[asterisk-users] MeetMe and dynamic_features

Hello, I am trying to use a dynamic_features during a MeetMe conference without any luck. The dynamic_features defined macro works great during a normal call, but is ignored while on a MeetMe conference. extensions.conf [macro-RaiseHand] exten => s,1,DumpChan(1) features.conf RaiseHand =>

[asterisk-users] Spandsp updated

I just noticed there is some sort of new spandsp library. http://www.soft-switch.org/downloads/spandsp/snapshots/ The version reported was still 0.0.6 and there is absolutely no "whats new" file. Is there anyone with more details? Leandro --

[asterisk-users] Pound and hash

Hello, am I wrong or the audio file for vm-rec-name in en_GB package says "pound" instead of "hash"? Pound should be for American while British use hash for the # key. Leandro -- _ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] Tricking asterisk to think the call has ended, but it was continuing on the other side

Unfortunately the only log messages regarding that channel are the "joined" and the "left" for both legs. VERBOSE[18771][C-066c] bridge_channel.c: Channel SIP/201-boxoffice-0f66 joined 'simple_bridge' basic-bridge <00bd58c3-3bce-4f1b-9d79-11eb96f37260> VERBOSE[18779][C-066c]

Re: [asterisk-users] Tricking asterisk to think the call has ended, but it was continuing on the other side

of "music on hold" for this channel? > > Greetings > Max > > > - Nachricht von Leandro Dardini <ldard...@gmail.com> - > Datum: Thu, 15 Sep 2016 18:06:14 +0200 >Von: Leandro Dardini <ldard...@gmail.com> > Antwort an: Asteris

[asterisk-users] Tricking asterisk to think the call has ended, but it was continuing on the other side

I am banging my head over a simple asterisk trick I was seeing on one asterisk server. An extension dials an international premium number, the called number answers, then the extension hangups, but the call continue to run on the international number side, generating an high profit for the

[asterisk-users] Different cachertclasses setting for different Music on Hold

As you know, there is the following settings [general] cachertclasses=yes ; use 1 instance of moh class for all users who are using it, ; decrease consumable cpu cycles and memory ; disabled by default It allows to use a single instance of MOH for all

Re: [asterisk-users] Impossible to use any recent asterisk version with chan_sip

No. I thank you for all the hard work done and dedication to the project. Leandro Il 06/Lug/2016 11:10 PM, "Joshua Colp" <jc...@digium.com> ha scritto: > Leandro Dardini wrote: > >> This is a great news, thank you. I have open the issue, >> https://issues.aster

Re: [asterisk-users] Impossible to use any recent asterisk version with chan_sip

This is a great news, thank you. I have open the issue, https://issues.asterisk.org/jira/browse/ASTERISK-26177 and added the relevant files, let me know if you need more info. Leandro 2016-07-06 21:46 GMT+02:00 Joshua Colp <jc...@digium.com>: > Leandro Dardini wrote: > >>

[asterisk-users] Impossible to use any recent asterisk version with chan_sip

Hello, I'd like to know if anyone of you is finding my same problems using any recent asterisk version, after 13.7 / 13.8 with chan_sip. If I use any recent asterisk version, after just few seconds asterisk completely locks up, stopping processing SIP/UDP packets. Nothing is written in the

[asterisk-users] Registration server with PJSIP

Hello, I am moving from realtime chan_sip to pjsip and one of the problem I am facing is the lack of "sipregs". With chan_sip, when an extension registers, the server where it has registered to is stored in sipregs. Is there something similar in pjsip? How can I find on which server the pjsip

[asterisk-users] Recording barged calls

Hi, I'd like to record the barged call... but whichever leg of the call I try to barge, my speaking is never recorded using MixMonitor. Any idea about the reason? Leandro -- _ -- Bandwidth and Colocation Provided by

[asterisk-users] Manager events when ringing multiple extensions at once and pickupExten is used

I run in a weird issue with a BLF application I have written... this application is just receiving events from Asterisk Manager Interface and blink the lights accordingly. All almost work perfectly, except when a pickupexen is used when multiple extensions are dialed. If extension 105 dials

