Re: [asterisk-users] Confbridge

John, What you see it's how it should be. The wait for admin means that all users join the conference room but the conference is not started and they all should hear MOH. When the admin will join then the conference will start and all will hear the admin (or all others if they are not muted)

Re: [asterisk-users] SIP Codec negotiation

The unwritten rule of SDP is that if possible you use the first codec of a type listed, but you don’t have to. If the sender says he can do something, he had better be prepared to handle media of that type no matter in what order it was listed. So when you send OK with ulaw as first priority and

Re: [asterisk-users] how to make International calls from asterisk PBX

International calls are exactly as local phones using the same lines/trunks. First check your outbound route to verify that your dial plan match your dialing international pattern. Sincerely, Sam Basan From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun

Re: [asterisk-users] asterisk-users Digest, Vol 160, Issue 5

Get Outlook for Android From: asterisk-users-boun...@lists.digium.com on behalf of asterisk-users-requ...@lists.digium.com Sent: Monday, November 6, 2017 6:00:01 PM To:

[asterisk-users] Tranfer the called number in 3 way call

Hey there, I have a question i want a dialplan to send the called number of the client instead of my callerID when making a 3way call or when transfering to an extension from a bridge to another pbx. The problem i add a variable and using thw two underscores but i still see the my calledID ,

Re: [asterisk-users] app_queue ringall - 2 agents answer same time problem

Your second call is not without sound, there is simply no call at all. As the first answer the call his channel and the external call channel connected. The second device simply off hook but his channel have no external channel to connect. It's looks like a simple telephony glare. Sam בתאריך 30

Re: [asterisk-users] How to custom the message on call busy or no answer in asterisk

Sincerely, Sam Basan From: Thyda ENG [mailto:ength...@gmail.com] Sent: Saturday, November 21, 2015 8:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Subject: Re: [asterisk-users] How to custom the message on call busy or no answer

Re: [asterisk-users] No sound with internal calls depending on which phones

Snom default configuration is SRTP enabled. You should disable the SRTP from the phone web GUI configuration Sincerely, Sam Basan From: Mitul Limbani [mailto:mi...@enterux.in] Sent: Thursday, November 12, 2015 5:25 PM To: Asterisk Users Mailing List - Non-Commercial

[asterisk-users] Update new IP address (move temporarily) for INVITE

Hello, How can I update asterisk to send back move temporarily with updated IP address to incoming INVITE. i.e, Incoming call from ITSP to server 1 with x DID and there is a need to update the ITSP that the specified x DID number is allocated in server 2. Thanks, Sam Basan

[asterisk-users] Reverse one way paging or silent monitoring

Hello, Paging a phone set the phone to auto answer and open the speaker. How can I set the phone to turn on just his microphone and the camera, if available, for security remote controlling? Thanks, Sam

Re: [asterisk-users] Help with voicemail

Check your phone codecs. It set to g729 while you don't have this codec in your asterisk nor files in this codec. בתאריך 17 באוק' 2015 18:34,‏ "Luca Bertoncello" כתב: > Hi list! > > My problem: I have three extensions in my Asterisk 1.8.30.0 and they have a > voicemail. >

[asterisk-users] Asterisk AMI events filtering

Hi folks, I have one server with multiple companies (multi-tenant). >From AMI I get all events of all extensions so any one that connect can see other extensions, from different company (context). How can I limit specific user to get just specific context?

Re: [asterisk-users] SIP domain different than provider's

On 08/21/2015 12:52 AM, Sam wrote: Hello, I have what I would think would be a common situation: I run asterisk at home simply as a land line. I started a new job working remotely and they gave me a SIP account with user name, domain, and proxy. I've never had to deal with sip domains before

[asterisk-users] SIP domain different than provider's

://tinyurl.com/ouy2ajr Regards, Sam -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk

Re: [asterisk-users] chan_sip.c: Retransmission timeout reached on transmission

for asterisk NAT configuration parameters. נשלח מטלפון נייד בתאריך 14 באוג' 2015 22:12,‏ Daniel - Asterisk earohua...@gmail.com כתב: Hello Sam, Do you have any recommendation to overcome these NAT issues? On 8/14/15, Sam Basan sba...@bluebe.net wrote: Hi, It's looks like you are having NAT

