Re: [asterisk-users] Configuring Opus Forward Error Correction in Asterisk 16 (FreePBX)?

/AST/Asterisk+19+Configuration_codec_opus https://www.asterisk.org/configuring-opus-encoder-asterisk/ https://www.asterisk.org/asterisk-opus-packet-loss-fec/ - Kevin On Wed, Jul 20, 2022 at 10:48 AM Brant Merryman wrote: > Hi. I am using Asterisk 16.27.0 in FreePBX 15.0.23.11. I installed

Re: [asterisk-users] Logging different verbosity levels

So this turned out more complicated than I originally thought! My expectation: Verbosity gets logged using an "at least" check against the current system's verbose level, which if passed subsequently gets checked against the logging channel's verbose level. Thus only verbose messages with a

Re: [asterisk-users] Setting up sipml5

On Fri, Sep 10, 2021 at 12:44 PM Jerry Geis wrote: > HI All, > > I am trying to get SIPml5 working with 18.6.0. > My http.conf file: > enabled=yes > bindaddr=myip > bindport=8088 > serverName=MyName > tlsenabled=true > tlsbindaddr=myip > tlscertfile=/etc/letsencrypt/live/mpname/fullchain.pem > >

Re: [asterisk-users] Asterisk Getting Crashed

d on your system so the backtrace doesn't have any extractable information. Please see the wiki [3] on how to get a useful backtrace. Before that though I recommend upgrading to the latest version of Asterisk [1]. Or if you're set on using a certified version [3]. The version you are on is quite old,

Re: [asterisk-users] error compiling current git

res_format_attr_g729.c -> res_format_attr_g729.o > > > Is this to be expected or should I make a bug report? > > When you pulled the lasted code this change would have forced a re-configure. If you haven't already try doing a full clean and rebuild, and see if you still have the error: $ mak

Re: [asterisk-users] pjsip startup errors when using "with-ssl" configure option

On Tue, Feb 25, 2020 at 4:02 PM Patrick Wakano wrote: > Hi Kevin! > Thanks very much for your reply! Much appreciated! > You're welcome! > So I just have a remaining question from this, if the with-ssl is not > mandatory to have the encryption support, what is i

Re: [asterisk-users] pjsip startup errors when using "with-ssl" configure option

e --with-ssl is > used? I could not find a clear explanation for this problem and how to fix > it > There appears to be a bug here. I configured, built, and ran with the same options mentioned (--with-ssl, etc...) and received similar pjsip module load errors

Re: [asterisk-users] [asterisk-app-dev] ARI Get Channel Variable

splay/AST/Asterisk+16+Function_CHANNEL -- Kevin Harwell Senior Software Developer Sangoma Technologies Check us out at: https://sangoma.com & https://asterisk.org ___ asterisk-app-dev mailing list asterisk-app-...@lists.digium.com http://lists.d

Re: [asterisk-users] PJSIP Setup Outbound SIP Trunk

; > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo

Re: [asterisk-users] Experiencing what I think are issues with the confbridge 'video_mode = follow_talker' and also the talk detection

nguage = en > > internal_sample_rate = 0 > > mixing_interval = 20 > > record_file_append = no > > max_members = 10 > > video_mode = follow_talker > > > > [4] > > type = user > > admin = no > > marked = no > > startmuted = no > > mus

Re: [asterisk-users] Does anyone know if there is a problem with the Chrome browser and asterisk cmp2k video

So you are probably seeing it work or not in Chrome vs Firefox due to browser, and codec support of such occurrences. -- Kevin Harwell Digium - A Sangoma Company | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: https://digium.com & https:

Re: [asterisk-users] Question on WebRTC configuration

; > > > Is the wiki web page mistaken or is this an actual http.conf setting that > is undocumented? > The page is mistaken. It should not be there. the 'tlscafile' option is not supported by the Asterisk http server. I've removed it from the wiki. Thanks for catching that! > > >

