[Asterisk-Users] [OT] Small SIP phones?

Hi. Does anyone know of any small SIP phones (and preferably have some experience of using them and happy to recommend them)? By 'small' I mean a single-piece phone, with dial buttons in the handset, so that it can be carried around easily in a laptop bag. Something like

Re: [Asterisk-Users] [OT] Small SIP phones?

On Sunday 12 December 2004 20:12, Clay Reiche wrote: I don't know of a small phone, but you use a WorlACCXX TA200 device (pretty small) along with any standard analogue phone. http://www.worldaccxx.com I have one and carry it around in my laptop bag. Demensions are 6x4.5x1.25 Thanks. In

Re: [Asterisk-Users] codec order in SIP doesn't work

On Wednesday 15 December 2004 14:21, Roy Sigurd Karlsbakk wrote: hi using the following in sip.conf, codec preferences aren't set, and asterisk uses alaw whatever I do, except force it to one specific in the [user] [general] disallow=all allow=g726 allow=g729 allow=gsm allow=alaw

Re: [Asterisk-Users] TDM400p FXO module always offhook

On Wednesday 15 December 2004 22:34, Carey Pillar wrote: I have a TDM400p with 3 FXS mods and 1 FXO mod. I have all set up with what seems to be correct settings (according to digium and asterisk wiki). As soon as I plug in my POTS line into FXO mod the line goes into offhook state (whether

Re: [Asterisk-Users] SIP Server question / recommendations

On Thursday 16 December 2004 01:09, Shahed wrote: Hello All, I am new to *, and this is my first post on the user list. I have had success with making / receiving calls to a SIP hardware Phone and the Console Channel Driver. Can anyone please suggest what would be a good SIP server to use,

Re: [Asterisk-Users] Polycom SIP Phones

On Thursday 16 December 2004 22:57, Jared Armstrong wrote: I found IP 500's for $170. Where? Antony -- The truth is rarely pure, and never simple. - Oscar Wilde Please reply to the list;

Re: [Asterisk-Users] My Boss wants background music!!!!

On Friday 17 December 2004 14:31, Steven Kalcevich (Lists) wrote: Why not just dial an extention for music when the user wants music from there desk. Because then the phone will be engaged on a call and will not ring when someone else wants to talk to the person at the desk? Antony. The

Re: [Asterisk-Users] Asterisk and HylaFax

On Friday 17 December 2004 16:22, Lee Howard wrote: On 2004.12.17 05:42 Sergio Serrano wrote: Hi all, again I try configure Hylafax with asterisk. I would like configure Asterisk in the next way: 1)An incoming fax go into through X100P 2)Asterisk detects Fax and

Re: [Asterisk-Users] application meetme

On Friday 17 December 2004 18:34, Geoffroy KOUMADI wrote: i have problem to setup application meetme. i'm using asteisk-1.0.3 and sjphone as client. Thanks for letting us know. If you want some help in solving the problem, perhaps you might tell us what the actual problem is? Useful

Re: [Asterisk-Users] Call on hold disconnects...

On Friday 17 December 2004 20:24, Ferguson, Michael wrote: G'Day All, How do I fix this: I receive a call at the extension. Press the hold button. Music on hold starts. When I place the handset back on the cradle, the call gets hung up/disconnected. The Phone is A GrandStream Budge Tone

Re: [Asterisk-Users] Call on hold disconnects...

On Friday 17 December 2004 20:43, Ferguson, Michael wrote: OK. I guess I was not clear. Sorry. The phone rings. The person picks up the handset and speaks to the caller. He then puts the call on hold by pressing the HOLD button on the GS 100 phone. The caller hears music on hold. So far,

Re: [Asterisk-Users] Total newbie here looking to do a VoIP conference call?

On Friday 17 December 2004 21:00, Patrick Campbell wrote: I am looking to help out my company find a more budget conscious but reliable way to hold conference calls between 5+ people. 4x a month we hold several hour long conference calls during non-business hours. All of the employees have

Re: [Asterisk-Users] Call on hold disconnects...

