found it worked pretty well. Failing that I suppose I would settle for running the routing/firewalling on linux. I've just found the linux networking tools very awkward up until now - perhaps someone know of a linux distribution - or tool - that makes routing/firewall/NAT as painless as on openbsd? Maybe I just need to sit down for a day and learn the tool properly ;) Anyways, I know there are a lot of questions in here but perhaps someone has done one or all of these things? Thanks for any advice or warnings! Steve Glaus ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] LIDB/CNAM STORAGE DATABASE NEEDED
Hi Folks, I am in need of someone who can provide me with a LIDB / CNAM storage database.I will set the pointcode on my numbers to point to your database, and then I need to be able to update my numbers in your database. If you are able to offer these services, please contact me. Matt Hoppes 1-866-678-6858 x 126 [EMAIL PROTECTED] Thanks! ___ --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] GXV3000 Speakphone
I have about 4 GrandStream GXV 3000 video phones. I have been trying to integrate them into our asterisk environment. All works fine until someone trys to use the speakphone function or the conference call function (conferencing to inbound call together). As soon as they try either function previously listed the sound quality goes to garbage. We are using gsm as the codec. I have not tried to use the video support as we are stilling on asterisk 1.2. Has anyone experienced this problem? ___ --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] Incase anyone wanted it - SNOM USA DST settings
add this to your config dst: 3600 03.02.07 02:00:00 11.01.07 02:00:00 -- --- Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! Hind sight is most always 20/20 or better. --- ___ --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] OT: Sipura DST Rules
Thanks a million! Just verified after putting it in my encrypted configs and it works like a charm! :) Curt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Fullerton Sent: Monday, March 12, 2007 12:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] OT: Sipura DST Rules Since we've had discussion about DST on polycom I thought I'd pass along the rule I used to configure DST on my sipura units as well (This way the date and time passed in caller ID will be correct). Under the admin view go to the regional tab. At the bottom under miscellaneous enter this in Daylight Saving Time Rule: start=3/8/7/2:0:0;end=11/1/7/2:0:0;save=1 This is based off information I found here: http://www.sipura.com/Documents/faq/Section_5.html -Dave ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] deprecated ALERT_INFO var andAMI's Originate command
Since 1.4 ALERT_INFO variable has been deprecated. I used to send this via AMI: Action: Originate Channel: Sip/1234 Application: AgentLogin Data: 1234 Variable: _ALERT_INFO=info=alert-autoanswer Callerid: AutoLogin[1234] In order to send an autologin and autoanswer call to the agent 1234 on an Aastra phone at extension 1234. (just for example). Now in * 1.4 with ALERT_INFO deprecated I don't have that option, rather than calling a Local/ channel to SetSIPheaders() and Dial(). I don't want to do it in that fashion 'cos I like (and have) to have completely separated dial plan logic (extensions.conf) and external applications via AMI. Regards, -- I'm having a MID-WEEK CRISIS! ___ --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] Re: Filter IDENT(Port 113) on Linksys router puts remote extensions to one way audio
One solution I read about it is to forward the port 113 to an unused IP address. I did so and now phones seems to be working again. ___ --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] Polycom: warble on registration?
Hi, all. I just upgraded my sip.cfg for my Polycoms, and that damn warble on registration(? -- maybe it's on acquiring an IP?) has started again. I still have the old sip.cfg, but can't figure out which option it is. Any help? Thanks! -Ken D'Ambrosio -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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
Re: [asterisk-users] GXP-2000 DST Change
Thanks for the info, Ken. I was about to research this tonight. Todd On Mar 12, 2007, at 12:53 PM, Ken Williams wrote: In case it hasn't been posted before, here's instructions to get the correct time to show up on your Grandstream GXP-2000's: 1. Login to phone 2. Go to Basic Settings tab 3. Change Daylight Savings Time to yes 4. Change Optional Rule to 3,2,7,2,0;11,1,7,2,0;60 (this means change clocks the second sunday of March and back again the first sunday of November - i.e., the new savings times). -snip-___ --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] deprecated ALERT_INFO var andAMI's Originate command
12 mar 2007 kl. 19.16 skrev Octavio Ruiz (Ta^3): Since 1.4 ALERT_INFO variable has been deprecated. I used to send this via AMI: Action: Originate Channel: Sip/1234 Application: AgentLogin Data: 1234 Variable: _ALERT_INFO=info=alert-autoanswer Callerid: AutoLogin[1234] In order to send an autologin and autoanswer call to the agent 1234 on an Aastra phone at extension 1234. (just for example). Now in * 1.4 with ALERT_INFO deprecated I don't have that option, rather than calling a Local/ channel to SetSIPheaders() and Dial(). I don't want to do it in that fashion 'cos I like (and have) to have completely separated dial plan logic (extensions.conf) and external applications via AMI. You can still add a variable like that, since sipaddheader is based on using channel variables. If you add _SIPADDHEADER55 it will be used. Feel free to replace 55 with any two-digit number, as long as it's unique. This is how the SIP addheader functionality works on the inside. /O ___ --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] Re: DTMF not being detected with 1 provider. Works with the other provider...