Re: [asterisk-users] SIPp scenario file for testing UAC Authentication with Asterisk ?

ded branchc and [1] but met no success yet > > Best regards > > [1] https://github.com/rkday/sipp-samples/blob/master/uac-auth.xml > > -- Kevin Harwell Digium - A Sangoma Company | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - U

[asterisk-users] pjsip trunk config question + DNS related error messages

Greetings, I am getting the following error (below) continually in my asterisk log, related to qualify_frequency I believe. I am trying to use sip trunking with the company flowroute. 3 questions if I may: 1) Is using qualify_frequency with a sip trunk a common or recommended practice? I

[asterisk-users] how to get "SMS" messages (http) into Asterisk "sip messages"

something like this. Outbound is the easy part. How are you handling inbound SMS->SIP ? Regards, Kevin Long -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community fo

[asterisk-users] pjsip trunking configuration issue

Greetings ! My goal is to get Twilio trunking working, and with TLS/SRTP. I see this concerning message in my log: [Feb 7 16:50:26] ERROR[20596] res_sorcery_config.c: Could not create an object of type 'endpoint' with id ’twilio' from configuration file ‘pjsip.conf’ Thus, ‘pjsip show

[asterisk-users] AUTO: Kevin Larsen is out of the office (returning Mon 01/08/2018)

I am out of the office from Thu 01/04/2018 until Mon 01/08/2018. I am out of the office and will have limited contact. For all emergencies/issues, please contact the helpdesk at helpd...@pioneerballoon.com or 316-688-8777. Note: This is an automated response to your message "[asterisk-users]

Re: [asterisk-users] Rewrite Outgoing Number

asterisk-users-boun...@lists.digium.com wrote on 12/14/2017 09:52:32 AM: > From: "basti" > To: asterisk-users@lists.digium.com > Date: 12/14/2017 09:52 AM > Subject: Re: [asterisk-users] Rewrite Outgoing Number > Sent by: asterisk-users-boun...@lists.digium.com > >

Re: [asterisk-users] Rewrite Outgoing Number

Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208 asterisk-users-boun...@lists.digium.com wrote on 12/14/2017 09:36:06 AM: > From: "basti" <mailingl...@unix-solution.de> > To: asterisk-users@lists.digium.com > Date: 12/14/2017 09:36 AM > Subject:

[asterisk-users] AUTO: Kevin Larsen is out of the office (returning 07/31/2017)

I am out of the office until 07/31/2017. I am out of the office and will have limited contact. For all emergencies/issues, please contact the helpdesk at helpd...@pioneerballoon.com or 316-688-8777. Note: This is an automated response to your message "[asterisk-users] [asterisk13] Multiple

[asterisk-users] RTP / NAT question with IPv6/IPv4 problem

Hello, All my asterisk systems use only IPv4 currently. I have one phone which is on T-Mobile network, and this network is only IPv6 now. The phone can register fine, because T-Mobile does NAT64 and it connects fine to my IPv4 asterisk server. But in the SDP for a call setup, this

Re: [asterisk-users] log incoming calls without answering

> I've already proposed your solution (is the most reasonable) but they > have more than 60 analogs lines (no faxes) and some of them terminate in > appliances like alarms, etc, so the solution must not touch in any way > the connection between the line and his termination: doing a analog to >

Re: [asterisk-users] log incoming calls without answering

> From: Fabio Moretti > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Date: 04/20/2017 03:26 PM > Subject: [asterisk-users] log incoming calls without answering > Sent by: asterisk-users-boun...@lists.digium.com > > Hi, >

Re: [asterisk-users] Crashes in jitterbuffer with framedata->timer_interval > 1000

lpful too if you have those. Which channel type (chan_sip, local channel, chan_pjsip) is involved, and how you are enabling the jitter buffer (dialplan function vs configuration) would be good to know as well. [1] https://issues.asterisk.org [2] https://wiki.asterisk.org/wiki/display/AST/Getting+a+Ba

[asterisk-users] packet loss stats - how does asterisk know about packets sent % lost ?

