Re: [asterisk-users] Alphabet character in destination number (CDR)

Hi Ikka, The last time I had this kind of problem the numbers in DB were altogether different and the reason for that was inappropriate columns data-types in DB. You can also print out some CDR(${variable}) AFAIK in the dialplan and verify that those are in original condition there or not.

Re: [asterisk-users] How to have callers not being billed when in waiting queue ? [SOLVED]

Hi, Just trying to figure out how is this solved ? by involving multiple telcos in the loop and asking them to not charge based on 200 OK/Answer!? As far as I know people have designed Queue/CallCenter platforms who upon entering a number in queue just state them their number in queue and approx

Re: [asterisk-users] Asterisk behind RTPproxy | On-Demand SDP engagement

That makes sense, so its not possible to have option 'tT' in DIAL() and have directmedia at the same time. Thanks Richard, Regards, Sammy On Thu, Feb 18, 2016 at 4:42 PM, Richard Mudgett <rmudg...@digium.com> wrote: > > > On Thu, Feb 18, 2016 at 3:05 PM, SamyGo <govoi..

[asterisk-users] Asterisk behind RTPproxy | On-Demand SDP engagement

Hi All, I've been wondering if I can instruct asterisk in the dialplan to engage the Media handling for a particular call or not. I've SIP users behind Kamailio & RTPProxy, and I can make use of sip.conf setting "directmediadeny|directmediapermit" to offload media from asterisk for peer-to-peer

Re: [asterisk-users] OpenSIPS, Asterisk and LocalAgents for Queues

Hi, I hope you already have fixed it . In case you didnt then here are my thoughts. Looking at the flow and keeping in mind that all devices are in same subnet you should never get one way audio issue since OpenSIP is not playing with SDP so Asterisk and PGW should just be able to have two way

Re: [asterisk-users] Adding Variable in all AMI events

Thanks Richard, Let me have it a go and get back on its effects since I've Asterisk 11.XX version.Hopefully it won't do much harm. Appreciate your time. Regards, Sammy. On Wed, Sep 9, 2015 at 1:48 PM, Richard Mudgett <rmudg...@digium.com> wrote: > > > On Wed, Sep 9, 2015 at 1

Re: [asterisk-users] Looking for Asterisk Consultants & Experts

Plot Twist: Shahid himself is a consultant and wants to find out where he can get some freelance project. :P Shahid: upwork.com, elance.com, freelancer.com, and google.com are your options. Regards, Sammy On Sep 2, 2015 10:10 AM, "Ganbold Tsagaankhuu" wrote: > Shahid, > > On

Re: [asterisk-users] How to dial extensions asynchronous-sequentially ?

Hi, Even you achieve that, what would be the objective? Do you want to just call the user and Hangup ? or Dial two users and connect them together ? Is this some sort of ring group implementation where users are dialled and first one to answer will get the call ?? Anyway here's one way of how I

Re: [asterisk-users] RES: How to dial extensions asynchronous-sequentially ?

message pointing to different sources, the caller's UAC should ideally pick only one of them, the latest one I believe. BR, Sammy On Mon, Jul 13, 2015 at 3:51 PM, Rodrigo Pimenta Carvalho pime...@inatel.br wrote: Hi SamyGo. Thank you for the replay. So, let me explain it better: I knew that I

Re: [asterisk-users] RES: RES: How to dial extensions asynchronous-sequentially ?

you think? Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979(Brasil) De: asterisk-users-boun...@lists.digium.com [ asterisk-users-boun...@lists.digium.com] em Nome de SamyGo [ govoi

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

Could this possibly mean that any person who has CF set should never be available as CF Destination. Simple db entry/check can have this done. On Tue, Jun 2, 2015 at 5:34 PM, d...@donkelly.biz wrote: *From:* asterisk-users-boun...@lists.digium.com [mailto:

Re: [asterisk-users] save the number of sip phone

Hi, exten = _0614.,n,MixMonitor(zap_g2_${EXTEN}_${UNIQUEID}_*${CALLER}* .wav|av(0}V(0)) This should append the caller number in your recorded file name. Ensure that you save the callerid in the variable before you're changing it to MY_CALLERID exten =

Re: [asterisk-users] AGI command

Hi, Please see my comments in line. Regards, Sammy On Wed, Jan 16, 2013 at 12:13 PM, Zohair Raza engineerzuhairr...@gmail.comwrote: On Wed, Jan 16, 2013 at 11:01 AM, Muhammad mohammad.ghaz...@gmail.comwrote: *Thanks Zohair! I wrote some php code to working with AGI, but it dosen't work.

