similar to: cdr table's "dst" column

Gentleman, Moderators, i don't know if this topic if OFF-Topic, if yes, please tell me. I had some difficult looking for a Asterisk software that provide me some functions (For exemple: CDR, Queue control, MeetMe Control) all-in-one. So i decided to develop than. In a few weeks i'll deploy a Beta version of this software and i'd like to know if is somebody available to try this

Hi Helviom I am interested to evaluate your product. What asterisk version you build this product around? -- regards, abdul basit | p: +92 32 1416 4196 | o: +92 30 0841 1445 On Tue, Jun 23, 2015 at 7:34 PM, Tech Support <asterisk at voipbusiness.us> wrote: > Please keep the ?me to? emails off the list. > > Regards; > > JV > > > > *From:*

Hello, ? I am interested, too. ? Att, Welinghton Citando Mitul Limbani <mitul at enterux.in>: > Hey Helvio, > > Would like to check it out as well. > > Do email me, > > Mitul > On 22-Jun-2015 9:05 AM, "Helvio Junior" <helvio.listas at gmail.com> wrote: > >> Gentleman, >> >> Moderators, i don't know if this topic

Hi, Try as a first step a tcpdump capture to verify if the softphone is actually sending the register message to the server. For me it seems like the softphone is not able to reach the server ! Best Regards, On Fri, Apr 24, 2015 at 10:55 AM, Helvio Junior <helvio.listas at gmail.com> wrote: > Hi Akhilesh, > > SIP protocol use port 5060 (default) and many other ports to stablish

Hi Guenther, Thanks for ur reply I have concern from long time I'm not able to login through softphone with AWS Cloud.Please let me know is there any document or guide line for the same. Regards Akhilesh On Fri, Apr 24, 2015 at 1:26 PM, Guenther Boelter <gboelter at gmail.com> wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > On 04/24/2015 03:34 PM, akhilesh

Use the command bellow to check if is Asterisk opening the port. netstat -nap | grep 4569 You need to see something like this output, otherwise your asterisk is not opening the port. udp 0 0 0.0.0.0:4569 0.0.0.0:* 10244/asterisk Att, H?lvio Junior dCAA - Digium Certified Asterisk Administrator SafeId - Gest?o de identidades e Acessos +55 41 | 9

Hi Helvio, Could you tell me what is process to setup an environment for IAX. Regards Akhilesh On Fri, Apr 24, 2015 at 4:25 PM, Helvio Junior <helvio.listas at gmail.com> wrote: > Hi Akhilesh, > > SIP protocol use port 5060 (default) and many other ports to stablish > calls. You need to check if there is AWS firewall rule that allow your > communication from your client

Hi, We are launching a new product to help-us to reduce mobile call costs using Asterisk. More informations you can see at http://asteriskdialer.com.br/en -- Att, H?lvio Junior SafeId - Gest?o de identidades e Acessos +55 41 | 9893-2694, single-sign-on.com.br helvio.junior at safetrend.com.br

1.8 or higher. Att, H?lvio Junior SafeId - Gest?o de identidades e Acessos +55 41 | 9893-2694, single-sign-on.com.br helvio.junior at safetrend.com.br On 29/06/2015 14:43, Abdul Basit wrote: > Hi Helviom > > I am interested to evaluate your product. > > What asterisk version you build this product around? > > -- > regards, > > abdul basit | p: +92 32 1416 4196 | o:

You can use tcpdump in your server to verify if it is receiving the packets. tcpdump -ni any port 4569 So you have more than one ip in the server? On 5 Jun 2017 9:13 pm, <thelma at sys-concept.com> wrote: > No, I don't think it is IP table issue, I've not upgraded dd-wrt for a > while and it was zoiper was working OK with my previous version of > asterisk. > >

Please keep the ?me to? emails off the list. Regards; JV From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Magno Guimar?es Sent: Monday, June 22, 2015 3:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Product CDR/Queue/Meetme Hello, I am interested, too. Att,

Hey Helvio, Would like to check it out as well. Do email me, Mitul On 22-Jun-2015 9:05 AM, "Helvio Junior" <helvio.listas at gmail.com> wrote: > Gentleman, > > Moderators, i don't know if this topic if OFF-Topic, if yes, please tell > me. > > I had some difficult looking for a Asterisk software that provide me some > functions (For exemple: CDR, Queue

Hi Akhilesh, SIP protocol use port 5060 (default) and many other ports to stablish calls. You need to check if there is AWS firewall rule that allow your communication from your client external IP and your AWS host. Also, think in use IAX intead of SIP, because SIP protocol has many trouble when used with NAT, also IAX protocol use only one port (4569) to everything. When i need allow

On Mon, 18 Jan 2016, Ethy H. Brito wrote: >> how to flush user input before READ()? How about a read() to a dummy variable with a 1 second timeout to consume the octothorpe and password? -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST

Hi I had a bad experience upgrading Ubuntu a few months ago. Today I made a "dd" copy to another harddisk and tried to dist-upgrade. I get "Illegal instruction (core dumped)" running "service asterisk debug" at random places. Any help will be appreciated to spot this problem. I think it is worth to mention that the dadhi hardware is not present at the copied

Hi All We've been reading this in the CLI a lot lately: Received SSL traffic on RTP instance '0x7fe7481faad8' without an SSL session How can we find details about this particular RTP instance? "rtp set debug" needs an IP which is precisely what I want to know (and I don't)! Cheers Ethy

Hi all I am using a pass trhu router and I need to QoS some clients output by its IP address. The problem is that QoS is due after NATing. Is there some clever way of doing this besides MARKing every packet with some IP hashing in POSTROUTING NAT table? Regards Ethy

Hi all These questions were asked back in 2009 at lists.digium.com and got unanswered: - Has someone been successful in using IMPORT on a Local channel ? - Is there a known limitation in doing so ? I run into the same problem. ${IMPORT(Local/1234 at example-abcd;2,CALLERID(dstchannel))} returns nothing. But I can read the dstchannel for it into the CDR. Asterisk is 11.7.0~dfsg-1ubuntu1

I visited yesican.chsoft.biz and the author proposes a way to match packets by less than some size . Here is the thing: match u16 0x0000 0xffb0 at 2 With this match he says that packet with less than 80 bytes will match the rule. Well, 0xffb0 translates to 1111 1111 1011 0000 (which is -80 BTW). So, if I am correct any packet with bits 4 and/or 5 set (together with any of the 4

Hi All After a Dial() I get: WARNING[7964][C-000075a8]: app_dial.c:2437 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent) if the subscriber is not registered. Is there a way from dialplan to know, *before* Dial(), if a destination Subscriber is a) not registered or b) busy ? I need to redirect a call to some other Subscriber if (s)he is not there