Re: [Asterisk-Users] No sound when bridging two single FXO cards
Ok, sorry for this lack of explanations. I run Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k. The log has recorded this during the last test of bridging two FXO lines : "","","s","default","","Zap/2-1","Zap/3-1","Dial","Zap/3/ww0164417888","2005-07-15" 12:34:25","2005-07-15 12:34:25","2005-07-15 12:35:49",84,84,"ANSWERED","DOCUMENTATION" The call progress normaly, but no sound. And, yes, the incoming and outgoing calls are placed normaly with my cards (3 cards, in fact). They are seen as X101P by Linux (at boot). Zaptel tell that : Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) Channel 03: FXS Kewlstart (Default) (Slaves: 03) 3 channels configured. My zapata.conf : cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=2000 rxgain=4 txgain=4 busydetect=yes busycount=5 group = 1 context = pstn signalling = fxs_ks callerid=asreceived amaflags = documentation channel => 1-3 My zaptel.conf : loadzone=fr defaultzone=fr fxsks=1-3 How to trace the bridge establishment from CLI ? I know how for IAX or SIP protocol, not for zap interface... TIA Best Regards, Francois BERGERET, France. ___ 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 sound when bridging two single FXO cards
Wow ! No reply... May be I must talk about X100P instead of X101P ? Is someone has yet encountered this kind of "no sound" problem when bridging two FXO lines like this (first as input, second as output) ? Any idea ? TIA. Best Regards, Francois BERGERET, France. - Original Message - From: "Francois BERGERET" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Saturday, July 16, 2005 7:41 PM Subject: Bridging two FXO cards (X101P) problem Hi the list :-) Wondering why I can't "bridge" two X101P FXO cards to forward an external call from a first X101P to another analog telephone outside my house throught a seconf X101P. [VACATION] exten => s,1,Answer exten => s,2,Dial(Zap/3/ww0161417888),120 exten => s,3,Voicemail(u1001) exten => s,4,Hangup exten => s,104,Voicemail(b1001) exten => s,105,Hangup I temps to do that to avoid missing calls during my summer vacations. Numbering is ok when receiving a call, but no sound is heard or only few peak distorsions in background if I speak loud in the phone. Normal calls are running well, I use this Asterisk box every day and this FXO cards are ok. What have I missed ? Thanks for any help. Best Regards, Francois BERGERET, France. ___ 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] modprobe wcfxo fails.
Hello Mike, Is your Asterisk box running for a HAM's project that I am working on ? 73, F6HQZ, Francois BERGERET. - Original Message - From: Michael J. Tubby G8TIC To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Monday, July 18, 2005 12:30 AM Subject: Re: [Asterisk-Users] modprobe wcfxo fails. Tim, Yes, it was resolved but cannot remember how... I'm using RedHat FC3, Asterisk 1.0.9 and it all just works assuming that you: - RTFM with respect to udev setup - use "make linux26" to build the kernel module for a 2.6 kernel Regards Mike - Original Message - From: Tim King To: [EMAIL PROTECTED] Sent: Sunday, July 17, 2005 8:09 PM Subject: [Asterisk-Users] modprobe wcfxo fails. I was reading a thread where you were helping someone out and noticed it ended without resolve. Was this issue ever taken care of?I seem to be having the exact same problem. Thanks Tim King Network Engineer Computer & Network Solutions LLC 1331 Plainfield Ave Grand Rapids MI 49505 Phone: 800-669-3290 ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo 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] Bridging two FXO cards (X101P) problem
Hi the list :-) Wondering why I can't "bridge" two X101P FXO cards to forward an external call from a first X101P to another analog telephone outside my house throught a seconf X101P. [VACATION] exten => s,1,Answer exten => s,2,Dial(Zap/3/ww0161417888),120 exten => s,3,Voicemail(u1001) exten => s,4,Hangup exten => s,104,Voicemail(b1001) exten => s,105,Hangup I temps to do that to avoid missing calls during my summer vacations. Numbering is ok when receiving a call, but no sound is heard or only few peak distorsions in background if I speak loud in the phone. Normal calls are running well, I use this Asterisk box every day and this FXO cards are ok. What have I missed ? Thanks for any help. Best Regards, Francois BERGERET, France. ___ 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] VPN's