Re: [asterisk-users] chan_sip.c: Retransmission timeout reached on transmission

Hi, It's looks like you are having NAT problem. Packets from the provider fail reaching your box. נשלח מטלפון נייד בתאריך 14 באוג' 2015 15:56,‏ Daniel - Asterisk earohua...@gmail.com כתב: Hello friends: I am facing cutoffs randomly when negotiating calls. The PBX dials the destination, the

Re: [asterisk-users] Looking for PRI Card with automatic fail over

Hi, What you need is PRI TAP (also called SWITCH box) Check this one http://www.voicetronix.com/openpri.htm נשלח מטלפון נייד בתאריך 3 באוג' 2015 17:53,‏ Eric Klein eric.kl...@greenfieldtech.net כתב: Hi all, Strange request, I have a customer where we are putting an Asterisk PBX in front of

Re: [asterisk-users] Outbound DID: in sip.conf or dialplan or db?

this is when multiple phones are grouped (multiple groups) to share a DID number for their callerid. *** Sam Lutgring Director of Informational Technology Services Calhoun Intermediate school district lutgr...@calhounisd.orgmailto:lutgr

Re: [asterisk-users] How to add prefix in Extensions.Conf

The short answer is yes you can. Now the longer answer is give us more detail if you want to know how. Are they asking you to add the 92 when you dial 5672531308, or is this question really about the callerid number? *** Sam Lutgring Director

Re: [asterisk-users] Phone Inventory

Thank you all You are a life saver Sam Dale Noll wrote: On 02/23/2012 08:49 AM, Danny Nicholas wrote: Here is a snippet that somebody smarter than I am can improve upon for a in `asterisk -rx sip show peers|cut -f1 -d/` ;do asterisk -rx sip show peer $a;done|grep Useragent

[asterisk-users] Replicating SIP registration Info between active to standby

allow double registration, we would like to make it simple by provision only one registration server at a time. How can I copy sip registration information from Active Server to Standby Server Sam -- _ -- Bandwidth and Colocation

Re: [asterisk-users] Replicating SIP registration Info between active to standby

Hi Takehiro Are you suggesting sharing the AstDB ? Sam Takehiro Matsushima wrote: Hi. How about place backend DB on shared disk, or make replication between them? 2012/02/24 13:58 Muro, Sam resea...@businesstz.com: I have a scenario whereby two servers are acting in active-standby mode

[asterisk-users] Phone Inventory

Hi there I have just took a support of a customer with hundreds of IP phones, mostly Polycom with mixed models. Is there a way to query asterisk or any other command to retrieve the inventory of all connected phones. i.e. Phone Type and Phone Model, say Polycom SPIP331 or so Thanks Sam

[asterisk-users] Asterisk Security: Allow only one phone per sip registration

Hi there Consider this. You have three SIP extension 200, 201 and 202 and you have configured your phones, say Polycom 331 to those accounts. 200 being one very sensitive individual. Lets say, an insider, get a new phone or perhaps an xlite and configure it with the same extension, 200. Asterisk

Re: [asterisk-users] Asterisk Security: Allow only one phone per sip registration

Terry Wilson wrote: - Original Message - From: Sam Muro resea...@businesstz.com To: asterisk-users@lists.digium.com Sent: Friday, October 14, 2011 2:02:01 AM Subject: [asterisk-users] Asterisk Security: Allow only one phone per sip registration Hi there Consider this. You have

Re: [asterisk-users] Asterisk Security: Allow only one phone per sip registration

Terry Wilson wrote: Is there a way one can bind sip account to specific mac-address (assume on the same subnet). In this way, even if you know the username/secret, you will still have to use the same physical phone, unless you play with mac-address. No. And mac addresses are easily

Re: [asterisk-users] Asterisk Security: Allow only one phone per sip registration

Thanks Terry! Let me think of all possibilities and shall holla. Can you be one? Terry Wilson wrote: Thanks. Let me see how best i can complicate them per phone. Ooops, 1000 sip phones If it were me, I would look into Asterisk Realtime for handling the SIP phones. I would then write a

Re: [asterisk-users] Asterisk Security: Allow only one phone per sip registration

Thanks A.J I know and I can assure you no one will get that physical access to the system. A J Stiles wrote: On Friday 14 October 2011, Muro, Sam wrote: Hi there Consider this. You have three SIP extension 200, 201 and 202 and you have configured your phones, say Polycom 331 to those

Re: [asterisk-users] Make asterisk cluster appear and operate as a single server?