! Kevin Long -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org

[asterisk-users] 256 bit SRTP ciphers in Asterisk 14.x , only works for outbound call ?

on the phone, the call fails. Perhaps this is just not documented, or may not be implemented yet. Anyone have a thought? Thank you,. Kevin Long -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com

[asterisk-users] TLS certificate warnings in softphone, but not until after successful registration and call placed ?

is complete, it then connects to the IP instead of the hostname, and the mismatch occurs ? Any help appreciated, Thanks, -Kevin -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new

Re: [asterisk-users] Cisco IP 8841 asterisk integration

> True agree, problem is somehow the people purchased am > supporting to overcome that. Trying level best... around 20 > phones has been purchased Ah, yes, the "we purchased these without consulting you, but it is up to you to make them work" school of thought. It often goes with,

Re: [asterisk-users] PJSIP missing objects

nt and res_pjsip_multihomed was removed as the bulk of its code was moved into the res_pjsip core. -- Kevin Harwell Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digiu

[asterisk-users] Specify "name" for Resource in RLS

Is there are way to specify the display name of a resource in a resource list? I have setup a resource list in Asterisk 13 for 1234. All is working on the device, but I want to show "Joe User" instead of "1234". Any thoughts? --

Re: [asterisk-users] FAX CNG detected but no fax extension

> Hello, > I have a question regarding incoming fax to local file (on the > Asterisk server). > While the fax is received properly (I have the tiff file generated > as expected) I get the warning 'FAX CNG detected but no fax > extension' on the consol. > > If the fax is received ok then what

Re: [asterisk-users] Problem "re-parking" calls

> All; > I have a problem with regards to “re-parking” calls and I was > hoping someone could shed some light on the topic. Consider this scenario: > > (1) An inbound call comes in and the attendant answers it > (2) The attendant places the call on hold and the caller is sent to >

Re: [asterisk-users] Asterisk inside network. What phone works well?

> I have Asterisk running well inside our network. I did some > experiments exposing it to internet but had some issues: > 1. NAT issues (voice one way, etc). From what I understand double- > NAT users will always have something like this > 2. Immediately I see people trying to hack into. I did

[asterisk-users] AUTO: Kevin Larsen is out of the office (returning 09/06/2016)

I am out of the office until 09/06/2016. I am out of the office and will have limited contact. For all emergencies/issues, please contact the helpdesk at helpd...@pioneerballoon.com or 316-688-8777. Note: This is an automated response to your message "Re: [asterisk-users] Need ISDN call

Re: [asterisk-users] Getting better Caller ID

> Hello, > > We use Asterisk and as per book we use MAC addresses as user names. > So, when call coming in from outside (SIP trunk) - caller id is good. > > But when users calling each other on extensions - they see MAC > addresses. How would I make it so we see actual names instead of MAC >

[asterisk-users] PJSIP/Realtime RLS

I see that you can configure RLS in pjsip.conf, but does this work with realtime? The wiki refers to pjsip.conf for configuration, but since many of the other items can be in the the DB, I was wondering if RLS can as well. -- _

[asterisk-users] Asterisk 13 with LDAP ? (single sign on )

of other applications, and am curious if anyone has a working example or if this is even possible? Thank you, Kevin Long -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us

Re: [asterisk-users] Need stronger SRTP ciphers (256 bit)

suite); > On May 30, 2016, at 11:49 AM, Kevin Long <kevin.l...@haloprivacy.com> wrote: > > > > Hi folks, > > > At least several endpoints (soft phone and desk phones) are supporting > various 256 bit ciphers for SRTP these days. I *believe* libs

[asterisk-users] Need stronger SRTP ciphers (256 bit)

with the know-how be willing/able to submit a patch ? Thank you, Kevin Long -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs

Re: [asterisk-users] open source pbx free

> Anyone have any experience running an open source pbx and call > center solution?Need to start a call center of 10 users and i need help > > I have already installer a server with Ubuntu Server 14.04 , E1 installed > > Please advice me how to process from here > > Regards > >

Re: [asterisk-users] Recommendations for free virtual server tech and Asterisk?