On Friday 17 December 2004 21:10, Ferguson, Michael wrote: Antony, Thanks. It seems that the GS will not keep the call on hold. In the real world though, when you place a call on hold, it is held until further action. Yes, although I might think that hanging up is a further action? The

Re: [Asterisk-Users] voicemail without prompt

On Friday 17 December 2004 21:25, Ross Kevlin wrote: this would still only work if the mailbox number was the same as the caller id. I need some way to get the actual mailbox number of the caller. Where / how are your mailbox numbers stored? It shouldn't be too difficult to create a script or

Re: [Asterisk-Users] Asterisk Hardware

On Friday 17 December 2004 21:42, Nihal wrote: Does some hardware just not work very well with Asterisk? Yes. (or, no, depending on how you view the question) I've got a fresh installation on a Fedora C2, P4x2, 2GB Ram. Some people have reported problems with FC3, I don't know if FC2

Re: [Asterisk-Users] Total newbie here looking to do a VoIPconfer ence call?

On Friday 17 December 2004 23:04, Patrick Campbell wrote: Come to think of it since the DTA310 uses DNS to find the SIP server, you could setup a DNS cache and override the DNS entry for what packet8 uses (proxy2-eqix-sjo.packet8.net : 15062, over here) to point to the IP of your own SIP

Re: [Asterisk-Users] [OT] Small SIP phones?

[mailto:[EMAIL PROTECTED] On Sun, 2004-12-12 at 21:27, Antony Stone wrote: Thanks. In fact I already have a Grandstream ATA-486, which I'm very pleased with: http://www.grandstream.com/y-ht486.htm This unit is even smaller - 105 x 75 x 25mm (or 4 x 2.75 x 1), however I'm just

Re: [Asterisk-Users] How expensive are the different codecs? (Regarding CPU time)

On Wednesday 15 December 2004 21:26, Michael Vogel wrote: Jim Van Meggelen schrieb: YIKES! What kind of processor have you got there? Its a: - Pentium II (Deschutes) 333MHz - 128mb memory I'm using it as: - Mailserver (IMAP, SMTP) - Webserver (mainly for webmail) - Newsserver -

Re: [Asterisk-Users] Newbie setup (Hardware questions)

On Wednesday 15 December 2004 23:24, Puddle wrote: Thanks, that makes a lot more sense. Would VoIP phones still require FXO units or would that not require any special telephony hardware? SIP phones connect by ethernet - no telephony hardware needed. You would want an FXO port if you want

Re: [Asterisk-Users] Open Ports

On Saturday 18 December 2004 10:17, Norman Zhang wrote: Hi, May I ask what ports are necessary for SIP communication through a firewall? I read somewhere that UDP/5060 alone is enough. Some recommends more ports to be opened for RTP. Both the above statements are correct. SIP uses port

Re: [Asterisk-Users] Open Ports

On Saturday 18 December 2004 10:58, Norman Zhang wrote: SIP uses port 5060 RTP uses multiple ports, typically in the range 1-2 Remember that SIP and RTP are different - SIP is used to set up the call; RTP is used to carry the audio once the call has been set up. Thanks. May

Re: [Asterisk-Users] Open Ports

On Saturday 18 December 2004 11:40, Rich Adamson wrote: But, to return to my initial question, what's the security risk in leaving your Asterisk server open to UDP packets from the world? I regard it like a mail server - a firewall allowing TCP packets through to port 25 cannot protect

Re: [Asterisk-Users] Re: Open Ports

On Saturday 18 December 2004 13:21, Tom Ivar Helbekkmo wrote: My home firewall allows my Asterisk PBX to send any UDP traffic to anyone, and keeps state, so they can answer. It also specifically allows anyone to connect to UDP port 5060 on the PBX. Interesting. Does that allow other people

Re: [Asterisk-Users] VoIP Termination

On Saturday 18 December 2004 18:07, Dorn Hetzel wrote: I wouldn't say I hate SIP, it sucks less than H.323 and so on by a large margin. But, having said that, if you can use IAX, it sucks even much than SIP does :) Um, are you saying IAX is good, or that it is not good? I'm not sure I

Re: [Asterisk-Users] My Boss wants background music!!!!