Hey, Why do you think using OpenSIPs is not going to work for you ? You can always add SIP trunks on openSips and based upon which trunks getting the call you can LB or/and FO to as many asterisk servers as you want ! Regards, -Sammy On Sun, Oct 2, 2011 at 7:12 AM, Michelle Dupuis mdup...@ocg.ca

Re: [asterisk-users] Add PinCode on my dialplan

, Malvin On 9/21/2011 1:01 PM, Sam Govind wrote: See core show application autheTAB If passwords are already the same as those of voicemail.conf go for application VMAuthenticate() - DIA generates a dial-tone which I don't think is suitable for dialling out from users(insiders) -= Info about

Re: [asterisk-users] invite authentication error !?

What Sip declaration are you using for the remote sip proxy in sip.conf? On Fri, Sep 30, 2011 at 12:30 PM, Alex Balashov abalas...@evaristesys.comwrote: This is just a speculative shot in the dark, but remember that the domain of the From URI is important, and that the authentication realm

Re: [asterisk-users] invite authentication error !?

Whatever that remote party is, you are most definitely using a username/secret declaration for that. So the sip attributes set for that proxy define the behaviour for this. On Fri, Sep 30, 2011 at 12:55 PM, cnasterisk cnaster...@163.com wrote: ** hi, Sam thanks for your kindly reply

Re: [asterisk-users] invite authentication error !?

...@163.com ** asterisk can register successfully on the remote party. so i think username password must be ok 2011-09-30 -- cnasterisk -- *发件人：* Sam Govind *发送时间：* 2011-09-30 16:05:21 *收件人：* Asterisk Users Mailing List - Non

Re: [asterisk-users] PSTN connectivity

, ) in new stack == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/199-003a' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/199-003a' On Wed, Sep 28, 2011 at 2:50 PM, Sam Govind govoi...@gmail.com wrote

Re: [asterisk-users] PSTN connectivity

? On Thu, Sep 29, 2011 at 11:35 AM, Sam Govind govoi...@gmail.com wrote: The Call at this point is not even looking for FXO/Dahdi/Zap.. See the CLI. there is some misconfiguration in FreePBX and your dialled number is not hitting any dial-able rule. See your FreePBX guide. On Thu, Sep 29, 2011

Re: [asterisk-users] PSTN connectivity

Hey Warren I thought that these are the complete CLI logs for one call. It started like == Using SIP RTP CoS mark 5 and from-internal priority-1 ..So that seemed legit to me. Yeah I too suspect that dialing rules are not being matched and thats why Gotoif's are failing. On Thu, Sep 29, 2011 at

Re: [asterisk-users] record calls of specific agnets

I guess that was this variable like SPYGROUP which needs to be set for specific extensions and then ask Chanspy to spy on that group. !! On Thu, Sep 29, 2011 at 9:37 PM, A J Stiles asterisk_l...@earthshod.co.ukwrote: On Thursday 29 September 2011, Lyle McKarns wrote: Hello Asterisk List!

Re: [asterisk-users] Features not working

Hey, Whats the output of command features show ? on CLI ? On Fri, Sep 30, 2011 at 1:51 AM, Mike Diehl mdi...@diehlnet.com wrote: Hi all. I could have sworn this working at one time... But it doesn't look like any of the functions provided by features.so is working for me. (one-touch

Re: [asterisk-users] Call does not pass through

as well as SIP traces combined for each failed successful call. Regards. -Sammy On Wed, Sep 28, 2011 at 10:59 AM, Malvin Rito mr...@mail.altcladding.com.ph wrote: Thanks Sam. Please see below CLI log: *[root@localhost ~]# asterisk -r Asterisk 1.6.2.7, Copyright (C) 1999 - 2010 Digium, Inc