Personally I am about to try asterisk on proxmox using containers since they run code "native". I've had timing issues on conference calls (stutter) with VMware esxi . Not sure about KVM I hope it's also better than esxi too. Sent from my iPhone > On Apr 6, 2016, at 9:13 AM, Markos Vakondios

Re: [asterisk-users] Is possible to use FXO Digium card like a Fax modem?

> There are also cheap USB fax modems that you can attach to an FXO > port and that works fine. All you have to do then is configure > asterisk to detect incoming faxes and route them to that port > (faxdetect=yes?). > > This worked great for me when I had all my incoming calls coming > over

[asterisk-users] Client TLS certificates for auth ?

a second factor of authentication besides the SIP secret , since in my current setup, despite using a TLS/SSL cert for the server, the server only verifies the client by the SIP secret. Regards, Kevin Long smime.p7s Description: S/MIME cryptographic signature

Re: [asterisk-users] what to do when a sip password includes a semicolon

asterisk-users-boun...@lists.digium.com wrote on 03/11/2016 01:43:47 PM: > From: Saint Michael > To: Asterisk Users Mailing List - Non-Commercial Discussion > , > Date: 03/11/2016 01:44 PM > Subject: [asterisk-users] what to do when a sip

Re: [asterisk-users] 2 devices same *actual* extension - can it be done

> Can someone tell me if this is possible? > > I currently have a VOIP phone registered on an Asterisk PBX at a > remote location (working fine). > I want to install an Asterisk PBX at the local location. I will be > porting the current POSTS lines to SIP trunking. > So now I want the remote

Re: [asterisk-users] conference call stuttering / clocking issue (?) - ESXi virtual environment

Thanks John, For anyone reading this using FreePBX - simply switching the default conference app from MeetMe to ConfBridge seems to be a drastic improvement, have not stress tested but running a conf now with no stutter on Confbrdige app. Cheers, Kevin Long > On Mar 9, 2016, at 12:17

[asterisk-users] conference call stuttering / clocking issue (?) - ESXi virtual environment

, and wondering if anyone has anything I could try to fix or mitigate the problem in ESXi environment . We have freepbx (asterisk 11 chan_sip) and test environments asterisk 13.7/8 pjsip . Thank you again, Kevin Long smime.p7s Description: S/MIME cryptographic signature

[asterisk-users] 2 devices same *actual* extension - can it be done

. Their provisioning system assumes that both devices will use the same SIP extension for auth however. Normally we would use separate extensions and a follow-me , but if there is any way to use the same extension, I need to figure it out. Thank you, Kevin Long smime.p7s Description: S/MIME

Re: [asterisk-users] PJSIP signaling question

Joseph <george.jos...@fairview5.com> wrote: > > > > On Fri, Mar 4, 2016 at 1:16 AM, Kevin Long <kevin.l...@haloprivacy.com> wrote: > Hi George the patch was from here , you wrote it I believe . I pulled > asterisk 13 from git, apply this patch which fixed RTP issue , bu

Re: [asterisk-users] PJSIP signaling question

rge Joseph <george.jos...@fairview5.com> > wrote: > > > >> On Thu, Mar 3, 2016 at 8:25 PM, Kevin Long <kevin.l...@haloprivacy.com> >> wrote: >> >> Thanks George I appreciate the info . Being able to see what codec is in >> use for call in p

Re: [asterisk-users] PJSIP signaling question

ot working and the internal IP being sent in the SDP from asterisk - I applied this patch to the codebase and recompiled I am seeing the TLS “new transport” issue again , I think. Regards, Kevin Long smime.p7s Description: S/MIME cryptographic

