Re: [asterisk-users] A good price for FXS 48 ports?
Xorcom has good produtcs for that:For 48 FXO you can combine Astribank-32 with Astribank-16 and they connect to the Asterisk server via a normal USB-2 cable.Take a look at:http://www.xorcom.com/products.html ThierryOn 8/10/06, Javier Matos Odut <[EMAIL PROTECTED]> wrote: Hello, I was looking for some FXS gateway with 48 ports. I find some products in different webs but prices are between 3.000$ and 4.000$. Someone that have search about that kind of hardware find some website with great prices?. Many thanks. ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Integrating Asterisk with a Panasonic D500 using MFC/R2
About the ANI problem, in Brazil I use the following parameter for protocolvariant.protocolvariant=br,20,16,16I have the following configuration:Astetrisk E1 - PBXE1Telco But Steve, after changing versions it really started to work without any modification in the .conf files. It was simple as that: before nothing worked and I had plenty of error messages from the MFC/R2 on the Asterisk log and after installing the different version it started to work. So I considered not to troubleshoot the problem because it had been resolved. As both files has different sizes, surely there is a difference between them. Thierry >> Hi Carlos,>> I had the same problem and spent a lot of time studying the MFC/R2 > protocol but the problem is in the libmfcr2 package version!!>> Try using the packages in:>> http://www.soft-switch.org/downloads/unicall/unicall-0.0.3pre7>> And not in pre9.>> Both "pre7" and "pre9" have libmfcr2-0.0.3.tar.gz> < http://www.soft-switch.org/downloads/unicall/unicall-0.0.3pre7/libmfcr2-0.0.3.tar.gz>> with the same name.. but in different sizes. They are different.pre9 works just fine. His problem is something completely different. His configuration does not match the requirements of the PBX.Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Jitterbuffer on SIP
Thank You Patrick,After some minor problems in some file paths I had success compiling.The only problem was the codec_g726 witch does an illegal call and Asterisk doesn´t come up. But I only use g729 or g723 so I´ve deleted it from the modules directory and asterisk came up. I´m going to test it now. Thanks again!!On Tue, 2006-08-08 at 10:45 -0300, Thierry Querette wrote: > Hi,>> Is that a way to patch a running asterisk 1.2.9.1 instalation with the> experimental SIP Jitterbuffer support ?Yes, see http://www.asterisk-backports.orghttp://asterisk-backports.org/downloads/ast_jb-1.2.9.1+rtp-keep-jb+fax +g726.patchThe jb seems to work fine on my setup (with low usage).Regards,Patrick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Integrating Asterisk with a Panasonic D500 using MFC/R2
Hi Carlos,I had the same problem and spent a lot of time studying the MFC/R2 protocol but the problem is in the libmfcr2 package version!!Try using the packages in: http://www.soft-switch.org/downloads/unicall/unicall-0.0.3pre7And not in pre9.Both "pre7" and "pre9" have libmfcr2-0.0.3.tar.gz with the same name.. but in different sizes. They are different.After that everything started do work !!Good Luck I am trying to get my Asterisk server to talk to a Panasonic D500 PBXusing an E1 connection. The card for the Panasonic uses MFC/R2 and Ihave installed Unicall. Calls from the Asterisk server to the Panasonic go through without a hitch and I can call any extension I want. Theproblem is that I cannot get any calls from the Panasonic. I have thefollowing log from a call: ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Jitterbuffer on SIP
Hi,Is that a way to patch a running asterisk 1.2.9.1 instalation with the experimental SIP Jitterbuffer support ?Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IP JitterBuffer for 1.2.5
Hi, I´ve read a post about SIP Jitter Buffer for 1.2.5. I´m runnig Asterisk 1.2.5 on a production server. Is that a way to patch my Asterisk server with the SIP Jitter Buffer support? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Anyone Know That !!!