On Saturday 18 December 2004 20:19, Bill Seddon wrote: Detecting the ringing state of a specific device from, say, a desktop running Windows or Linux AGI is trivial. Care to share a trivial example with us? Sounds like a useful link for several applications... Antony. -- Software

Re: [Asterisk-Users] modprobe wcfxo crashes my IBM NetFinity5000 after few seconds

On Saturday 18 December 2004 20:27, Rodolfo Grave wrote: Hi and thanks once more. I moved the card around, and it kept the same IRQ. Then I went into setup and changed it. This is the output of lspci -v now: 01:04.0 Communication controller: Tiger Jet Network Inc. Intel 537

Re: [Asterisk-Users] Q about IAX (and IAXy)

On Sunday 19 December 2004 00:26, Nabeel Jafferali wrote: I have heard many times that IAX is NAT-transperant. I am unsure how it accomplishes this. I do know that SIP works like this: your SIP device send a request to the SIP server (usually on port 5060) with whatever command. The SIP

Re: [Asterisk-Users] Re: Asterisk Crackly Bad quality

On Sunday 19 December 2004 02:00, Keith O'Brien wrote: Since the incoming stream is using VAD, my assumption is that it is losing the timing during the pauses in the speech. Does anyone know of a way to just turn off VAD in *? This would have multiple benefits (if you have the bandwidth).

Re: [Asterisk-Users] Q about IAX (and IAXy)

On Sunday 19 December 2004 01:41, Nabeel Jafferali wrote: Thanks for all the info so far! Therefore a NAT device between two IAX systems has only a single channel, on a well-known port number, to deal with, and this is simple to do. So then how does IAX deal with the equivalent of SIP

Re: [Asterisk-Users] SMS - how to send one

On Sunday 19 December 2004 20:18, Brian West wrote: The SMS in asterisk is not SMS like you're thinking... Its not for sending to mobile phones and not something usable in the US. Um, sorry, but if SMS is not for sending to mobile phones, then what is it for (if it matters, I'm not in the US)

Re: [Asterisk-Users] SMS - how to send one

On Sunday 19 December 2004 21:35, Antony Stone wrote: On Sunday 19 December 2004 20:18, Brian West wrote: The SMS in asterisk is not SMS like you're thinking... Its not for sending to mobile phones and not something usable in the US. Um, sorry, but if SMS is not for sending to mobile

Re: [asterisk-users] Tesco Internet Phone

On Friday 02 March 2007 07:46, Alan Chandler wrote: On Thursday 01 March 2007 20:33, bails wrote: plug it in a linux box and tell us what it is please, generic-usb-audio or what? Bails Julian Lyndon-Smith wrote: Yeah, that's where firefly comes from, doesn't it. I've got the

Re: [asterisk-users] Tesco Internet Phone

On Wednesday 21 March 2007 11:57, bails wrote: Antony Stone wrote: On Friday 02 March 2007 07:46, Alan Chandler wrote: On Thursday 01 March 2007 20:33, bails wrote: plug it in a linux box and tell us what it is please, generic-usb-audio or what? Bails Julian Lyndon-Smith wrote

Re: [asterisk-users] Tesco Internet Phone

On Wednesday 21 March 2007 15:11, asterisk wrote: I use this driver for the SJ phone with the USB tesco internet phone: http://www.sjphone.org/usbphone/SJphoneDriverWebPoint.exe Yes, but that's a corded phone which plugs into the USB socket. # cat /proc/bus/usb/devices P:  Vendor=19af

Re: [asterisk-users] Voicemail mailbox number passed in connection?

On Wednesday 21 March 2007 19:54, Lutgring, Sam wrote: Does anyone know how to configure a SIP phone to pass the mailbox number to the voicemail service when dialing? I would like to press the message waiting lamp and be prompted for my password instead of mailbox number. Can this be passed

Re: [asterisk-users] "Your call is not allowed. P U A M I"

On Thursday 20 April 2017 at 21:29:58, Steve Edwards wrote: > Not an Asterisk question, but... > > A bunch of our 8xx numbers started playing this recording when dialed. Our > provider (Inteliquent) says it's not them. Where are Inteliquent feeding the calls (assuming they connect instead of

Re: [asterisk-users] asterisk as non root

On Wednesday 19 April 2017 at 15:44:39, Atux Atux wrote: > hello there. i am running debian 8 in my swerver and i would like to run > asterisk as non root. i did follow the > https://www.voip-info.org/wiki-Asterisk+non-root without any success. Did you do the very first step:

Re: [asterisk-users] asterisk as non root

On Thursday 20 April 2017 at 12:31:14, Atux Atux wrote: > Hi. thanks a lot for your replies. I did stop the services and i did issued > the the "chown" and "chmod" commands listed in the guide. > It is necessary to compile it, instead if using the apt-get version > What am i missing? Let's go

Re: [asterisk-users] asterisk as non root

On Wednesday 19 April 2017 at 18:48:29, Atux Atux wrote: > Hi. > Here is the output of the command > > root@pbx: ~ $ find / -name asterisk -exec ls -ld '{}' \; > > drwxr-xr-x 3 root root 4096 Apr 19 17:32 /usr/include/asterisk > > drwxr-x--- 3 asterisk asterisk 4096 Apr 19 17:32