Re: [asterisk-users] RTP / NAT question ( pjsip )

again, Kevin Long smime.p7s Description: S/MIME cryptographic signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs

Re: [asterisk-users] RTP / NAT question ( pjsip )

Hi Joshua, This Asterisk 13 was pulled from git master branch just 2-3 days ago: GIT-13-d1495b . I used this very recent source code to overcome a pjsip problem (you can see my email list post from a few days ago) Thanks again smime.p7s Description: S/MIME cryptographic signature --

Re: [asterisk-users] RTP / NAT question ( pjsip )

Hi Joshua, Looking at the transmitted SIP packets from Asterisk, it looks like Asterisk is only sending it’s own internal IP (it is behind a NAT too, with proper port forwarding) . I did set in my transport the external_signaling_address and external_media_address , and I have now put

Re: [asterisk-users] RTP / NAT question ( pjsip )

Thank you for the response Joshua . I had rtp_symmetric=yes before I wrote the email, then I set it to no, restart asterisk, and tried to make the call from the remote endpoint again but still tcpdump is showing me the RTP packets are being sent from Asterisk to the private IP. tcpdump

[asterisk-users] RTP / NAT question ( pjsip )

t;sip:4...@dev1.domain.com> From: "Kevin"<sip:6...@dev1.domain.com>;tag=0af40611 Call-ID: MGE5OWFhMDY5OGFhYzM4ZDIxNjA5OGRjY2M5OWE3ZGY CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, REFER, INFO, NOTIFY, UPDATE, PRACK, MESSAGE, OPTIONS, SUBSCRIBE, OPTIONS Content-Type: applicati

Re: [asterisk-users] PJSIP signaling question

, firewall, or Asterisk/pjsip that is the culprit . Regards, Kevin Long smime.p7s Description: S/MIME cryptographic signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us

[asterisk-users] PJSIP signaling question

that asterisk is still trying to create *new* TLS outbound connections to my endpoints, which are failing. Thank you for your time Kevin - My simple pjsip config file: [transport-tls] type=transport protocol=tls bind=0.0.0.0:5061 local_net=10.50.55.0/24 external_media_address=x.x.x.x

Re: [asterisk-users] Passing Caller ID through Digium Gateway

> Hi All, > > I've setup a Digium G100 VoIP gateway to replace an internal PCI VoIP > card in our Asterisk PBX. When using the VoIP card the callerid entries > listed in sip.conf were displayed when calling someone over the PSTN. > Now, however, though the gateway it just displays the

[asterisk-users] Determining and setting TLS cipher ?

k you, Kevin Long output from “openssl ciphers” on my Asterisk box: ECDHE-RSA-AES256-GCM-SHA384 ECDHE-ECDSA-AES256-GCM-SHA384 ECDHE-RSA-AES256-SHA384 ECDHE-ECDSA-AES256-SHA384 ECDHE-RSA-AES256-SHA ECDHE-ECDSA-AES256-SHA DHE-DSS-AES256-GCM-SHA384 DHE-RSA-AES256-GCM-SHA384 DHE-RSA-AES256-SHA256 DH

[asterisk-users] NAT traversal for mobile app softphones - best strategy?

unclear as to whether I truly need 2 separate public IPs for the turn server to work, which I have seen mentioned in some of the documents. Thank you for your time. Regards, Kevin Long smime.p7s Description: S/MIM

Re: [asterisk-users] Queue logfile txt format in mySQL needed

> From: Thomas > To: asterisk-users@lists.digium.com, > Date: 01/21/2016 04:17 AM > Subject: [asterisk-users] Queue logfile txt format in mySQL needed > Sent by: asterisk-users-boun...@lists.digium.com > > Hello, > > Iam using queues and agents, thats OK. > > I have

Re: [asterisk-users] Statsd Dialplan Application

s did not go out with the latest release of 13.7.0. Actually the new StatsD Dialplan application currently resides in master only. A small change to the res_statsd api was made and got tagged with that issue number for some reason, thus making it look as if the StatsD application feature was added to 13

