Re: [asterisk-users] Any way of limiting incoming caller connection time without making 2 active calls for each incoming call?

Jonathan, Have you tried: same => n,Set(TIMEOUT(absolute)=3540) You could override later if required. Thanks, Dan On 17 Apr 2017 04:49, "Jonathan H" wrote: The following setup prevents callers from going over 59 minutes: --

Re: [asterisk-users] HP DL360 G5 better than HP DL360 G7 ?

On 24 July 2012 15:18, Hans Witvliet aster...@a-domani.nl wrote: On Sun, 2012-06-03 at 23:23 -0400, Tom Browning wrote: Any tips on solving the following performance conundrum: Asterisk 1.8.12.2 running on HP DL360 G5 and G7s tcpdump running to capture UDP 5060/SIP signaling to .pcap

Re: [asterisk-users] Realtime Issue

How about... exten = _X.,1,NoOp() exten = _X.,2,Set(DEVICE=${CUT(CHANNEL,,1)}) exten = _X.,3,Set(NULL=${REALTIME(agents,device,${DEVICE})}) exten = _X.,4,Set(usernamepair=${CUT(NULL,\,,1)}) exten = _X.,5,Set(username=${CUT(usernamepair,=,2)}) exten = _X.,6,NoOp(DEVICE is ${DEVICE}) exten =

Re: [asterisk-users] Got Anonymous from DID incoming call and can't re-send to another asterisk with new callerid

On 26 February 2010 12:38, Trevor Peirce tpei...@digitalcon.ca wrote: Charles Wang wrote: The sip.conf of MYE1 likes below: [MYPBX] type=peer host=mypbx.abc.com http://mypbx.abc.com nat=no disallow=all allow=g729 canreinvite=yes qualify=no context=default

Re: [asterisk-users] Redirect call based on CLI???

Or, alternatively using the 's' priority... exten = 845,1,Verbose(3, Incoming call from ${CALLERID(all)}) ; this will be priority 1 exten = 845/12345678,n,Goto(blacklist) ; the n will make this priority 2 exten = 845/23456789,s,Goto(blacklist) ; the s will make this also priority 2 exten =

Re: [asterisk-users] problem in upgrading to 1.6.1.0

In the 1.6.1.* branch the line type=peer seems to be required on each user... d 2009/9/19 Örn Arnarson o...@arnarson.net Sorry I wasn't more specific. The error message is just the standard 'Can't find that extension'. The problem is, however, that asterisk parses users.conf (and doesn't

Re: [asterisk-users] Call getting stucked !!

Check your VICIdial logs and try to debug the VICIdial side of things... It could be something along the lines of agent hits hangup, web interface goes to add hangup command into the manager queue, fails due to a lock on the table so call stays up... This wouldn't be an asterisk issue... You need

Re: [asterisk-users] Meaning of requested special control 20, passing it to SIP

2009/8/15 John Novack jnov...@stromberg-carlson.org Received this on the console -- IAX2/76.21.238.129:4569-4986 requested special control 20, passing it to SIP/magicjack-08225a58 Did a Google search, but reached a dead end Can anyone explain? Something need to be changed in my

Re: [asterisk-users] Cisco 7960 Multiline phone

With that phone what you really probably want to do is just configure them all with the same details... i.e. # Line 1 appearance line1_name: incoming line1_shortname: Incoming (Line1) line1_authname: incoming line1_password: password # Line 2 appearance line2_name: incoming line2_shortname:

Re: [asterisk-users] 7940g

2009/8/11 Chuck Coleman p...@2cci.com I have 6 Cisco 7940g phones and I would like to add them to my Asterisk 2.6.2 box. My SNOM 320 work just fine but I cannot get the Cisco’s to register. They pull the TFTP P0S3-08-9-00 just fine. I change the NAT to * no* but it still does not register.