Re: [asterisk-users] Crash asterisk res_odbc

Which operating system are you using? I have experienced the same problem on several OS except for CentOS 6. I suppose an ODBC problem on newer OS version. Leandro Il 24/Feb/2016 05:30 PM, "Maxime" ha scritto: > Dear list, > > i have a issue > > Asterisk crash (Module

Re: [asterisk-users] Best Asterisk Platform

Please chech also MiRTA PBX http://www.mirtapbx.com ... it is a multitenant realtime multiserver interface. Leandro Il 23/Dic/2015 09:06 AM, "er ic" ha scritto: > Although, I do like the OS information. I personally am a fan of CentOS. > > I realize now that the

Re: [asterisk-users] Network range in trunk definition

I see, really thank you ... I have just migrated my config. By the way ... is pjsip realtime supporting realtime registrations? Leandro 2015-09-08 21:23 GMT+02:00 Joshua Colp <jc...@digium.com>: > On 15-09-08 04:21 PM, Leandro Dardini wrote: > >> I have some problem finding

[asterisk-users] Network range in trunk definition

I have some problem finding a smart way to add inbound trunks ip authentication. I don't want to set allowguests=yes Some of my providers just list some IP and I add them like: [provider](!) context=fromoutside type=friend insecure=port,invite disallow=all allow=g729 allow=ulaw allow=alaw

[asterisk-users] Escaping parameter for ODBC function

Hello, I just noticed a weird behavior when using ODBC functions. If the content of any of the paramter has a "=" inside, then the function is not processed correctly by asterisk. Let's take for example the following ODBC function in func_odbc.conf [LOG_SMS] dsn=asterisk1,asterisk2 synopsis=Log

[asterisk-users] Stopping recordings on all legs

Hello,I'd like to use a feature code for stopping recordings. Things are quite easy when the call is received from the outside or just dialed from inside to outside, but it can go really crazy when there are blind and attended transfer going on. It ends I don't know on which call leg is the

[asterisk-users] Realtime peers and mailbox not existant

Some time to time, usually after an asterisk restart or a sip reload, some realtime sip peers are loaded in memory without their mailbox. I was not able to replicate the issue on a constant basis, but after adding some additional logs to asterisk, it seems the add_peer_mailboxes is run correctly,

[asterisk-users] Realtime peers, mailbox and MWI problem

Hello, I am facing a problem I can't understand. I have several realtime SIP peers and from time to time, the mailbox field is not loaded in asterisk memory. The mailbox field is correctly populated in the database, but often, after an asterisk restart, the mailbox is not associated to the peer

[asterisk-users] Mixing HASH() and LOCAL()

The HASH function is really useful when you have to deal with values loaded using func_odbc, but how do you use with the LOCAL function? Is it possible to define a HASH as LOCAL? Leandro -- _ -- Bandwidth and Colocation Provided

[asterisk-users] Realtime followme and channel variables

Followme is perfect to handle FMFM and it is now also realtime, but it seems impossible to assign some value to a variable, from within the followme to store info for example about the tenant the followme is running under, like instead happen for example in the queue with the setinterfacevar

[asterisk-users] Dialing multiple channels with confirm

I'd like to dial two extensions (or external number) and ask for confirmation to accept the call. Dialing an extension, asking for confirmation and then dialing a second extension if the call has not been accepted is easy by using the dial option U(...), but if I dial two extensions at once, when

[asterisk-users] Weird callerid when getting call from Parking lot

Hello, I am experiencing a weird problem on asterisk when I place an outbound call, park it and then retrieve it. I am using extensions.ael with macro and switch and I get something as SW_456_... that is autogenerated by asterisk when compiling the extensions.ael This doesn't happen when the call

[asterisk-users] Inline transfer

Hello, while most of the physical phones have keys to handle attended and blind transfer, most soft phones have no support for it. Asterisk offers a featuremap to assign a key to blindxfer and atxfer and they work fine if the call is still in the same starting context, but if the call has moved in