Re: [asterisk-users] Forwarding call if extension busy

asterisk-users-boun...@lists.digium.com wrote on 01/04/2016 08:55:40 AM: > My question: > > - two extensions: and > - an active call on > - incoming calls to should be forwarded to (call advice!) and > > I know how can I forward an incoming call to more than an

Re: [asterisk-users] Forwarding call if extension busy

> Kevin Larsen <kevin.lar...@pioneerballoon.com> schrieb: > > > I am not sure if I completely understand what you are trying to do, but it > > sounds like you want to query the DEVICE_STATE function. > > IT WORKS > > Thank you very much! > Glad I

[asterisk-users] How exactly does asterisk know what IP to send RTP traffic to?

assigned the client device. Does asterisk send RTP traffic to the IP which is in the IP headers of the SIP REGISTER , or can a client “specify” it’s truly reachable IP ? I hope this makes sense. Regards, Kevin Long

[asterisk-users] repeating TLS error in log file

Greetings, I use TLS and SRTP on all my extensions. I use openssl and distribute my root certificate to my endpoints. Most of the time my calls work just fine. Sometimes I receive a repeating error in my log files however, and I don’t know why this is happening. I’m wondering if this

[asterisk-users] AUTO: Kevin Larsen is out of the office (returning 10/24/2015)

I am out of the office until 10/24/2015. I am working in Mexico with limited availability. If the matter is urgent, please contact the Pioneer Helpdesk. Note: This is an automated response to your message "Re: [asterisk-users] Live Recording on the NAS?" sent on 10/15/2015 1:55:13 PM. This

Re: [asterisk-users] Asterisk => Mediant 1000 (AudioCodes) => PSTN (E1)

> > Does anyone have any information for me? > > > Welinghton. > > > > Citando Welinghton Magno Guimaraes : > Hello! > > I am setting up an Asterisk server with a Mediant 1000 (Audiocodes) > to make external links. Does anyone have any manual or

Re: [asterisk-users] Share agents state?

> Is it possible to share all agents state? if an agent is on the > phone on a queue on one of the Asterisk servers, other servers will > need to about it and therefore, will be able to operate adequately? > For instance, an agent is a member of two queues (app_queue > realtime) and those

Re: [asterisk-users] How to integrate Asterisk with XMPP

> > How to integrate Asterisk with XMPP ? > What you are asking for isn't a simple question to answer. What exactly do you want to accomplish by integrating XMPP? Shared states among multiple extensions? Passing messages between extensions? Depending on what you want and what infrastructure

Re: [asterisk-users] Receiving faxes with spandsp question

I’m trying to add fax functionality to my asterisk installation. Right now I’m focusing on receiving faxes. This is not explained in a book, but I assume that I can use same context, add “fax” extension and if someone calls to send fax - it will autodetect. Right? Per book, I made

Re: [asterisk-users] asterisk email to fax

Since the O.P. said he's using it for his home office, I think he'll be able to control user expectations :-) I provide tech support to my parents on all their computers. The amount of annoyance I have dealt with in the last few months over the fact that a recipe program and various card

Re: [asterisk-users] FXS Solutions for modems and other non jitter tolerant devices

The legal and medical communities still seem to prefer faxing, in the ( mistaken? ) belief that it is more secure. In fact the medical community is fearful of the legal beagles. These groups are really slow to change. At least in the USA The couple of times I have received medical faxes

Re: [asterisk-users] small homebrew pbx

I don't know this 'translates' to Italy, but this is what I would advise somebody in the US to consider, assuming you have a reliable Internet connection. 0) I hope you mean you want to run Asterisk at home instead of 'Asterisk at Home.' A@H was an ancient distribution from around

Re: [asterisk-users] Am I cracked?