Re: [asterisk-users] Going to VM after 180 seconds in queue

2009/8/8 Dan Pilcheck pilch...@gmail.com Hello all, This is a VICIDial server and I am looking to send calls to VM box 2100 after 3 minutes of sitting in the queue(via the VICIDial AGI). This would be inserted between exten = s,8,Background(open) and exten = s,9,AGI. From what voip-info

Re: [asterisk-users] Voicemail feature: enable or disable the ability to leave a message

Perhaps it's only basic in certain parts of the world... I know I've never experienced a voicemail system with such a feature... I'm not saying having the option would be bad... but... I'd prefer voicemail to get some more common, more requested of me, features first and that's personally where

Re: [asterisk-users] how to implement CLONED LINE Feature in asterisk?

2009/8/4 Faheem faheem_...@yahoo.com how to implement CLONED LINE Feature in asterisk Hey, I want to implement Clone Line feature in asterisk. I am using SPA-2100. The feature should work in this way. There are two ports in the SPA-2100 both are registered with asterisk with same

Re: [asterisk-users] Message Waiting Indicator on DAHDI line

2009/8/5 Mike asterisk-us...@norgie.net On Tue, Aug 04, 2009 at 03:35:22PM -0500, Doug Bailey wrote: This code is designed to handle Message Waiting Indication (MWI) incoming on FXO line. This data could very well be embedded in your CID spill as part of an MDMF message that also

Re: [asterisk-users] how to implement CLONED LINE Feature in asterisk?

point here is in chain_sip.c what are variables or structure that need to maintain so that we can consider all registered users as active users. Thanks! Faheem --- On *Wed, 8/5/09, D Tucny d...@tucny.com* wrote: From: D Tucny d...@tucny.com Subject: Re: [asterisk-users] how to implement

Re: [asterisk-users] asterisk 1.6 call forwarding

2009/7/31 pepesz76 pepes...@o2.pl Dear All, I'n trying to make a simple call forwarding, however I have small problem when evaluating an expresion. Here is my extensions.conf ... ; Unconditional Call Forward exten = _#21*X.,1,Set(DB(CFIM/${CALLERID(num)})=${EXTEN:4}) exten =

Re: [asterisk-users] MeetMe Options Enter Leave Sound

2009/7/24 Stefan Schmidt s...@sil.at Hello, i´ve a question about the Meetme Options. How could i play a enter and leave sound but without recording the user name first. I just want a User joined conferenc and a user leaved. With the i or I Option i have to record the name first. Is

Re: [asterisk-users] Not getting inbound CallerID name on Asterisk

Looking at the code, asterisk doesn't know how to handle RPID in an INFO message, so it just responds with an OK and goes on with it's business... The fact that the message has the name of the called party, rather than the calling party probably wouldn't help even if Asterisk did understand it...

Re: [asterisk-users] PHP AGI Problems

2009/5/27 Atlanticnynex atlanticny...@gmail.com (Accidentally posted this to asterisk-dev, should be here) fgets is only returning one character... either when run as an AGI or run as a test on PHP on CLI... Example, enter , then fgets returns '3'. It's not... There are problems with

Re: [asterisk-users] Question with Asterisk and call waiting ${CALLERID(name/num)}

2009/4/30 Justin Piszcz jpis...@lucidpixels.com Hello, I am using an SPA3102, all is working with asterisk 1.4, voice mail, outbound calling etc, and it even passes the cid name/num to my analog phone. However, when someone is calling me, I hear the beeps but the caller-id information is

Re: [asterisk-users] Faxing and TIFF files

2009/4/22 Michael mich...@networkstuff.co.nz I use GPL Ghostscript 8.6.2 to produce the TIFF files for faxing. Does anyone know of a way, either while producing the file, or after, to tell how many pages have been produced? (without manually viewing the file) tiffinfo? then count the

Re: [asterisk-users] run dialplan when open line

2009/4/22 michel freiha mich...@gmail.com Hi all, Does asterisk support the following scenario? I need when a customer who own an endpoint registered on asterisk open the line, the asterisk will run a specific AGI script inside the endpoint context? You mean when they pick up the phone

Re: [asterisk-users] opening 2 and more channels on 1 SIP account

2009/4/18 Tamer Higazi th9...@googlemail.com Scenario: I have a Asterisk PBX with a cologne chipset ISDN BRI card on it a DSP cpu to take out the echo cancellation. Communication is done through the chan_capi interface module. If a call comes inside, and I forward it to the SIP account