Evert wrote: No one...? This problem is really bugging me... :-/ Regards, Evert Evert wrote: Hi all! Working on the following brain-scratcher. I am setting up a Trixbox system for someone who uses 'provider A'. Everything works fine, except for the IVR: keypresses by callers are not being detected. Just for testing I added my own provider, 'provider B' to their system. And then the IVR works! Is there any possibility that the config on the provider-side is causing this difference? If yes, what could it be, and is there a way for me to fix this? Yes, there's a strong possibility. DTMF problems are common with VOIP providers; I've had personal experience with them. You might try talking nicely with them to see if they might help you diagnose the problem; alternatively, explore moving your customer over to provider B. _Stephen- ___ --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] Polycom: warble on registration?
Ken D'Ambrosio wrote:
Hi, all. I just upgraded my sip.cfg for my Polycoms, and that damn warble
on registration(? -- maybe it's on acquiring an IP?) has started again.
I still have the old sip.cfg, but can't figure out which option it is.
Any help?
Ken: It's really smart to follow Polycom's guide for managing
configuration files. Keep all your customized settings in a pre-loaded
.cfg file and make sure that your {MACADDR}.cfg files specify it.
Then you can do upgrades with a greatly reduced possibility that they
will break your setup.
-Stephen-
___
--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] Re: Help: CallerID Name not being sent on outbound PRI trunk
OK, this makes sense, as I have seen this here in the US as well. On 3/12/07, Tomislav Parcina [EMAIL PROTECTED] wrote: C F wrote: Tomislav, really? and how does it show up on my POTS line? It only can be seen if other end is also on Optima provider. Ant it is shown exactly as originator has define it. It's strange when you, for the first time, get the phone call from unknown number and you see his name at your display :)) -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] Rebooting all Aastra phones
Hi Matt, Probably you already found it - but I think it could help others: http://www.voip-info.org/wiki/view/Aastra+Failsafe+Reboot+Script You have to give the password - but for us it was OK. Best regards, ## nini @ www.modulo.ro ## Matt wrote: Is there a command in Asterisk that will cause all Aastra phones to reboot and/or recheck for new firmware? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FW: Seamless Multi Office Asterisk Deployment
Brandon,
You're certainly inviting a challenge.
All you describe is possible with PBXware from www.bicomsystems.com a lot, lot
more of course and some helping hands.
Please contact me offline if you prefer for more detail.
Regards,
Steve
steve 'at' bicomsystems {dot} com
- Original Message -
From: Brandon Comouche
To: asterisk-users@lists.digium.com
Sent: Monday, March 12, 2007 6:11 PM
Subject: [asterisk-users] FW: Seamless Multi Office Asterisk Deployment
Hello
I have a brief and a long question about a possible Asterisk deployment I am
planning.
Long Story Short:
I have four total offices, one main and three remote. All offices are
connected using dedicated network T1 lines creating one unified network across
offices. I would like to know if it is possible to set up an Asterisk system
with the following capabilities:
- Branch Unification (I know this can be done)
- Branch Independence (In case of T1 network Failure, PSTN line failover at
each branch)
- Roaming Extensions (A user can go to any office and log in to a phone -
hopefully check voice mail as well)
Basically, I am asking if Asterisk can be a system that will seamlessly
operate as one big system and handle failovers as well.
After spending hours playing with Asterisk, reading voip-info.org, and
watching this list, it seems that Asterisk can handle anything. I just would
like re-assurance that I am not chasing a lost cause. A simple Yes or No answer
is acceptable to me. Below I have a long version of what I am trying to do if
anyone is in the mood to give me more pointers J
Brandon
(Long Version Follows)
Long Story Version:
Here is what I have to work with:
- Four Offices (One main and three remote)
- Dedicated T1 lines connecting three remote offices to one main office (all
connections made through the main office)
- Will have a T1 Voice line at the main office
- Three PSTN lines at each remote office
Essentially what I would like to do is create a system comparable to the
ShoreTel ShoreGear product line (if you are familiar with it). This system will
seamlessly unite all offices as one and provide failover in the case of line
outage. It also allows users to roam from phone to phone across offices
seamlessly. It has many more features, but those are two main features I am
looking for. About 40 total phones will be deployed. To make it even more
difficult, I would like all user extensions to start the same (i.e. across
offices all extensions are 5### with no discernable pattern).