Re: [asterisk-users] asterisk as non root

On Thursday 20 April 2017 at 18:31:03, Atux Atux wrote: > root@PBX: /var/www/html $ /etc/init.d/asterisk start > [ ok ] Starting asterisk (via systemctl): asterisk.service. I'm somewhat puzzled that your root-user prompt is "$" instead of the more normal "#", but never mind... > root@PBX:

Re: [asterisk-users] IAX2 getting stuck

On Wednesday 19 April 2017 at 23:35:24, Carlos Chavez wrote: > On 4/19/17 4:23 PM, Antony Stone wrote: > > > > You say the USB ethernet adapter got unplugged and then reconnected... > > > > 1. What's the name of the network device for this adapter? Is it the > &g

Re: [asterisk-users] IAX2 getting stuck

On Wednesday 19 April 2017 at 23:14:46, Carlos Chavez wrote: > On 4/19/17 4:09 PM, Antony Stone wrote: > > On Wednesday 19 April 2017 at 22:54:51, Carlos Chavez wrote: > >>I have a server that had been operating for a few years now with > >> > >>

Re: [asterisk-users] IAX2 getting stuck

On Wednesday 19 April 2017 at 22:54:51, Carlos Chavez wrote: > I have a server that had been operating for a few years now with > IAX2 trunks to several other servers. Since yesterday all IAX2 trunks > now say UNREACHABLE. ...snip... > So far the only thing different is that the receive

Re: [asterisk-users] asterisk name in mysql

On Saturday 22 April 2017 at 22:25:52, Atux Atux wrote: > Thanks a lot for the reply. > I did follow that already, but i do have a problem. Here is my > extensions.conf part for that particular number > exten => 6912345678,1,Answer() > exten => 6912345678,n,MYSQL(Connect connid 127.0.0.1 root

[asterisk-users] Dial() using full SIP account details

call: Conflicting extension values given. Using '832+ios' and not '0203yyy' == Using SIP RTP CoS mark 5 -- Called SIP/832+ios:31onpmlq_9d_x...@remote.server.com/0203yyy [2017-02-28 11:43:47] NOTICE[11692][C-0d22]: chan_sip.c:23010 handle_response_invite: Failed to authenticate on

Re: [asterisk-users] [asterisk13] Multiple transport objects of same protocol in pjsip.conf

On Saturday 29 July 2017 at 19:03:55, Joshua Colp wrote: > On Sat, Jul 29, 2017, at 02:55 PM, O. Hartmann wrote: > > Scenario: > > > > Our Asterisk 13 PBX (on network 192.168.254.0/24, bound to > > 192.168.254.1:5060) is behind > > a NAT, acting as a client to our ITSPs SIP server. But also,

[asterisk-users] AMI column widths

Hi. I'm trying to get a list of the channels currently in use on an Asterisk server (1.8.32.1 if it matters) over AMI. I send the command "sip show channels", and I get back a response along the lines of (* used to protect the innocent...): Peer User/ANR Call ID

Re: [asterisk-users] AMI column widths

On Saturday 08 July 2017 at 10:16:19, Antony Stone wrote: > On Saturday 08 July 2017 at 07:15:08, Marcelo Terres wrote: > > There are no sip show channels on AMI. Also, the output that you sent is > > not a AMI output. Are u using AMI ou running commands on console? > >

Re: [asterisk-users] AMI column widths

achieve your goals > > https://wiki.asterisk.org/wiki/display/AST/AMI+Actions Hm, I don't see anything there which will give me a list of the SIP channels currently in use - what command should I be using for that? Thanks, Antony. > On 7 Jul 2017 10:32 pm, Antony Stone wrote: > > Hi.

Re: [asterisk-users] choppy calls version Asterisk 13.17 on CentOS 7

On Friday 14 July 2017 at 23:34:37, Motty Cruz wrote: > Since the upgrade our remote users' conversions are choppy. > Monitoring using CLI, I noticed the device always select ulaw > for codec. What's the device? What are its codec settings? What's your available & used bandwidth on the

Re: [asterisk-users] Asterisk 13.16.0 segfault

On Thursday 20 July 2017 at 20:46:30, Marcelo Terres wrote: > I don't have much knowledge about freepbx, but if some day I had to use it, > I would prefer to use the Asterisk compiled from source, unless it comes > with an Asterisk package (rpm, supposing it is running CentOS). FreePBX (as a

Re: [asterisk-users] Simplest way of executing a non-blocking (async) python AGI script?