Re: [asterisk-users] PSTN connectivity

Some CLI logs will get you better help on the issue ! also paste the FXO configurations and how you configured it ! On Wed, Sep 28, 2011 at 2:11 PM, michael k mich...@inapp.com wrote: Hi All, I am trying to connect my asterisk box with freepbx to PSTN. I have purchased x100p FXO

Re: [asterisk-users] number of calls simultaneous from AMI

If you can post any relevant code sections and CLI output for this then it'll be lot better to determine whats causing this. I never got any problem initiating as many call as u can say from AMI ! On Tue, Sep 27, 2011 at 5:36 PM, Jerry Geis ge...@pagestation.com wrote: I am starting up 4 calls

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

Correct me if I'm wrong or don't know anything other than AMI Originate Event or a call file to kick start a call from asterisk ! So making a new or modifying asterisk call-file cron job/poller seems like a nice idea but why put on extra load on Asterisk. (See pbx_spool.c if still want to modify).

Re: [asterisk-users] Set (MONITOR_FILENAME=.................) for queuing recording calls

:P I'd this very similar situation/ project Carl - and guess what. The filename is created before the call actually hits QUEUE application so these Queue variables are not populated by then so filename won't contain the Agent Number. UNLESS you move the file after queue to a new filename

Re: [asterisk-users] Receiving musinc on hold instead of ring

Very strange indeed! post the dialplan lines as well. Seems like a very normal Dial command execution. Also complete SIP packets for this particular behaviour can show some insider. Which version of Asterisk you are using? On Wed, Sep 28, 2011 at 6:44 AM, Alejandro Recarey

Re: [asterisk-users] Call does not pass through

I see a couple of conflicting extensions as well as something I assume copy-paste malfunction. Please paste the CLI logs of the call. On Wed, Sep 28, 2011 at 8:26 AM, Malvin Rito mr...@mail.altcladding.com.phwrote: Thanks All. Here is my config: *On my Firewall NAT:* *I allowed the

Re: [asterisk-users] Asynchronous AGI Problems (Asterisk 1.8.7.0), ubuntu-server

Oh! I was informed that Async:AGI is an AGI that is called in from AMI. Do tell more about it. On Sun, Sep 25, 2011 at 5:26 PM, Mehmet Avcioglu meh...@activecom.netwrote: Actually it doesn't say AGI(async:script) it says AGI(async:agi) and than continues further to setting up an AMI user so

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

Hmmm..interesting..I haven't came across anything like this so far..How about making a new table for the insertion of a new call data..and trigger some script to activate AMI/Call file according to new call data. http://dev.mysql.com/doc/refman/5.0/en/faqs-triggers.html#qandaitem-B-5-1-10 On

Re: [asterisk-users] AGI Problem

How much time your AGI is taking? Check if it is completing its task and not killed by asterisk. I guess we've 6~7 seconds before asterisk kills all call channel and related tasks. On Sat, Sep 24, 2011 at 3:21 PM, Mehmet Avcioglu meh...@activecom.netwrote: On Sep 23, 2011, at 8:01 PM, Mehmet

Re: [asterisk-users] Set (MONITOR_FILENAME=.................) for queuing recording calls

yes you are right. You should put it before calling in the queue. Set monitor filename as you want. Also you can set directory in the filename here too. So when the queu is triggered and mixmonitor starts it'll use that filename and record inot the file. On Sat, Sep 24, 2011 at 2:08 AM, bilal

Re: [asterisk-users] AGI Problem

and script completed messages within the same second, so less than a second. -- Mehmet On Sep 24, 2011, at 3:34 PM, Sam Govind wrote: How much time your AGI is taking? Check if it is completing its task and not killed by asterisk. I guess we've 6~7 seconds before asterisk kills all call

Re: [asterisk-users] AGI Problem

to replicate it under test conditions. Thanks -- Mehmet On Sep 24, 2011, at 4:00 PM, Sam Govind wrote: Thats wicked !! hmmm stop your asterisk (if u can afford) and run it like asterisk -cvg and then make a call.. see whats your AGI doing in there !! On Sat, Sep 24, 2011 at 5:43

Re: [asterisk-users] Change of default IVR prompt for meetme conference bridge.