Re: [asterisk-users] php programming for working with asterisk

Hi, Johan has referred you to a very good resource for doing that. Alternatively you can achieve this all by making an AGI in your preferred language. Thanks, Sammy On Mon, Jan 14, 2013 at 2:01 PM, Johan Wilfer li...@jttech.se wrote: It's written for asterisk 1.4, but I think that will get

Re: [asterisk-users] php programming for working with asterisk

Hi, If your caller is using softphone then you can create a simple dialplan which consults a DB and verifies that the dialled number is in the allowed-caller-list and if the result is OK just let the call dial through. Use Mixmonitor() application in your dialplan to record this call at some

Re: [asterisk-users] monitoring asteriks

Hi, I had a zabbix http://zabbix.org/wiki/Main_Page monitoring server with zabbix_agent installed on the asterisk server. Zabbix server requests the agent to execute an AMI script and pull information about the phone in the given argument to that function. That AMI script returned the 1 from the

Re: [asterisk-users] Counting calls in progress from AMI

Hi, Though I've moved this functionality to freeSWITCH but here is the logic. *AMI approach:* Use perl POE and when received Bridged event increment the counter, and similarly on Hangup decrement the counter. Since I see that there are some issues mentioned by OP so I assume using plain

Re: [asterisk-users] DTMF digits are coming through twice

Why am I feeling like I'm the only one here who is not able to see any pastebin link or attachments in this thread ! On Fri, Oct 12, 2012 at 6:18 PM, Vik Killa vipki...@gmail.com wrote: The trace is attached 3 emails back. --

Re: [asterisk-users] DTMF digits are coming through twice

Can you share your pcap trace ! On Thu, Oct 11, 2012 at 5:16 PM, Vik Killa vipki...@gmail.com wrote: Only callers calling from Earthlink internet connection On Wed, Oct 10, 2012 at 5:18 PM, Don Kelly d...@donkelly.biz wrote: Is this happening for all callers, or just iPhone callers?

Re: [asterisk-users] DTMF digits are coming through twice

Hi, Not exactly a solution, but I'm sure you must've taken pcap traces of a few such sample calls. See in their RTPs that you are receiving repeatedly same RTPs which will tell you that any DTMF packet is coming in twice by the source or not ! just one such simple pcap will help you identify at

Re: [asterisk-users] SIP Retransmitting REGISTER message

and not in Asterisk server. I just want to isolate things why I am not getting any response, or somewhere the response is getting lost! :( Regards, Gopal. On Wed, Sep 26, 2012 at 6:32 PM, SamyGo govoi...@gmail.com wrote: Hi, How are you connected to server ? How have you configured your

Re: [asterisk-users] SIP Retransmitting REGISTER message

Hi, How are you connected to server ? How have you configured your asterisk server to register to other side ? What about any NAT involved in your scenario ?Turn on sip debug and share your registrations. BR Sammy On Sep 26, 2012 5:54 PM, Danny Nicholas da...@debsinc.com wrote: Another

Re: [asterisk-users] AGI HANGUP PROBLEM

Hi, Just following this thread for few days, I've some basic troubleshooting questions for you. 1- What do you mean by calling from landline? How is your Landline /mobile reaching your asterisk box ? is there a Hardware card ! or a VoIP provider. 2- Enable SIP traces and keep an eye on the

Re: [asterisk-users] AGI HANGUP PROBLEM

Hi, So basically the FXO cards configurations need to be tweaked i.e hanguponpolarityinverse=yes etc. Since this is a Hangup request initiated by the SIP client, Asterisk then atleast it should close all the media streams and channel should get deleted. Keeping an eye on BYE : *CLI sip set debug

Re: [asterisk-users] Help with GotoIf Command

Hi, This is exactly why it is very important and wise to attach CLI verbose logs and other traces so the real issue gets resolved quicker than making all the people figure out the possibilities. BR Sammy On Mon, Sep 10, 2012 at 7:52 AM, David Klaverstyn da...@klaverstyn.com.auwrote: Hi

Re: [asterisk-users] Cascading macros in 1.8. Bug or feature ?