Hello did you noticed that http://www.asterisk.org is just pointing to a web cvs directory best regards Thierry ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Unable to reach voip-info.org from france
Hello There is a loopback between these IP adress 63.216.31.38ge3-1.br01.atl01.pccwbtn.net 63.216.31.174 wvfiber.ge2-2.br01.atl01.pccwbtn.net 63.216.31.173 ge2-2.br01.atl01.pccwbtn.net 63.216.31.174 wvfiber.ge2-2.br01.atl01.pccwbtn.net 63.216.31.173 ge2-2.br01.atl01.pccwbtn.net 63.216.31.174 wvfiber.ge2-2.br01.atl01.pccwbtn.net 63.216.31.173 ge2-2.br01.atl01.pccwbtn.net 63.216.31.174 wvfiber.ge2-2.br01.atl01.pccwbtn.net 63.216.31.173 ge2-2.br01.atl01.pccwbtn.net Etc Best regards Thierry > -Message d'origine- > De : [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] De la part > de James H Thompson > Envoyé : samedi 27 août 2005 17:56 > À : Asterisk Users Mailing List - Non-Commercial Discussion > Objet : Re: [Asterisk-Users] voip-info - is it alive > > If its NOT working for you, please send a traceroute to: > [EMAIL PROTECTED] > Thanks. > Jim > [EMAIL PROTECTED] > > - Original Message - > From: "Bob Goddard" <[EMAIL PROTECTED]> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Sent: Saturday, August 27, 2005 3:15 AM > Subject: Re: [Asterisk-Users] voip-info - is it alive > > > > On Friday 26 Aug 2005 14:54, Julian Lyndon-Smith wrote: > >> I cannot reach voip-info - is it just me or is the site > not available ? > > > > There is a bad route being propogated. > > > > > > B > > ___ > > --Bandwidth and Colocation sponsored by Easynews.com -- > > > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SPA 841 form SIPURA
> -Message d'origine- > De : [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] De la part > de Paul Dugas > Envoyé : dimanche 7 août 2005 16:11 > À : Asterisk Mailing List > Objet : Re: [Asterisk-Users] SPA 841 form SIPURA > > On Sun, August 7, 2005 2:07 am, [EMAIL PROTECTED] said: > > How good is :SPA 841 form SIPURA. > > Not good if voice quality is a requirement. Talking on the > handset sounds to the caller and callee like you're on one of > those really old speakerphones that clips the beginning of > each phrase after a pause. > There's a ramp-down of white-noise at the end of each phrase. This is not true You have to switch to last firmware and/or disable silent suppression > The speakerphone is totally useless as the user is completely > inaudable unless yelling with their face directly in front of > the unit. Have a look at last firmware and user setup Best Reagrds Thierry ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IP-ID in RTP/UDP/IP packets
Hello Did you tried a Sysctl -w net.ipv4.ip_no_pmtu_disc=1 So /proc/sys/net/ipv4/ip_no_pmtu_disc is set to 1 and the kernel won't do any mtu discovery Best regards Thierry > -Message d'origine- > De : [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] De la part de Aj > Envoyé : vendredi 29 juillet 2005 06:25 > À : asterisk-users@lists.digium.com > Objet : [Asterisk-Users] IP-ID in RTP/UDP/IP packets > > Hi All, > > I am doing some testing with the asterisk server and have > been monitoring the packets exchanged during a SIP-ZAPTEL phone call. > > I see that the IP-ID in all of the RTP/UDP/IP packets are set to zero. > After some googling, I have learnt that some of the linux > implementations set the IP-ID to 0 (if the DF bit is set in the IP > header) if the two hosts exchanging data are on the same subnet. > > I see this happening in the RTP packets of the phone calls being made. > > Is there a way to: > - force linux to send the right IP-ID? > - or to set the DF bit to 0 in an attempt to make the IP > stack to put in the right IP-ID? > > Linux kernel being used: 2.4.20-8. > > Please let me know, > Thanks, > Ajay > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Interface with mobile phone