On Friday 30 June 2017 at 19:11:08, Jonathan H wrote: > I use a python AGI which pulls some info from a web service, which should > take half a second. > > Sometimes, it takes 5-10 seconds which blocks the dialplan execution, but > the dialplan should continue immediately as it's not dependent

[asterisk-users] PJSIP list of peers online/offline?

Hi. I have some Nagios / Icinga monitoring plugins I've created for Asterisk, and one of them checks the percentage of SIP accounts which are currently registered on an Asterisk server. It does this by running "sip show peers" via AMI and analysing the summary line at the end: 1066 sip peers

Re: [asterisk-users] Writing CDR's to two database servers

On Monday 19 June 2017 at 18:12:35, Sebastian Gutierrez wrote: > use replication 1. Agreed - use replication. 2. If you want an HA (High Availability, not dependent on a single Master DB server replicating to a slave) solution, consider setting up Master-Master replication, with an LVS (Linux

Re: [asterisk-users] Autodialer - call simultaneously to both ends

On Monday 26 June 2017 at 14:06:10, Harel wrote: > Hello List, > I'm working on an autodialer project. > At the moment I use the Originate application then I "throw" it to an > extension where I Dial() the other party and then both legs are bridged. > > The problem is that the Dial() will only

Re: [asterisk-users] Autodialer - call simultaneously to both ends

On Monday 26 June 2017 at 18:01:22, J Montoya or A J Stiles wrote: > On Monday 26 Jun 2017, Harel wrote: > > Hello List, > > I'm working on an autodialer project. > > At the moment I use the Originate application then I "throw" it to an > > extension where I Dial() the other party and then both

Re: [asterisk-users] Automatically dial a number, then an extension

On Tuesday 23 May 2017 at 19:20:25, Tech Support wrote: > All; > > What I did was add a line in the dialplan that used the SendDTMF() > application and that worked great. The problem that I’ve run into now is > that dialing the extension screwed up the answering machine detection. The >

Re: [asterisk-users] Automatically dial a number, then an extension

On Tuesday 23 May 2017 at 20:01:14, Tech Support wrote: > Ok, the purpose of the answering machine detection (AMD) is to > determine when the audio file should start playing *after* the call has > been picked up. Typically, if a call has been picked up by a person, they > say a short

Re: [asterisk-users] Upgraded server crashes on voicemail storage

On Tuesday 06 June 2017 17:54:59 Mike Diehl wrote: > Hi all, > > I'm upgrading to Asterisk 13.14.0 x86_64. During my beta testing, I've > discovered that my server crashes as soon as I leave a voicemail message. > I'm using odbc voicemail storage as well as mysql dynamic configuration. > >

Re: [asterisk-users] asterisk server - no sound

On Tuesday 06 June 2017 15:18:32 andre castro wrote: > I just installed asterisk in a debian server. > All seems to be running fine, but the audio sent by the server. > But I hear nothing at the peer's end. > > When one peer calls another, sound comes through just fine. Tell us about your

Re: [asterisk-users] asterisk server - no sound

On Tuesday 06 June 2017 16:57:07 andre castro wrote: > On 06/06/2017 04:36 PM, Antony Stone wrote: > > > > Tell us about your networking arrangement - are both phones and the > > Asterisk machine on the same network? > > Nop. They are in 2 different

Re: [asterisk-users] CM for menuselect choices

On Friday 05 May 2017 at 16:21:20, Richard Kenner wrote: > I'd like to be able to save the choices made in menuselect in a way > that they can be tracked in a CM system and applied to a later release > of Asterisk using an automated tool like Ansible. What's the best > way to do that?

Re: [asterisk-users] CM for menuselect choices

On Friday 05 May 2017 at 16:52:39, Richard Kenner wrote: > > Of course, you might run into problems if the later release introduces > > new options (or deprecates old ones) which then aren't going to be in > > your makeopts file > > That's my question: how do I reflect the changes that I made to

Re: [asterisk-users] SwitchVox and Asterisk

On Monday 08 May 2017 at 16:43:24, Luca Pradovera wrote: > Hello, > I need to have an extension on a SwitchVox server dial out to one on an > Asterisk (FreePBX actually) box which will host a voice directory. What's a voice directory? > The Asterisk server will then need to dial one of the

Re: [asterisk-users] Dial an extension to modify dialplan

On Monday 08 May 2017 at 15:44:47, Marcelo Terres wrote: > https://wiki.asterisk.org/wiki/display/AST/Asterisk+14+ManagerAction_Dialpl > anExtensionAdd > > Is it enough? Is there a similar call to delete an extension, or to modify an existing one? On the basis that the OP already has extension

Re: [asterisk-users] Need to restart Asterisk if remote server not working?

