Re: [asterisk-users] tcpbind and source IP address
Hi, I recommend you to install from sources, especially because the latest Asterisk 13 has several bugfixes for pjsip. To my knowledge, nobody proposes ppa or Debian backports for Asterisk. Wazo has Debian packages, but it's only for Debian Jessie and with extra patches for Wazo. -- Ludovic Gasc (GMLudo) Lead Developer Architect at ALLOcloud https://be.linkedin.com/in/ludovicgasc 2017-04-16 21:36 GMT+02:00 Kseniya Blashchuk: > Hi! > > Unfortunately pjsip is broken in Ubuntu Asterisk installed from repo. Yes > I also thought to try with pjsip, just to know if it's also affected. I'll > try to make a test next days. > > On Sun, Apr 16, 2017, 8:18 PM Ludovic Gasc wrote: > >> Hi Kseniya, >> >> You might test with chan_pjsip: We have less production experience with >> chan_pjsip than chan_sip, however, for now, we are more and more confident >> in this new stack while we're digging in documentation and we're testing on >> production. >> >> However, I've no idea if you'll have the same issue with pjsip, but more >> chances of support on the issues tracker of Asterisk to have help. >> >> Regards. >> >> >> -- >> Ludovic Gasc (GMLudo) >> Lead Developer Architect at ALLOcloud >> https://be.linkedin.com/in/ludovicgasc >> >> 2017-03-13 14:41 GMT+01:00 Kseniya Blashchuk : >> >>> Ok, thank you for the assistance! >>> >>> пн, 13 мар. 2017 г. в 16:38, Joshua Colp : >>> On Mon, Mar 13, 2017, at 10:32 AM, Kseniya Blashchuk wrote: > Tested with latest Asterisk 14.3.0 on Ubuntu 16 kernel 4.4.0-66-generic > and > Centos 7 kernel 3.10.0-514.10.2.el7.x86_64. Absolutely the same behavior. > Joshua, maybe you can advice what can be done further? You can file an issue but chan_sip is a community supported module, so there is no guarantee of when it would be looked at and resolved. Ultimately though someone has to spend the time to replicate what is going on, look into the code, and understand what is going on. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users >>> >>> -- >>> _ >>> -- 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/wiki/display/AST/Getting+Started >>> >>> asterisk-users mailing list >>> To UNSUBSCRIBE or update options visit: >>>http://lists.digium.com/mailman/listinfo/asterisk-users >>> >> >> -- >> _ >> -- 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/wiki/display/AST/Getting+Started >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users > > > -- > _ > -- 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/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- 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/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to build with cdr_adaptive_odbc ?
Thank you, but unixodbc and odbcinst are installed... end even unixodbc-dev But I get the same need for "generic odbc(E)". On 17/04/2017 10:48, Marcelo Terres wrote: You need unixodbc and odbcinst packages too, to configure the odbc. []s Marcelo H. TerresIM: mhter...@jabber.mundoopensource.com.br https://www.mundoopensource.com.br https://twitter.com/mhterres https://linkedin.com/in/marceloterres On 13 April 2017 at 19:41, Pierre Couderc wrote: I use debian stretch and I have installed unixodbc-dev but I have a dependency on genreric_odbc in make menuselect What am I missing ? Is there an howto ? Thanks PX -- _ -- 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/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voicemail asking for login
We have a template for extensions and voicmail. They look like this: exten => %ACCOUNT%,1,Verbose(0,Entering extension %ACCOUNT%) same => n(DialDesk),Verbose(0,${CALLERID(all)} Calling ${EXTEN}) same => n,Dial(SIP/%ACCOUNT%,30) same => n(VoiceMail),Set(CDR(userfield)=VoiceMail) same => n,Verbose(0,${CALLERID(all)} going into voice mail for %ACCOUNT%) same => n,Set(_ACCOUNT=%ACCOUNT%) same => n,VoiceMail(%ACCOUNT%@VoiceMail,u) same => n,Hangup() And for voicemail.conf: %ACCOUNT% => %VM_PASSWORD%,%NAME%,%log...