Progress so far:
At this time I have determined that I will need a server at each office as
well as a T1 card (TE110P) at the Main office and the four port TDM PSTN cards
at each remote office. I plan on using the Polycom IP 430 or 501 (Undecided,
501 if required). I have been using TrixBox to this point, would like to
continue if possible. It appears that I will want to use DunDi in some fashion
to unite the branches.
My main roadblock right now is trying to figure out how to get all the
information across the offices at the same time (extensions, voicemail). I have
successfully had two boxes communicate, but what I am looking for is much more
complex I feel. I have thought of synchronized MySQL databases, but I do not
know if that will work the way I wish.
If anyone reads this far ;) I am looking for suggestions for routes I might
consider or places I should/could look for more information. I am relatively
new to Asterisk, but I am not afraid to get my hands dirty. If something I said
did not make any sense or if there is more information I could provide, I am
happy to help where I can. Thank you for your time and assistance.
Brandon Comouche
An IT Guy
--
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] FW: Seamless Multi Office Asterisk Deployment
If you're up for it, I've done this a few times before , and with asterisk. Contact me offlist, and I can help. - Original Message - From: Brandon mailto:[EMAIL PROTECTED] Comouche To: asterisk-users@lists.digium.com Sent: Monday, March 12, 2007 6:11 PM Subject: [asterisk-users] FW: Seamless Multi Office Asterisk Deployment Hello I have a brief and a long question about a possible Asterisk deployment I am planning. Long Story Short: I have four total offices, one main and three remote. All offices are connected using dedicated network T1 lines creating one unified network across offices. I would like to know if it is possible to set up an Asterisk system with the following capabilities: - Branch Unification (I know this can be done) - Branch Independence (In case of T1 network Failure, PSTN line failover at each branch) - Roaming Extensions (A user can go to any office and log in to a phone - hopefully check voice mail as well) Basically, I am asking if Asterisk can be a system that will seamlessly operate as one big system and handle failovers as well. After spending hours playing with Asterisk, reading voip-info.org, and watching this list, it seems that Asterisk can handle anything. I just would like re-assurance that I am not chasing a lost cause. A simple Yes or No answer is acceptable to me. Below I have a long version of what I am trying to do if anyone is in the mood to give me more pointers :-) Brandon (Long Version Follows) Long Story Version: Here is what I have to work with: - Four Offices (One main and three remote) - Dedicated T1 lines connecting three remote offices to one main office (all connections made through the main office) - Will have a T1 Voice line at the main office - Three PSTN lines at each remote office Essentially what I would like to do is create a system comparable to the ShoreTel ShoreGear product line (if you are familiar with it). This system will seamlessly unite all offices as one and provide failover in the case of line outage. It also allows users to roam from phone to phone across offices seamlessly. It has many more features, but those are two main features I am looking for. About 40 total phones will be deployed. To make it even more difficult, I would like all user extensions to start the same (i.e. across offices all extensions are 5### with no discernable pattern). Progress so far: At this time I have determined that I will need a server at each office as well as a T1 card (TE110P) at the Main office and the four port TDM PSTN cards at each remote office. I plan on using the Polycom IP 430 or 501 (Undecided, 501 if required). I have been using TrixBox to this point, would like to continue if possible. It appears that I will want to use DunDi in some fashion to unite the branches. My main roadblock right now is trying to figure out how to get all the information across the offices at the same time (extensions, voicemail). I have successfully had two boxes communicate, but what I am looking for is much more complex I feel. I have thought of synchronized MySQL databases, but I do not know if that will work the way I wish. If anyone reads this far ;) I am looking for suggestions for routes I might consider or places I should/could look for more information. I am relatively new to Asterisk, but I am not afraid to get my hands dirty. If something I said did not make any sense or if there is more information I could provide, I am happy to help where I can. Thank you for your time and assistance. Brandon Comouche An IT Guy _ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FW: Seamless Multi Office Asterisk Deployment