[asterisk-users] Mailbox password change problem on realtime engine

Hello, I am struggling with what seems a common unresolved problem, changing the password from voicemailman when using a realtime engine (adaptive_odbc in my case, connected to mysql). I have seen messages dating back to 2007 with this problem and the last one was bug 5168, reported as closed,

Re: [asterisk-users] Showing sip subscriptions in Manager

show subscriptions as a parameter -- Alex Epshteyn email: a...@thirdlane.com web: www.thirdlane.com phone +1 415.261.6601 - Original Message - From: Leandro Dardini ldard...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com

[asterisk-users] Showing sip subscriptions in Manager

Hello, almost any useful CLI command has an analogue on Asterisk Manager Interface, but I cannot find a way to get the list of subscriptions using AMI. Which is the command, if any? The CLI command is sip show subscriptions Leandro --

[asterisk-users] Propagating channel driver flag

Starting with asterisk 1.8, when you dial multiple channels at once and one of them is answered, all other channels were canceled with the cause 200 - Call completed elsewhere, so modern phones don't display the call as missed. Do you know a way to transmit this cause over multiple channels? Let

[asterisk-users] SLA (Shared Line Appearance) and realtime

Hello, do you know if it is possible to define the SLA configuration in the database for realtime usage with asterisk? Leandro -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us

[asterisk-users] SPA504G auto answer

Hello, I am struggling to have a SPA504G to auto answer (for intercom/paging). I have tried the following SIP headers (not all together), but without luck: SIPAddHeader(Call-Info:\;answer-after=0); SIPAddHeader(Call-Info: answer-after=0); SIPAddHeader(Alert-Info: info=intercom);

[asterisk-users] Asterisk 12.6 and MWI, no more working

Hello, while moving from asterisk 12.3 to asterisk 12.6, I see the MWI support for voicemail has stopped working. If I check sip show peer 104-DEVEL on asterisk 12.3, I can clearly see the Mailbox option set, while on asterisk 12.6 it appears empty. Is there anything to do more for having MWI to

[asterisk-users] Voicemail message number off by one when using ODBC storage

Hello, have you noticed the message num (VM_MSGNUM) is off by one? For example, I receive the following message: Just wanted to let you know you were just left a 0:03 long message (number 7) but in attach there is the msg0006.wav Leandro --

Re: [asterisk-users] features.conf and mixmonitor stop and start

to the same file (using the a option) On 27 August 2014 21:20, Leandro Dardini ldard...@gmail.com wrote: Hello, I have a recording started in the dialplan with the MixMonitor application. I want to be able to stop it during a call and maybe restart it. I tried using the value defined

[asterisk-users] features.conf and mixmonitor stop and start

Hello, I have a recording started in the dialplan with the MixMonitor application. I want to be able to stop it during a call and maybe restart it. I tried using the value defined in [featuremap] but it starts another MixMonitor application even if there already one instead of stopping it. Any

[asterisk-users] Calls to voicemail drops after 41 seconds due to no rtp packets

Hello, I have my provider dropping the calls after 41 seconds of not receiving any RTP from my asterisk. Obviously there is no RTP back when the caller is leaving a message in the voicemail. Is it possible to have asterisk generate some RTP packet back? Leandro --

Re: [asterisk-users] Realtime integration: Unregistered clients showing as registered?

It is the way it works. First the phone sends a REGISTER without any password. Asterisk answers with a Unauthorized and provide a nonce to be used for the next registration attempt, using it to encrypt the password. Leandro 2014-05-14 13:12 GMT+02:00 Olli Heiskanen

[asterisk-users] 302 Moved Temporarily and channel variable

When a call is transferred to another extension using a blind transfer, asterisk keeps traces of who is transferring in the BLINDTRANSFER variable. If instead the call is forwarded using most phone call forward feature, a 302 Moved Temporarily is sent back to asterisk -- Called SIP/104-DEVEL

Re: [asterisk-users] Strange incoming call issue.

About a call not being hang up for asterisk while the client hang up, please remember SIP is based on UDP and UDP packets get easily lost... they are retransmitted but sometime they are lost as the previous... For the ghost calls, are the SIP port of the phones reachable from the Internet...