Re: [asterisk-users] opening 2 and more channels on 1 SIP account

2009/4/18 Tamer Higazi th9...@googlemail.com D Tucny schrieb: 2009/4/18 Tamer Higazi th9...@googlemail.com mailto:th9...@googlemail.com Scenario: I have a Asterisk PBX with a cologne chipset ISDN BRI card on it a DSP cpu to take out the echo cancellation

Re: [asterisk-users] dialling multiple extensions in an internal context

2009/4/19 Matthew Pounsett m...@conundrum.com -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi there. I've done some googling around to try and find an example of what I'm trying to do, but it's one of those things that just seems hard to find the right terms to search for. If there's

Re: [asterisk-users] dialling multiple extensions in an internal context

2009/4/19 Matthew Pounsett m...@conundrum.com -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On 18-Apr-2009, at 17:17, John F. Ervin wrote: Can't you handle that by defining an outbound route? set it to hit a trunk or set of trunks when the correct dial pattern is detected? Probably,

Re: [asterisk-users] 2B Channel Transfer on XO-based T1

2009/4/15 Olivier oza-4...@myamail.com May I ask : beside saving channels, what are the benefits of TBCT over bridging calls inside Asterisk ? I'm not aware of anything apart from saving channels... What about caller ids ? I would say caller id should passed over to final callee after

Re: [asterisk-users] duration of rfc2833 generated dtmf

2009/4/15 John covici cov...@ccs.covici.com Its not there and the link you gave me says its for sip originating rather than calls to a sip channel. on Tuesday 04/14/2009 Brent Davidson(br...@texascountrytitle.com) wrote It's been around awhile. I've used it in 1.4 Check out this link for

Re: [asterisk-users] Asterisk-beginner : cannot make phonecalls using Asterisk (update)

2009/4/14 jonas kellens jonas.kell...@telenet.be [r...@asterisk asterisk]# netstat -a -n -p | grep 5060 udp0 0 0.0.0.0:50600.0.0.0:* 3047/asterisk [r...@asterisk asterisk]# /usr/sbin/tcpdump port 5060 tcpdump: verbose output suppressed, use -v or -vv for full

Re: [asterisk-users] Siemens Gigaset Phones get mute function.

2009/4/8 Olivier oza-4...@myamail.com 2009/4/8 Alan Lord (News) alansli...@gmail.com * DHCP Option 114 implemented.* * DHCP Option 120 implemented.* http://lists.digium.com/mailman/listinfo/asterisk-users What does it imply ? Provisionning from DHCP server ? 114 is for passing

Re: [asterisk-users] New CentOS 5 repo: dahdi, asterisk, freepbx RPMs

2009/3/28 Jason Parker jpar...@digium.com D Tucny wrote: 2009/3/26 John Morris aster...@zultron.com mailto:aster...@zultron.com Hi, Axel. Axel Thimm wrote: How about merging in your changes/improvements/new packages with ATrpms (and automatically later

Re: [asterisk-users] Early Media

I can't say it's always been like this, as I don't recall, but, Background in 1.0 behaved like this, answering the channel if it wasn't already answered and playing the sound file/s until they finished or an exten was dialed... in 1.0 the 'skip' option would cause playback to be skipped if the

Re: [asterisk-users] New CentOS 5 repo: dahdi, asterisk, freepbx RPMs

2009/3/26 John Morris aster...@zultron.com Hi, Axel. Axel Thimm wrote: How about merging in your changes/improvements/new packages with ATrpms (and automatically later into rpmrepo.org)? That way we won't have further fragmentation and a larger user base to test bits (which will be

Re: [asterisk-users] Asterisk multi-cpu

From your figures, it would appear that if you double the load you will be potentially starting to see problems... FYI, not sure if it's of use to you... but... The digium tc400b is a transcoder card that can offload upto 120 channels of transcoding for g729 - ulaw... It's available as PCI only,

Re: [asterisk-users] incoming call problem from pri

2009/3/19 Oguzhan Kayhan oguzh...@bilkent.edu.tr Hi, i managed to connect to Ericsson MD110 with PRI at last. And made a successful call thru asterisk to ericsson. But when i try to call from ericsson to asterisk i got an error on asterisk side. And i couldnt figure out why. Here's my