Brandon Your on the right track with what is can do. It will also be good to look into what kind of QOS you can do on the T-1 connections between offices. I have an 8 office setup similar to this and many of your goals I have achieved and would be glad to offer ideas and such if you want to email me off list. On 3/12/07, Brandon Comouche [EMAIL PROTECTED] wrote: Hello I have a brief and a long question about a possible Asterisk deployment I am planning. Long Story Short: I have four total offices, one main and three remote. All offices are connected using dedicated network T1 lines creating one unified network across offices. I would like to know if it is possible to set up an Asterisk system with the following capabilities: - Branch Unification (I know this can be done) - Branch Independence (In case of T1 network Failure, PSTN line failover at each branch) - Roaming Extensions (A user can go to any office and log in to a phone – hopefully check voice mail as well) Basically, I am asking if Asterisk can be a system that will seamlessly operate as one big system and handle failovers as well. After spending hours playing with Asterisk, reading voip-info.org, and watching this list, it seems that Asterisk can handle anything. I just would like re-assurance that I am not chasing a lost cause. A simple Yes or No answer is acceptable to me. Below I have a long version of what I am trying to do if anyone is in the mood to give me more pointers J Brandon (Long Version Follows) Long Story Version: Here is what I have to work with: - Four Offices (One main and three remote) - Dedicated T1 lines connecting three remote offices to one main office (all connections made through the main office) - Will have a T1 Voice line at the main office - Three PSTN lines at each remote office Essentially what I would like to do is create a system comparable to the ShoreTel ShoreGear product line (if you are familiar with it). This system will seamlessly unite all offices as one and provide failover in the case of line outage. It also allows users to roam from phone to phone across offices seamlessly. It has many more features, but those are two main features I am looking for. About 40 total phones will be deployed. To make it even more difficult, I would like all user extensions to start the same (i.e. across offices all extensions are 5### with no discernable pattern). Progress so far: At this time I have determined that I will need a server at each office as well as a T1 card (TE110P) at the Main office and the four port TDM PSTN cards at each remote office. I plan on using the Polycom IP 430 or 501 (Undecided, 501 if required). I have been using TrixBox to this point, would like to continue if possible. It appears that I will want to use DunDi in some fashion to unite the branches. My main roadblock right now is trying to figure out how to get all the information across the offices at the same time (extensions, voicemail). I have successfully had two boxes communicate, but what I am looking for is much more complex I feel. I have thought of synchronized MySQL databases, but I do not know if that will work the way I wish. If anyone reads this far ;) I am looking for suggestions for routes I might consider or places I should/could look for more information. I am relatively new to Asterisk, but I am not afraid to get my hands dirty. If something I said did not make any sense or if there is more information I could provide, I am happy to help where I can. Thank you for your time and assistance. Brandon Comouche An IT Guy ___ --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 -- Bruce Nortex Networks ___ --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] _ALERT_INFO replacement in 1.4?
Bruce Reeves wrote: Does SIPAddHeader(Alert-Info:) not do it? No, but from another thread, setting the _SIPADDHEADER variable works. ___ --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] Fix for TZ values updates for DST
Somebody could help me with a call back implementation, please? I mean, I just want call to my Asterisk, hung up the phone, and wait it calls me back... Somebody ever did that for local or international calls? Thanks. LC. ___ --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] Call Back
Somebody could help me with a call back implementation, please? I mean, I just want call to my Asterisk, hung up the phone, and wait it calls me back... Somebody ever did that for local or international calls? Thanks. LC. ___ --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] AMI - DBPut
Tomislav Parcina wrote: I'm using AMI on Asterisk 1.2.15 and I'm having problems with DBPut action. If I execute this: Action: DBPut Family: checkin Key: 316 Val: yes Try putting quotes around the value. I played with it a while back only a little, but I can't remember if quotes did it or I ended up having stripping the quotes off myself when I retrieved the value ... -- Warm Regards, Lee ___ --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] Fix for TZ values updates for DST
Please start new threads for new questions. Alex On 3/12/07, Luis Claudio Santos [EMAIL PROTECTED] wrote: Somebody could help me with a call back implementation, please? I mean, I just want call to my Asterisk, hung up the phone, and wait it calls me back... Somebody ever did that for local or international calls? Thanks. LC. ___ --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 -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: Spam? Re: [asterisk-users] cmd page crashes AsteriskSVN-branch-1.4-r57207
