Re: [Asterisk-Users] How can I use MySQL in the dialplan?
What the hell? NO! show application MySql app_addon_mysql is the name of the module. load app_addon_mysql.so -Matthew Quoting Ronald Wiplinger [EMAIL PROTECTED]: Matthew Boehm wrote: Ronald_Wiplinger wrote: I would like to put / get some data from an MySQL database. I want to use this MySQL database also via a web page. bye Ronald app_addon_mysql or use RealTime. *CLI show application app_addon_mysql Your application(s) is (are) not registered I want to use it for putting stored speed dial numbers into the per phone stored register, ... I guess I cannot get that with realtime done!!! bye Ronald Wiplinger ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This message was sent using IMP, the Internet Messaging Program. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] strange dial problem with polycom 501
You should take a look at Section 2.1.5 of http://www.faqs.org/rfcs/rfc3435.html This is the basis for the Polycom digit maps. At 07:31 AM 7/28/2005, you wrote: I am having a strange problem with polycom 501 and dailing. I've read the archives and no one there specifically mentions this problem, so I thought I'd ask here. The problem is that when the user picks up the receiver or pressed new call, sometimes they will enter a number (for example 95072091234) and in the middle of the number the cursor might jump back one digit. So the call above, if just typed into the phone, might end up: 9507291234. Other times the cursor might jump right back to the beginning of the number. This doesn't happen when they enter the number and the press dial, so it seems to be a digitmap problem. However, the digitmap is nearly the same as what I've used on IP-500s in the past. It is: [0]|*8|*2xx|#5|#7x|[278]xx|[9]11|9411|9x.T [Actually it was [0]|*8|*2xx|#5|#7x|[278]xx|[9]11|9411|9x.T -- I don't know where that space came from, but I'll take it out and test again today.] Are there any obvious problems with that digitmap? Anything else that I should take a look at? Thank you. -- -M ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] problem calling SIP accounts
Hi I have configured sip accounts and they work some times. when i make a call to another SIP account it works right but some times i get the following error Jul 29 07:17:00 WARNING[802]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Critical Request) this happence when i register the SIP users and stay for some time and dial.but no problem with out going calls, can call any time. Regards Kanishka ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Auto loading of qozap module
Apologies for being a bit of a Linux newbie... I have got a working * system but each time I reboot my box I need to: modprobe qozap ztcfg asterisk Now I realise this is really a Linux question but I am struggling with the problem and any help would be much appreciated. There is a module qozap.ko - which if I do a find I see in /lib/modules/2.6.11.4-11.4.21.7-default/misc/qozap.ko Is this the module? If it is here, then why do I need to modprobe qozap? I have looked at /etc/init.d/rc - but this seems to be all about services! Wrong place to look? So somehow how do I load this module so it runs at startup? Angus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Messaging - Asterisk presence
Is there some configuration or specific extension in the asterisk to send instant messages between two SIP clients? I'm using the eyebeam and this service is not working! Thank you. []'s Wendell ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] [Asterisk-Dev] Digium to Sponsor a Pizza party atCluecon
Brian West [EMAIL PROTECTED] wrote: Digium, the creator and primary developer of Asterisk, the industrys first Open Source PBX, will be hosting a pizza party from 4pm to 6pm on the first day of Cluecon. We look forward to everyone coming out to enjoy this opportunity to meet fellow developers and users in a more casual environment. Thanks. I missed the first 600 copies of that announcement. :-) -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 360 record button?
Christian Stredicke wrote: It would be nice if the PBX can acknowlegdge the Record header - then it would have the chance to paint a record icon on the screen. In the next release.-) Right. Is there another header for turning off recording? Anyway, we should not send unsupported media type... /O ;-) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CDR disposition field always says ANSWERED on inbound calls
Hi Jerry,
this is the cdr result of four tests (sorry for bad format):
+--+---+-+-+---+--+-+--+--++---+-+--+-+-+--+
| uniqueid | userfield | accountcode | src | dst |
dcontext | clid|
channel | dstchannel | lastapp| lastdata |
calldate| duration | billsec | disposition | amaflags |
+--+---+-+-+---+--+-+--+--++---+-+--+-+-+--+
|20126 | | | | 293504780 |
inbound | |
Zap/1-1 | | BackGround | fga_main_menu |
2005-07-29 10:09:00 | 10 | 10 | ANSWERED|2 |
|20127 | | | | 104 |
inbound | |
Zap/1-1 | SIP/104_fga-281d | Dial | SIP/104_fga|21|t |
2005-07-29 10:09:27 | 15 | 15 | ANSWERED|2 |
|20128 | | | 104 | s |
outbound_qualcuno_in_ufficio | Giorgio Incantalupo 104 |
SIP/104_fga-7fa0 | Zap/1-1 | Dial | Zap/g3/0331551932 |
2005-07-29 10:28:09 | 54 | 24 | ANSWERED|3 |
|20129 | | | 104 | s |
outbound_qualcuno_in_ufficio | Giorgio Incantalupo 104 |
SIP/104_fga-3f24 | Zap/1-1 | Dial | Zap/g3/0293256288 |
2005-07-29 10:30:23 | 11 | 0 | NO ANSWER |3 |
+--+---+-+-+---+--+-+--+--++---+-+--+-+-+--+
The first test was to call the number 293594780 and hanging up the phone
during inbound menu message. The result ia ANSWERED and I think it is ok
because channel Zap/1-1 answered as I asked it using the Answer command.
BUT
The second line shows I dialled 104 during Background message to call
104_xxx SIP user: the telephone rang but I didn't pick up the phone. I
expected some kind of result like NOANSWER because I didn't answer but
Asterisk wrote ANSWERED in the disposition field. The same is if the
phone is busy but the dialstatus variable shows another status (bust,
etc..). Another strange thing is the billsec: I didn't answer...so why
billsec and duration are the same??
Then I made an outbound call:
third test: I called my mom and she answered as you can see from
disposition field: she took 24 sec to pick up the phone and then we
spoke for a while.
Now last test: I called another number but after a bit I hung up the
phone and the CDR is right!! I didn't speak (billsec = 0) and duration
is 11, right!!!
In your opinion ,is this the right behaviour for an inbound call?
I do not think so but I may be wrong.
TIA
Giorgio
[EMAIL PROTECTED] wrote:
Hi,
Quoting Giorgio Incantalupo [EMAIL PROTECTED]:
My inbound context is:
[inbound_menu]
include = internals ; very strange: include doesn't work!!!
exten = _X.,1,DigitTimeout(2)
exten = _X.,2,Answer
exten = _X.,3,NoOp(DS:${DIALSTATUS}) ; my debug purpose
exten = _X.,4,Background(fga_main_menu)
exten = _X.,5,Background(3-sec-pause)
exten = _X.,6,Background(fga_main_menu)
exten = _X.,7,Hangup
exten = 101,1,Macro(interni,${PIPPO},${RING_TIME})
The Macro executes a Dial commandso nothing strange.
anybody knows why the CDR field named disposition always says ANSWERED
on inbound calls even if nobody picks up the phone (we are using various
I do believe this might clarify; if not, you are explicitly answering
the line. I don't see any Dial commands (might be in your include),
but if your call hits this extension it will be considered answered
after priority 2 (if I understand how the CDR works). You are then
having them go through an IVR, but the call is already deemed connected.
I think you want to play with NoCDR or one of it's cousins to get the
effect you desire.
J.
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--
GIORGIO INCANTALUPO
Tel. +39 02 9350 4780 (104)
FGA Software
20017 Rho - Via Puccini, 8
E-Mail :
[EMAIL PROTECTED]
Internet:
http://www.fgasoftware.com
RE: [Asterisk-Users] Play Dialtone - get digits
Here is a crude hack, but it requires the user to press # at the end.
exten = s,1,Playtones(dial)
exten = s,2,Read(1stnumber,,1)
exten = s,3,StopPlaytones
exten = s,4,Read(restofnumber)
exten = s,5,SetVar(totalnumber=${1stnumber}${restofnumber})
Hope that helps.
B. J.
-Original Message-
From: Peter Svensson [mailto:[EMAIL PROTECTED]
Sent: Thursday, July 21, 2005 1:37
To: Ed Greenberg; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Play Dialtone - get digits
On Wed, 20 Jul 2005, Ed Greenberg wrote:
I'd like to write a snippet of dialtone that plays dialtone and collects a
specific number of digits into a variable.
Sort of like READ but with a generated dialtone.
Naturally, I want the dialtone to stop playing after the first digit.
I can't find this anywhere.
Only thing I can think of is a no-password DISA. Is this the correct
method? Is there a better one?
DISA would proably work, though it may be a hassle since the call will be
sent into the disa context. Another option is to use READ with a
filecontaining a recording of the dialtone.