Hello try to setup a gsm gateway it will do what you want best regards Thierry De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de chawki hammoudEnvoyé : samedi 16 juillet 2005 20:55À : Asterisk-Users@lists.digium.comObjet : [Asterisk-Users] Asterisk Interface with mobile phone Hi: I live in a country where calls from landline phone to a mobile phones is more expensive than mobile to mobile. I have FXO card connected to the landline. All the calls from IAX goes through this interface to the pstn and mobile phones. I want to save money by transfering mobile calls through a mobile phone. Is there some interface between the FXO card and the mobile phone so asterisk can dial the mobile phone? Lnadlines and mobile phones can be differntaited by their prefix. Thanks Start your day with Yahoo! - make it your home page ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Last CVS -> High Load
Good morning on our Test Machine based on RedHat 9 on a Pentium 4 SMP we are experiencing a 99% CPU load for asterisk with the CVS of tonight does anyone noticed that best regards Thierry ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sipura SPA-841 Volume Oscillation Problem
Yes i we know it but as it is an intermediate patched version i think it's better way for the final release A++ > -Message d'origine- > De : Javier Ergas [mailto:[EMAIL PROTECTED] > Envoyé : vendredi 8 juillet 2005 18:51 > À : [EMAIL PROTECTED] > Objet : RE: [Asterisk-Users] Sipura SPA-841 Volume Oscillation Problem > > Thank you very much. Here you have another 3.1.4(a): > > http://www.sipura.com/download/temp/phone/spa841-03-01-03-a-vo > l-fix.zip > > Cheers > -Mensaje original- > De: Thierry Wehr [mailto:[EMAIL PROTECTED] Enviado el: Viernes, > 08 de Julio de 2005 9:57 > Para: 'Javier Ergas' > Asunto: RE: [Asterisk-Users] Sipura SPA-841 Volume Oscillation Problem > > We are reselling sipura equipments and i'm in touch with the support > > You can download the previous version on our web site > > http://www.widevoip.com/download/firmware/spa841-3.1.2d.zip > > Thierry > http://www.widevoip.com > > Best regards > > > > -Message d'origine- > > De : Javier Ergas [mailto:[EMAIL PROTECTED] Envoyé : vendredi > 8 juillet > > 2005 15:21 À : [EMAIL PROTECTED] Objet : RE: [Asterisk-Users] Sipura > > SPA-841 Volume Oscillation Problem > > > > Hi, > > > > How do you know? > > Where can I find the previous version? I have 3.1.3(a) Do you know > > when will be an update? > > > > Thanks, > > > > Jergas > > > > -Mensaje original- > > De: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] En nombre > de Thierry > > Wehr Enviado el: Viernes, 08 de Julio de 2005 3:44 > > Para: 'Asterisk Users Mailing List - Non-Commercial Discussion' > > Asunto: RE: [Asterisk-Users] Sipura SPA-841 Volume > Oscillation Problem > > > > Hello > > > > Go back to the firmware before and all will be ok or wait until the > > next one > > > > Best Regards > > Thierry > > > > > -Message d'origine- > > > De : [EMAIL PROTECTED] > > > [mailto:[EMAIL PROTECTED] De la part > > de Vahan > > > Yerkanian Envoyé : vendredi 8 juillet 2005 07:23 À : > Asterisk Users > > > Mailing List - Non-Commercial Discussion Cc : > > > [EMAIL PROTECTED] Objet : Re: > > [Asterisk-Users] Sipura > > > SPA-841 Volume Oscillation Problem > > > > > > Greetings, > > > > > > I'm experiencing the same problem. It manifests itself > > mostly in noisy > > > environments - as soon as there is some increase of the > > ambient noise > > > the volume in the headpiece or the "speakerphone" decreases > > > immediately, and starts to randomly increase/decrease for > some time > > > after the ambient noise gets low. This is 100% repeatable > > if you start > > > the conversation by using speakerphone. As soon as you > > switch to the > > > handset, the defect disappears. Now the problem is that > 5% of calls > > > via headset have the same problem. > > > > > > I am using the latest firmware for the SPA-841. > > > > > > Javier Ergas wrote: > > > > Hi all, > > > > > > > > > > > > > > > > The problem is on the volume of the voice sent by the > > > SPA-841. I think > > > > the echo cancel algorithm sets a limit to the microphone > > > when detects > > > > sounds or noise from the earphone. This problem generates an > > > > oscillation on the