On Saturday 06 May 2017 at 09:21:16, Luca Bertoncello wrote: > Antony Stone schrieb: > > > 4. Did the IP address of Telekom's end of the connection change? > > I really don't know, but I suppose not I suspect this may in fact have been the cause of your problem. Firstly, I

Re: [asterisk-users] Need to restart Asterisk if remote server not working?

On Saturday 06 May 2017 at 08:37:39, Luca Bertoncello wrote: > Yesterday Deutsche Telekom had a really big problem and Asterisk couldn't > connect to the remote Server (by Telekom) until today about 7:30. > > Well, it could happen... > What I find really annoying was that I needed to restart

Re: [asterisk-users] A bit OT - Configure GoIP for Asterisk

On Monday 02 October 2017 at 20:58:33, Steve Edwards wrote: > I recently received a GoIP-32 for a client project -- primarily outbound > calling. > > How should a GoIP be configured for Asterisk? Have you tried http://www.hybertone.com/en/solutionsClass.asp?Id=78 Antony. -- Police have

[asterisk-users] Discovering ring time immediately after call is answered

Hi. Does anyone know of a way to find out the ring time of a call as soon as it has been answered (ie: without waiting for the call to be completed, when it's part of the standard CDR record)? I'm looking for a way to place a call, wait for it to be answered, and then perform different

Re: [asterisk-users] Asterisk bugs make a right mess of RTP

On Friday 01 September 2017 at 16:48:17, Dovid Bender wrote: > On Fri, Sep 1, 2017 at 9:13 AM, Joshua Colp wrote: > > On Fri, Sep 1, 2017, at 09:01 AM, Dave Topping wrote: > > > http:/www.theregister.co.uk/2017/09/01/asterisk_admin_patch/ > As Josh mentioned this is an issue

Re: [asterisk-users] ERROR during high volume MoH dialplan

On Thursday 31 August 2017 at 18:15:54, Joseph Smith wrote: > I was hoping Asterisk would handle more than 4k simultaneous calls. I know from experience that Asterisk can handle more than 4k simultaneous calls, however it's an extreme case to have all of them playing music on hold. I think

[asterisk-users] Adding custom commands to AMI

Hi. https://www.voip-info.org/wiki/view/Asterisk+manager+API says that "There are a finite (but extendable) set of actions available to the client, determined by the modules presently loaded in the Asterisk engine." Can anyone point me at some appropriate documentation for adding custom

Re: [asterisk-users] Adding custom commands to AMI

On Sunday 12 November 2017 at 18:27:56, Tzafrir Cohen wrote: > On Sun, Nov 12, 2017 at 04:45:45PM +0000, Antony Stone wrote: > > Hi. > > > > https://www.voip-info.org/wiki/view/Asterisk+manager+API says that "There > > are a finite (but extendable) set of

Re: [asterisk-users] asterisk 13.18.0: pjsip: unnecessary 603 decline because of wrong codec decision

On Wednesday 01 November 2017 at 08:10:36, Michael Maier wrote: > Hello! > > I'm facing the following scenario: > > - Initial call opened to asterisk: SDP g722,alaw,ulaw > > - Outgoing call to provider started with Invite / SDP alaw, g726 and > g729. So, you're claiming to the provider that

Re: [asterisk-users] Looking for the carrier that owns a particular DID

On Thursday 02 November 2017 at 16:33:04, Tech Support wrote: > I have a customer who is looking for a particular DID. (I dialed it and > it is not in service). I searched through my preferred upstream provider's > list but I came up empty. I wrote them, and this is their reply. > > "We

Re: [asterisk-users] asterisk 13.18.0: pjsip: unnecessary 603 decline because of wrong codec decision

On Wednesday 01 November 2017 at 12:15:08, Michael Maier wrote: > On 11/01/2017 at 10:14 AM Antony Stone wrote: > > On Wednesday 01 November 2017 at 08:10:36, Michael Maier wrote: > >> > >> I'm facing the following scenario: > >> > >> - Initia