Re: [asterisk-users] Asterisk and PBX internal numbers

2009/3/19 Oguzhan Kayhan oguzh...@bilkent.edu.tr Hi, i know i am asking a lot of questions lately in this forum..sorry about that first of all. :) Ok, here is the deal.. I am trying to make a hybrid system with an ericsson MD110 and asterisk. Internally we have 4 digit phone extensions on

Re: [asterisk-users] Asterisk and PBX internal numbers

2009/3/20 D Tucny d...@tucny.com 2009/3/19 Oguzhan Kayhan oguzh...@bilkent.edu.tr Hi, i know i am asking a lot of questions lately in this forum..sorry about that first of all. :) Ok, here is the deal.. I am trying to make a hybrid system with an ericsson MD110 and asterisk. Internally

Re: [asterisk-users] Asterisk and G.726 Codec

2009/3/18 Kevin P. Fleming kpflem...@digium.com Le'an Liu wrote: My questions: 1. G.726 16/24/32/40 supported in asterisk-1.6.0.5? No. Only G726-32 is supported in all Asterisk versions. Perhaps the confusion in the voip-info page mentioned is due to the other G726 rates being supported

Re: [asterisk-users] Good phone near $125

2009/3/17 Marc Charbonneau timebandit...@gmail.com I was looking at the aastra 9133i, however I was informed that this phone is no longer supported. What are good phones around the $100 - $125 price point? (Need POE) I like the Polycom IP-330. 2 lines, nice speakerphone, dual ethernet,

Re: [asterisk-users] Is it possible to get full callin number from E1?

2009/3/12 ssmax ss...@126.com Hi all i have just set up a asterisk in china, using DE410P and one E1 line and get a phone number like: +86 020 87654321 from my sp when somebody dial +86 020 87654321 , the asterisk will get the call in number by ${EXTEN} variable, but it can only

Re: [asterisk-users] Outgoing call drops

2009/3/12 Danny Nicholas da...@debsinc.com Greetings Listers, I’m running 1.4.21.2 on SUSE 11.0 with and zaptel 1.4.12.1 on a TDM400P. Most of my calls work great, but occasionally we try to connect to a customer or vendor external conference call and the call

Re: [asterisk-users] CISCO 2950 - 4 connections - Cap of 512 Kbps - How to bond ?

A 2950 can be configured to limit the speed per port... I guess the ISP here is operating this way because they are out of the way and have limited bandwidth themselves, so, they are trying to split up the bandwidth provided into smaller, more manageable chunks to avoid overloading things at

Re: [asterisk-users] Caller ID replacement

The hotdesking section of the asterisk book may also be of interest... d 2009/2/13 David Ruggles da...@safedatausa.com Some googling lead me to this: http://hans.fugal.net/blog/tag/astdb Which looks like it has an answer. Thanks all! Thanks, David Ruggles CCNA MCSE (NT) CNA A+

Re: [asterisk-users] CISCO 2950 - 4 connections - Cap of 512 Kbps - How to bond ?

2009/2/13 Vikas topg...@gmail.com My questions are: 1. The black wire coming into the Mc Manstel box is that a fibre optic cable ? 2. What is the Mc Manstel box doing ? 3. What CISCO router do I need to buy to do bandwidth aggregation at my end ? 1) Yes 2) It's stopping you from poking

Re: [asterisk-users] meetme application

2009/2/9 邱磊 qiulei...@163.com i reload the app_meetme.so in CLI: - Reloading module 'app_meetme.so' (MeetMe conference bridge) == Parsing 'etc/asterisk/meetme.conf': Found All the sip message show that there is no fault, and i dont know why the meetme application can't work. i

Re: [asterisk-users] escaping regular expression

2009/2/4 Klaus Darilion klaus.mailingli...@pernau.at Hi! I am going nuts using REGEXP. I just want to verify if a variable contains a valid +E164 phone number. These, the the pattern is ^\+[0-9]+ First I tried: Set(pattern=^\+[0-9]+); if (${REGEX(${pattern} ${${var}})}) but that