Hi, Though I cant say its a bug or feature, but the way you think it should work then try setting correct priority..i.e 1 in custom and then 2 in the other context. Somehow I'm having a feeling that this won't work. Also AFAIK macros are being replaced by Go-SUB thing. Macros cant be called in

Re: [asterisk-users] Asterisk on Rackspace, My SIP phone behind NAT

Oh, I see - check if your country blocks the SIP port 5060 ? try changing the default poert from 5060 to something else like and then try this. I think your ISP is blocking the SIP. On Fri, Aug 10, 2012 at 1:10 PM, A J Stiles asterisk_l...@earthshod.co.ukwrote: On Thursday 09 August 2012,

Re: [asterisk-users] Asterisk on Rackspace, My SIP phone behind NAT

Ok Good. It always feel good to add +1 to the list of resolved user-list issues. As I'm not a POWERFUL guy, I can't simply ask my ISP to unblock it. It takes a VPN or in near future WebRTC(in other words Knowledge) to become one powerful guy. With these technologies you don't need to care what

Re: [asterisk-users] Asterisk on Rackspace, My SIP phone behind NAT

Hi, Asterisk is quite good with resolving the NAT issues specially the kind of issue you are facing ,as I see it, shouldn't be a problem. A few steps you can troubleshoot this problem. 1-a:Are your SIP packets from PC/SoftPhone reaching the server !! On Asterisk CLI execute *CLIsip set debug on

Re: [asterisk-users] Block outbound calls based on IP address

Hi, How many Public IPs connect to you ? If they are less than 15 or 10 , I suggest you make sip.conf peers for them with host=Publicip and then decide if you want that to be blocked or rerouted to some other direction ! If that isn't doable then try extracting/parsing some IP using the

Re: [asterisk-users] Kamailio 3.3.x and Asterisk 10.7.0 Realtime Integration Tutorial

Thats a great tutorial with very good conceptual details like SIP messages flow. Thanks Daniel :) On Mon, Aug 6, 2012 at 6:48 PM, Daniel-Constantin Mierla mico...@gmail.comwrote: Hello, I released an update to my series of Kamailio and Asterisk Realtime Integration, using the latest stable

Re: [asterisk-users] Showing the name of the called number at the source IP Phone, how?

Hi, You need to set rpid on the calling phone settings, if that phone knows what to do with RPID. Then you need to set allowrpid=yes in the sip peer settings of A party and B party. I did that on CISCO 79X0 phones and it worked perfectly, Regards, Sammy On Tue, Aug 7, 2012 at 3:43 AM, bilal

Re: [asterisk-users] PBX, IVR and Conferencing Platforms From the Same Installation of Asterisk

Hi, If you have no custom requirements beside these then yes, Asterisk provides these all by default. You can use the same asterisk for all the functionality it offers in just one box. But thats not wise. Regards, Sammy On Mon, Jul 23, 2012 at 9:22 AM, Kannan vasdevelo...@gmail.com wrote: Hi

Re: [asterisk-users] FreePBX: using context other than the default context and the generation for the configuration

if you want to block them In freepbx there is a field in outbound route page to select callerid that the route applies to Cheers Duncan On 12/07/2012, at 4:52 PM, SamyGo govoi...@gmail.com wrote: See Route-Permissions module, It lets you restrict certain phones/extensions to follow a dial

Re: [asterisk-users] FreePBX: using context other than the default context and the generation for the configuration

See Route-Permissions module, It lets you restrict certain phones/extensions to follow a dial-plan pattern and dial out to the defined trunk etc meanwhile not breaking any other functionality or features of FPBX- though you can restrict the features from this too.