Re: [asterisk-users] SPA112 Won't stay up

How long is the registration timeout? If the device is behind a router/firewall, then you need to set a registration timeout lower than the state table life in the router/firewall. I usually set my devices to just 2 minutes and it works almost all the time. Most Cisco devices have a very long

[asterisk-users] CDR(start) returns nothing in Asterisk 12

Hello, I am migrating my dialplan from asterisk 11 to asterisk 12 and it seems the ${CDR(start)} is not returning any data. Other functions, like ${CDR(duration)} or ${CDR(src)} or ${CDR(accountcode)} are returning correct values. Where is my mistake? Has this function being renamed? Leandro --

Re: [asterisk-users] CDR(start) returns nothing in Asterisk 12

I love you all :-) Leandro 2014-02-05 Richard Mudgett rmudg...@digium.com: On Wed, Feb 5, 2014 at 2:46 PM, Leandro Dardini ldard...@gmail.comwrote: Hello, I am migrating my dialplan from asterisk 11 to asterisk 12 and it seems the ${CDR(start)} is not returning any data. Other

Re: [asterisk-users] Parking in Asterisk 12.0.0

I have converted the normal Park application and I can only alert you about the syntax change. I suspect also in the ParkAndAnnounce command, the parameters are ordered completely different. Leandro 2014-01-30 Anders Larsson aster...@adev.se: Hi I'm trying to get the rebuilt parking

[asterisk-users] CDR and Transfer, an asterisk scaring bug lasting from 1.4 version...

When you use a product which version number is 11 or even 12, you might go with the assumption all big bugs are fixed and then you find there is a huge, important, expensive bug still running in the code we are relaying upon... The problem is simple. If you transfer a call, that dialing will be

Re: [asterisk-users] CDR and Transfer, an asterisk scaring bug lasting from 1.4 version...

2014/1/23 Matthew Jordan mjor...@digium.com On Thu, Jan 23, 2014 at 12:44 PM, Leandro Dardini ldard...@gmail.com wrote: When you use a product which version number is 11 or even 12, you might go with the assumption all big bugs are fixed and then you find there is a huge, important

Re: [asterisk-users] Asterisk Fax detection *11.7

It is really more interesting the receiving part. Can you paste here? Leandro 2014/1/21 Jakob-Matthias Böttger ja...@j-mb.de Hello everybody I'm trying to enable the Digium res_fax app at my *11.7 Server. a fax show stats comes up with FAX Statistics: --- Current Sessions

Re: [asterisk-users] Dialing a SIP URI with an ;ext= parameter

I am going to try a Lync server/asterisk integration, so I really appreciate! Leandro 2014/1/21 Lincoln King-Cliby linc...@controlworks.com Ok, so now I just feel kind of stupid. After I got home I decided to play with this a little more. After far too long I realized that part of the

Re: [asterisk-users] Asterisk Fax detection *11.7

I am not sure, but try to add a wait(2) as first command. When I want fax detection, I insert always a small delay for letting the fax detection routine to detect it. Leandro 2014/1/21 Jakob-Matthias Böttger ja...@j-mb.de Hi The log i've posted == Using SIP VIDEO CoS mark 6 == Using

Re: [asterisk-users] Asterisk Fax detection *11.7

Please paste the actual code. First has to be the Wait and then any other thing. Leandro 2014/1/21 Jakob-Matthias Böttger ja...@j-mb.de i already added a Progess() and Wait(5) and it still does not detect faxes. Am 21.01.2014 16:53, schrieb Leandro Dardini: I am not sure, but try

Re: [asterisk-users] Asterisk ignoring nat settings

...@pack-net.co.uk Is directmedia set to no? On 15 January 2014 23:11, Leandro Dardini ldard...@gmail.com wrote: Hello, I have an asterisk box with a peer configured with nat=force_rport,comedia, but asterisk keeps sending the audio to the private IP address and ignoring the client peer nat