Re: [asterisk-users] escaping regular expression

2009/2/4 Klaus Darilion klaus.mailingli...@pernau.at D Tucny schrieb: 2009/2/4 Klaus Darilion klaus.mailingli...@pernau.at mailto:klaus.mailingli...@pernau.at Hi! I am going nuts using REGEXP. I just want to verify if a variable contains a valid +E164 phone number

Re: [asterisk-users] Contact lookup

2009/2/4 Gordon Henderson gordon+aster...@drogon.netgordon%2baster...@drogon.net On Tue, 3 Feb 2009, Geoff Lane wrote: Hi All, Asterisk 1.4.12 on CentOS 5 I'd like to be able to look up each incoming CLI to retrieve an associated name, if available, and then pass that to the

Re: [asterisk-users] Out of Office: Out of Office: Out of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Re : som e kind of t i

Please get this out of office reply disabled, or at the very least, fixed... It currently seems to have generated a loop, sending out of office replies to the out of office replies it's already sent to the asterisk-users mailing list... It's bad that it sent a reply to the list anyway, but this

Re: [asterisk-users] Quiet 24 port POE gig switch

2009/2/2 Steve Underwood ste...@coppice.org Bernd Felsche wrote: Ian Cowley i...@moffat.co.uk wrote: Beware PoE switches that can't handle Class 3 (15W) on all ports. Most have fans because 24 (or 48) x 15W is hot! That's the power supplied .. which'd be at the far end of the

Re: [asterisk-users] Trunk with Polocom Video Conferencing Unit

2009/2/2 Daniel Harper dan...@harper.net.nz I was wondering if anyone can help me with a problem we have at one of our sites. We have setup a Asterisk Trunk to a Avaya PBX, ie ... Avaya - Asterisk (1.2.30) - External ISDN Network BUT They also have a Polycom VSX 7000 that with some

[asterisk-users] Improving asterisk documentation - sources and what the community can do

Hi All, There have been a number of comments recently about a shortage of documentation on Asterisk, so I wanted to cover briefly the documentation options available and suggest what they are useful for and how they can be improved... Documentation sources: http://www.asteriskdocs.org/ - Not

Re: [asterisk-users] Ntework Card

This is still very off topic... Someone's already suggested you look to somewhere for centos help... if you had, you'd have found this... http://www.centos.org/modules/newbb/viewtopic.php?forum=39topic_id=10098 Which is an RPM containing an update driver, so you wouldn't have to mess about

Re: [asterisk-users] Auto Detect

on CentOS you can use lshal to return which driver is in use... e.g. lshal -s |grep pci_ |xargs -n1 lshal -l -u |grep -E udi|info.product|info.linux.driver get's a list of items, filters on pci, gets the long output for those pci devices and outputs lines containing udi, info.product and

Re: [asterisk-users] [SOLVED] Nortel IP phone i2002 - DHCP server unreachable

Perhaps this would help... http://blog.michaelfmcnamara.com/2007/10/dhcp-options-voip/ Gives details on the dhcp option string needed for the phones and explains that without it the phone will not accept a DHCP response... d 2009/1/24 Joseph syscon...@gmail.com Thanks for the input. Yes, I

Re: [asterisk-users] registration problem using asterisk 1.6

2009/1/22 Laurent Bonny laurent.bo...@gmail.com Hello, I am trying to connect an asterisk 1.6 to a trunking plate forme. With asterisk 1.4.x I added to sip.conf a line asking for registration in the form of: register =

Re: [asterisk-users] OT - Is Netgear ProSafe FS108P with PoE silent ?

2009/1/23 Olivier oza-4...@myamail.com Hi, I need to locate a Asterisk server and a PoE-enabled switch on someone's desk. I've seen this Netgear ProSafe FS108P with 4 10/100 and 4 10/100 w/PoE ports silent enough ? Any recommendation ? Specs on netgear.com say it's silent... 'Acoustic

Re: [asterisk-users] No Ring on Analog Phone using Rhino ChannelBank in China

2009/1/23 Lee, John (Sydney) john@compuware.com There's nothing special about analogue phones in China, they are fully interchangable with analogue phones elsewhere... Perhaps you have a configuration problem, or, hardware problem on the Rhino Channel Bank, perhaps the ports are