Re: [asterisk-users] sip set debug on always showing error

Hi, *CSeq: 245 OPTIONS * * * This is just SIP keep-alive. It has nothing to do with any Call-media degradation. If you are not getting clear voice check the codecs, network latency/delay/loss/jitter parameters. BR Sammy On Thu, Jul 5, 2012 at 2:34 PM, alok srivastava alok...@gmail.com wrote:

Re: [asterisk-users] FreePBX: using context other than the default context and the generation for the configuration

Hey, If you want to have all the dialplan features for your extensions and still need to implement some outbound calling restrictions then you need to look for some modules in freePBX. i've used that module exactly for this purpose and it works..can't remember its name. Just google it or lookup

Re: [asterisk-users] FreePBX: using context other than the default context and the generation for the configuration

umm Warren, yes including from-internal is the way of getting all the features,,,but in my experience the calls going out using the dialplan script we manually enter in our custome context don't get inserted into the FreePBX CDR and recording stuff !! On Fri, Jul 6, 2012 at 10:01 AM, Warren

Re: [asterisk-users] touch command not behaving for future calls in asterisk 1.4.41

Hi, Did you get anything working on it !! See the permission for the user running asterisk process and see if that user can touch files like that. Regards, Sammy On Thu, Jul 5, 2012 at 10:47 PM, virendra bhati virbh...@gmail.com wrote: Hi All, It's small issue but making a big problem for my

Re: [asterisk-users] port 5060 is blocked by ISP

:122.163.193.94:1801 --- - Really destroying SIP dialog 'qajoimvsihxpsff@alok-Inspiron-N5110' Method: REGISTER regards abhi On Mon, Jul 2, 2012 at 5:22 PM, SamyGo govoi...@gmail.com wrote: actually its a one-way audio issue due to NAT ! alok , please explain your

Re: [asterisk-users] Queue Member login from IAX trunk

Hi, exten = 105,n,Read(AGENT_SIP,agent-**newlocation) exten = 105,n,Set(AGENT_SIP=${DB(**agent_sip/${agent-newlocation}**)}) Above two lines are very suspicious: AGENT_SIP is a variable which is getting some DTMF from caller. agent-**newlocation is the message you want to be played while

Re: [asterisk-users] Queue Member login from IAX trunk

(AGENT_SIP=${DB(IAX2/intranet/agent_ip)}) so the agent enters the number from the phone he is connected. Then Asterisk adds IAX2/serverb to the number and saves it as agend phone number... Regards Jakob Am 04.07.2012 11:45, schrieb SamyGo: Hi, exten = 105,n,Read(AGENT_SIP,agent

Re: [asterisk-users] How to play different different hold music.

Hi, if possible for you put some header in SIP which mentions the music on hold flag on Server-A. The Dial the call to Server-B. On Server-B extract the value of that header and change the music on hold class based on the value. Regards, Sammy On Tue, Jul 3, 2012 at 5:30 PM, akhilesh chand

Re: [asterisk-users] How to play different different hold music.

Hi, The method Danny suggested is simple except I guess he swapped the priority and exten field. The idea is to dial a different extension on B server if you need to use some other MOH class. If you don't want to change the dialled extension you can always add a single digit prefix in Server-A

Re: [asterisk-users] Outbound Asterisk calls default directmedia specifications

I don't think you can set SIP properties in some variables anywhere in asterisk dialplan or call file. What you can do is change the directmedia options of the SIP or any other channel you're using. i.e if your call file has CHANNEL=SIP/12345@latestgateway Then change the properties of the

Re: [asterisk-users] urgent

whats the output when you do this on mysql *mysql show databases;* If there is no database defined then you definitely need to go through the installation steps and see if you've missed to create the A2billing Database. On Wed, Jun 20, 2012 at 5:59 PM, b.ti...@pinguin.ag wrote: did you put

Re: [asterisk-users] Asterisk 10.4.0 GotoIf to label problem when DUNDi active

Hi, You might have already tried but can you try reducing the label name and exclude the underscore in it ! Regards, Sammy On Wed, May 30, 2012 at 11:02 PM, Noah Engelberth n...@directlinkcomputers.com wrote: I have a hotdesking environment at my main office, and up until today, the GotoIf