[asterisk-users] Asterisk ignoring nat settings

Hello, I have an asterisk box with a peer configured with nat=force_rport,comedia, but asterisk keeps sending the audio to the private IP address and ignoring the client peer nat settings. If I check the sip show peer extension, I see both symmetric RTP and Force Rport are set to yes, but

Re: [asterisk-users] screen capture for asterisk call center solution

Just use VNC... 2013/12/20 Goke M Aruna gok...@gmail.com Thanks AJ, The capturing of agent activities on their desktop by the supervisor. Regards On 20 Dec 2013 12:18, A J Stiles asterisk_l...@earthshod.co.uk wrote: On Friday 20 December 2013, Goke M Aruna wrote: Thank you AJ, Just

Re: [asterisk-users] [NEWBIE] Right dect to buy to use with asterisk

Hello Mario, nice to meet you on this mailing list! Gigaset phones are a very high quality/price ratio, so I'll suggest you to go with the dect ip models. Then you'll need to configure asterisk to act as IVR, configure a queue and a failover to ring all hunt list. Drop me a phone call and I'll be

[asterisk-users] Answering agent

Hello friends, when a call arrives in the queue, a CDR record is created, but there is no info about which agent has picked up the call. I can find that info only in queue_log. Is there a way to have that info in the CDR or maybe in a variable in the h context, when the call is ended? Leandro --

Re: [asterisk-users] Asterisk 11.6.0 not starting up

On which kind of processor are you trying to run asterisk? Is it a real or emulated CPU? Leandro 2013/11/25 Daniel - Asterisk earohua...@gmail.com Hello Friends: I've just installed Asterisk 11 on my Linux (debian) server but it is not starting up when trying with asterisk -vvc and

[asterisk-users] Dialing directly with username and password

-0001689e]: chan_sip.c:22914 handle_response_invite: Failed to authenticate on INVITE to 'Leandro Dardini sip:100@91.11.22.33;tag=as1c0d8470' -- SIP/78.11.22.33-000144c3 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) Which is the correct syntax to use to dial directly

Re: [asterisk-users] SIP Presence across two servers

Aligning presence over multiple servers is not simple and require some changes on the dialplan and some custom code to transmit the state from one server to the other. The BLF on the phone is displayed using the hint of an extension. To be able to manually manage the hint of an extension, you

Re: [asterisk-users] SIP Presence across two servers

, 2013 at 3:54 AM, Leandro Dardini ldard...@gmail.comwrote: Aligning presence over multiple servers is not simple and require some changes on the dialplan and some custom code to transmit the state from one server to the other. The BLF on the phone is displayed using the hint of an extension

[asterisk-users] Queue linear unordered feature when using realtime

Hello, I was trying to use a queue in linear order and to provide the exact order of members to dial by adjusting the uniqueid value. Obviously it doesn't work and it seems an old problem: https://issues.asterisk.org/jira/browse/ASTERISK-18480 Realtime configuration can't identify orders in the

Re: [asterisk-users] Asterisk Realtime Static Voicemail

2013/11/11 John T. Bittner j...@xaccel.net Guys, I need you help on this one. Don’t know when this broke but we have a custom gui that runs on top of Asterisk running a real-time static for configurations. Nothing has changed with the database other than upgrades of Asterisk 10.

[asterisk-users] Disable the Connected Line info

When you set sendrpid=yes in sip.conf, a very nice feature is activated. When dialing an extension, the callerid of the dialed extension is returned back on the display of the calling phone. So if you call extension 100, you can see you are calling Ann (for example). I want to selectively disable

[asterisk-users] Sending 603 Declined message

In my dialplan I'd like to send a 603 Declined message to the user placing the call. I see the commands for the Busy and Congestion, but not the one for the Declined. Any help? Leandro -- _ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] Loopback question

Is the echo application suitable to you? Leandro 2013/5/20 CDR vene...@gmail.com Dear friends I need to loopback the audio on my channel. Did anybody on the development team thought about a function or app that would do that? If it is not clear, I mean that whatever audio I get, I send