Sure, I have more than 18 tunnels to manage here, and the only blocking effects are thuse that I have volontary encoded . ;-) I believe that Peter has missed something in the VPN parmeters themselves or not correctly understood how are his IPtables onto this two IPSec secure gateway... Peter, could you post us the content of your "/etc/ipsec.conf" file ? We can take a look here and verify what is not good. Best Regards, Francois BERGERET, France. - Original Message - From: "Shamsul Arefin" <[EMAIL PROTECTED]> To: "Francois BERGERET" <[EMAIL PROTECTED]>; "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Friday, July 15, 2005 11:48 PM Subject: Re: [Asterisk-Users] VPN's Hi, We use firewall and VPN togather to connect around 5 remote sites, and never encounter these problems. Make sure that port 10,000 and above mentioned in ur rtp conf files are opened in ur vpn and firewall. also when u connect from remote site don't use public ip use privte behind firewall. If still have problem send me more detail and i will be more then happy to sort this out . Regards Shamsul Arefin Saktek Broadband telephony experts On 7/16/05, Francois BERGERET <[EMAIL PROTECTED]> wrote: Hi men, You have some IP ports blocked ! I use SuperFreeSwan and I encounter no problem with this kind of configuration. Do you have open all ports on your IPsec gateways ? Think to have a look to your IPchains or any kind of firewall you are running in your IPSec gateway. I use shorewall and it is possible to miss some rules or to let pass few ports only for protections between sites. You must describe more your configurations to see what... Good luck ! Francois BERGERET, [EMAIL PROTECTED], France. - Original Message - From: "Armin Schindler" <[EMAIL PROTECTED]> To: "Peter Osborne" <[EMAIL PROTECTED]> Cc: Sent: Friday, July 15, 2005 8:35 PM Subject: Re: [Asterisk-Users] VPN's > On Fri, 15 Jul 2005, Peter Osborne wrote: >> Hi All, >> >> I'm using Asterisk for my PBX, I have a remote office that is connected >> by a >> VPN link. I am using Openswan on my side and a Linksys box on the >> remote >> side. I have a Polycom IP300 on the remote side configured with a >> static >> IP >> address. When I call the phone on the remote side, it rings and >> establishes >> the call fine. The problem I am having is that the remote side can hear >> the >> call find but the local side hears nothing. Because of the VPN there >> are >> no >> firwalls in the way. Does anyone have some ideas or atleast how I can >> track >> down the problem. > > I had the same problem with VPN using 'netscreen' (or a similar name) > boxes. > When I switched from SIP to IAX protocol, it worked perfectly. > > I think the SIP voice UDP packets are blocked somehow, but I didn't > investigated it further. > > Armin > ___ > 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 -- Best Regards Shamsul Arefin Saktek , Broadband Telephony experts ___ 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] VPN's
Hi men, You have some IP ports blocked ! I use SuperFreeSwan and I encounter no problem with this kind of configuration. Do you have open all ports on your IPsec gateways ? Think to have a look to your IPchains or any kind of firewall you are running in your IPSec gateway. I use shorewall and it is possible to miss some rules or to let pass few ports only for protections between sites. You must describe more your configurations to see what... Good luck ! Francois BERGERET, [EMAIL PROTECTED], France. - Original Message - From: "Armin Schindler" <[EMAIL PROTECTED]> To: "Peter Osborne" <[EMAIL PROTECTED]> Cc: Sent: Friday, July 15, 2005 8:35 PM Subject: Re: [Asterisk-Users] VPN's On Fri, 15 Jul 2005, Peter Osborne wrote: Hi All, I'm using Asterisk for my PBX, I have a remote office that is connected by a VPN link. I am using Openswan on my side and a Linksys box on the remote side. I have a Polycom IP300 on the remote side configured with a static IP address. When I call the phone on the remote side, it rings and establishes the call fine. The problem I am having is that the remote side can hear the call find but the local side hears nothing. Because of the VPN there are no firwalls in the way. Does anyone have some ideas or atleast how I can track down the problem. I had the same problem with VPN using 'netscreen' (or a similar name) boxes. When I switched from SIP to IAX protocol, it worked perfectly. I think the SIP voice UDP packets are blocked somehow, but I didn't investigated it further. Armin ___ 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] Looking for a consutant in France about Asterisk.
Hi all the list, Bonsoir, It could be nice to give a correct email adress to contact you, instead of an "over quota" one... If you want a direct reply out of this list, please, give us one other. Best Regards, a bientot, Francois BERGERET, France. [EMAIL PROTECTED] - Original Message - From: Aref Cheikhrouhou To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Tuesday, July 12, 2005 1:08 AM Subject: [Asterisk-Users] Looking for a consutant in France about Asterisk. Hi All, Looking for a consultant in France to build a VoIP offer. Thanks for the reply Aref ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo 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