Just wanted to update the group I updated asterisk to SVN-branch-1.4-r58833M and page no longer crashes Asterisk. My below example works great. Thanks! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hall, Eric M. Sent: Friday, March 02, 2007 3:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: Spam? Re: [asterisk-users] cmd page crashes AsteriskSVN-branch-1.4-r57207 Did that. No change -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lee Jenkins Sent: Friday, March 02, 2007 10:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Spam? Re: [asterisk-users] cmd page crashes Asterisk SVN-branch-1.4-r57207 Hall, Eric M. wrote: Group I'm having some trouble with asterisk and the page cmd. Any help would be great! This is what's in my extensions.conf exten = _**2,1,SIPAddHeader(Call-Info: answer-after=0) exten = _**2,2,Page(SIP/36651)|d exten = _**2,3,Hangup Looks like you have at least a syntax error. You have: _**2,2,Page(SIP/36651)|d And it should be _**2,2,Page(SIP/36651|d) Try fixing the d option by placing it within the right parenthesis and try it again. -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FW: Seamless Multi Office Asterisk Deployment
Why does everyone want to go off-list? Is this not information that could benefit others? On 3/12/07, Bruce Reeves [EMAIL PROTECTED] wrote: Brandon Your on the right track with what is can do. It will also be good to look into what kind of QOS you can do on the T-1 connections between offices. I have an 8 office setup similar to this and many of your goals I have achieved and would be glad to offer ideas and such if you want to email me off list. On 3/12/07, Brandon Comouche [EMAIL PROTECTED] wrote: Hello I have a brief and a long question about a possible Asterisk deployment I am planning. Long Story Short: I have four total offices, one main and three remote. All offices are connected using dedicated network T1 lines creating one unified network across offices. I would like to know if it is possible to set up an Asterisk system with the following capabilities: - Branch Unification (I know this can be done) - Branch Independence (In case of T1 network Failure, PSTN line failover at each branch) - Roaming Extensions (A user can go to any office and log in to a phone – hopefully check voice mail as well) Basically, I am asking if Asterisk can be a system that will seamlessly operate as one big system and handle failovers as well. After spending hours playing with Asterisk, reading voip-info.org, and watching this list, it seems that Asterisk can handle anything. I just would like re-assurance that I am not chasing a lost cause. A simple Yes or No answer is acceptable to me. Below I have a long version of what I am trying to do if anyone is in the mood to give me more pointers J Brandon (Long Version Follows) Long Story Version: Here is what I have to work with: - Four Offices (One main and three remote) - Dedicated T1 lines connecting three remote offices to one main office (all connections made through the main office) - Will have a T1 Voice line at the main office - Three PSTN lines at each remote office Essentially what I would like to do is create a system comparable to the ShoreTel ShoreGear product line (if you are familiar with it). This system will seamlessly unite all offices as one and provide failover in the case of line outage. It also allows users to roam from phone to phone across offices seamlessly. It has many more features, but those are two main features I am looking for. About 40 total phones will be deployed. To make it even more difficult, I would like all user extensions to start the same (i.e. across offices all extensions are 5### with no discernable pattern). Progress so far: At this time I have determined that I will need a server at each office as well as a T1 card (TE110P) at the Main office and the four port TDM PSTN cards at each remote office. I plan on using the Polycom IP 430 or 501 (Undecided, 501 if required). I have been using TrixBox to this point, would like to continue if possible. It appears that I will want to use DunDi in some fashion to unite the branches. My main roadblock right now is trying to figure out how to get all the information across the offices at the same time (extensions, voicemail). I have successfully had two boxes communicate, but what I am looking for is much more complex I feel. I have thought of synchronized MySQL databases, but I do not know if that will work the way I wish. If anyone reads this far ;) I am looking for suggestions for routes I might consider or places I should/could look for more information. I am relatively new to Asterisk, but I am not afraid to get my hands dirty. If something I said did not make any sense or if there is more information I could provide, I am happy to help where I can. Thank you for your time and assistance. Brandon Comouche An IT Guy ___ --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 -- Bruce Nortex Networks ___ --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 -- sean ___ --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: Spam? Re: [asterisk-users] cmd page crashesAsteriskSVN-branch-1.4-r57207
I take it back. It will not work if you hang up the calling phone first. Still crashes -- Executing [EMAIL PROTECTED]:1] SIPAddHeader(SIP/36651-b7d1cf48, Call-Info: answer-after=0) in new stack -- Executing [EMAIL PROTECTED]:2] Page(SIP/36651-b7d1cf48, SIP/36651SIP/36652sip36655sip/36653sip/36651h|d) in new stack -- Called 36652 [Mar 12 19:25:27] WARNING[7784]: app_page.c:129 page_exec: Incomplete destination 'sip36655' supplied. -- Called 36653 -- Called 36651h -- SIP/36651-b7d1cf48 Playing 'beep' (language 'en') -- SIP/36653-09f68f78 is ringing -- SIP/36652-09f679f8 is ringing -- SIP/36651h-09f7fbd8 is ringing -- SIP/36652-09f679f8 answered -- Created MeetMe conference 1023 for conference '1689628562d' -- SIP/36651h-09f7fbd8 answered -- SIP/36653-09f68f78 answered [Mar 12 19:25:31] WARNING[7784]: channel.c:1686 ast_hangup: Hard hangup called by thread 21883824 on SIP/36652-09f679f8, while