Re: [asterisk-users] Secure Calling

I think it can be worth checking the authenticate function. http://www.voip-info.org/wiki/view/Asterisk+cmd+Authenticate 2013/5/20 Felix Vazquez felix.vazq...@theboshgroup.com How do I make a user dial a passcode to make calls through asterisk? We would like to place a phone at a client’s

Re: [asterisk-users] Passcode

Again, the authenticate function can help you Leandro 2013/5/20 Felix Vazquez felix.vazq...@theboshgroup.com How do I make a user dial a passcode if he wants to make an international call? -- This electronic message contains information from BOSH Global

Re: [asterisk-users] Dynamic realtime + queues

You need a name column. This is my queue table: CREATE TABLE IF NOT EXISTS `queue` ( `name` varchar(128) NOT NULL, `musiconhold` varchar(128) DEFAULT NULL, `announce` varchar(128) DEFAULT NULL, `context` varchar(128) DEFAULT NULL, `timeout` int(11) DEFAULT NULL, `monitor_join`

Re: [asterisk-users] Dynamic realtime + queues

|| +-+---+--+-+++ 35 rows in set (0.00 sec) - Forwarded Message - *From:* Leandro Dardini ldard...@gmail.com *To:* Tommy Cooper tomcoope...@yahoo.com; Asterisk Users Mailing List - Non

[asterisk-users] rtcachefriends and rtautoclear on change password

Hello friends, I am using from a long time rtcachefirends=yes and rtautoclear=yes in my sip.conf for asterisk 11.2.1. I have found the data of the peers are never reloaded from the database, so if you change the password for a peer, it will continue to work with the old password. Do you think it

Re: [asterisk-users] rtcachefriends and rtautoclear on change password

: - Original Message - From: Leandro Dardini ldard...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, March 26, 2013 5:28:22 AM Subject: [asterisk-users] rtcachefriends and rtautoclear on change password

Re: [asterisk-users] Optimizing Asterisk Environment

I dont apply any secret recipe while installing asterisk, but maybe you can share yours... I am typing from my mobile phone... Il giorno 23/mar/2013 14:34, Nick Khamis sym...@gmail.com ha scritto: Hello Everyone, We are getting some rather poor results (relative) with our Asterisk setup. Not

Re: [asterisk-users] Asterisk disconnecting SIP Calls after 15 Minutes

2013/3/21 Florian Wolters flor...@florian-wolters.de: Hi @ll, I just moved my Asterisk Box and changed the Provider and Internet Access to a full IP Access by Deutsche Telekom. I set up my sip.conf as I found various examples throughout the Net. Calls and some other stuff is basically

Re: [asterisk-users] Need help understanding CDR

Top replying ... In the CDR you have two fields, duration and billed. Duration is the total time from Dial command to end of calls. It is the time the Dial command is running. Billed is the time from when the other party answered and the end of the call. In your example, duration and billsec

Re: [asterisk-users] Need help understanding CDR

You can add custom fields in the CDR, so your dialplan can store start time, end time and duration whenever you like. Just use something like the Set(CDR(customfield)=100); Leandro 2013/3/18 RSCL Mumbai rscl.mum...@gmail.com: Thank you every one. Now I understand why I was confused. I have

Re: [asterisk-users] asterisk sizing for play and dtmf detection

2013/3/8 nik600 nik...@gmail.com Dear all i'm planning a migration to asterisk for a high volume IVR service (from 1000 to 1500 concurrent call) The IVR service is based only on DTMF tones so the features required is - play feature - dtmf detection Asterisk will receive calls via VOIP

Re: [asterisk-users] asterisk with 1000 extensions

2013/3/7 Steve Edwards asterisk@sedwards.com Please don't top-post. On Thu, 7 Mar 2013, Bharat Lalcheta wrote: You can use ATA box with pstn phone to reduce cost. Are you wiring a building where multiple-line SIP gateways make sense? How about a description of what you are trying

Re: [asterisk-users] asterisk with 1000 extensions

2013/3/7 Duncan Turnbull dun...@e-simple.co.nz On 7/03/2013, at 9:29 PM, Kamlesh Kumar kamlesh_...@hotmail.com wrote: On Thu, 7 Mar 2013, Bharat Lalcheta wrote: You can use ATA box with pstn phone to reduce cost. Are you wiring a building where multiple-line SIP gateways make sense?