Re: [asterisk-users] add new sip account in sip.conf with API Action UpdateConfig with php

Hi, 1- try putting absolute filepath in source and destination field. 2- verify that the permissions of the files you're changing. Regards, Sammy. On Mon, May 21, 2012 at 5:10 PM, virendra bhati virbh...@gmail.com wrote: Hi List, I am trying to add new SIP account in new file

Re: [asterisk-users] Asterisk and the media path

Hi, Can you check if there is any transcoding involved with these calls, or maybe some NAT handling needs to be done by asterisk so it's not stepping out of the media-path !? Regards, Sammy On Mon, May 21, 2012 at 5:03 PM, David Wessell da...@ringfree.biz wrote: I am attempting to get an

Re: [asterisk-users] Realtime peers and trunks coming from the same IP

Hello Ricardo, The reason why your asterisk refused the calls from phone registering on SIP proxy is that it only gets INVITE of the call from: a user that is defined BUT Not Registered within asterisk. The easy way of solving this is 1- Stop asterisk SIP realtime and let only the SIP proxy handle

Re: [asterisk-users] Why did it Hangup?

Hi Shahid, I am in favor of asterisk for what it is doing to your call. When you send an AMI event like the one you wrote it sends the A party invite right away w/o going into any context/extension. As soon as the A-party answers the call Asterisk manager connects/lands A-channel to the test

Re: [asterisk-users] Asterisk 8 and mixmonitor

I can't figure out if it's a known issue, or a new bug. Or a new feature !! Can you share the dialplan code where you are executing the mixmon application ! Regards, Sammy. On Wed, May 2, 2012 at 5:09 PM, ik ido...@gmail.com wrote: Hello, I have weird issue with Asterisk 8 lately.

Re: [asterisk-users] Music as ringtone

Hi, I don't think even putting a Progress() at the 1st priority will help you alot give it a try. The reason why I think this will not work that the clients/UACs have timers and those timers are working exactly the way they are supposed to and hence when a certain time expires and No-Answer/Ack

Re: [asterisk-users] No UDPTL ports remaining

Hi, Which version of asterisk is this !? I had the same situation in which asterisk just consumes UDP ports and don't release the ports on call hangup hence this error appears after a while. So I just upgraded asterisk version and everything worked better than expected. Regards, Sammy On Fri,

Re: [asterisk-users] No extension found ?

, i receive a call from ~20 ip .. I can't put a subnet ? best regards Le 23 avril 2012 07:57, SamyGo govoi...@gmail.com a écrit : Hi, No matching peer for '+331MYCLID' from '84.xx.xx.72:5060' This line is telling you everything. The peer you've declared isn't being matched

Re: [asterisk-users] No extension found ?

needs to define multiple peers for all incoming IPs inorder to goto the correct context and match the desired extension. On Tue, Apr 24, 2012 at 5:06 PM, Administrator TOOTAI ad...@tootai.netwrote: Le 24/04/2012 12:37, SamyGo a écrit : Thats not gonna work TOOTAi, that's just ACL thing you

Re: [asterisk-users] Strange problem on ougoing call

Hi, Lots of mixing and confusing stuff - Can you re-explain the topology you are trying to achieve with proper IP addresses and declared sip ext. names. When i call with the phone connected to I-User01, no problems, that's work but when i call with the second phone (use I-User02) i have a

Re: [asterisk-users] No extension found ?

Hi, No matching peer for '+331MYCLID' from '84.xx.xx.72:5060' This line is telling you everything. The peer you've declared isn't being matched for the incoming call and hence it tries to look in default context (I assume allowguest=yes in your sip.conf) Make sure that your peer is matched,

Re: [asterisk-users] HELP!! Caller ID unknown for all inbound call

I see extra/additional fields in the pasted configuration dahdi-channels.conf, try removing these. Then do a module reload chan_dahdi callerid= group= Regards, Sammy On Sun, Apr 22, 2012 at 12:49 PM, Satria Anamarta anam.satri...@gmail.comwrote: This is a very strange problem (at least