UmmmWhen I was a child I replaced the prompts to do that, Now I'd suggest you to try creating a new directory in /sounds folder like /en i.e /meetme and put in corresponding prompts there. Then just before going into the meetme application change the Language for the current call in dial plan

Re: [asterisk-users] Add PinCode on my dialplan

DISA and DB based Auth could be an overkill. Kyle showed the very simplistic dial plan if Dial-out pin is common for the whole system. See application *Authenticate(password[,options[,maxdigits[,prompt]]] *and if Voicemail PIN are required to be used use application

Re: [asterisk-users] RESEND: Mixmonitor command parameter problem on Asterisk 1.8.4

+1 Dale Alternatively I'd troubles using the MixMonitor() command execution, so what I did is used System(my commands here) just after the StopMixMonitor(). Using StopMixMonitor() is always recommended to guarantee save the recorded file and using any commands via System() is easy. On Wed, Sep

Re: [asterisk-users] oddity with CISCO CCM and Asterisk

Hi Danny, If you explain some more about this phantom process !! I've never seen asterisks doing this before. This initiation of a new call is always dependent upon arrival of an INVITE. I doubt its CCM that is doing some re-INVITES or sort of keepalive for this call and thus a phantom call is

Re: [asterisk-users] Message recorder

stupified. Thanks, - Sammy On Sun, Sep 18, 2011 at 2:13 AM, Steve Edwards asterisk@sedwards.comwrote: On Sat, 17 Sep 2011, Sam Govind wrote: Requirement: Two copies of the recorded message are required. [Recorder-A] One will contain only the last message recorded(final) [Recorder

Re: [asterisk-users] DTMF problem

Hey there, I don't think that its DTMF mode issue ! OP say pressing 9 asterisk ignores while pressing 6 is OK. Using expensive PBX solutions should be never be the first priority. So I'd a similar experience in some asterisk version when I used to enter 2 asterisk always took 3-4 seconds to do

Re: [asterisk-users] single registration per user

Hmmm..this could be a complex solution - Use OpenSIPS to handle registration. On each new register attempt see if a user AOR or other records exists already - if yes deny registration. On Mon, Sep 19, 2011 at 1:23 AM, Catalin S. jonsonpla...@gmail.com wrote: Hello Eric, Is about outgoing

Re: [asterisk-users] Sip re-register / delay problem.

Hey, I don't think there could be any solution to this. Even a SIP Proxy...I don't think so you'll get enough control there to get re-registers from lagging users only. SIP Timers adjustment on each user level is something atleast I haven't cam across so far. SIP Timers are global params for all.

[asterisk-users] Message recorder

Hello List, Good day. I'm trying to create a message recorder. User will be prompted with some soundfile (instructions on how to use recorder * to restart and # to submit). *Requirement:* Two copies of the recorded message are required. [Recorder-A] One will contain only the last message

Re: [asterisk-users] testing simultaneous calls

A little look at the dialplan which rings your extension, or get dtmf, and plays DTMF will help better understand. btw you can set the context/extension/priority in a call file to skip some priorities of a particular extension set. On Fri, Sep 16, 2011 at 12:18 AM, ERIC HERRON e...@lanline.com

Re: [asterisk-users] Monitoring second leg being dialed?

The image you provided didn't open so I'm not sure about the design. If you can send some SIP flow diagram and Asterisk CLI logs maybe it'll help understand the problem. On Fri, Sep 16, 2011 at 1:28 AM, Gilles codecompl...@free.fr wrote: Hello My ISP provides an FXS port to plug a

Re: [asterisk-users] Inter-astersik dialling encounteres no audio

This obviously is pointing to NAT issue. see if you've configured both asterisk servers with externip= PUBLICIPOFAsterisks. Studying SIP traces on each console and specially looking at the SDPs in INVITE will help you find out exact problem. I expect that one of the asterisk box is sending the

Re: [asterisk-users] Monitoring second leg being dialed?