fd is blocked by thread 49327024 in procedure ast_waitfor_nandfds! Expect a failure [Mar 12 19:25:31] WARNING[7784]: channel.c:1686 ast_hangup: Hard hangup called by thread 21883824 on SIP/36653-09f68f78, while fd is blocked by thread 116542384 in procedure ast_waitfor_nandfds! Expect a failure [Mar 12 19:25:31] WARNING[7784]: channel.c:1686 ast_hangup: Hard hangup called by thread 21883824 on SIP/36651h-09f7fbd8, while fd is blocked by thread 95366064 in procedure ast_waitfor_nandfds! Expect a failure == Spawn extension (amaxx, **2, 2) exited non-zero on 'SIP/36651-b7d1cf48' VoIP-PBX*CLI Disconnected from Asterisk server -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hall, Eric M. Sent: Monday, March 12, 2007 7:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: Spam? Re: [asterisk-users] cmd page crashesAsteriskSVN-branch-1.4-r57207 Just wanted to update the group I updated asterisk to SVN-branch-1.4-r58833M and page no longer crashes Asterisk. My below example works great. Thanks! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hall, Eric M. Sent: Friday, March 02, 2007 3:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: Spam? Re: [asterisk-users] cmd page crashes AsteriskSVN-branch-1.4-r57207 Did that. No change -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lee Jenkins Sent: Friday, March 02, 2007 10:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Spam? Re: [asterisk-users] cmd page crashes Asterisk SVN-branch-1.4-r57207 Hall, Eric M. wrote: Group I'm having some trouble with asterisk and the page cmd. Any help would be great! This is what's in my extensions.conf exten = _**2,1,SIPAddHeader(Call-Info: answer-after=0) exten = _**2,2,Page(SIP/36651)|d exten = _**2,3,Hangup Looks like you have at least a syntax error. You have: _**2,2,Page(SIP/36651)|d And it should be _**2,2,Page(SIP/36651|d) Try fixing the d option by placing it within the right parenthesis and try it again. -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Create meetme conference rooms on the flight.
Wai Wu wrote: Hi all, Anyone know how to dynamically create meetme conference rooms on the flight? I remembered a while ago there was a switch that tell meetme to create the conference room is the room is not defined in the meetme.conf. It doen't seem to be working for me anymore. Thnx Not sure about the switch but remember to turn off all electron devices during takeoff and landing. Thanks, Steve ___ --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] Voicemails with occasional speeded up portions
Greetings, Very occasionally, we have a complaint from a user that a portion of a voicemail message is very speeded up - like when you press the fast- forward button on an old-fashioned tape dictaphone. This affects both the server-stored and emailed copies of the message. I have a sample if anyone is interested. Has anyone else experienced this? CP ___ --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] TDM-400, Polycom SIP phones, and echo problems
Hi: I am working on a new system with a TDM-400P card with three FXO modules and one FXS module. The system has been in place for a week. Users are complaining of echo problems. I have noticed this echo myself. It varies in severity. It is sometimes bad enough to make it difficult to converse, but the users find it generally unacceptable. They miss their old phones, which just worked. As you can imagine, this is not the way to get them excited about this technology. I have used the 1.4 branch version of fxotune to tune the card. These are the echo statistics I get on the affected channels. asterisk1 asterisk # fxotune -d -b 1 -w 1004 Dumping module /dev/zap/1 echo ratio = 0.0077 (85.5 / 11145.0) Done! asterisk1 asterisk # fxotune -d -b 3 -w 1004 Dumping module /dev/zap/3 echo ratio = 0.0296 (330.2 / 11145.0) Done! According to the wiki, both those echo ratios should be more than sufficient to let the echo canceller handle the echo. Residual echo is present, however. It doesn't interfere with conversation, but I could see how it would be irritating, especially if you never had it before. The call experience should get better, rather than worse -- especially if we're using Polycom IP 650s. Questions: - What settings might I tweak to eliminate this remaining echo? - Is there a hardware or software echo canceller that will do a more thorough job of it? - How is the HPEC in dealing with this echo? ___ --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] Call Back
Luis Claudio Santos wrote: Somebody could help me with a call back implementation, please? I mean, I just want call to my Asterisk, hung up the phone, and wait it calls me back... Somebody ever did that for local or international calls? Luis, Check out this article: http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out -- Warm Regards, Lee ___ --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] Voicemails with occasional speeded up portions
Anthony Rodgers wrote: Very occasionally, we have a complaint from a user that a portion of a voicemail message is very speeded up - like when you press the fast-forward button on an old-fashioned tape dictaphone. This affects both the server-stored and emailed copies of the message. I have a sample if anyone is interested. Has anyone else experienced this? No, but it suggests a timing problem somewhere. Difficult to solve if it happens rarely :| -Stephen- ___ --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] Playback 5% Too Fast?