Re: [asterisk-users] HELP!! Caller ID unknown for all inbound call

? BR, Anam On 4/23/12, SamyGo govoi...@gmail.com wrote: I see extra/additional fields in the pasted configuration dahdi-channels.conf, try removing these. Then do a module reload chan_dahdi callerid= group= Regards, Sammy On Sun, Apr 22, 2012 at 12:49 PM, Satria Anamarta

Re: [asterisk-users] Video Conference in Asterisk1.4 (using asterisk gui)

app. On Tue, Apr 10, 2012 at 12:46 AM, Paul Belanger pabelan...@digium.comwrote: On 12-04-09 05:58 AM, SamyGo wrote: Hi, Actually asterisk don't provide video conference. In simple terms, the setting which zohair told just enables two end points to use video codecs and establish a one-one

Re: [asterisk-users] Video Conference in Asterisk1.4 (using asterisk gui)

Hi, Actually asterisk don't provide video conference. In simple terms, the setting which zohair told just enables two end points to use video codecs and establish a one-one video session for video capable phones. For making your asterisk do a video conferencing you may need to look into Vmukti

Re: [asterisk-users] Asterisk 1.8 and DeadAGI

exten = _X.,1,Hangup() its wrong causing priority conflict. NOOP is just for fun no benefit aside from printing something for you info on CLI. On Wed, Apr 4, 2012 at 6:35 PM, bilal ghayyad bilmar...@yahoo.com wrote: Dears; In asterisk 1.8, it is not more possible to use DeadAGI? Also,

Re: [asterisk-users] Asterisk 1.8 and DeadAGI

Thanks for the lesson, I appreciate that. On Wed, Apr 4, 2012 at 8:02 PM, Steve Edwards asterisk@sedwards.comwrote: On Wed, 4 Apr 2012, SamyGo wrote: NOOP is just for fun no benefit aside from printing something for you info on CLI. It's not fun, it's misleading. Especially

Re: [asterisk-users] All circuits are busy now on outgoing trunk call

configured a route on the fxo to send all incoming sip traffic to the fxo ports. I will try set the specific digits and see. On 3/21/12, SamyGo govoi...@gmail.com wrote: 404 NOT FOUND means that they were unable to find any destination/route/rule/prefix match corresponding to your

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

At first I was like *DIAL(SIP/92${NUM}@SIP_PROVIDER)* But Danny and Sam are right, the information is incomplete to give any answer at all !! BR/ Sammy On Fri, Mar 23, 2012 at 8:16 PM, Lutgring, Sam lutgr...@calhounisd.orgwrote: The short answer is yes you can. Now the longer answer is give

Re: [asterisk-users] All circuits are busy now on outgoing trunk call

404 NOT FOUND means that they were unable to find any destination/route/rule/prefix match corresponding to your dialled number. See your FXO gateway configuration Web-UI for outbound patterns OR verify that the FXO has its outbound line configured and working properly. On Wed, Mar 21, 2012 at

Re: [asterisk-users] Rate sheet normalization

could MS-Excel possibly be the easiest way to do that normalization ! just merge two rate sheets put some formulas in there and use it in your A2billing or XYZ tool ! On Thu, Mar 15, 2012 at 10:07 AM, Ast Coder asteriskcod...@gmail.comwrote: A2Billing doesn't do that. A2Billing in fact has a

Re: [asterisk-users] Rate sheet normalization

So, maybe a subscription service where a dialler system continuously tests routes with a list of 10 providers so that it's established which routes actually work and then allow that data to be downloaded for usage. I think that it may not be humanly possible and also not possible to have a

Re: [asterisk-users] Rate sheet normalization

. --Don Don Kelly PCF Corp People Come First 651 842-1000 651 842-1001 fax ** ** ** ** *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *SamyGo *Sent:* Thursday, March 15, 2012 11:45 PM *To:* Asterisk Users Mailing

Re: [asterisk-users] Rate sheet normalization

with passage of time. On Fri, Mar 16, 2012 at 10:06 AM, SamyGo govoi...@gmail.com wrote: Good Idea but that means all the members of the coop use the same vendors and it may not be suitable for servers having a bad network with a premium quality provider and thus mark it as bad, whereas others