asking an army Have you guys ever worked with guns!! :P Please try producing SIP traces so your problem could be identified. Which asterisk and DAHDI version you are using btw? On Fri, Sep 16, 2011 at 2:51 PM, Gilles codecompl...@free.fr wrote: On Fri, 16 Sep 2011 11:13:16 +0500, Sam Govind

[asterisk-users] Temporarily disable DTMF on a call

Hello List, I need help on disabling DTMF from a caller for a specific set of dialplan commands and enable DTMF for some other dialplan part. This is not a SIP peer - just live incoming call on SIP. Please help. Thanks -Sammy --

Re: [asterisk-users] Temporarily disable DTMF on a call

,prompt, …) Exten = s,n,process.. Exten =s(end),n,hangup ** ** *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Sam Govind *Sent:* Friday, September 16, 2011 1:03 PM *To:* asterisk-users@lists.digium.com *Subject

Re: [asterisk-users] secret=pw in sip.conf affecting inter-asterisk sip call

Hey, The callee server is complaining too loud Call from '2765' to extension '* 1166:password*' rejected because *extension not found*. Try changing the Dial string as DIAL(SIP/asterisk-callee/${EXTEN}) or w/e extension you require in place of ${EXTEN} Let me know what changes. Also this is a

Re: [asterisk-users] Mysql dialplan statement not executed

I expect that your same query when executed directly on MySQL console executes successfully ! Normally errors in DB queries are printed on CLI but apparently there is nothing wrong. On Wed, Sep 14, 2011 at 5:51 PM, Jonas Kellens jonas.kell...@telenet.bewrote: ** Hello, I do the following in

Re: [asterisk-users] broadcast

it .call file start playback at it's own channels but I am not able to hear anything into conf. As i know localdial is not joining into the conf. but how I will do it so that I will be able to hear any played file into conference ? On Mon, Sep 12, 2011 at 3:36 PM, Sam Govind govoi...@gmail.com

Re: [asterisk-users] broadcast

to hear anything into conf. As i know localdial is not joining into the conf. but how I will do it so that I will be able to hear any played file into conference ? On Mon, Sep 12, 2011 at 3:36 PM, Sam Govind govoi...@gmail.com wrote: Good to know, I think it'll be a feedback score or a poll

Re: [asterisk-users] Asterisk Manager Interface (AMI)

Hey, I think I remember the same post before. previously I heard someone telling to use vicidial or some other thing like that.But I don't think that those are totally AMI based call-generators. What I'd recently done is make a php page which connects to Asterisk's AMI port. I send page request

Re: [asterisk-users] sox: Failed reading obd-demo.mp3: Do not understand format type: mp3

1- *-bash: obd-demo.ulaw: No such file or directory* // try use absolute file path i.e /usr/src/mymp3.mp3 . I guess that's why its saying no such file or directory. 2- http://lists.digium.com/pipermail/asterisk-users/2006-March/144689.html Go through this thread. 3- When everything fails from

Re: [asterisk-users] Asterisk is keep on sending Register request

Hey krishnan, Everything happens for a reason. The most intuitive cause of this issue seems to be network change. Can you confirm that no change in networking happened! because your server is sending register requests but not getting responses. Meanwhile the same server replying to scenarios2 can

Re: [asterisk-users] Call drop in 10 seconds without disconnecting a-party call

Thats goood ! :) thanks for updating. On Fri, Sep 9, 2011 at 2:16 PM, Ishfaq Malik i...@pack-net.co.uk wrote: Hi Can you please provide an excerpt of your logs when this happens? Regards Ish On Fri, 2011-09-09 at 09:05 +, Vinod Dharashive wrote: Hi sam, Have solved the problem

Re: [asterisk-users] (no subject)

See absolute timeout. I think yours' a complex thing to achieve I guess absolute timeout may be the thing that can help. In older versions absoluteTimeoute(n) could take you to exten T when time n elapsed. now I guess funtion Timeout() is used as replacement. here's an excerpt from somewhere: ;

Re: [asterisk-users] Queue agent login notification

and queue_log doesn't exist either (as a file or as a db table). We're using AsteriskNOW, so maybe these files/tables were not created. How should we add them? Thanks. On Wed, Sep 7, 2011 at 8:54 AM, Sam Govind govoi...@gmail.com wrote: See this link: http://www.voip-info.org/wiki/view/Asterisk