[asterisk-users] Measuring total end-to-end latency

Hi. Does anyone have some recommendations for measuring total end-to-end latency (by which I mean: the time between person A saying something and person B hearing it) when there are both SIP and PSTN/analogue/mobile legs in the call path? Examples: Person A has a SIP phone registered to

Re: [asterisk-users] Using ControlPlayback with AWS S3

On Wednesday 06 June 2018 at 12:02:38, Dovid Bender wrote: > Hi, > > I have tested ControlPlayback and grabbed files via an apache server with > no issue. ControlPlayback is an Asterisk dialplan function. How have you integrated this with Apache? > I want to be able to grab files via aws S3

Re: [asterisk-users] How to execute priorities following a caller hangup in a successful Dial?

On Tuesday 05 June 2018 at 08:33:26, David P wrote: > We're using Asterisk 14.7.6 and I have a dialplan that ends like this: > > same => n,Dial(SIP/${EXTEN:0:4}@peer1) > same => n,Set(GLOBAL(EpochAtCallEnd)=${EPOCH}) > same => n,Hangup() > > When peer1 hangsup, the priorities after the Dial

Re: [asterisk-users] Using ControlPlayback with AWS S3

On Wednesday 06 June 2018 at 16:30:08, Dovid Bender wrote: > On Wed, Jun 6, 2018 at 6:18 AM, Antony Stone wrote: > > On Wednesday 06 June 2018 at 12:02:38, Dovid Bender wrote: > > > Hi, > > > > > > I have tested ControlPlayback and grabbed files via

Re: [asterisk-users] AMI manager logins - omitting from logging output?

On Thursday 07 June 2018 at 10:44:15, Tony Mountifield wrote: > In article <201806070119.51560>, Antony Stone wrote: > > > > Is there any way to tell AMI that I don't want it to log login attempts - > > or, to put it another way, is there any way to tell the logg

[asterisk-users] AMI manager logins - omitting from logging output?

Hi. Is there any way to eliminate AMI manager logins from the logging output (without just turning the log level down and thereby losing lots of other stuff as well)? I'm running Asterisk 13.14.1 as a backend service to LVS/IPVS, and using the AMI login as the "service alive" check to see

Re: [asterisk-users] doing dnsmgr_lookup for

On Thursday 31 May 2018 at 15:52:53, Jonas Kellens wrote: > Hello list > > is there a way to limit the number of dns lookups for 1 and the same host? > > I see on Asterisk CLI a flood of : > > [May 31 15:45:37]> doing dnsmgr_lookup for 'proxy1.sip.x2reg.be' > [May 31 15:45:37]>

Re: [asterisk-users] SIP invite timeouts : how is someone sending invites from our server ??

On Sunday 31 December 2017 at 00:49:17, sean darcy wrote: > I've been getting a lot of timeouts on non-critical invite transactions. > So how is someone on a Dutch ISP using my server to mess with a US DoD > ip address ? What's your setting for "allowguest" (under [general]) in

Re: [asterisk-users] sip trunk with social media

On Thursday 04 January 2018 at 01:27:59, bilal ghayyad wrote: > Hello > It will be amazing if possible to do sip trunk with any of social media > providers like: whatsapp, facebook, imo, viber, ... etc To the best of my knowledge none of the services you mention either operate over SIP or

Re: [asterisk-users] remote Asterisk console

On Tuesday 16 January 2018 at 18:19:30, Paul Neuwirth wrote: > On Tue, 16 Jan 2018 18:18:18 +0200 Tzafrir Cohen wrote: > > > Anyway, as mentioned before: you should probably use AMI. > > Thank you both. That was (most likely) what I was looking for - but > still some worries about sending

Re: [asterisk-users] Can anyone help with a quick app_record.c module improvement and can explain over-riding modules?

On Saturday 20 January 2018 at 18:45:49, Jonathan H wrote: > Oh, what a good idea! That's exactly the kind of lateral thinking I > was hoping someone would come up with. > > I thought it was called MixMonitor, and tried to wrap my head around > it but couldn't. MixMonitor is related, but

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

On Monday 12 February 2018 at 12:25:00, Uzma Anjum wrote: > Hello... > > I'm running asterisk-13 and international calls are not working in it.How > can I make it work.Can anyone please tell me. We are sorry, but all our telepaths and clairvoyants are busy dealing with other queries right now.