Peter
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RE: [Asterisk-Users] Cisco Call manager
I've gotten CME to talk to *, but have not used the plain Call Manager. I'd guess you could use a SIP trunk like the wiki talks about to configure call managed to talk to a SIP termination service. Rick -Original Message- From: Anton Krall [mailto:[EMAIL PROTECTED] Sent: Thursday, July 28, 2005 1:24 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Cisco Call manager Anybody using Cisco Call Manager and connecting to any SIP termination service like voipjet, voxee, etc? Please msg me offlist. AK ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The information in this communication is intended to be confidential to the Individual(s) and/or Entity to whom it is addressed. It may contain information of a Privileged and/or Confidential nature, which is subject to Federal and/or State privacy regulations. In the event that you are not the intended recipient or the agent of the intended recipient, do not copy or use the information contained within this communication, or allow it to be read, copied or utilized in any manner, by any other person(s). Should this communication be received in error, please notify the sender immediately either by response e-mail or by phone, and permanently delete the original e-mail, attachment(s), and any copies. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Marc Spindt is out of the office
I will be out of the office starting 07/29/2005 and will not return until 08/09/2005. Thank you, Marc ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Queue/Agents
Looking for a good web app that will show agents that are login to queue. I tried the operator panel but I'm unable to get the LED to change color per the doco I have.. It works well for everything else but no luck on the agent part.. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sipura SPA-1001: Bad Outgoing Call Quality
Greetings, I have a Sipura SPA-1001. When I make outgoing calls, I have very jittery sound. Incoming calls work fine. This wasn't the case a few months ago, I am running head as of yesterday. Any suggestions? Thanks, Erik ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] what is the problem with gmail and the list.
I have not been receiving mail from the list 29th July, what is the problem with gmail and the list. No problem here. Check you Spam folder, and if you find email there from this list, select them all and click Not spam hth ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] List
Is it my imagination or did I just drop off the list for several days somehow... I didn't get any posts since Friday... rhuddleston.vcf Description: Binary data ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Potential reboot problem with Polycom IP600 phones
I can confirm that my IP 600 phone will reboot on the slightest electrical glitch, even though it does not affect any servers, workstations, other phones or any other equipment. I think the IP 600s are very close to the maximum power output of the PSU. Perhaps the easiest solution would be a larger power supply. -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] test message - ignore me
Haven't seen email since the 29th.. just testing. begin:vcard fn:Matt Hess n:Hess;Matt org:LiveWire Networks adr;dom:;;4577 Pecos St;Denver;CO;80211 email;internet:[EMAIL PROTECTED] title:Sr. Network Engineer tel;work:303-458-5667 x 106 tel;fax:303-458-5725 x-mozilla-html:FALSE url:http://www.livewirenet.com/ version:2.1 end:vcard ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] [EMAIL PROTECTED]/Grandstream Call Transfer
OK, now this should be really simple, but I am a bit of a newbie so please bear with me. I have an [EMAIL PROTECTED] box setup with TDM04B and two POTS lines. On the SIP side, I have GXP2000 phones. Most things seem to work, but the users cannot figure out how to transfer incoming calls from one extension to another. Now I am not sure that I have things setup correctly, but is there something special that needs to be done in order to transfer calls?? The GXP2000 has a Transfer button on the keypad but that doesnt seem to allow a transfer. Is there something that I am missing?? Any help would be greatly appreciated. Marc ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] g729 liscence question
I have a TDM400P with one FXS and one FXO.. how many liscence(2) I will have to buy? Thanks___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom IP500 Ringtone howto
Hi Guys, I thought some of you might be interested in a minimalistic Polycom ringtones howto. I assume this works with the ip600 (501/601) but not sure about the 300. http://www.voipphreak.ca/archives/82-My-Little-Howto-for-Polycom-IP500-Ringtones.html Matt -- Matt Gibson Telecommunications Director Voxip.ca / NJ Tech Solutions Inc. Mobile: 1.613.868.9318 Tel: 1.314.480.4550 ex 6400 Toll Free: 1.888.999.4678 ex 6400 Email: [EMAIL PROTECTED] Fax: 1.613.761.1828 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] call transfer
Hi! I have searched answer how can I transfer calls with asterisk,with no result. Can you advice me and show some example file how can I use SIP phone to transfer calls by hitting # and get the Transfer prompt and enter an extension I want to transfer to? Thanks for your answers This mail sent through L-secure: http://www.l-secure.net/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX Devices Recommendation
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Does anyone have any recommendations on an IAX Desktop Telephone or ATA Device. I currently have 2 of the SIPURA-841's on my local network and now I am wanting to try an IAX Device at my remote office since I think that it would be easier to configure through various routers than a SIP Device. I just started to look at the Digium IAXy Single FXS Adapter but unable to find a Telephone that supports the IAX Protocol. Any Recommendations or is the Digium FXS Adapter the way to go. -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (MingW32) Comment: GnuPT 2.6.2.1 by EQUIPMENTE.DE Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQFC7nLuC9Rk5Ie0reIRApaJAJ9NsOquK9qu+adee3rtT/43TEeoRgCfYh/P OuuibBI5wt5a2pt28I7pvds= =C+jj -END PGP SIGNATURE- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Record() permission problem
On Sat, Jul 30, 2005 at 09:51:22PM -0400, Jim Archer wrote:
Hi All...
I'm trying to use the record() app and it complains that it can't open it's
file because permission was denied. I'm running the released Asterisk on
Debian Linux. The target directory is workd writable. Here is the
relevant part of the dialplan:
exten = 1,1,Playback(leave-message)
exten = 1,2, Record(/var/local/whois-messages/whois-${contactid}:wav|6|120)
Here is the output from Asterisk:
-- Executing Record(Zap/9-1,
/var/local/whois-messages/whois-321:wav|6|120) in new stack
-- Playing 'beep' (language 'en')
Jul 30 21:44:20 WARNING[4206]: file.c:910 ast_writefile: Unable to open
file /var/local/whois-messages/whois-321.wav: Permission denied
Jul 30 21:44:20 WARNING[4206]: app_record.c:299 record_exec: Could not
create file /var/local/whois-messages/whois-321
Do you use official debs (where sound resides under
/usr/share/asterisk/sound )?
Does Record(/full/path) record to /full/path or to /full/path
relativly to the default sound files path?
And natuarally: can the asterisk user write there? Please provide the
output of the following:
groups asterisk
ls -la /var/local/whois-messages/
--
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il | | a Mutt's
[EMAIL PROTECTED] | | best
ICQ# 16849755 | | friend
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Re: [Asterisk-Users] most stable linux to build business
On Thursday 28 July 2005 18:28, snacktime wrote: On 7/28/05, wassim darwish [EMAIL PROTECTED] wrote: what is the most stable linux that we can build business on it, i mean the best linux a linux without problems . I have Suse 9.1. I had no problems installing it. It is not the latest which is 9.3. Installing it was just putting a cd in the drive and booting. It recognized everything in my system and everything worked. Bob ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Astcc Configuration Problem
Hi: I used astcc to create database. After I get the message database created, I save the configuration and I move to the next step to assign trunk and route. But I get the message: Database unavailable -- please check configuration Cannot edit routes until database is configured I checked the databse and it's in mysql, the file /var/lib/astcc/astcc-config.conf is empty. astcc-admin.cgi is supposed to write a file based on the configuration, but for some reason doesn't I inserted data manually into the databse tables and astcc works fine on asterisk and write data into the database. Why the web browser can't see the database it created? could apache be the problem although it's running? Regards; Chawki Start your day with Yahoo! - make it your home page http://www.yahoo.com/r/hs ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How can I use MySQL in the dialplan?
You'll have a much more flexible solution if you keep your MySQL access out of the * dialplan, and put it in AGI. Matthew Boehm wrote: What the hell? NO! show application MySql app_addon_mysql is the name of the module. load app_addon_mysql.so -Matthew Quoting Ronald Wiplinger [EMAIL PROTECTED]: Matthew Boehm wrote: Ronald_Wiplinger wrote: I would like to put / get some data from an MySQL database. I want to use this MySQL database also via a web page. bye Ronald app_addon_mysql or use RealTime. *CLI show application app_addon_mysql Your application(s) is (are) not registered I want to use it for putting stored speed dial numbers into the per phone stored register, ... I guess I cannot get that with realtime done!!! bye Ronald Wiplinger ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This message was sent using IMP, the Internet Messaging Program. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk 1.0.9 and PostreSQL DB
Hello everybody, now I'm using MySQL for SIP/IAX friends and CDR. Is it also possible to use PostreSQL instead of MySQL? Regards Bastian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Queue/Agents
Commercial plug. Signate is the North American distributor for XC-AST, call queue monitoring and reporting software for Asterisk. It allows managers to monitor queues and agents in real time, or to analyze queue activity for given periods. Real time facilities allow managers to monitor: -Agents logging on and off -Calls by agent -Calls in queue with wait times -The launch of queue URLs like external CRM applications. XC-AST is free for up to two agents. A ten agent system is $900 USD. Installation is available for an additional charge. For more information, http://www.signate.com/xcast.php or call Signate at 415.442.4011. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hall, Eric M. Sent: Monday, August 01, 2005 1:38 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Queue/Agents Looking for a good web app that will show agents that are login to queue. I tried the operator panel but I'm unable to get the LED to change color per the doco I have.. It works well for everything else but no luck on the agent part.. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.9.8/61 - Release Date: 8/1/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.9.8/61 - Release Date: 8/1/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] what is the problem with gmail and the list.
No. It's not that. I know that was a problem previously, but I've had the same problem as the user mentioned and those emails aren't in my Spam folder. It's like they've completely disappeared. I guess that's why they call it Beta. :-) On 8/1/05, Time Bandit [EMAIL PROTECTED] wrote: I have not been receiving mail from the list 29th July, what is the problem with gmail and the list. No problem here. Check you Spam folder, and if you find email there from this list, select them all and click Not spam hth ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] what is the problem with gmail and the list.
--- Time Bandit [EMAIL PROTECTED] wrote: I have not been receiving mail from the list 29th July, what is the problem with gmail and the list. No problem here. Mine stopped on the same data, July 29. I had to subscribe as a new account to get mail from the list. I checked the log of my mail server, no mail seen since than. Chris __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] List
Huddleston, Robert wrote: Is it my imagination or did I just drop off the list for several days somehow... I didn't get any posts since Friday... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hello list, We've same problem. ~Madhawa ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] what is the problem with gmail and the list.