Re: [asterisk-users] Beginner Question: Remote access

There could be as easy solutions as using teamviewer or use tools like Hamachi used in combination with dyn-dns etc. IP-tunneling I guess needs static public IPs for the sake of completing the route. On Wed, Sep 7, 2011 at 5:30 AM, A Dunor alsta...@gmail.com wrote: Thanks for the speedy pointer

Re: [asterisk-users] Queue agent login notification

See this link: http://www.voip-info.org/wiki/view/Asterisk+queue_log+on+MySQL You'll find similar pages where you can setup to store queue logs/events(as Alex mentioned) in MySQL DB and further do your triggers or functions on them. On Wed, Sep 7, 2011 at 10:46 AM, Michael

Re: [asterisk-users] how to add-edit-delete entery into asterisk conf files

AM, Sam Govind govoi...@gmail.com wrote: Though this might have been resolved/accomplished already but I've couple of questions for Virendra Bhati. 1- If you are doing this to make new accounts for new users, why couldn't you use Asterisk realtime(DB) based configurations of Voicemail/MoH/SIP

Re: [asterisk-users] how to add-edit-delete entery into asterisk conf files

and last things which hurt me. On Mon, Sep 5, 2011 at 12:48 PM, Sam Govind govoi...@gmail.com wrote: 1- Per my experience I've used DB with configuration files and I was amazed that Asterisk was taking a union of DB + conf file configurations and accepting both.So if you just make a simple

Re: [asterisk-users] how to add-edit-delete entery into asterisk conf files

Though this might have been resolved/accomplished already but I've couple of questions for Virendra Bhati. 1- If you are doing this to make new accounts for new users, why couldn't you use Asterisk realtime(DB) based configurations of Voicemail/MoH/SIP/dialplan etc wouldn't it be much easier than

Re: [asterisk-users] Dialing multiple endpoints and CallerID presentation

Alternative work around to this could be: 1- Make two different dialplan extensions. One to dial DAHDI numbers with setting for DAHDI and other extension for SIP dialing. Both extensions setting different CallerID presentation 2- Create a queue with Local extensions as static members

Re: [asterisk-users] Wanted a modified SIP message body

Use the *SIPAddHeader(Header:Content)* application in dialplan. I don't think Method specific SIP headers can be done via asterisk. On Fri, Aug 26, 2011 at 3:05 PM, Jaime Lozano jaimelozan...@gmail.comwrote: Hello everybody, I want Asterisk Server to send packets (SIP packets) to some 3Com

[asterisk-users] spam blacklist

Just a note, the asterisk mailing list server continually gets blacklisted over at http://www.uceprotect.net/rblcheck.php?ipr=216.207.245.17 for delivering mail to spamtraps. Perhaps something needs to be looked into... Regards, Sam

[asterisk-users] Passing parameter from executable program to asterisk dialplan

0117473789 NAME: Franklin John STATUS: Active Can someone advice on how i can catch this values from AGI or directly on dialplan. Thanks Sam -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk

Re: [asterisk-users] Passing parameter from executable program to asterisk dialplan

Kyle Kienapfel wrote: On Sun, Jul 25, 2010 at 8:18 AM, Muro, Sam resea...@businesstz.com wrote: I am having a problem understanding the way to retrieve some parameters to asterisk via AGI or what ever method that fits. I have an executable program that accept one parameter (CALLERID

Re: [asterisk-users] Passing parameter from executable program to asterisk dialplan

Steve Edwards wrote: On Sun, 25 Jul 2010, Muro, Sam wrote: I am having a problem understanding the way to retrieve some parameters to asterisk via AGI or what ever method that fits. I have an executable program that accept one parameter (CALLERID) and return customer status from

Re: [asterisk-users] Passing parameter from executable program to asterisk dialplan

Kyle Kienapfel wrote: On Sun, Jul 25, 2010 at 9:04 AM, Muro, Sam resea...@businesstz.com wrote: Kyle Kienapfel wrote: On Sun, Jul 25, 2010 at 8:18 AM, Muro, Sam resea...@businesstz.com wrote: I am having a problem understanding the way to retrieve some parameters to asterisk via AGI