Hi All I have a problem with IVR scripts which consist mainly of Playback of audio files, driven from an AGI application. There are clicks every few seconds or more frequently that is audible on the remote end (PSTN), but not on the Asterisk recording of the call. If I record the remote end and compare it to the local recording, it appears to be about 5%-7% too fast - i.e. if I synchronise the starts, the remote end finishes sooner. I can find points in the remote recording where parts of the waveform have been missed out, leading to jumps in the waveform, which correspond to the audible clicks. These jumps seem like dropped packets, and I'm deducing that Asterisk is sending data slightly too fast (i.e. more frequently than 50x160 sample per second) for the remote end, which has to drop data to keep up. This is a VoIP-only set up - no Zap hardware. Thinking this was a timing issue, I have installed Zaptel to get ztdummy, which is loaded OK, but that hasn't made any difference. I have tried it with different VoIP providers and observed the same problem. Behaviour has persisted from 1.2 to 1.4 and now 1.4.1. CentOS 4.4 (2.6.9 kernel), Dell 1950. Any ideas how to progress? Is this a timing issue or am I wide of the mark? Thanks for any help David ___ --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] RE: Playback 0.5% Too Fast?
Just checked my figures, and I mean 0.5%-0.7%. Anyway, it is the resulting clicks that are the problem. Any help still appreciated. David -Original Message- From: David Brazier Sent: 13 March 2007 00:33 To: asterisk-users@lists.digium.com Subject: Playback 5% Too Fast? Hi All I have a problem with IVR scripts which consist mainly of Playback of audio files, driven from an AGI application. There are clicks every few seconds or more frequently that is audible on the remote end (PSTN), but not on the Asterisk recording of the call. If I record the remote end and compare it to the local recording, it appears to be about 5%-7% too fast - i.e. if I synchronise the starts, the remote end finishes sooner. I can find points in the remote recording where parts of the waveform have been missed out, leading to jumps in the waveform, which correspond to the audible clicks. These jumps seem like dropped packets, and I'm deducing that Asterisk is sending data slightly too fast (i.e. more frequently than 50x160 sample per second) for the remote end, which has to drop data to keep up. This is a VoIP-only set up - no Zap hardware. Thinking this was a timing issue, I have installed Zaptel to get ztdummy, which is loaded OK, but that hasn't made any difference. I have tried it with different VoIP providers and observed the same problem. Behaviour has persisted from 1.2 to 1.4 and now 1.4.1. CentOS 4.4 (2.6.9 kernel), Dell 1950. Any ideas how to progress? Is this a timing issue or am I wide of the mark? Thanks for any help David ___ --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] Nomination for Coolest App in 2007
On Tue, Mar 06, 2007 at 11:14:15PM -0500, Steve Totaro wrote: Mine goes to chan_bluetooth. Somewhat of a pain getting it going but I am totally floored with how cool it is! Thanks, Steve Totaro ___ --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 My, that is a cool app. I look forward to running it when it's a bit more stable. While the outgoing call ability seems of limited use since cell call quality is not that exciting to even unlimited night and weekend minuets are probably not too attractive compared to 1 cent/minute SIP terminations, there are a number of interesting possibilities: a) If your target has an unlimited calls to other customers/family/etc. plan, you would want to call them this way to save minutes. b) Handy on some carriers for checking cell voice mail. (I have found that with many US carriers, however, you can call your cell phone with CID set to your cell number, and it goes directly to voice mail Make sure you have a password!) c) Incoming calls, obviously handy. d) During daytime, program to receive incoming calls and say, I am at my desk. Please call me at xxx- or press 1 to have me call you back at CID so you get better quality and don't bill cell minutes. In the evening, assuming unlimited weekends, you might forward directly. Can it send and receive SMS via bluetooth too? I also like a lot the talk of coming softphones with bluetooth headset support. This would allow you to use your bluetooth headset as an extension on your Asterisk pbx. I happen to have a bluetooth headset that plugs into my hard phone -- I wish more hardphones supported them natively -- and that's handy. This could be just as good. To really get it right you would want some speech recognition so you could place calls from the bluetooth headset by saying names and digits, as many cell phones can already do. Of course, a linux softphone could reside right on the asterisk box. You could multi-dial your bluetooth headset and your hard phones and answer where you like. ___ --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] Call Back