voice volume sent by the phone and > > even turns it > > > > off completely for very little lapses of time making the > > > communication > > > > very uncomfortable. I manage three different > implementations with > > > > Asterisk and Sipura SPA-841 on different clients and network > > > > topologies, and on every one we are experiencing the same > > situation. > > > > > > > > > > > > > > > > Thanks, > > > > > > > > jergas > > > > > > > > > > > > > > > > > > > > > > > > > > -- > > > > -- > > > > > > > > ___ > > > > Asterisk-Users mailing list > > > > Asterisk-Users@lists.digium.com > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > To UNSUBSCRIBE or update options visit: > > > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > ___ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sipura SPA-841 Volume Oscillation Problem
Hello Go back to the firmware before and all will be ok or wait until the next one Best Regards Thierry > -Message d'origine- > De : [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] De la part > de Vahan Yerkanian > Envoyé : vendredi 8 juillet 2005 07:23 > À : Asterisk Users Mailing List - Non-Commercial Discussion > Cc : [EMAIL PROTECTED] > Objet : Re: [Asterisk-Users] Sipura SPA-841 Volume Oscillation Problem > > Greetings, > > I'm experiencing the same problem. It manifests itself mostly > in noisy environments - as soon as there is some increase of > the ambient noise the volume in the headpiece or the > "speakerphone" decreases immediately, and starts to randomly > increase/decrease for some time after the ambient noise gets > low. This is 100% repeatable if you start the conversation by > using speakerphone. As soon as you switch to the handset, the > defect disappears. Now the problem is that 5% of calls via > headset have the same problem. > > I am using the latest firmware for the SPA-841. > > Javier Ergas wrote: > > Hi all, > > > > > > > > The problem is on the volume of the voice sent by the > SPA-841. I think > > the echo cancel algorithm sets a limit to the microphone > when detects > > sounds or noise from the earphone. This problem generates an > > oscillation on the voice volume sent by the phone and even turns it > > off completely for very little lapses of time making the > communication > > very uncomfortable. I manage three different implementations with > > Asterisk and Sipura SPA-841 on different clients and network > > topologies, and on every one we are experiencing the same situation. > > > > > > > > Thanks, > > > > jergas > > > > > > > > > > > -- > > -- > > > > ___ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How to prevent log files from eating my harddrive?
Have a look in /etc/logrotate.d and use apache file for example Then have a look at /etc/logrotate.conf and chose for example day with 30 days You won't have to do anything, logrotate will do the job for you And all will be fine Best regards Thierry > -Message d'origine- > De : [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] De la part > de Michael Stahl > Envoyé : mardi 5 juillet 2005 22:04 > À : Asterisk Users Mailing List - Non-Commercial Discussion > Objet : RE: [Asterisk-Users] How to prevent log files from > eating my harddrive? > > Use a script to rotate the logs every night (by cron). If > you need one, I'll post what I have. > > That's only have the solution of course...you would have to > write another script to delete logs > X days old. > > Anyone have one of those handy? > > -Original Message- > From: Leo Burd [mailto:[EMAIL PROTECTED] > Sent: Tuesday, July 05, 2005 12:28 PM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] How to prevent log files from > eating my hard drive? > > Hello there, > > Somehow, Asterisk log files are consuming all the space that > I have in my hard disk... They've already eaten 14GB and are > still hungry!! What shall I do? I'm not even logging > anything in verbose mode!! > > Help really appreciated!! > > Best, > > Leo > > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sipura SPA2000 behind NAT