@vex.net Here is the sip.conf template: [%ACCOUNT%](client-phone) secret=%PASSWORD% callerid=%NAME% <%CLID%> mailbox=%ACCOUNT%@VoiceMail context=%CONTEXT% Every user gets set up using these templates so I know that everyone is identical other than the '%' variables above. I have looked and I don't see any significant differences. The ACCOUNTs are strings with most having digits appended. Obviously NAME, PASSWORD and LOGIN are different but not in kind. My issue is with users picking up their VM from an external phone. They call themselves and press '*' during the playback message. Normally they are asked for their password and then get dropped into the proper menu. One user (that we know of so far) has a different experience. In that case they are asked for a mailbox number first. I can't seem to find any significant difference in their configuration to account for that. Every other user that we have tested works as expected. Some of them have extension that are all letters, some have trailing digits. Some have associated cell phones and some don't. I have tried searching for this issue but nothing seems to apply. Most discussions are about "*97" vs. "*98". Can anyone suggest another field of enquiry? TIA. -- D'Arcy J.M. Cain Vybe Networks Inc. http://www.VybeNetworks.com/ IM:da...@vex.net VoIP: sip:da...@vybenetworks.com -- _ -- 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/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PBX selection
On 2017-04-17 12:41 PM, Victor Villarreal wrote: * Asterisk is build to work on Linux. So your team needs some skills like setting up a basic Linux server (Debian, Centos, etc), donwload software from Internet, compile and install software manually. It may be that the developers mostly use Linux but Unix (i.e. BSD) works perfectly fine as well. I run it on NetBSD and it is rock solid. As for Asterisk vs. FreeSwitch, as a data point I started out with the latter but converted to Asterisk before going live. FS really was more Linux oriented and I had many problems getting it working on NetBSD. I also find the support here better. If you run NetBSD (and probably any other BSD) you can install from their package system. I compile from source using the package but you can also install a pre-compiled version just as easily. Cheers. -- D'Arcy J.M. Cain Vybe Networks Inc. http://www.VybeNetworks.com/ IM:da...@vex.net VoIP: sip:da...@vybenetworks.com -- _ -- 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/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PBX selection
Hi Speed Boy. I agree with Emiliano Vazquez too. Additionally, you and your team must think others points before choose Asterisk: * Asterisk is build to work on Linux. So your team needs some skills like setting up a basic Linux server (Debian, Centos, etc), donwload software from Internet, compile and install software manually. * Your team must know how to configure Linux networking. And solve NAT issue if apply. Basic network protocols like UDP, SIP and SDP/RDP are welcome. * If Asterisk needs interact with external world via VOIP provider, then you must know how to configure SIP or IAX2 trunks. If you have analog (like FXO) or digitals lines (like ISDN or similar), then you need ti know how to install and configure hardware on the Linux server like telephony cards (PCI-e or PCI) or configure VOIP gateways. * Security: How to install and configure a basic firewall (using iptables), o Fail2Ban. And best practices in Asterisk about this topics. Cheers El 17 abr. 2017 13:03, "Emiliano Vazquez"escribió: > I prefer Asterisk for my projects. > > On Mon, Apr 17, 2017 at 11:57 AM, Speed Boy > wrote: > >> Hi all, I'm new to VoIP, now we have a project that needs a >> PBX with client APPs. >> In our team we have argument for choosing PBX. By so far, we >> have following candidates: >> >> A: Open source >> >> 1) Asterisk PBX (http://www.asterisk.org) (with longest >> history that almost every one knows it, now the last version using the >> PJSIP stack) >> 2) FreeSwitch (http://www.freeswitch.org) (A lot people >> recommended it to us) >> >> >> B: Commercial >> >> 1) Vodia PBX (http://www.vodia.com). It comes from SNOM, now >> acquired by a HongKong company now >> 2) PortSIP PBX (http://www.portsip.com/portsip-pbx). It >> also includes VoIP SDK, WebRTC and offer rebranding app for free. >> >> My boss prefers the Open Source PBX since they are free, but >> our CTO prefers the commercial editions, according to whom >> the business PBX has better support, and the performance is >> good, and easy to use - considering our team all are new to VoIP/PBX. >> > > Hire a team with knowledge about VOIP, without your prefer if you use > Asterisk or whatever you want > You will win a brand new full responsibility with VOIP. The learning > process is long and hard. You will find a lot of problems like NAT, > intrusions. Consider learn before you pain this. > > > >> >> We have did some searching of Asterisk, here are my questions: >> >> 1. Does the last Asterisk using PJSIP stack ? >> > > Yes. > > >> 2. Does there has the comparison of PJSIP and reSIProcate, sofia(using by >> FreeSwicth) ? >> > did you google about this? > > > > >> 3. Is it easy to compile and setup Asterisk? >> > You need some skills but today is really simple. > > > >> 4. Which Asterisk version is recommended? And does Asterisk support >> Windows ? >> >> The latest stable release. > > > > >> Thanks in advance . >> >> Best regards. > > >> > > -- > _ > -- 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/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- 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/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PBX selection