Re: [asterisk-users] Extension cant pickup calls but can transfer.

If I was in your shoes, I'll check in the elastix mailing list... Asterisk itself can't be blamed. Leandro I am typing from my mobile phone... Il giorno 07/mar/2013 19:06, Luis H. Forchesatto luisforchesa...@gmail.com ha scritto: Greetings. I got an extension on my Elastix who cannot pick

Re: [asterisk-users] Delay before audio starts

I think a simple tcpdump of the traffic will show the mystery. It can be your provider doing something nasty. Have you tried using some other cheap SIP termination? or arrange a fake termination yourself on another server? Leandro 2013/3/1 Gerard gsara...@rarcoa.com I thought it was the

[asterisk-users] AEL Macro are evil :-)

I just discover an hidden problem with AEL macro I want to have your feedback. If you use a macro to dial out, like dialout(${EXTEN}), the leg extension will became s and if it happens you transfer the call, that will be the callerid appearing on the other phone display. I am just

Re: [asterisk-users] AEL Macro are evil :-)

...@enterux.in Hi, You might want to use ${MACRO_EXTEN} variable inside to preserve exten variable of the original dialplan exten variable. Mitul On Feb 24, 2013 4:04 PM, Leandro Dardini ldard...@gmail.com wrote: I just discover an hidden problem with AEL macro I want to have your feedback. If you

Re: [asterisk-users] Remove Abandoned call

2013/2/21 akhilesh chand omakhileshch...@gmail.com hello all, i have two asterisk server for call transfer and one more asterisk server for agent login(server_X) where agent take the call. server_A and server_B server_A is connected with pri and configure with 60 channel for call

Re: [asterisk-users] Playback on h exten

2013/2/21 Enrico Pasqualotto e.pasqualo...@netspin.it Hi all, I'm trying to setup a Quiz/feedback for caller of call center when a agent hangup. I use Asterisk 1.8.16 and I'm trying with Queue and Dial with options c and g but every time I try to play something I got: -- Executing

Re: [asterisk-users] Playback on h exten

The h exten is triggered when the channel is hangup, so you cannot send any voice data on it. Leandro 2013/2/21 Enrico Pasqualotto e.pasqualo...@netspin.it Yes, correct now it works for Dial. I think is the same with c option on Queue, do you think there's a way to do it on h exten? My goal

Re: [asterisk-users] Asterisk question

2013/2/20 Nguyễn Công nguyencong.1...@gmail.com Hello everyone, I’m new to Asterisk and I have a question. There is a phone call between two users, then they are talking to each other directly or by the server. I mean all packets from the user A to user B will be send directly to each other

Re: [asterisk-users] problem to socket programming in AGI

Check if you have selinux enforcing anf try to disable it I am typing from my mobile phone... Il giorno 04/feb/2013 18:43, C. Savinovich c.savinov...@itntelecom.com ha scritto: I would just type in the web service url manually in a browser, and if the browser displays the response, then there

Re: [asterisk-users] sip register peer (the quest for near 100% availability)

2013/1/31 Ishfaq Malik i...@pack-net.co.uk On Wed, 2013-01-30 at 14:10 -0700, Carlos Alvarez wrote: On Wed, Jan 30, 2013 at 12:05 PM, XBrian bobo...@yahoo.co.uk wrote: Thanks - I was hoping there was some silver bullet to use out there. Thanks anyway. There

Re: [asterisk-users] Asterisk Messaging Refuses To Work!