Very strange... I ran the Asterisk CLI for other tasks, and suddenly I got this message: == Using SIP RTP CoS mark 5 -- Executing [000972592603325@default:1] Verbose(SIP/192.168. 20.120-002a, 2,PROXY Call from 0123456 to 000972592603325) innew stack == PROXY Call from 0123456

Re: [asterisk-users] Am I cracked?

OK, I set alwaysauthreject = yes and I discovered a allowguest, which I set to no, too. The PBX is behind a Firewall and I just allow UDP 5060 and 1-10100. Now I log the SIP-pakets coming from Internet, too... Hopefully I solved my problem... Make sure you have solved the problem. You

Re: [asterisk-users] Am I cracked?

Make sure you have solved the problem. You don't want to get hit with a phone bill for calls from your location to Israel. Basically, they are hoping that you are running the equivalent of a mail server open relay. They are trying to use you to dial out to another number. You don't

Re: [asterisk-users] RES: RES: How to invoke a binary file from the dial plan?

I love this question, simply because it allows me to talk about one of the neatest features I programmed into my system that barely anyone knows exists. Plus it lines up pretty much exactly with what you are trying to do. We have our gate control system tied into our Asterisk phone

Re: [asterisk-users] Forward loop protection...

Deciding on the mailbox to use is problematic! The dialed-party may be away for an extended period and wants voice mail handled by the forwarded-to party. And then you have the users who would work around this by sharing their voicemail passwords. Not quite as bad as sharing your computer

Re: [asterisk-users] RES: RES: How to invoke a binary file from the dial plan?

Hi Kevin. Thank you very much for the hint! It worked very well! Your example ' exten = 1234,1,System(echo This is a test / var/log/asterisk/test.txt) ' executes when the SIP client (my softphone Jitsi) sends a SIP INVITE to asterisk. So, the softphone tries to establish

Re: [asterisk-users] RES: RES: RES: How to invoke a binary file from the dial plan?

Hi Kevin. Thank you again for help me! In my case, in the final application for smartphones or in a softphone for PCs, there will be a button on the GUI and the user will have just to touch it, and the door or gate will open. I mean, during an ongoing call, the callee will see

Re: [asterisk-users] Forward loop protection...

Ia had a server overload today because someone did a call forward to their own extension. To do a call forward I write a key called CFWD with the extensión number and number to dial . The main script tests if the key/value exists and dials the number stored in the database. What

Re: [asterisk-users] How to invoke a binary file from the dial plan?

Hi everyone. I'm new with Asterisk and I have to create a dial plan that will invoke a binary code. That is, asterisk will execute a program in the same machine. How to do it? Let me explain what I have to do: In the project that I am currently working, there is smartphones, SIP

Re: [asterisk-users] RES: How to invoke a binary file from the dial plan?

Ok. Thanks for the hint. But, what exactly is a System() dialplan application? Is it a kind of command that i can call in dial plan? I will look for System() related to dial plans. From the Asterisk CLI type: core show application System It will print out the syntax for the command. One

Re: [asterisk-users] Forward loop protection...

The loop checking is a bit more challenging than that. If Bob forwards to Fred and Fred forwards to Sue, all is well when Bob and Fred head out for a beer. A little later, we’re in deep doo-do0 when Sue forwards to Bob. Could this possibly mean that any person who has CF set should never

Re: [asterisk-users] Signaling incoming call

Hi Kevin! Thanks! It works! I can set the name of the line with CALLERID(name) and see the caller number, too. And, it the number is in the address book, I see the name, too. Perfect! Glad it worked for you. I usually leave the number untouched, but will manipulate the name to suite

Re: [asterisk-users] Signaling incoming call

Hi Steve! Thank you very much! It seems to run! I wrote that: exten = _0049351333,n,Set(__ALERT_INFO=Bellcore-r3) exten = _0049351333,n,SIPAddHeader(Alert-Info: http://www.notused.com \;info=alert-external\;x-line-id=0) and the phone rings with another melody. Very curious

Re: [asterisk-users] Peer is UNREACHABLE

What kind of phone are we talking about, both yours that works and your wife's that does not? Right! Can you ping the unreachable phone and does it respond to a ping? I can ping both phones from the VM Many phones will have a network test function built in to them to help you

Re: [asterisk-users] Peer is UNREACHABLE

I have a problem and I hope someone can help me... I configured an Asterisk on a VM to serve more accounts and act as a proxy to other SIP-providers. The first account running on my phone works without any problem. A second account, running on the phone of my wife, is always UNREACHABLE.