Time Bandit wrote: I have not been receiving mail from the list 29th July, what is the problem with gmail and the list. Suddenly as is well and I am getting mail again on my normal account. -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] what is the problem with gmail and the list.
I have not been receiving mail from the list 29th July, what is the problem with gmail and the list. No problem here. Check you Spam folder, and if you find email there from this list, select them all and click Not spam The list server took a dump last week and has been off line since then. Apparently the server is off-site and support personnel couldn't be reached to correct the problem. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] what is the problem with gmail and the list.
I got 23 emails since friday. And im NOT using gmail. Kyle Time Bandit wrote: I have not been receiving mail from the list 29th July, what is the problem with gmail and the list. No problem here. Check you Spam folder, and if you find email there from this list, select them all and click Not spam hth ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] List
Huddleston, Robert wrote: Is it my imagination or did I just drop off the list for several days somehow... I didn't get any posts since Friday... Apparently, a LOT did. Including myself. Doug ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Queue/Agents
Hall, Eric M. wrote: Looking for a good web app that will show agents that are login to queue. I tried the operator panel but I'm unable to get the LED to change color per the doco I have.. It works well for everything else but no luck on the agent part.. I can share mine. Shows a list of callers and agent status. -- respectfully, Joseph ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] [EMAIL PROTECTED]/Grandstream Call Transfer
1: Upgrade your GXP to the latest firmware. See www.grandstream.com 2: [line1] number [send] speak [hold] [line2] number [send] speak [transfer] --Rob ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] test message - ignore me
Same here... Chris HARIGA -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Hess Sent: Monday, August 01, 2005 3:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] test message - ignore me Haven't seen email since the 29th.. just testing. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] test message - ignore me
well over 10,000 users getting 80+ emails a day, it was bound to go down. I wonder how this ranks in the size of mailing lists. Other than LKML what other lists would be this size? On 8/1/05, Matt Hess [EMAIL PROTECTED] wrote: Haven't seen email since the 29th.. just testing. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- sig Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) WWW: http://lathama.com Email: [EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! /sig ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] test message - ignore me
Me neither.. but just started receiving now. WEIRD. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Hess Sent: Monday, August 01, 2005 12:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] test message - ignore me Haven't seen email since the 29th.. just testing. smime.p7s Description: S/MIME cryptographic signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] what is the problem with gmail and the list.
I have no spam lists. :P It died for many people I know. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Time Bandit Sent: Monday, August 01, 2005 2:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] what is the problem with gmail and the list. I have not been receiving mail from the list 29th July, what is the problem with gmail and the list. No problem here. Check you Spam folder, and if you find email there from this list, select them all and click Not spam hth ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users smime.p7s Description: S/MIME cryptographic signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to install PHPAGI?
Hello everyone, Where can I find instructions on how to install PHPAGI? BTW, what's the difference between PHPAGI and PHPAGI2? Are they different products? It's hard to tell from voip-info.org... Best, Leo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] List
On Mon, Aug 01, 2005 at 02:46:55PM -0400, Huddleston, Robert wrote: Is it my imagination or did I just drop off the list for several days somehow... I didn't get any posts since Friday... See the IRC channel. The list has been broken for a couple days. If you look at the archives on lists.digium.com you can see there have been new threads, but no discussions to speak of :-) Ray ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] List
Nope, I have the same problem, nothing. I jumped on the ISP for not being able to get my mail. Ooops. Race -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Huddleston, Robert Sent: Monday, August 01, 2005 2:47 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] List Is it my imagination or did I just drop off the list for several days somehow... I didn't get any posts since Friday... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] List
Nope, I have the same problem, nothing. I jumped on the ISP for not being able to get my mail. Ooops. Race -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Huddleston, Robert Sent: Monday, August 01, 2005 2:47 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] List Is it my imagination or did I just drop off the list for several days somehow... I didn't get any posts since Friday... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] g729 liscence question
I have a TDM400P with one FXS and one FXO.. how many liscence(2) I will have to buy? Been discussed several times on the list (check the archives) and on the wiki. Essentially, need a license for each codec translation, including gsm sounds - g729, etc. The TDM card does not support g729, therefore you would need to count the number of destinations that _only_ use g729 to determine the number of licenses. An FXS phone on the TDM card listening to asterisk sounds won't need any, but the same phone talking to another g729 user will need one. Likewise, the TDM FXO port doesn't support g729. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] g729 liscence question
On Monday 01 August 2005 14:53, Innocent Evil wrote: I have a TDM400P with one FXS and one FXO.. how many liscence(2) I will have to buy? The licenses don't work that way. If Asterisk has to rip apart or assemble a g729 stream for any reason, you'll need a license to do so. If you need to do it twice in the same period of time, you'll need two. Typically speaking you will need a license any time Asterisk needs to convert betwen g729 and any other codec. Your FXS and FXO ports use the slinear audio format so any time you want to use one of the ports and connect to a VOIP provider using g729 you'll be using a license. Also, a license will be required any time Asterisk needs to hear the audio stream from a g729 source. This means if you want Asterisk to listen for silence or voice, detect DTMF or mix audio from several sources. If Asterisk is able to take a g729 frame and pass it along without doing anything to it, no license is required. Those last three paragraphs can be summed up by saying a g729 license is required any time Asterisk needs to transcode to or from a g729 audio format. HTH, -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicemail envelope time is 4 hours ahead
I'm running a recent CVS build under Solaris 10. In the shell than I'm running the Asterisk console I have TZ=US/Eastern and in my voicemail.conf I have tz=eastern and eastern=America/New_York|'vm-received' Q 'digits/at' IMp. The voicemail envelope information seems to be exactly 4 hours ahead. No matter what I try I can't seem to find the cause. Any ideas? Frank ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] what is the problem with gmail and the list.
I'm not on gmail, and also haven't received messages since 7/29 -- just now beginning to see a few trickle in. -Original Message- From: Time Bandit [mailto:[EMAIL PROTECTED] Sent: Monday, August 01, 2005 4:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] what is the problem with gmail and the list. I have not been receiving mail from the list 29th July, what is the problem with gmail and the list. No problem here. Check you Spam folder, and if you find email there from this list, select them all and click Not spam hth ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Toshiba Integration - MWI Light
I received an off-list follow-up to this, so I figured I'd post some
more info about how I got it to work:
exten = _91XXX, 1, Voicemail(u${EXTEN:2})
exten = _91XXX, 2, HasNewVoiceMail(${EXTEN:2})
exten = _91XXX, 3, Hangup
exten = _91XXX, 103, System(sed 's/__EXTEN__/${EXTEN:2}/'
/etc/asterisk/vmon.call /var/spool/asterisk/outgoing/vmon-`date +s`.call);
exten = _91XXX, 104, Hangup
exten = _92XXX, 1, VoicemailMain(${EXTEN:2})
exten = _92XXX, 2, HasNewVoiceMail(${EXTEN:2})
exten = _92XXX, 3, System(sed 's/__EXTEN__/${EXTEN:2}/'
/etc/asterisk/vmoff.call /var/spool/asterisk/outgoing/vmoff-`date
+s`.call);
exten = _92XXX, 4, Hangup
exten = _92XXX, 103, Hangup
vmon.call:
Channel: Zap/1/#63__EXTEN__
MaxRetries: 5
RetryTime: 15
WaitTime: 30
Application: NoOp
vmoff.call:
Channel: Zap/1/#64__EXTEN__
MaxRetries: 5
RetryTime: 15
WaitTime: 30
Application: NoOp
Create those two files, and put those extensions in your context that
you're answering or going to, and you should be good to go.
-Matt
Karl H. Putz wrote:
Use a Call file to dial back to the PBX.
In voicemail.conf set the externnotify value to something like:
externnotify=/usr/local/sbin/mwi.pl
where the perl script creates the Call file. I set up a specific group and
dedicated a port to making these calls instead of chancing the glare with
the pbx. Also, my specific pbx needed some delay between dialing a feature
access key # and the MWI dial code itself so that is why my string is
#www91$ext.
Here is the perl script:
#!/usr/bin/perl
my ($context,$ext,$msgs,@junk) = @ARGV;
my $tmpcallpath = /var/tmp;
my $astpath = /var/spool/asterisk/outgoing;
my $tmpname = mwi- . time();
my $tmpcallfile = $tmpcallpath/$tmpname;
my $callfile = $astpath/$tmpname;
$ext =~ s/[EMAIL PROTECTED]//;
if ($msgs 0) {
$channel = Zap/g3/#www91$ext;
} else {
$channel = Zap/g3/#www90$ext;
}
sleep 2;
print STDERR channel: $channel\n;
open (CALLFILE,$tmpcallfile);
print CALLFILE qq(
Channel: $channel
MaxRetries: 0
WaitTime: 5
Context: mwi
Extension: s
Priority: 1
);
close(CALLFILE);
rename($tmpcallfile, $callfile);
Good luck!
Karl Putz
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Matthew
Drobnak
Sent: Wednesday, July 27, 2005 4:58 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Toshiba Integration - MWI Light
Hi All,
On our Toshiba PBX, to light the MWI, one dials #63__EXTENSION__ --
how is it possible to easily trigger this after a voicemail is sent?