Re: [asterisk-users] Sip cause and response codes in dialplan

On Tuesday 20 February 2018 at 14:09:05, Marcus Kvarsell wrote: > Hi, > > I am experimenting with getting hold of the sip cause and sip response from > outgoing call. How could i make a userevent printing the sip cause and/or > sip response. I have tried using hangupcause, sip_cause and such ,

Re: [asterisk-users] Set external CID but retain internal extension in CDR...

On Thursday 22 February 2018 at 21:41:41, Carlos Chavez wrote: > On 2/22/18 1:07 PM, Antony Stone wrote: > > On Thursday 22 February 2018 at 19:10:51, Carlos Chavez wrote: > >> Usually phone companies set the outgoing CallerID for you but > >> > &

Re: [asterisk-users] Set external CID but retain internal extension in CDR...

On Thursday 22 February 2018 at 20:07:47, Antony Stone wrote: > On Thursday 22 February 2018 at 19:10:51, Carlos Chavez wrote: > > Usually phone companies set the outgoing CallerID for you but > > > > recently we got control over that and are now setting the ou

Re: [asterisk-users] Set external CID but retain internal extension in CDR...

On Thursday 22 February 2018 at 19:10:51, Carlos Chavez wrote: > Usually phone companies set the outgoing CallerID for you but > recently we got control over that and are now setting the outgoing > Calleir ID ourselves. My problem now is that the CDR will put the > outgoing CID in the CDR

Re: [asterisk-users] Set external CID but retain internal extension in CDR...

On Thursday 22 February 2018 at 23:44:43, Carlos Chavez wrote: > On 2/22/18 4:40 PM, Carlos Chavez wrote: > > On 2/22/18 3:54 PM, Carlos Chavez wrote: > >> On 2/22/18 3:46 PM, Antony Stone wrote: > >>> On Thursday 22 February 2018 at 21:41:41, Carlos Chavez wr

Re: [asterisk-users] Increasing timeout before ending call from AMI

On Tuesday 31 July 2018 at 12:38:04, Raimundo Pérez Nieves wrote: > Hi guys, I sent a dial to asterisk Which verson? > with a specific timeout, I want to increase it for some users if it is > approaching to the end, but when I send AbsoluteTimeout action Show us what command you are sending?

Re: [asterisk-users] How to know the IP of "manager show connected" in dialplan

On Wednesday 25 July 2018 at 19:53:47, Saint Michael wrote: > ​I need to launch a remote process at the machine that has the dialer. I > could hard-code the IP address in a global variable, but It would be much > more elegant if the dialplan would have a "manager" object where I could > read

Re: [asterisk-users] Disable asterisk ssl how to

On Wednesday 08 August 2018 at 22:30:52, Saint Michael wrote: > I am trying to install Asterisk 11 Why? > on debian 9 Have you tried installing https://packages.debian.org/jessie/asterisk from Debian 8 to see if it'll go onto Debian 9? Antony. -- Programming is a Dark Art, and it will

Re: [asterisk-users] Issues with install DAHDI

On Wednesday 15 August 2018 at 23:10:21, Dovid Bender wrote: > Hi, > > I am trying to install wanpipe Tell us how you are trying to install things. > with dahdi on a CentOS7 box and I am running in to a few issues. My setup. > > CentOS 7 > asterisk-15.5.0 > libpri-1.6.0 > dahdi linux and

Re: [asterisk-users] change dialing process on live call

On Sunday 19 August 2018 at 14:20:35, Khalil Khamlichi wrote: > Thanks for your response, this works but we cannot hardcode this in the > dialplan, we need this to be done from an external application connected > either via manager or stasis. Have you considered using Asterisk Realtime to store

Re: [asterisk-users] Merging 2 conference bridges

On Wednesday 22 August 2018 at 23:49:29, Ahmed Chohan wrote: > Hi, > > I would like to know how can I achieve merge 2 conference rooms in same > asterisk server. For example 10 users joined bridge A and max user limit is > set to 10. If more than 10 users try to join this bridge A, 11th user >

Re: [asterisk-users] STUN re-evalutation every 2 minutes ??

On Saturday 01 September 2018 at 22:12:50, sean darcy wrote: > 13.21.0 > > Every 2-3 minutes: Does it really vary, or is it more like "every 150 seconds"? > Sep 1 16:00:57] WARNING[150257]: res_stun_monitor.c:140 > stun_monitor_request: STUN poll got no response. Re-evaluating STUN > server

  1   2   3   4   >