Hello This iptables setup won't work You need specific rules for the incoming UDP packets with status ESTABLISHED and RELATED like these simple ones Remember it's a statefull firewall. In the nat section -A POSTROUTING -p udp -m udp -m state --state RELATED -j MASQUERADE -A POSTROUTING -p udp -m udp -m state --state ESTABLISHED -j MASQUERADE And in the filter section -A FORWARD -p udp -m udp -m state --state RELATED -j ACCEPT -A FORWARD -p udp -m udp -m state --state ESTABLISHED -j ACCEPT Best regards Thierry > -Message d'origine- > De : [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] De la part > de Guillermo Salas M > Envoyé : samedi 2 juillet 2005 22:56 > À : asterisk-users@lists.digium.com > Objet : RE: [Asterisk-Users] Sipura SPA2000 behind NAT > > Carlos, > > Thank you for your fast response :) , this is the output of > iptables -nL on my linux box: > > [EMAIL PROTECTED]:/home/guillermo # iptables -nL Chain INPUT > (policy ACCEPT) > target prot opt source destination > > Chain FORWARD (policy ACCEPT) > target prot opt source destination > ACCEPT all -- 192.168.0.0/24 0.0.0.0/0 > ACCEPT all -- 0.0.0.0/0192.168.0.0/24 > > Chain OUTPUT (policy ACCEPT) > target prot opt source destination > > [EMAIL PROTECTED]:/home/guillermo # iptables -nL -t nat Chain > PREROUTING (policy ACCEPT) > target prot opt source destination > > Chain POSTROUTING (policy ACCEPT) > target prot opt source destination > MASQUERADE all -- 192.168.0.0/24 0.0.0.0/0 > > Chain OUTPUT (policy ACCEPT) > target prot opt source destination > > > This is my very-small and simple firewall script: > [EMAIL PROTECTED]:/home/guillermo # cat /etc/init.d/firewall # > Cargar Modulos modprobe ip_tables modprobe ip_nat_ftp > modprobe ip_conntrack_ftp modprobe ip_nat_irc modprobe > ip_conntrack_irc > > # Habilitar el forward > echo 1 > /proc/sys/net/ipv4/ip_forward > > # Flush > iptables -X > iptables -F > iptables -X -t nat > iptables -F -t nat > > # Habilitar nat para 192.168.0.0/24 > iptables -t nat -A POSTROUTING -o eth0 -s 192.168.0.0/24 -j > MASQUERADE # Permitir el forward para 192.168.0.0/24 iptables > -A FORWARD -s 192.168.0.0/24 -j ACCEPT iptables -A FORWARD -d > 192.168.0.0/24 -j ACCEPT > > # EOF > > > On Sat, 2005-07-02 at 16:39 -0400, Carlos Alperin wrote: > > Guillermo, > > > > This is an issue with your router. Do you have open the > ports 5060 for SIP? > > Also, RTP needs to be open from 16384 to 32767. > > > > Saludos, > > > > Carlos Alperin > > Senior System Engineer > > Seneca Communications, LLC > > [EMAIL PROTECTED] > > > > > > -Original Message- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] On Behalf Of > > Guillermo Salas M > > Sent: Saturday, July 02, 2005 4:13 PM > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: [Asterisk-Users] Sipura SPA2000 behind NAT > > > > Hi, I've one Sipura SPA2000 at home behind a linuxbox with > two network > > adapters (eth0 for WAN and eth1 for LAN) doing NAT/DHCP: > > > > > > ___ HOME ___ OFFICE > > SPA2000 <---> Linux Box <--> Asterisk Box > > 192.168.0.253192.168.0.1 eth1 200.93.xxx.a > > 200.93.xxx.b eth0 > > > > My problem is when I try to call to any trunk or extention > I can the > > audio when the destination is ringing, but I can hear the > voice of the > > person when it reponds. The person in the other side can > hear me, but > > I can not hear anything from him. I can not hear the voice > prompts for > > the voicemail (*98) or the operator voice, but can leave voice > > messages to other SIP devices and they can hear my messages. > > > > This is my sip.conf > > [105] > > username=105 > > type=friend > > secret=105 > > qualify=no > > port=5060 > > nat=yes > > [EMAIL PROTECTED] > > host=dynamic > > dtmfmode=rfc2833 > > context=from-internal > > canreinvite=no > > callerid="Guilllermo Salas HOME" <105> > > > > My ext on line 1 of the Sipura is 105, and is registred > with the * box: > > -- Registered SIP '105' at 200.93.220.27 port 5060 expires 3600 > > > > asterisk*CLI> sip show peer 105 > > asterisk*CLI> > > > > * Name : 105
RE: [Asterisk-Users] Fwd: JE TROUVE QUE VOUS N'ETES PAS HONETE!