I prefer Asterisk for my projects. On Mon, Apr 17, 2017 at 11:57 AM, Speed Boywrote: > Hi all, I'm new to VoIP, now we have a project that needs a > PBX with client APPs. > In our team we have argument for choosing PBX. By so far, we > have following candidates: > > A: Open source > > 1) Asterisk PBX (http://www.asterisk.org) (with longest > history that almost every one knows it, now the last version using the > PJSIP stack) > 2) FreeSwitch (http://www.freeswitch.org) (A lot people > recommended it to us) > > > B: Commercial > > 1) Vodia PBX (http://www.vodia.com). It comes from SNOM, now > acquired by a HongKong company now > 2) PortSIP PBX (http://www.portsip.com/portsip-pbx). It > also includes VoIP SDK, WebRTC and offer rebranding app for free. > > My boss prefers the Open Source PBX since they are free, but > our CTO prefers the commercial editions, according to whom > the business PBX has better support, and the performance is > good, and easy to use - considering our team all are new to VoIP/PBX. > Hire a team with knowledge about VOIP, without your prefer if you use Asterisk or whatever you want You will win a brand new full responsibility with VOIP. The learning process is long and hard. You will find a lot of problems like NAT, intrusions. Consider learn before you pain this. > > We have did some searching of Asterisk, here are my questions: > > 1. Does the last Asterisk using PJSIP stack ? > Yes. > 2. Does there has the comparison of PJSIP and reSIProcate, sofia(using by > FreeSwicth) ? > did you google about this? > 3. Is it easy to compile and setup Asterisk? > You need some skills but today is really simple. > 4. Which Asterisk version is recommended? And does Asterisk support > Windows ? > > The latest stable release. > Thanks in advance . > > Best regards. > -- _ -- 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/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PBX selection
Hi all, I'm new to VoIP, now we have a project that needs a PBX with client APPs. In our team we have argument for choosing PBX. By so far, we have following candidates: A: Open source 1) Asterisk PBX (http://www.asterisk.org) (with longest history that almost every one knows it, now the last version using the PJSIP stack) 2) FreeSwitch (http://www.freeswitch.org) (A lot people recommended it to us) B: Commercial 1) Vodia PBX (http://www.vodia.com). It comes from SNOM, now acquired by a HongKong company now 2) PortSIP PBX (http://www.portsip.com/portsip-pbx). It also includes VoIP SDK, WebRTC and offer rebranding app for free. My boss prefers the Open Source PBX since they are free, but our CTO prefers the commercial editions, according to whom the business PBX has better support, and the performance is good, and easy to use - considering our team all are new to VoIP/PBX. We have did some searching of Asterisk, here are my questions: 1. Does the last Asterisk using PJSIP stack ? 2. Does there has the comparison of PJSIP and reSIProcate, sofia(using by FreeSwicth) ? 3. Is it easy to compile and setup Asterisk? 4. Which Asterisk version is recommended? And does Asterisk support Windows ? Thanks in advance . -- _ -- 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/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to build with cdr_adaptive_odbc ?
You need unixodbc and odbcinst packages too, to configure the odbc. []s Marcelo H. TerresIM: mhter...@jabber.mundoopensource.com.br https://www.mundoopensource.com.br https://twitter.com/mhterres https://linkedin.com/in/marceloterres On 13 April 2017 at 19:41, Pierre Couderc wrote: > I use debian stretch and I have installed unixodbc-dev > > but I have a dependency on genreric_odbc in make menuselect > > What am I missing ? Is there an howto ? > > Thanks > PX > > -- > _ > -- 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/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users