Thanks,
-Matthew Drobnak
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Re: [Asterisk-Users] [EMAIL PROTECTED]/Grandstream Call Transfer
There are 2 methods blind and announced here you go: Blind:Call someone, or receive a call. Hit 'Trnf' The screen displays TRANSFER TO? and you hear a dial tone. The other end can still hear you, so don't say anything nasty. Dial the number and hit 'Send', caller is transferred (blind) Announced: a.. Be on a call b.. Push a LINE button that isn't in use (this puts the call on hold) c.. Dial the extension you wish to transfer to d.. Speak e.. Push TRNF - Original Message - From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, August 01, 2005 2:03 PM Subject: [Asterisk-Users] [EMAIL PROTECTED]/Grandstream Call Transfer OK, now this should be really simple, but I am a bit of a newbie so please bear with me. I have an [EMAIL PROTECTED] box setup with TDM04B and two POTS lines. On the SIP side, I have GXP2000 phones. Most things seem to work, but the users cannot figure out how to transfer incoming calls from one extension to another. Now I am not sure that I have things setup correctly, but is there something special that needs to be done in order to transfer calls?? The GXP2000 has a Transfer button on the keypad but that doesnt seem to allow a transfer. Is there something that I am missing?? Any help would be greatly appreciated. Marc ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Queue/Agents
the flash operator panel does it great. You need to use the agent channels instead of the zap/sip channels. Try this: [Agent/101] Position=3 ; Button number in the console Label=Steve 101 Extension=101; Extension to reach that channel Context=localext ; Context where that extension is defined Voicemail_Context=default Icon=4 Cheers, Jon. On Monday 01 August 2005 03:38 pm, Hall, Eric M. wrote: Looking for a good web app that will show agents that are login to queue. I tried the operator panel but I'm unable to get the LED to change color per the doco I have.. It works well for everything else but no luck on the agent part.. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Issue with zapata.conf immediate setting
I currently have two channel groups in my zapata.conf file. I would like one group to be immediate=yes and the other immediate=no Does not seem to matter which way I go, the first entry in overrides my explicit setting for the second group. I am running * 1.0.9 on FC1 [trunkgroups] ;trunkgroup = 1,24 trunkgroup = 1,48,72 ;spanmap = 1,1,0 spanmap = 2,1,0 spanmap = 3,1,1 spanmap = 4,1,2 [channels] ; Tie line to Nortel context=tie_line_01 signalling=em_w rxwink=300 usecallerid=yes hidecallerid=no usecallingpres=yes rxgain=0 txgain=0 overlapdial=yes transfer=yes immediate=no group=1 callgroup=1 pickupgroup=1 amaflags=billing accountcode=tie_line_01 callprogress=yes busydetect=yes channel = 1-24 ; Qwest DID Lines context=qw_pri_01 switchtype=national signalling=pri_cpe pridialplan=national callerid=XX nsf=sdn rxwink=300 usecallerid=yes immediate=no hidecallerid=no usecallingpres=yes rxgain=0 txgain=0 group=2 callgroup=2 faxdetect=both pickupgroup=2 amaflags=billing accountcode=qwe_pri_01 callprogress=yes channel = 25-47,49-71,73-96 the purpose of this is to bridge our traditional voice PBX and connected digital phones to our * box with a tieline, as well as allow incoming DIDs to flow through the * box into the traditional PBX using the same tieline. In extensions.conf, I have a dialplan set up for the qw_pri_01 circuit/context for calls coming in to hit the tie line device. This works fine. Going from the PBX to *, I have an issue. We have an ACOD of 777 to hit that trunkgroup. After I dial 777, a simple switch is started (I can see it on the console). As soon as I dial any other number (like 83028 as a SIP phone), the 8 is usually the only number that gets picked up. There is no 8 extension in the tie line context, so I get a not in service message. If I set immediate to yes, I COULD default the call to the 's' extension and attempt to handle the additional characters/digits after answering (perhaps a Read cmd). If I have immediate set to yes for this channel group, than the qw_pri_01 group also acts like I set immediate yes in that group - regardless of immediate=no being set. This screws everything up as the calling party does not get anything returned to them for an extension to dial. I suppose I could set up a forced call - but I think setting up immediate=yes on my tieline and immediate=no on my DIDs is a better plan. Perhaps there is a better way? Something I am missing? Thank you in advance Jason ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] g729 liscence question
On Mon, 2005-08-01 at 10:53 -0800, Innocent Evil wrote: I have a TDM400P with one FXS and one FXO.. how many liscence(2) I will have to buy? Short answer: None. Long answer: Zap interfaces use G711 and do not need G729 to work. Only if you plan to connect SIP or IAX phones from outside your local network do you really need voice compression. -- Telecomunicaciones Abiertas de Mexico Carlos Chavez Director de Tecnologia +52-55-91169161 Ext. 2001 signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] what is the problem with gmail and the list.
I do not think the problem with the lists since sometime July 29th was specific to Gmail... If you check the web archives, you'll see both regular posts and others (non-Gmail) asking about problems. My best guess is a subscriber's domain expired or some other similar problem which clogs mailserver queues. Hopefully the list admin will post an update once the problem is solved. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Queue/Agents
Or you could get it (or at least something similar) for free from www.asteriskguru.com. A small preview is available here: http://www.asteriskguru.com/tutorials/queue_stats.html Its 100% ready, just waiting to be uploaded. (Should be there in the next few days). Zoa. William Boehlke wrote: Commercial plug. Signate is the North American distributor for XC-AST, call queue monitoring and reporting software for Asterisk. It allows managers to monitor queues and agents in real time, or to analyze queue activity for given periods. Real time facilities allow managers to monitor: -Agents logging on and off -Calls by agent -Calls in queue with wait times -The launch of queue URLs like external CRM applications. XC-AST is free for up to two agents. A ten agent system is $900 USD. Installation is available for an additional charge. For more information, http://www.signate.com/xcast.php or call Signate at 415.442.4011. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hall, Eric M. Sent: Monday, August 01, 2005 1:38 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Queue/Agents Looking for a good web app that will show agents that are login to queue. I tried the operator panel but I'm unable to get the LED to change color per the doco I have.. It works well for everything else but no luck on the agent part.. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.9.8/61 - Release Date: 8/1/2005 signature.asc Description: OpenPGP digital signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] dialplan defenition
Joao Pereira wrote:
Hello list,
Im writing my dial plan, in witch every SIP phone begins with 74 and has
more 3 numbers (like 74XXX).
So, I want to route all 74XXX calls to my sip channel. For this I wrote
this line:
exten = s,1,Dial(SIP/[EMAIL PROTECTED],30,r)
What is happening is that capi is sending it to s.
You will need to either set up an IVR, asking which number to send it to.
So, you would do the following:
exten = s,1,Answer()
exten = s,2,Background(pls-entr-extn)
exten = _74XXX,1,Dial(SIP/${EXTEN})
exten = _74XXX,2,Goto(s|1)
exten = _74XXX,102,Goto(s|1)
You will obviously need to record the pls-entr-extn sound.
You can do this by making an exten like this:
exten = 678,1,Record(pls-entr-extn)
--
Cheers,
Matt Riddell
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Re: [Asterisk-Users] Astcc Configuration Problem
Check and make sure that astcc-config.conf is owned by the same process that owns apache. Usually the problem is that astcc-admin cannot write to the file due to permission problems. Darren Wiebe [EMAIL PROTECTED] chawki hammoud wrote: Hi: I used astcc to create database. After I get the message database created, I save the configuration and I move to the next step to assign trunk and route. But I get the message: Database unavailable -- please check configuration Cannot edit routes until database is configured I checked the databse and it's in mysql, the file /var/lib/astcc/astcc-config.conf is empty. astcc-admin.cgi is supposed to write a file based on the configuration, but for some reason doesn't I inserted data manually into the databse tables and astcc works fine on asterisk and write data into the database. Why the web browser can't see the database it created? could apache be the problem although it's running? Regards; Chawki Start your day with Yahoo! - make it your home page http://www.yahoo.com/r/hs ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] what is the problem with gmail and the list.