> -Message d'origine- > De : [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] De la part > de harry gaillac > Envoyé : lundi 27 juin 2005 22:51 > À : Asterisk Users Mailing List - Non-Commercial Discussion > Objet : Re: [Asterisk-Users] Fwd: JE TROUVE QUE VOUS N'ETES > PAS HONETE! > > finally i'm feeling tired, > > Asterisk users have to know i'm lazy because of i'm french. I don't think writing down this is a good idea. I'm french and don't like these types of "short ways" Best regards Thierry ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] French Audio Files
> -Message d'origine- > De : [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] De la part > de Asterisk > Envoyé : jeudi 23 juin 2005 22:47 > À : asterisk-users@lists.digium.com > Objet : [Asterisk-Users] French Audio Files > > Hello - sorry for my bad english. > I will make a list of all sound files on asterisk and i'll > record then on professional studio. > the french prompts from sineapps sounds bad... sorry for her... > tell me if their is many peoples want it ! > > thank's. > > en francais: > > dites moi si ca vaut le coup que j'investissexr dans > l'enregistrement des messages en francais. La voix sera la > "voix off" d'M6... > merci ! Hello This is a great idea Ceci est une tres bonne idee A++ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] PHP/AGI Problem
Hello Did you tried a deadagi in place of agi A++ > -Message d'origine- > De : [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] De la part > de Jon Farmer > Envoyé : mercredi 25 mai 2005 11:40 > À : asterisk-users@lists.digium.com > Objet : [Asterisk-Users] PHP/AGI Problem > > The problem is as follows. If the caller hangs up at any time > during the application the following happens. > > 1. Asterisk console reports the call hung up. As follows > > == Spawn extension (default, 502, 3) exited non-zero on > 'SIP/3753684-fabd' > > 2. However when I look in the server process list the PHP app > is still running. > > ps -ax > ... > 8029 ?S 0:00 /usr/bin/php -q > /usr/share/asterisk/agi-bin/test.php > > They only way to get rid of it is to killall -9 it. > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Grandstream ATA 286 and ilbc
Our ATA286 and 486 using 1.0.6 have all a broken ILBC A++ > -Message d'origine- > De : [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] De la part > de Daniel Nylander > Envoyé : vendredi 20 mai 2005 01:48 > À : Asterisk Users Mailing List - Non-Commercial Discussion > Objet : Re: [Asterisk-Users] Re: Grandstream ATA 286 and ilbc > > Speaking of Grandstream firmware. Anyone noticed that they > removed the BETA firmware from their site? > Wonder why. > > Daniel > (CISSP) > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Compile problem on last CVS
Good evening from the CVS of the 2005/05/14 it's impossible to build asterisk* on a redhat 7.3 i get this at compile time chan_sip.c: In function `build_user':chan_sip.c:10007: parse error before `struct'chan_sip.c:10029: `userflags' undeclared (first use in this function)chan_sip.c:10029: (Each undeclared identifier is reported only oncechan_sip.c:10029: for each function it appears in.)chan_sip.c:10029: `mask' undeclared (first use in this function)chan_sip.c:10094: warning: type defaults to `int' in declaration of `__s'chan_sip.c:10094: warning: comparison of distinct pointer types lacks a castchan_sip.c: In function `build_peer':chan_sip.c:10176: parse error before `struct'chan_sip.c:10221: `peerflags' undeclared (first use in this function)chan_sip.c:10221: `mask' undeclared (first use in this function)chan_sip.c:10391: warning: type defaults to `int' in declaration of `__s'chan_sip.c:10391: warning: comparison of distinct pointer types lacks a castmake[1]: *** [chan_sip.o] Erreur 1make[1]: Quitte le répertoire `/usr/src/asterisk-cvs/asterisk/channels'make: *** [subdirs] Erreur 1 may be someone have a clue to fix it best rehards Thierry ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Formatting problem in cmd sip show peers
Good afteroon i had found a special issue while using "sip show peers" sometimes i get 6115/6115 xx.xx.xx.xx 6109/6109 xx.xx.xx.xx 6001/6001 xx.xx.xx.xx6107/6107 xx.xx.xx.xx 00/500 xx.xx.xx.xx 7000/7000 xx.xx.xx.xx in fact it must be 6115/6115 xx.xx.xx.xx 6109/6109 xx.xx.xx.xx 6001/6001 xx.xx.xx.xx6107/6107 xx.xx.xx.xx 500/500 xx.xx.xx.xx 7000/7000 xx.xx.xx.xx after 3 or 4 "sip show peers" it shows right things best regards Thierry PS i'm using latest CVS from 05/11/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] NEED HELP PROGRAMING ASTERISK VoIP NETWORK
> -Message d'origine- > De : [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] De la part > de Jean-Michel Hiver > Envoyé : lundi 25 avril 2005 07:54 > À : Asterisk Users Mailing List - Non-Commercial Discussion > Objet : Re: [Asterisk-Users] NEED HELP PROGRAMING ASTERISK > VoIP NETWORK > > Franz wrote: > > >Please contact me Urgent... > > > > > Hi Frantz, > > I can do custom programming. Here is some information about > my company: > > http://ykoz.net/intl/ > > Let me know what you're after and I'll send you a preliminary quote. > > Cheers, > Jean-Michel. > > -- > Ykoz Un Max - La VoIP en pré-payé! > Essayez gratuitement - 5 crédits offerts. > ---> http://ykoz.net/voip/max <--- Hi Can you please do advertising for your company in Asterisk-Biz Aniway where are the legal notices and the RCS on you'r web site Seem's to be quit strange Thierry Wehr ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Realtime UPDATE