In general how I implemented is as follows: - Caller calls asterisk. - From AGI, asterisk gets the caller ID. - Without answering, play back beeps, to simulate busy. - When the user hangs up, asterisk detects broken connection, cannot send beeps, and Originates a call back to the caller. - After the caller hangs up, the Originate from Asterisk rings his phone. - As soon as he picks up, asterisk ask him to enter destination. - The caller enter destination #, asterisk dials the destination. - The two channels are connected. That's it. Works great. Luis Claudio Santos wrote: Somebody could help me with a call back implementation, please? I mean, I just want call to my Asterisk, hung up the phone, and wait it calls me back... Somebody ever did that for local or international calls? Thanks. LC. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Call Back
Can you provide some specific details as I would like to implement something like this. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ivo Zivkov Sent: Tuesday, 13 March 2007 12:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call Back In general how I implemented is as follows: - Caller calls asterisk. - From AGI, asterisk gets the caller ID. - Without answering, play back beeps, to simulate busy. - When the user hangs up, asterisk detects broken connection, cannot send beeps, and Originates a call back to the caller. - After the caller hangs up, the Originate from Asterisk rings his phone. - As soon as he picks up, asterisk ask him to enter destination. - The caller enter destination #, asterisk dials the destination. - The two channels are connected. That's it. Works great. Luis Claudio Santos wrote: Somebody could help me with a call back implementation, please? I mean, I just want call to my Asterisk, hung up the phone, and wait it calls me back... Somebody ever did that for local or international calls? Thanks. LC. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Back
All the code is in AGI. Take a look at the Originate application. (http://www.voip-info.org/tiki-index.php?page=Asterisk+Manager+API+Action+Originate) Klaverstyn, David C wrote: Can you provide some specific details as I would like to implement something like this. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ivo Zivkov Sent: Tuesday, 13 March 2007 12:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call Back In general how I implemented is as follows: - Caller calls asterisk. - From AGI, asterisk gets the caller ID. - Without answering, play back beeps, to simulate busy. - When the user hangs up, asterisk detects broken connection, cannot send beeps, and Originates a call back to the caller. - After the caller hangs up, the Originate from Asterisk rings his phone. - As soon as he picks up, asterisk ask him to enter destination. - The caller enter destination #, asterisk dials the destination. - The two channels are connected. That's it. Works great. Luis Claudio Santos wrote: Somebody could help me with a call back implementation, please? I mean, I just want call to my Asterisk, hung up the phone, and wait it calls me back... Somebody ever did that for local or international calls? Thanks. LC. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Back
Ivo Zivkov wrote: All the code is in AGI. Take a look at the Originate application. (http://www.voip-info.org/tiki-index.php?page=Asterisk+Manager+API+Action+Originate) I take it this is to make ultra-cheap calls from anywhere, right? -Stephen- ___ --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] Call Back
This doesn't make sense to me. Are you able to give some example dial plans? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ivo Zivkov Sent: Tuesday, 13 March 2007 2:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call Back All the code is in AGI. Take a look at the Originate application. (http://www.voip-info.org/tiki-index.php?page=Asterisk+Manager+API+Actio n+Originate) Klaverstyn, David C wrote: Can you provide some specific details as I would like to implement something like this. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ivo Zivkov Sent: Tuesday, 13 March 2007 12:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call Back In general how I implemented is as follows: - Caller calls asterisk. - From AGI, asterisk gets the caller ID. - Without answering, play back beeps, to simulate busy. - When the user hangs up, asterisk detects broken connection, cannot send beeps, and Originates a call back to the caller. - After the caller hangs up, the Originate from Asterisk rings his phone. - As soon as he picks up, asterisk ask him to enter destination. - The caller enter destination #, asterisk dials the destination. - The two channels are connected. That's it. Works great. Luis Claudio Santos wrote: Somebody could help me with a call back implementation, please? I mean, I just want call to my Asterisk, hung up the phone, and wait it calls me back... Somebody ever did that for local or international calls? Thanks. LC. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