2013/1/30 XBrian bobo...@yahoo.co.uk I am pulling my hairs out here. This is my dialplan. exten = 100,1,Set(AGISIGHUP=no) exten = 100,n,AGI(a2billing.php,4,callingcard) exten = 100,n,Set(__APP_MSG_IND=${APP_MSG_IND}) exten = 100,n,Set(__APP_MESSAGE=${APP_MESSAGE}) exten = 100,n,Hangup()

Re: [asterisk-users] sip register peer (the quest for near 100% availability)

You can just shorten the time the phone device register on the asterisk server. It is up to the peer to send the registration command. It cannot be triggered or forced in any way. Leandro 2013/1/30 XBrian bobo...@yahoo.co.uk I am aware that the direction is from peer to asterisk. Its a valid

Re: [asterisk-users] round-robin in asterisk 1.4

The simplest way is to use the Random function and to pickup one number from 1 to 3 and use that line. Leandro I am typing from my mobile phone... Il giorno 29/gen/2013 11:35, Salaheddine Elharit salah.elharit...@gmail.com ha scritto: I am installing asterisk 1.4 with 2 ISP and i have one

Re: [asterisk-users] Complex Call Distribution

2013/1/26 RSCL Mumbai rscl.mum...@gmail.com Hello, I have Elastix ISO install (FreePBX 2.7.0.3) My current Setup is as follows: Inbound Route Queue (Dynamic Agents) The queue distributes calls based on rrMemory. I have been asked to redesign the call distribution as follows: Calls

Re: [asterisk-users] Realtime vs Static Files

It is a shame we were unable to find the solution to your problem. Do you want to setup a test system like the good one and let me access it to check what is going on? I am really really curious. Leandro Il giorno 26/gen/2013 19:49, Dan Journo d...@keshercommunications.com ha scritto: It is

Re: [asterisk-users] Realtime vs Static Files

2013/1/25 Dan Journo d...@keshercommunications.com Upgrading to the latest version didn't help. After about 30 minutes, Asterisk2 tries to send out OPTIONS keepalive packets to peers listed as Registered on Asterisk1. It is something really amazing... Can you run sip show peers on each

Re: [asterisk-users] Realtime vs Static Files

2013/1/24 Dan Journo d...@keshercommunications.com I am curious, is your version of asterisk correctly compiling the regserver field? Each server needs to have a distinct server name. ** ** Upgrading to the latest version didn't help. After about 30 minutes, Asterisk2 tries to send

Re: [asterisk-users] Realtime vs Static Files

2013/1/23 Dan Journo d...@keshercommunications.com Hi, ** ** We're trying to decide whether to switch back to a static file for sip.conf. Currently we use mysql realtime but can't see any real benefit.* *** ** ** Why would someone choose realtime sip over static files? ** **

Re: [asterisk-users] Realtime vs Static Files

2013/1/23 Dan Journo d...@keshercommunications.com We have never experienced that and use realtime with multiple asterisk servers. We've only recently started seeing the problem. To simplify the issue, assuming we have two servers, Asterisk1 and Asterisk2... Asterisk1 is a primary

Re: [asterisk-users] Realtime vs Static Files

2013/1/23 Dan Journo d...@keshercommunications.com Maybe you are lacking some of the configuration. These is the relevant part. ** ** rtcachefriends=yes rtsavesysname=yes rtupdate=yes rtautoclear=yes ** ** We have rtcachefriends=yes

Re: [asterisk-users] Realtime vs Static Files

2013/1/24 Dan Journo d...@keshercommunications.com Its probably an issue with the version of Asterisk we are using because I haven't had this problem in the past. I am running the latest 1.8 version. Which version are you running? ** ** ** ** 1.8.15.0. I'll upgrade it to

Re: [asterisk-users] [SOLVED] Blind transfer behavior - Asterisk 1.8 and 10

Can you please post a dialplan excerpt about using these variables. I just tried using them, but they are all empty. Maybe I am making the same mistake of you. Leandro 2013/1/22 Administrator TOOTAI ad...@tootai.net Please forget this message, BLINDTRANSFER is working, I had a typo in the

Re: [asterisk-users] g729 codec over SIP Trunk between CCM and Asterisk

2013/1/17 Onur Cem Çelebi occel...@gmail.com Hello, My problem is, outgoing calls (from asterisk to CCM) work fine but incoming (from CCM to Asterisk) does not work because of CCM is trying to use g729 over SIP trunk. I have found that link after a quick search. Problem is the same as in

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