Rich Adamson wrote: I have not been receiving mail from the list 29th July, what is the problem with gmail and the list. No problem here. Check you Spam folder, and if you find email there from this list,select them all and click Not spam The list server took a dump last week and has been off line since then. Apparently the server is off-site and support personnel couldn't be reached to correct the problem. Curious that some of us have received little or no postings, others have, and that many messages that show up in the archive haven't been seen here or elsewhere. I suspect that any dump the list might or might not have taken isn't the complete story. John Novack ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Queue/Agents
Joseph - I would love to see something like this if you are willing to share. Thanks. Joseph wrote: Hall, Eric M. wrote: Looking for a good web app that will show agents that are login to queue. I tried the operator panel but I'm unable to get the LED to change color per the doco I have.. It works well for everything else but no luck on the agent part.. I can share mine. Shows a list of callers and agent status. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] X100P/Caller ID: clidtest works, * complains [repost]
Hi, I'm running Asterisk 1.0.8 on Gentoo, with an X100P (clone) card, and I'm having problems with Caller ID. I have run clidtest, and it seems happy enough, returning:- server clidtest # ./clidtest /dev/zap/1 Number: 041222, Name: MOBILE (that number's fake.) However, I'm not getting the caller ID passed through with *. Sometimes I get a failed checksum like this:- -- Starting simple switch on 'Zap/1-1' Jul 30 16:06:14 NOTICE[9597]: callerid.c:306 callerid_feed: Caller*ID failed checksum Jul 30 16:06:15 NOTICE[9597]: chan_zap.c:5405 ss_thread: Got event 2 (Ring/Answered)... Jul 30 16:06:16 NOTICE[9597]: chan_zap.c:5405 ss_thread: Got event 2 (Ring/Answered)... Jul 30 16:06:18 NOTICE[9597]: chan_zap.c:5405 ss_thread: Got event 2 (Ring/Answered)... -- Executing Wait(Zap/1-1, 2) in new stack snip and sometimes I get an error that I _really_ don't understand:- -- Starting simple switch on 'Zap/1-1' Jul 30 16:25:02 NOTICE[9616]: chan_zap.c:5405 ss_thread: Got event 2 (Ring/Answered)... Jul 30 16:25:02 NOTICE[9616]: chan_zap.c:5405 ss_thread: Got event 2 (Ring/Answered)... Jul 30 16:25:04 ERROR[9616]: callerid.c:260 callerid_feed: fsk_serie made mylen 0 (-62) Jul 30 16:25:04 WARNING[9616]: chan_zap.c:5434 ss_thread: CallerID feed failed: Success Jul 30 16:25:04 WARNING[9616]: chan_zap.c:5476 ss_thread: CallerID returned with error on channel 'Zap/1-1' snip This seems to be a common topic in the archives! I have tried adjusting the gain to no avail. This is a Telstra (Australia) CLID service, and I have ADSL on the same line (a line filter is installed.) The fact that clidtest works suggests that the card's getting the CLID fine, but there's a problem after that. Sorry for the repeat post - I managed to post the original during the recent list 'blackout', so I guess it didn't get to many people. Any ideas would be greatly appreciated. Cheers, Jon ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] List
Neither did I.. So I called digium this afternoon and they said they would have someone look at it.. -Gerard Huddleston, Robert wrote: Is it my imagination or did I just drop off the list for several days somehow... I didn't get any posts since Friday... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: [Asterisk-ss7] Asterisk - ss7
www.footnotess7.com is now open to begin the creation of SS7 that can be used with asterisk. Sign up and list which and what parts you would like to work on. Race V. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael D Schelin Sent: Friday, July 01, 2005 2:23 PM To: CARDOSO Jorge Miguel; asterisk-users@lists.digium.com; asterisk-ss7@lists.digium.com Subject: [Asterisk-Users] Re: [Asterisk-ss7] Asterisk - ss7 I thought everyone should know this. Jorge, After reading your page in the http://voip-info.org/tiki-index.php?page=Asterisk+SS7 please advise Your U.S. customers that SS7 is not done the same way as in the rest of the world and the requirements are different. The U.S carrier's require 2 redundant links. I know this first hand because we run an SS7 network. CARDOSO Jorge Miguel wrote: http://voip-info.org/tiki-index.php?page=Asterisk+SS7 ___ Asterisk-SS7 mailing list Asterisk-SS7@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-ss7 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP WEB Phone (Wanna implement Click to Call)
Hello list, This sounds interesting. Has anyone looked at the source code of these phone clients. I would be reluctant to download and install software that could be a trojan software. Thanks, Bill Wesson -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kanuri, Seshu (Company IT) Sent: Thursday, July 28, 2005 12:05 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] SIP WEB Phone (Wanna implement Click to Call) The download link is in the url pasted in the email. You can test it from here. Click on the first link: http://www.geocities.com/babarnazmi/ Seshu -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Guillermo Salas M Sent: Thursday, July 28, 2005 12:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] SIP WEB Phone (Wanna implement Click to Call) On Thu, 2005-07-28 at 10:48 -0400, Kanuri, Seshu (Company IT) wrote: Try babar nazmi's IAX web phone. This does not have G729 or G723 but it has high bit rate codecs. http://www.geocities.com/babarnazmi/ Have you the url where can I download it? I need to test it. We at iareanet use this product as part of our virtual office solution where remote customers dial in and dial out using the IAX SoftPhone. Seshu __ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Walid Azab Sent: Thursday, July 28, 2005 10:07 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] SIP WEB Phone (Wanna implement Click to Call) Hi, I appreciate it if someone knows what is available for SIP web phones out there. I am interested in putting a soft phone on a website that registers with Asterisk using SIP. Then, when someone uses it, it directly calls into an asterisk call queue.. Any ideas? __ NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Guillermo Salas M. Telconet S.A. Manta Calle 15 y Av. 24 Esq. Phone : 593 5 262 8071 Mobile: 593 9 985 5138 SIP : [EMAIL PROTECTED] e-mail: [EMAIL PROTECTED] www : http://www.telconet.net http://www.telcocarrier.net Linux User: 255902 Soporte en Linea en http://www.manta.telconet.net Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] E1 and SS7
www.footnotess7.com is now open to begin the creation of SS7 that can be used with asterisk. Sign up and list which and what parts you would like to work on. Race V. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike M Sent: Thursday, June 09, 2005 9:46 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] E1 and SS7 On Thu, Jun 09, 2005 at 06:11:06PM -0600, Michael Welter wrote: VOIP Consultant wrote: I have the exact same problem.It would ideal if we could set an astersik box with 2 E1 ports to do an IP-to-SS7 conversion. Anyone has done this before? I'm looking a signaling gateways--does anyone have any words of wisdom? http://www.sigtran.org What's your - budget - application - connection count - traffic volume - growth plan - protocol on the IP side - link type: A or F This can go deep and wide and way OT. I'll dispense what I know off-list to anyone interested. -- Mike ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] A bit of a survey: What do do if you needmorethan 4 C.O. lines
www.footnotess7.com is now open to begin the creation of SS7 that can be used with asterisk. Sign up and list which and what parts you would like to work on. Race V. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Critchfield Sent: Sunday, February 20, 2005 4:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] A bit of a survey: What do do if you needmorethan 4 C.O. lines On Sun, 2005-02-20 at 11:47 -0500, Jim Van Meggelen wrote: Well, I appreciate everyone's input, and I'll give the matter some more thought. Just so no one stays up at night worrying, this is not a situation I am facing, it is simply a hypothetical scenario. As with so many things, there is always a trade-off between economy and functionality. The Adit 600 and T1 integration is certainly quality, but I have not been able find an economical way to do this (purchasing used equipment on eBay is fine for smaller deployments and lab gear, but not a very sound logistics strategy, and awfully difficult to explain to a customer). This would be one of those cases where you keep a couple in stock and watch the ebay auctions when your stock goes low. You will find that your customers that are looking for the cheapest solutions possible will not baulk at used equipment. It is highly likely that they will price you against a used key system or pbx. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Warning: We're Zap/XX-1,
I have the following problem: I have installed two T1 digium card (old T100P cards), plus a TDM400 with 4 fxo modules. Several times in the week I have thousands of warnings like these in the log Aug 1 08:54:47 WARNING[2243]: We're Zap/18-1, not G\u\uG Aug 1 08:54:47 WARNING[2243]: We're Zap/18-1, not G\u\uG Aug 1 08:54:47 WARNING[2243]: We're Zap/18-1, not G\u\uG Aug 1 08:54:47 WARNING[2243]: We're Zap/18-1, not G\u\uG Aug 1 08:54:47 WARNING[2243]: We're Zap/18-1, not G\u\uG Aug 1 08:54:47 WARNING[2243]: We're Zap/18-1, not G\u\uG Aug 1 08:54:47 WARNING[2243]: We're Zap/18-1, not G\u\uG Aug 1 08:54:47 WARNING[2243]: We're Zap/18-1, not G\u\uG Aug 1 08:54:47 WARNING[2243]: We're Zap/18-1, not G\u\uG I found something similar in the list: http://lists.digium.com/pipermail/asterisk-users/2004-September/064956.html but; nobody I answer this message. This it is the state of my interrupts cat /proc/interrupts CPU0 0: 510485494IO-APIC-edge timer 1:337IO-APIC-edge i8042 7: 0IO-APIC-edge parport0 8: 1IO-APIC-edge rtc 9: 2 IO-APIC-level acpi 14:1217453IO-APIC-edge ide0 16: 0 IO-APIC-level uhci_hcd, uhci_hcd 17: 510391706 IO-APIC-level Intel ICH5, t1xxp 18: 14143151 IO-APIC-level uhci_hcd, libata, eth0 19: 510477191 IO-APIC-level uhci_hcd, wctdm 22: 510396103 IO-APIC-level t1xxp NMI: 0 LOC: 510532759 ERR: 0 MIS: 0 I am thinking to replace the two T1 cards by a new TE205P Thanks in advance for any comment Jorge Verastegui redcetus.com binXHHRmGDesb.bin Description: PGP Public Key binpXE2kN59dv.bin Description: Clave PGP pública ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] call transfer
You must use the 't' 'T' options in the Dial() command when placing calls to and from the device. We had extensions that were combinations of SIP and IAX devices and didn't want/need this behavior on all of our devices so we setup our extensions with something as follows: Exten = 1000,1,Dial(Local/IAX-1000/[EMAIL PROTECTED]Local/SIP-1000/[EMAIL PROTECTED], 60, r) [devices] Exten = SIP-1000,1,Dial(SIP-XYZ, 60, tr) Exten = IAX-1000,1,Dial(IAX-ABC, 60, r) That will ring both devices using different dial statements for each. Cullin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Monday, August 01, 2005 2:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] call transfer Hi! I have searched answer how can I transfer calls with asterisk,with no result. Can you advice me and show some example file how can I use SIP phone to transfer calls by hitting # and get the Transfer prompt and enter an extension I want to transfer to? Thanks for your answers This mail sent through L-secure: http://www.l-secure.net/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to install PHPAGI?