Re: [asterisk-users] Packet8 hacked

2009/1/23 Dean Collins d...@cognation.net Looks like www.packet8.com has been hacked L The phone service is offline as well. Anyone else on this list using packet8? Not using packet8, but, the website looks normal to me... What are you seeing? d

Re: [asterisk-users] No Ring on Analog Phone using Rhino Channel Bank in China

2009/1/21 Lee, John (Sydney) john@compuware.com I am testing analog phone and fax machine plugged into Rhino Channel Bank which is connected to TE412P card. This site is in China. I am running RHEL 5, Asterisk 1.4.21.2, Zaptel 1.4.11 and libpri 1.4.4 I ran into a problem which is

Re: [asterisk-users] Need help registering Cisco 7960 Phones on Asterisk

something wrong which I don't know what it is. Zeeshan On Tue, Jan 20, 2009 at 1:58 AM, D Tucny d...@tucny.com wrote: That's not my experience... e.g. SIP Phone show register LINE REGISTRATION TABLE Proxy Registration: ENABLED, state: REGISTERED line APR state timer expires

Re: [asterisk-users] Fring and Asterisk

2009/1/20 Olivier oza-4...@myamail.com 2009/1/20 Gordon Henderson gordon+aster...@drogon.netgordon%2baster...@drogon.net On Tue, 20 Jan 2009, Olivier wrote: GTalk seems to fill the bill of requirements, though, I don't think it's available on Nokia mobile phones .. My Nokia

Re: [asterisk-users] Fring and Asterisk

2009/1/20 Olivier oza-4...@myamail.com One thing to note about fring, the device establishes a connection using fring's proprietary protocols to fring servers, fring then establishes SIP connections from those servers... So, even if connected to the office Wifi connection, you could

Re: [asterisk-users] Problem with TDM808

If your provider provides any signalling to indicate answer, such as a polarity reversal, this could be detected easily... ; Use a polarity reversal to mark when a outgoing call is answered by the ; remote party. ; ;answeronpolarityswitch=yes This isn't very common though... alternatively, there

Re: [asterisk-users] Fring and Asterisk

2009/1/20 Olivier oza-4...@myamail.com Hi, Is anyone using Fring as a SIP client to an Asterisk server ? Yes, testing it... A prospective customer of mine is asking to integrate its iphones with an Asterisk server and after googling, I still have some unanswered questions : 1. Which

Re: [asterisk-users] Need help registering Cisco 7960 Phones on Asterisk

2009/1/20 Zeeshan Zakaria zisha...@gmail.com Hi everyone, I googled this followed the instructions, but it hasn't work for me yet. I have universal setting in SIPDefault.cnf and phone specific settings in SIPXX.cnf. But it doesn't get registered. I need to register it on two

Re: [asterisk-users] Fring and Asterisk

2009/1/20 John Todd jt...@digium.com On Jan 19, 2009, at 6:29 PM, D Tucny wrote: 2009/1/20 Olivier oza-4...@myamail.com Hi, Is anyone using Fring as a SIP client to an Asterisk server ? Yes, testing it... A prospective customer of mine is asking to integrate its iphones

Re: [asterisk-users] followme order field

2009/1/19 Thomas Stein thomas.st...@knowledgetools.de Hello. Does someone know what order field means in followme.conf? The Doku says: number= number to call[2nd #[3rd #]] [, timeout value in seconds [, order in follow-me] ] So an example would be: number= 123124125,10,? It would be

Re: [asterisk-users] Need help registering Cisco 7960 Phones on Asterisk

register to the second box, only to the first one. Why? god knows... __Yehavi: 2009/1/20 D Tucny d...@tucny.com 2009/1/20 Zeeshan Zakaria zisha...@gmail.com Hi everyone, I googled this followed the instructions, but it hasn't work for me yet. I have universal

Re: [asterisk-users] How to hangup a call manually...

2009/1/17 Carlos Chavez cur...@telecomabmex.com I have this call: SIP/protel-525512047 default 90445528885371 1 Ringing AppDial (Outgoing Line) 90445528885371 264:24:2 (None) I cannot use the soft hangup commando from the CLI because I do not