Re: [asterisk-users] Passing parameter from executable program to asterisk dialplan

Steve Edwards wrote: On Sun, 25 Jul 2010, Muro, Sam wrote: I am having a problem understanding the way to retrieve some parameters to asterisk via AGI or what ever method that fits. I have an executable program that accept one parameter (CALLERID) and return customer status from the database

[asterisk-users] Corba interface

Hi there Has anyone know how to configure asterisk to be able to query Corba interface directly from the dialplan Sam -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us

[asterisk-users] Corba interface

Hi there Does anyone know how to configure asterisk to be able to query Corba interface directly from the dialplan Sam -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us

Re: [asterisk-users] Using asterisk as avaya definity recordingserver

(currently is done via Nice) on asterisk - This's the problem Sam -- Moises Silva Senior Software Engineer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.com

Re: [asterisk-users] Using asterisk as avaya definity recording server

with alternative solution? Muro, Sam wrote: Oh.. I didnt know that. Thanks Sam Muro, Sam escribió: What do you mean chief? What am looking at is ability for asterisk to receive a call and recording until it tier down without bridging it to the physical device Sam Would you like the advice in all

Re: [asterisk-users] asterisk-users Digest, Vol 68, Issue 33

What do you mean chief? What am looking at is ability for asterisk to receive a call and recording until it tier down without bridging it to the physical device Sam Would you like the advice in all caps? On 03/15/2010 01:20 AM, RESEARCH wrote: Hi there I remember to ask this question

Re: [asterisk-users] USING ASTERISK AS AVAYA DEFINITY RECORDING SERVER

What do you mean chief? What am looking at is ability for asterisk to receive a call and recording until it tier down without bridging it to the physical device Sam Would you like the advice in all caps? On 03/15/2010 01:20 AM, RESEARCH wrote: Hi there I remember to ask this question

Re: [asterisk-users] Using asterisk as avaya definity recording server

Oh.. I didnt know that. Thanks Sam Muro, Sam escribió: What do you mean chief? What am looking at is ability for asterisk to receive a call and recording until it tier down without bridging it to the physical device Sam Would you like the advice in all caps? He means that you put

Re: [asterisk-users] VERY HIGH LOAD AVERAGE: top - 10:27:57 up 199 days, 5:18, 2 users, load average: 67.75, 62.55, 55.75

Hi Steve Even though you shouldn't have to, have your rebooted? 200 days of uptime and this just started? It seems this problem is common as i have three boxes of the same capacity with exactly the same problem. So reboot should only solve the problem for a while Have you recently updated

Re: [asterisk-users] VERY HIGH LOAD AVERAGE: top - 10:27:57 up 199 days, 5:18, 2 users, load average: 67.75, 62.55, 55.75

Hi Team Can someone advice me on how i can lower the load average on my asterisk server? dahdi-linux-2.1.0.4 dahdi-tools-2.1.0.2 libpri-1.4.10.1 asterisk-1.4.25.1 2 X TE412P Digium cards on ISDN PRI Im using the system as an IVR without any transcoding or bridging

Re: [asterisk-users] VERY HIGH LOAD AVERAGE: top - 10:27:57 up 199 days, 5:18, 2 users, load average: 67.75, 62.55, 55.75

** top - 10:27:57 up 199 days, 5:18, 2 users, load average: 67.75, 62.55, 55.75 Hi Sam! Hello Steve! Are there any side-effects from the high load average? The system doesn't seem to be CPU or disk bound from the look of the CPU stats. System %age is high by way

[asterisk-users] VERY HIGH LOAD AVERAGE: top - 10:27:57 up 199 days, 5:18, 2 users, load average: 67.75, 62.55, 55.75

:37.83 events/1 28 root 10 -5 000 S 0.0 0.0 0:15.67 events/2 29 root 10 -5 000 S 0.0 0.0 0:40.36 events/3 30 root 10 -5 000 S 0.0 0.0 0:16.45 events/4 * Thanks Sam

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