> -Message d'origine- > De : [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] De la part > de Rod Bacon > Envoyé : jeudi 7 avril 2005 01:06 > À : asterisk-users@lists.digium.com > Objet : [Asterisk-Users] Realtime UPDATE > My problem is that upon registration, the UA's IP address and > port information isn't being written to the MYSQL realtime database. > Subsequently, calls to the UA fail if they originate from > another * server (The server DOES attempt a lookup, but > obviously gets no value for IP address / PORT). Hi I'd the same problem and discovered that you have to put rtnoupdate=no in you'r sip.conf Hope it helps you Thierry ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] using unixODBC
> -Message d'origine- > De : [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] De la part > de Kamran Ahmad > Envoyé : vendredi 1 avril 2005 11:08 > À : asterisk-users@lists.digium.com > Objet : [Asterisk-Users] using unixODBC > > hi list > > i know i am asking question out of the scope of this list. > actualy i cant find any place to ask question like this. may > be someone using ODBC with asterik. > actualling i want to make ODBC connection for asterisk on my > new fedora core 2. i have tried every thing. > tried rpms. compiled code nothing works here. > i have already done this kind of connection on my other > mechine. i dont know why i am getting error. > > actually when i am doing > > isql asteriskdsn > [ISQL]ERROR: Could not SQLConnect Hello Just give my own config that works well /etc/odbc.ini [MySQL-asterisk] Description = MySQL asterisk database Trace = Off TraceFile = stderr Driver = MySQL SERVER = 127.0.0.1 USER= connecting-user PASSWORD= user-password PORT= 3306 DATABASE= asterisk /etc/odbcinst.ini [MySQL] Description = MySQL driver for Linux Driver = /usr/lib/libmyodbc.so FileUsage = 1 /etc/asterisk/res_odbc.conf [asterisk] dsn => MySQL-asterisk username => connecting-user password => user-password pre-connect => yes Hop i'll help you as it works great here Best regards Thierry Wehr ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] New Warning in CVS: Format for authentication entry is user[:secret]@realm
Good evening since the cvs of the 22 i get this message in my logs WARNING[31865]: chan_sip.c:9599 in add_realm_authentication: Format for authentication entry is user[:[EMAIL PROTECTED] at line 0 may be someone now where it is coming from also the IP address and Port are still not being updated in the sipfriend database (using odbc + mysql) bets regards thierry ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] No more updates of IP address and port in CVS HEAD
Good afternoon Since the cvs version of yesterday, the ip address and the port of the sipfriend are no more updated in the realtime database Regards Thierry Wehr ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ERROR: compile asterisk(from CVS HEAD) and got anerror
Hi All Just made a post before but seem's not to appear in the list I replaced at line 62 of include/asterisk/app.h file struct ast_ivr_option options[]; /* All options */ With struct ast_ivr_option *options; /* All options */ It works but as i'm not very good a C don't know if it's ok Bets regards thierry -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Charles Wang Envoyé : samedi 26 février 2005 15:58 À : asterisk-users@lists.digium.com Objet : [Asterisk-Users] ERROR: compile asterisk(from CVS HEAD) and got anerror Dear ALL: I got an error message lists below. Does anyone have the same problem? How to solve it? Best Regard Charles In file included from config.c:34: include/asterisk/app.h:62: array size missing in `options' make: *** [config.o] Error 1 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk billing solution
Hi I have something like this but it's in french and it uses teh res_config Best regards Thierry wehr -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Nabeel Jafferali Envoyé : mercredi 22 décembre 2004 22:57 À : asterisk-users@lists.digium.com Objet : [Asterisk-Users] Asterisk billing solution Hello. I am looking for a simple Asterisk billing solution. I expect about 50-100 users (a mix of IAX and SIP) through 3-5 outgoing providers (all IAX). I need something that can handle monthly fees and per call charges (depending on destination, obviously), and should provide a web interface for customers and administrators. Something that can tie in to one of the existing management GUIs would be a big plus. Any ideas? -- Nabeel Jafferali tel: 647.722.8457 x201 718.606.4190 x201 fwd: 46990 x201 email/msn: nabeeljafferali.net ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicemail Question
Hi Just for a quick question. I'm using voicemail with mysql but i cannot modifiy the password in the sql table from the phone with option 5 of the mailbox option Is it because i must change it through the database only Best regards Thierry ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk / VOIP Employment Opportunity