let me know if phpagi is a product, i tought it was just a php class for programming agi php scripts. best regards On 8/1/05, Leo Burd [EMAIL PROTECTED] wrote: Hello everyone, Where can I find instructions on how to install PHPAGI? BTW, what's the difference between PHPAGI and PHPAGI2? Are they different products? It's hard to tell from voip-info.org... Best, Leo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Voicemail envelope time is 4 hours ahead
I had the same problem in 1.0.9. We fixed it by moving the [zonemessages] section above the [general] section so that it gets processed first. Cullin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Frank Tarczynski Sent: Monday, August 01, 2005 6:53 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Voicemail envelope time is 4 hours ahead I'm running a recent CVS build under Solaris 10. In the shell than I'm running the Asterisk console I have TZ=US/Eastern and in my voicemail.conf I have tz=eastern and eastern=America/New_York|'vm-received' Q 'digits/at' IMp. The voicemail envelope information seems to be exactly 4 hours ahead. No matter what I try I can't seem to find the cause. Any ideas? Frank ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ast_config not updating voicemail password
I've been using realtime to store my voicemail configuration in a mysql table for several months now, and have had no problems...until today. A few weeks ago, I upgraded to the latest CVS and today I noticed voicemail is not updating the password when the user changes it through option 0. I'm not sure when this started happening, but I assume it was sometime after I upgraded. Has anyone else seen a problem like this, and if so, what's the solution? TIA Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MFC/R2
i have a very problem , how to configure MFC/R2 with asterisk, I'am install o module but while asterisk loaded is module is broken ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Configuring [EMAIL PROTECTED] with Analog Phones
Working [EMAIL PROTECTED] 1.3 two 4 port TDM100 WildCards, 3 port FXS, 4 port FXO. I've been able to work the FXO ports out and been able to make and receive calls using softtel PC phones. I'm having difficulty with configuring 4 line non-PBX analogs to function on the FXS side tho.. I've tried using ZAP protocols as some techs have suggested, but all I get are slow busy signals. If someone has the procedures to configure [EMAIL PROTECTED] for analog phones, this would be greatly appreciated. RC ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Configuring [EMAIL PROTECTED] with Analog Phones UPDATED
Er, make that TDM400P cards... X.X rc ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dialplan to dial SIP, but stop dial on analog pick up?
Most of the documentation I have read through shows dial plan examples that
dial the SIP phones and stop if one is picked up. I have not seen an example
of or read how to stop the SIP dial when an analog phone is answered.
How can the extension be set up so that when an analog phone is picked up
the SIP dial stops?
extensions.conf
_
[incoming]
exten = s,1,Dial(SIP/2001SIP/2002,20,tr)
exten = s,2,Answer
exten = s,3,Hangup
[outgoing]
exten = _9X.,1,NoOp(Call for ${EXTEN:1})
exten = _9X.,2,Dial(Zap/1/${EXTEN:1})
[default]
; desktop
exten = 2001,1,Dial(SIP/2001,30)
exten = 2001,2,Hangup
exten = desktop,1,goto(2001,1) ; To be able to dial with text, sonya
; Laptop
exten = 2002,1,Dial(SIP/2002,30)
exten = 2002,2,Hangup
exten = laptop,1,goto(2002,1) ; To be able to dial with text, laptop
include = outgoing
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[Asterisk-Users] Questions on Asterisk and CallerID
Hello,
I have few questions about Asterisk.
I installed Asterisk from CVS on FreeBSD and I made cvsup 2 days ago.
1.I couldn't find Asterisk version using asterisk -V command.
How can I to find version information?
2.I am using Wildcard X101P (FXO) and Wildcard TDM400P REV E/F (FXS)on
it.
I tried Asterisk CallerID feature, but unable to get it.
I tried callerid signaling V23, Bell202, DTMF, no success. Finally, I
found in our country (Mongolia) PSTN/Cellular provider send FSK/ETSI
type of CallerID.
Is Asterisk support such type of CallerID signaling?
If no, is there any way to get it?
3.I enjoyed Asterisk most of feature until now. I registered X-Pro
softphone, SIP analog and analog phone connected to FXS port too.
There one problem is I am unable to make outgoing call from SIP phone,
softphone, analog phone through FXO port.
Following is my Asterisk configuration:
--
zaptel.conf
loadzone=us
defaultzone=us
fxsks=1
fxoks=2
zapata.conf
context=bell
signaling=fxs_ks
group=1
channel = 1
context=home
group=2
signalling=fxo_ks
channel = 2
sip.conf
[]
type=friend
username=
;secret=
host=dynamic
nat=yes
defaultip=192.168.1.5
context=bell
reinvite=no
canreinvite=no
callerid=
[EMAIL PROTECTED]
allow=g729
allow=g723
allow=all
[]
type=friend
username=
;secret=
host=dynamic
nat=yes
defaultip=192.168.1.1
context=bell
reinvite=no
canreinvite=no
callerid=
[EMAIL PROTECTED]
allow=g729
allow=g723
extensions.conf
[bell]
exten = s,1,Wait
exten = s,2,Answer
exten = s,3,Playback(greetings)
exten = s,4,WaitExten
; used to record prompts
exten = 205,1,Wait(2)
exten = 205,2,Record(/tmp/greetings:alaw)
exten = 205,3,Wait(2)
exten = 205,4,Playback(/tmp/greetings)
exten = 205,5,Wait(2)
exten = 205,6,Hangup
exten = 111,1,Dial(CONSOLE/dsp)
exten = 111,2,Hangup
exten = 100,1,Answer
exten = 100,2,MusicOnHold()
exten = 100,4,Hangup
exten = 200,1,VoicemailMain
exten = 300,1,Dial(Zap/2)
exten = 400,1,Voicemail(9)
exten = 800,1,MeetMe(100|Mp)
exten = 800,2,Hangup
exten = 601,1,WaitMusicOnHold(30)
exten = 700,1,Dial(SIP/,20,rt)
exten = 900,1,Dial(SIP/,20,rt)
exten = _ZXXX,1,Answer
exten = _ZXXX,2,Dial(Zap/g1/${EXTEN})
exten = _Z,1,Answer
exten = _Z,2,Dial(Zap/g1/${EXTEN})
exten = _NX,1,Answer
exten = _NX,2,Dial(Zap/g1/${EXTEN})
exten = _NXXX,1,Answer
exten = _NXXX,2,Dial(Zap/g1/${EXTEN})
[home]
exten = s,1,Playback(greetings)
exten = 100,1,Answer
exten = 100,2,MusicOnHold()
exten = 100,4,Hangup
exten = 111,1,Dial(CONSOLE/dsp)
exten = 111,4,Hangup
exten = 700,1,Dial(SIP/,20,rt)
exten = 900,1,Dial(SIP/,20,rt)
exten = _ZXXX,1,Answer
exten = _ZXXX,2,Dial(Zap/g1/${EXTEN})
exten = _Z,1,Answer
exten = _Z,2,Dial(Zap/g1/${EXTEN})
exten = _NX,1,Answer
;exten = _NX,2,SetVar(TIMEOUT(AbsoluteTimeout)=10)
exten = _NX,3,Dial(Zap/g1/${EXTEN})
exten = _NXXX,1,Answer
exten = _NXXX,2,Dial(Zap/g1/${EXTEN})
I can to see following in /var/log/messages when I make outgoing call.
Jul 20 00:50:26 boldsoft kernel: Zapata Telephony Interface Registered
on major 196
Jul 20 00:50:26 boldsoft kernel: ZapTel device: vendor=e159 device=1
subvendor=8085
Jul 20 00:50:26 boldsoft kernel: wcfxo0: Wildcard X101P port
0xe800-0xe8ff mem 0xfaffe000-0xfaffefff irq 18 at device 9.0 on
pci2
Jul 20 00:50:26 boldsoft kernel: ZapTel Attach for wcfxo0: deviceID :
0xe159
Jul 20 00:50:26 boldsoft kernel: wcfxo: DAA mode is 'FCC'
Jul 20 00:50:26 boldsoft kernel: Found a Wildcard FXO: Wildcard X101P
Jul 20 00:50:26 boldsoft kernel: ZapTel device loaded.