Re: [asterisk-users] Peer is UNREACHABLE

Darryl Moore dar...@moores.ca schrieb: I'd start by turning on sip debugging in asterisk sip set debug ip [your_phone_ip] Really destroying SIP dialog '490d1996593c8e11217828b71aae5c4d@172. 16.34.133' Method: OPTIONS Reliably Transmitting (no NAT) to 192.168.200.11:5060: OPTIONS

Re: [asterisk-users] Peer is UNREACHABLE

No, I'm not sure. And no, I can't make any call, right now... At least, not connected to my Asterisk... If I connect it to the other VM with AsteriskNOW I can call my Twinkle, but NOT my phone connected on my Asterisk, using the proxy. I can see that in the log: [May 28 22:49:51]

Re: [asterisk-users] Asterisk as Proxy and more device for a number

I'm very new in Asterisk and VoIP, and of course I have a problem... :) Well, my problem is, that Deutsche Telekom wants me to change my ISDN to VoIP... :( I must do that, since I have no alternative. Well, I have now two VoIP-phones (Thomson ST2022 and KE1020A). I can configure my two

Re: [asterisk-users] Asterisk as Proxy and more device for a number

Maybe I got it... I installed an asterisk on a VM with Ubuntu 10.04 and I got it connecting to another Test-VM with AsteriskNOW and with an italian VoIP-provider. The very difficult was to understand, that my phone just can manage ONE profile at time, so I had to configure Asterisk to

Re: [asterisk-users] Phone provisioning template Snoms

I am looking for a phone provisioning template for Snom phones, Yealinks and Polycoms. I am always doing deployments of many phones and usually configure each phone one by one for each installation. Any help will be highly appreciated There’s some excellent documentation about

Re: [asterisk-users] Multicast to polycom from asterisk

I am using 11.17.0 - and MulticastRTP. Doesnt seem to work with polycom phones as other devices receive my multicast just fine. Is there something special to do to get multicast working with polycom phones? (other than enable multicast on the actual phone). Didn't see if anyone had

Re: [asterisk-users] Multicast to polycom from asterisk

I hesitate to promote the name here since this is non-commercial discussion... but Polycom... Polycom phones... If mentioning Polycom is OK, I think mentioning a possible commercial solution is OK. In that case, the product in question is the Algo 8180 SIP Audio Alerter. I will

Re: [asterisk-users] Determining if a queue member is paused in Dialplan logic. [1.8]

asterisk-users-boun...@lists.digium.com wrote on 03/25/2015 01:38:26 PM: I'm looking at enabling autopause on one of my queues where my queue members are bad about leaving their desks without pausing. The problem I see is that when the queue pauses an Member it doesn't jump into the dialplan

Re: [asterisk-users] [OT] switches

so how does a client pc find the server if there's no NAT? by IP address?? That makes no sense, to me, if the switch isn't assigning addresses. Switches have a MAC table that keeps track of which MAC addresses are on which ports. That's how they decide where to route packets.

Re: [asterisk-users] res_pjsip endpoint config object's 'identify_by' option needs new value uri.

/display/AST/Configuring+res_pjsip [2] https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Configuration_res_pjsip_endpoint_identifier_ip [3] https://wiki.asterisk.org/wiki/display/AST/Asterisk+PJSIP+Troubleshooting+Guide Hope that helps, -- Kevin Harwell Digium, Inc. | Software Developer 445

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