Title: Asterisk / VOIP Employment Opportunity Hi Sukhi it's late here and i need to update it (old one) i'll send it to you tomorrow morning Regards Thierry PS: do you have a direct email De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Sukhi BhullarEnvoyé : mercredi 8 décembre 2004 02:17À : Asterisk Users Mailing List - Non-Commercial DiscussionObjet : RE: [Asterisk-Users] Asterisk / VOIP Employment Opportunity Hi Thierry, We would certainly consider them as we can potentially organize work visa for the appropriate candidates. Do you have a resume you can send me please? Regards, Sukhi From: Thierry [mailto:[EMAIL PROTECTED] Sent: Wednesday, 8 December 2004 10:45 AMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [Asterisk-Users] Asterisk / VOIP Employment Opportunity Hello do you accept french candidates best regards Thierry Wehr De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Sukhi BhullarEnvoyé : mercredi 8 décembre 2004 00:42À : [EMAIL PROTECTED]Objet : [Asterisk-Users] Asterisk / VOIP Employment Opportunity Dear All, I am based in Australia and have a client looking to hire a VOIP Specialist with Asterisk experience to join their technical/engineering team. The company specialise in providing corporate and government grade data comms solutions. They are moving into the VOIP space, hence the need for the Asterisk/VOIP specialist. Would anyone be keen to explore this opportunity of working in Australia further? If so, please respond to this email and I will be more than happy to discuss in more detail. Kind regards Sukhi _ Sukhi Bhullar Director (O) 07 3233 9000 (F) 07 3233 9099 (M) 0402 467 741 www.mindworx.com.au NOTICE: This email, and any attachments transmitted with it, may contain privileged and confidential information, which is intended solely for the use of the individual or individuals named above. If you are not the intended recipient of this email you have received this email in error and are hereby notified that any use, dissemination, distribution or copying of this email is strictly prohibited. If you have received this email in error, please delete it and notify Mindworx Pty Ltd immediately by return email or by telephoning 3233 9000. The views expressed in this email are those of the individual sender and do not necessarily reflect the views of Mindworx Pty Ltd. A candidate referred by Mindworx remains a candidate of Mindworx for a period of twelve months after the initial referral. Should a candidate subsequently be appointed in any position, a Fee will be payable in accordance with Mindworx standard terms of business. If the details of a candidate or potential candidate, referred by Mindworx, are passed on to a third party who subsequently hires that candidate, a Fee will be payable to Mindworx in accordance with the Mindworx standard terms of business. Thank You. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk / VOIP Employment Opportunity
Title: Asterisk / VOIP Employment Opportunity Hello do you accept french candidates best regards Thierry Wehr De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Sukhi BhullarEnvoyé : mercredi 8 décembre 2004 00:42À : [EMAIL PROTECTED]Objet : [Asterisk-Users] Asterisk / VOIP Employment Opportunity Dear All, I am based in Australia and have a client looking to hire a VOIP Specialist with Asterisk experience to join their technical/engineering team. The company specialise in providing corporate and government grade data comms solutions. They are moving into the VOIP space, hence the need for the Asterisk/VOIP specialist. Would anyone be keen to explore this opportunity of working in Australia further? If so, please respond to this email and I will be more than happy to discuss in more detail. Kind regards Sukhi _ Sukhi Bhullar Director (O) 07 3233 9000 (F) 07 3233 9099 (M) 0402 467 741 www.mindworx.com.au NOTICE: This email, and any attachments transmitted with it, may contain privileged and confidential information, which is intended solely for the use of the individual or individuals named above. If you are not the intended recipient of this email you have received this email in error and are hereby notified that any use, dissemination, distribution or copying of this email is strictly prohibited. If you have received this email in error, please delete it and notify Mindworx Pty Ltd immediately by return email or by telephoning 3233 9000. The views expressed in this email are those of the individual sender and do not necessarily reflect the views of Mindworx Pty Ltd. A candidate referred by Mindworx remains a candidate of Mindworx for a period of twelve months after the initial referral. Should a candidate subsequently be appointed in any position, a Fee will be payable in accordance with Mindworx standard terms of business. If the details of a candidate or potential candidate, referred by Mindworx, are passed on to a third party who subsequently hires that candidate, a Fee will be payable to Mindworx in accordance with the Mindworx standard terms of business. Thank You. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