Jul 20 00:50:33 boldsoft kernel: FXS device: vendor=e159 device=1
subvendor=b100
Jul 20 00:50:33 boldsoft kernel: wcfxs0: Wildcard TDM400P REV E/F
port 0xec00-0xecff mem 0xfafff000-0xfaff irq 17 at dev
ice 8.0 on pci2
Jul 20 00:50:33 boldsoft kernel: FXS Attach for wcfxs0: deviceID :
0xe159
Jul 20 00:50:33 boldsoft kernel: Freshmaker version: 63
Jul 20 00:50:33 boldsoft kernel: Freshmaker passed register test
Jul 20 00:50:35 boldsoft kernel: Module 0: Installed -- AUTO FXS
Jul 20 00:50:35 boldsoft kernel: ProSLIC sanity check failed
Jul 20 00:50:35 boldsoft kernel: Module 1: Not installed
Jul 20 00:50:35 boldsoft kernel: ProSLIC sanity check failed
Jul 20 00:50:35 boldsoft kernel: Module 2: Not installed
Jul 20 00:50:35 boldsoft kernel: ProSLIC sanity check failed
Jul 20 00:50:35 boldsoft kernel: Module 3: Not installed
Jul 20 00:50:35 boldsoft kernel: Found a Wildcard TDM: Wildcard
TDM400P
REV E/F (4 modules)
Jul 20 00:50:39 boldsoft kernel: Registered tone zone 0 (United States
/ North America)
Jul 21 02:36:28 boldsoft kernel: DIAL: T345598w
Jul 21 02:39:43 boldsoft kernel: DIAL: T345598w
Jul 21 02:45:35 boldsoft kernel: DIAL: T345598w
Jul 21 02:45:56 boldsoft kernel: DIAL: T99114909w
Jul 21 02:47:09 boldsoft kernel: DIAL: T345598w
Jul 21 02:47:56 boldsoft kernel: DIAL: T345595w
Jul 21 02:48:16 boldsoft
Re: [Asterisk-Users] Cisco 7940 - Disappearing Clock
The clock on cisco phones 'disappears' when it fails to receive updates from the ntp server. This is most likely due to your ntp server configuration. By default the ntp mode on your cisco phone is directedbroadcast. If your ntp server doesn't support this you will need to change the mode on your phone to unicast. This is well documented by Cisco - Cisco SIP IP Phone Administrator Guide http://www.cisco.com/en/US/products/sw/voicesw/ps2156/products_administration_guide_book09186a00801d1978.html Please read! Cheers Sophus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco Call manager
Anybody using Cisco Call Manager and connecting to any SIP termination service like voipjet, voxee, etc? Please msg me offlist. AK ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Asterisk-Users Digest, Vol 12, Issue 171
If the SC420 is sharing interrupts, can you go around that by chaning slots or maybe, I don't know if it can do APIC? Or how about disabled the shared devie like USB? |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Joe McConnaughey |Sent: Lunes, 25 de Julio de 2005 12:48 p.m. |To: asterisk-users@lists.digium.com |Subject: [Asterisk-Users] Re: Asterisk-Users Digest, Vol 12, Issue 171 | |The cheap ones on EBay won't work with the SC420 server. I |have one and can't make any of the clones work. I do have one |TDM40B card for analog stations that works well. The problem |with the SC420 is that it won't let you set the interrupts |yourself and you end up with interrupts being shared. | |=== | |Message: 26 |Date: Mon, 25 Jul 2005 08:31:47 -0500 |From: Eric Wieling aka ManxPower [EMAIL PROTECTED] |Subject: Re: [Asterisk-Users] Need Advice |To: [EMAIL PROTECTED], Asterisk Users Mailing List - |Non-Commercial Discussion asterisk-users@lists.digium.com |Message-ID: [EMAIL PROTECTED] |Content-Type: text/plain; charset=us-ascii; format=flowed | |Nathan Pralle wrote: | However, for FXO ports, I'm using the Digium Wildcard X100P's which | can be obtained on eBay for $9-$20, usually. Much cheaper | price-per-port, although the TDM would give better expandibility. | |You mean NON Digium X100P's. Digium no longer sells the |X100P. The cheap ones on eBay are clone cards. | |-- |Eric Wieling * BTEL Consulting * 504-210-3699 x2120 | | |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom SoundPoint 600 : 10 seconds of delay when answering a call.
Hello everyone, I have just received 3 brand new Polycom SoundPoint IP 600 from voisupply.com and I have the exact same problem on all of them. When I receive a call, the phone is ringing correctly but when I answer it, it takes exactly 10 seconds before I can hear the caller. I also have SoundPoint 300 and 301 but I don't have that problem with those. I'm using Asterisk 1.0.7. I checked the user guide and admin guide from Polycom but didn't see anything interesting. Does anyone encounter this problem? Any idea? Ken ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] test message - ignore me
You are duly ignored. Matt Hess wrote: Haven't seen email since the 29th.. just testing. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] g729 liscence question
Thanks everybody for answering me. Yes, I plan to connect SIP phone outside of my network. Infact, I am going to use Asterisk as my PSTN gateway and voice mailbox. Also, I have plan to add two more FXO card when I will have bigger network. Sounds like, I should get two liscences at this moment. Thanks again. -Original Message- From: [EMAIL PROTECTED] Sent: Mon, 01 Aug 2005 18:19:03 -0500 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] g729 liscence question On Mon, 2005-08-01 at 10:53 -0800, Innocent Evil wrote: I have a TDM400P with one FXS and one FXO.. how many liscence(s) I will have to buy? Short answer: None. Long answer: Zap interfaces use G711 and do not need G729 to work. Only if you plan to connect SIP or IAX phones from outside your local network do you really need voice compression. -- Telecomunicaciones Abiertas de Mexico Carlos Chavez Director de Tecnologia +52-55-91169161 Ext. 2001___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Configuring [EMAIL PROTECTED] with Analog Phones
Please look on the [EMAIL PROTECTED] Fourms On Mon, 1 Aug 2005, Robert Chapin wrote: Working [EMAIL PROTECTED] 1.3 two 4 port TDM100 WildCards, 3 port FXS, 4 port FXO. I've been able to work the FXO ports out and been able to make and receive calls using softtel PC phones. I'm having difficulty with configuring 4 line non-PBX analogs to function on the FXS side tho.. I've tried using ZAP protocols as some techs have suggested, but all I get are slow busy signals. If someone has the procedures to configure [EMAIL PROTECTED] for analog phones, this would be greatly appreciated. RC ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Configuring [EMAIL PROTECTED] with Analog Phones
Have you used the automatic configuration script for the zaptel drivers? IF so, have you added ZAP extensions in AMP for your analog phones? Tom -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Chapin Sent: Monday, August 01, 2005 9:13 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Configuring [EMAIL PROTECTED] with Analog Phones Working [EMAIL PROTECTED] 1.3 two 4 port TDM100 WildCards, 3 port FXS, 4 port FXO. I've been able to work the FXO ports out and been able to make and receive calls using softtel PC phones. I'm having difficulty with configuring 4 line non-PBX analogs to function on the FXS side tho.. I've tried using ZAP protocols as some techs have suggested, but all I get are slow busy signals. If someone has the procedures to configure [EMAIL PROTECTED] for analog phones, this would be greatly appreciated. RC ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] [EMAIL PROTECTED]/Grandstream Call Transfer
I may be mistaken, but in [EMAIL PROTECTED], can't you just press # and dial the extension number , speak, and hang up? Tom -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Phoneguy Sent: Monday, August 01, 2005 7:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] [EMAIL PROTECTED]/Grandstream Call Transfer There are 2 methods blind and announced here you go: Blind:Call someone, or receive a call. Hit 'Trnf' The screen displays TRANSFER TO? and you hear a dial tone. The other end can still hear you, so don't say anything nasty. Dial the number and hit 'Send', caller is transferred (blind) Announced: a.. Be on a call b.. Push a LINE button that isn't in use (this puts the call on hold) c.. Dial the extension you wish to transfer to d.. Speak e.. Push TRNF - Original Message - From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, August 01, 2005 2:03 PM Subject: [Asterisk-Users] [EMAIL PROTECTED]/Grandstream Call Transfer OK, now this should be really simple, but I am a bit of a newbie so please bear with me. I have an [EMAIL PROTECTED] box setup with TDM04B and two POTS lines. On the SIP side, I have GXP2000 phones. Most things seem to work, but the users cannot figure out how to transfer incoming calls from one extension to another. Now I am not sure that I have things setup correctly, but is there something special that needs to be done in order to transfer calls?? The GXP2000 has a Transfer button on the keypad but that doesnt seem to allow a transfer. Is there something that I am missing?? Any help would be greatly appreciated. Marc ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Configuring [EMAIL PROTECTED] with Analog Phones
Yes and yes. Zap Trunks and Extensions were added for the analogs. rc Tom Rymes wrote: Have you used the automatic configuration script for the zaptel drivers? IF so, have you added ZAP extensions in AMP for your analog phones? Tom -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Chapin Sent: Monday, August 01, 2005 9:13 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Configuring [EMAIL PROTECTED] with Analog Phones Working [EMAIL PROTECTED] 1.3 two 4 port TDM100 WildCards, 3 port FXS, 4 port FXO. I've been able to work the FXO ports out and been able to make and receive calls using softtel PC phones. I'm having difficulty with configuring 4 line non-PBX analogs to function on the FXS side tho.. I've tried using ZAP protocols as some techs have suggested, but all I get are slow busy signals. If someone has the procedures to configure [EMAIL PROTECTED] for analog phones, this would be greatly appreciated. RC ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New digium TE406 411
We will start installing TE411 next week, I'll keep the list informed ! jack Eric Rees wrote: Has anyone on the list tried one of these new cards with built-in echo cancellation? This electronic message transmission, including attachments, is for the exclusive use of the individuals to which this e-mail is addressed and is to be reviewed and used exclusively for authorized company purposes. This transmission may contain proprietary, confidential or privileged information. If you are not the intended recipient of this transmission, you are hereby notified that any use, copying, disclosure, dissemination, distribution or taking of any action in reliance upon the contents of this transmission is strictly prohibited. If you believe you may have received this electronic message in error, please notify the sender immediately by return email and delete or destroy the original message and/or any copy of it from your computer system and/or your files. Thank you. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: IAX Devices Recommendation
Hi We purchased the AT320-EE IAXtalk phone from www.iaxtalk.com which ocnnects to our own asterisk server. Good value, a little tricky to set up - the instructions they supply to which they give you a link on their web site are OK, but their are some gaps which the asterisk wiki pages fill well - cannot find this at the moment but it explains how to do resets. IN summary you buy the phone and then upload the firmware for IAX2 protocol. Configuration is via web browser which works well. Automaticlaly logs in. Works well. Slightly slower to respond than (say) firefly softphone which we use for most users - the hardphone is for reception and as backup in case of computer failure. Paul Redstone ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM400P REV I issues - ProSLIC vs TDM400P
The REV I card shows up in the PCI table as:
02:05.0 Network controller: Tiger Jet Network Inc. Intel 537 (or
02:05.0 Class 0280: e159:0001)
Subsystem: Unknown device b119:0001
But the REV E/F shows up as:
02:0d.0 Communication controller: Tiger Jet Network Inc. Tiger3XX
Modem/ISDN interface (or
02:0d.0 Class 0780: e159:0001)
Subsystem: Unknown device b100:0003
One is class 0780, one is class 0280. I don't know if this normal,
but it might be an indication of the problem.
I managed to probe it with zaptel 1.0.8 correctly once after which
the box paniced.
$ sudo /sbin/modprobe wcfxs zaptel
kernel: Zapata Telephony Interface Registered on major 196
kernel: Freshmaker version: 73
kernel: Freshmaker passed register test
kernel: Module 0: Installed -- AUTO FXS/DPO
kernel: Module 1: Installed -- AUTO FXO (FCC mode)
kernel: Module 2: Installed -- AUTO FXO (FCC mode)
kernel: Module 3: Installed -- AUTO FXO (FCC mode)
kernel: Found a Wildcard TDM: Wildcard TDM400P REV I (4 modules)
kernel: Registered tone zone 0 (United States / North America)
One restart later, it fails:
kernel: Zapata Telephony Interface Registered on major 196
/lib/modules/2.4.21-4.EL/misc/wcfxs.o: init_module: No such device
Hint: insmod errors can be caused by incorrect module parameters, including
invalid IO or IRQ parameters.
You may find more information in syslog or the output from dmesg
/lib/modules/2.4.21-4.EL/misc/wcfxs.o: insmod
/lib/modules/2.4.21-4.EL/misc/wcfxs.o failed
/lib/modules/2.4.21-4.EL/misc/wcfxs.o: insmod wcfxs failed
Very confusing. Adding some printk()s to wcfxs.c, for example in
wcfxs_init() just before the call to pci_module_init() and as the
first command in wcfxs_init_one(), shows that pci_module_init()
gets called but wcfxs_init_one() never gets called.
In the PCI table of wcfxs_pci_tbl, if I add
{ 0xe159, 0x0001, 0xb119, PCI_ANY_ID, 0, 0, (unsigned long) wcfxsi },
(as done revision 1.116 of wctdm.c)
to it, the modprobe works (at least wcfxs_init_one() gets called):
kernel: Freshmaker version: 73
kernel: Freshmaker passed register test
But then, the full story is:
kernel: Zapata Telephony Interface Registered on major 196
kernel: Freshmaker version: 73
kernel: Freshmaker passed register test
kernel: Module 0: Installed -- AUTO FXS/DPO
kernel: Module 1: Installed -- AUTO FXO (FCC mode)
kernel: Module 2: Installed -- AUTO FXO (FCC mode)
kernel: Module 3: Installed -- AUTO FXO (FCC mode)
kernel: Found a Wildcard TDM: Wildcard TDM400P REV I (4 modules)
and the next attempt gives me:
kernel: Freshmaker version: 73
kernel: Freshmaker passed register test
kernel: ProSLIC on module 0, product 0, version 2
kernel: ProSLIC on module 0 seems sane.
kernel: ProSLIC on module 0 powered up to -74 volts (c6) in 10 ms
kernel: Loop current set to 20mA!
kernel: Post-leakage voltage: 48 volts
kernel: ProSLIC on module 0 powered up to -75 volts (ca) in 0 ms
kernel: Loop current set to 20mA!
kernel: Calibration Vector Regs 98 - 107:
kernel: 98: 11
kernel: 99: 10
kernel: 100: 00
kernel: 101: 00
kernel: 102: 06
kernel: 103: 34
Does this still make sense to anybody?
Edwin
--
Edwin Groothuis |Personal website: http://www.mavetju.org
[EMAIL PROTECTED]| Weblog: http://weblog.barnet.com.au/edwin/
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RE: [Asterisk-Users] New digium TE406 411
I have just installed the TE4110P card, found no real issues. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of pbx Sent: Tuesday, August 02, 2005 4:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] New digium TE406 411 We will start installing TE411 next week, I'll keep the list informed ! jack Eric Rees wrote: Has anyone on the list tried one of these new cards with built-in echo cancellation? This electronic message transmission, including attachments, is for the exclusive use of the individuals to which this e-mail is addressed and is to be reviewed and used exclusively for authorized company purposes. This transmission may contain proprietary, confidential or privileged information. If you are not the intended recipient of this transmission, you are hereby notified that any use, copying, disclosure, dissemination, distribution or taking of any action in reliance upon the contents of this transmission is strictly prohibited. If you believe you may have received this electronic message in error, please notify the sender immediately by return email and delete or destroy the original message and/or any copy of it from your computer system and/or your files. Thank you. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.9.8/61 - Release Date: 8/1/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.9.8/61 - Release Date: 8/1/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400P REV I issues - ProSLIC vs TDM400P
On Tue, Aug 02, 2005 at 02:48:00PM +1000, Edwin Groothuis wrote: The REV I card shows up in the PCI table as: 02:05.0 Network controller: Tiger Jet Network Inc. Intel 537 (or 02:05.0 Class 0280: e159:0001) Subsystem: Unknown device b119:0001 But the REV E/F shows up as: 02:0d.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface (or 02:0d.0 Class 0780: e159:0001) Subsystem: Unknown device b100:0003 One is class 0780, one is class 0280. I don't know if this normal, but it might be an indication of the problem. I managed to probe it with zaptel 1.0.8 correctly once after which the box paniced. What is it exactly? IIRC zaptel 1.0.9 is basically 1.0.8 with the added support for TDM REV I. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Can you caculate with me?
And where did you get your rate? The 11/2004 rates from nufone show: Taiwan 886 0.0469 Taiwan - Mobile/Special Services886 60 0.1006 Taiwan - Mobile/Special Services886 70 0.1006 Taiwan - Mobile/Special Services886 9 0.1006 Taiwan - Taipei 886 2 0.0469 0.0469 * 15.5 = 0.72695 Your multiplication skills are great, now if you only had the right numbers. -Original Message- From: Ronald Wiplinger [mailto:[EMAIL PROTECTED] Sent: Thursday, July 28, 2005 10:48 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Can you caculate with me? Bob Goddard wrote: On Thursday 28 Jul 2005 13:07, Ronald Wiplinger wrote: before I accuse somebody to overbill I would like you to calculate with me: Rate: 0.0189 for calling Taiwan via NuFone Duration: 930 seconds Lets vote for the answers:0.7269 or 0.2929 ??? Assuming it is per minute; 930 * 0.0189 / 60 = 0.29295 Thanks for your help. NuFone invoiced for me 0.7269 Asking for an answer from them, ... Guess what??? NO ANSWER at all. bye Ronald Wiplinger ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/aster isk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] what is the problem with gmail and the list.
On 8/2/05, John Novack [EMAIL PROTECTED] wrote: Rich Adamson wrote: I have not been receiving mail from the list 29th July, what is the problem with gmail and the list. I suspect that any dump the list might or might not have taken isn't the complete story. m using gmail and suddenly got all the mails after 29th july... on 2nd august. --gurmi John Novack ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: List
I lost a few too. I jumped from Vol. 12 Issue 199 to Vol. 13 Issue 3. Anybody know exactly how many issues to a volume? - I've seen it vary quite a bit (i.e. 208 issues in Vol. 11 / 268 in vol. 9). Would be kinda' nice if they'd pick a number (like maybe 200?) and stick with it??? If somebody's already addressed this, sorry for the duplication. Gary Guthary ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: TDM400P REV I issues - ProSLIC vs TDM400P
On Mon, Aug 01, 2005 at 11:56:26PM -0500, [EMAIL PROTECTED] wrote: On Tue, Aug 02, 2005 at 02:48:00PM +1000, Edwin Groothuis wrote: The REV I card shows up in the PCI table as: 02:05.0 Network controller: Tiger Jet Network Inc. Intel 537 (or 02:05.0 Class 0280: e159:0001) Subsystem: Unknown device b119:0001 But the REV E/F shows up as: 02:0d.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface (or 02:0d.0 Class 0780: e159:0001) Subsystem: Unknown device b100:0003 One is class 0780, one is class 0280. I don't know if this normal, but it might be an indication of the problem. I managed to probe it with zaptel 1.0.8 correctly once after which the box paniced. What is it exactly? IIRC zaptel 1.0.9 is basically 1.0.8 with the added support for TDM REV I. The TDM400P REV I card. Edwin -- Edwin Groothuis |Personal website: http://www.mavetju.org [EMAIL PROTECTED]| Weblog: http://weblog.barnet.com.au/edwin/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: binding asterisk-h323 on two interfaces
I have cvs-head of Aug-2. README has no information on how to bind asterisk-h323 on multiple interfaces. actually this was my question that can we bind asterisk-h323 on multiple interfaces ? as h323.conf says that bindaddr should contain a single valid IP. if we bind h323 to 0.0.0.0 as we do in SIP, it sends 0.0.0.0 as it is to the caller and callee. Use cvs -head code from the last day or two and read the README. Jeremy McNamara ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] register Every user without auth
hello is there any way to register all user without declaring them in sip.conf. because i want all users to auth. thanks in advance Kamran Start your day with Yahoo! - make it your home page http://www.yahoo.com/r/hs ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Queue/Agents
Looking for a good web app that will show agents that are login to queue. I tried the operator panel but I'm unable to get the LED to change color per the doco I have.. It works well for everything else but no luck on the agent part.. How are your agents loging into queues? Depending on that you should use slightly different configurations. Contact me off list if you need assistance. Regards, -- Nicolás Gudiño Buenos Aires - Argentina ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
