RE: [Asterisk-Users] 2 x Eicon BRI ISDN devices (UK)
Hi Does that mean that I should set up the msn=* and add DDI's to the extensions.conf? I think that BT class the 1+1 as 'auxilary line working'. Rgds -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of asterisk Sent: 18 May 2005 14:28 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] 2 x Eicon BRI ISDN devices (UK) MSN will only work on 1 ISDN2 line and cannot be spread across 2 ISDN2 lines. From your description I assume you have 2 calls up and the 3rd call fails. This is because you can only have 2 concurrent calls using MSN on ISDN2. You will find you have a different number range for the second ISDN2 If you want to use both ISDN lines for incoming calls with the same number range then you will need to have the lines converted to 1 + 1 Auxiliary working and have the numbers delivered as DDI. Neil -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lee Norvall Sent: 18 May 2005 13:27 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] 2 x Eicon BRI ISDN devices (UK) Hi I can see what seems to be both devices in use, so I guess it must be down to the capi.conf (below), does this look correct ??? [interfaces] msn=292880 incomingmsn=292880, 292881, 292882, 292883, 292884, 292885, 292886, 292887, 292888, 292889 outgoingmsn=292880 controller=1 softdtmf=1 ;accountcode= context=demo ;echosquelch=1 ;echocancel=yes ;echotail=64 ;callgroup=1 ;deflect=12345678 devices=2 msn=292xxx incomingmsn=292880, 292881, 292882, 292883, 292884, 292885, 292886, 292887, 292888, 292889 outgoingmsn=292880 controller=2 softdtmf=1 ;accountcode= context=demo ;echosquelch=1 ;echocancel=yes ;echotail=64 ;callgroup=1 ;deflect=12345678 devices=2 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Armin Schindler Sent: 18 May 2005 12:50 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] 2 x Eicon BRI ISDN devices (UK) On Wed, 18 May 2005, Lee Norvall wrote: Hi I have installed 2 x Eicon BRI ISDN cards into a Suse 9.2 server. We can use all 4 lines for out going calls fine, but on incoming we can only use 2. On calling in using the main msn, the 3rd line gives a an engaged signal. I have unplugged 1 of the cards, and the other card takes the 2 calls. I then swapped this around, and this also works fine. But when using both cards, we can only use 2 line in. There are two possibilities: 1) your Telco doesn't send the 3rd call to your other line. You can verify that by using divactrl mlog -c 1 -o (diva_idi module must be leaded) and see if an incoming call is shown. (use -c 2 for the second card) 2) your configuration of chan_capi is not correct and the 3rd call is ignored/rejected. If you don't use DIVA Server cards with CAPI, forget this mail. 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 ___ 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] 2 x Eicon BRI ISDN devices (UK)
Hi I have installed 2 x Eicon BRI ISDN cards into a Suse 9.2 server. We can use all 4 lines for out going calls fine, but on incoming we can only use 2. On calling in using the main msn, the 3rd line gives a an engaged signal. I have unplugged 1 of the cards, and the other card takes the 2 calls. I then swapped this around, and this also works fine. But when using both cards, we can only use 2 line in. Any ideas??? Rgds ___ 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] 2 x Eicon BRI ISDN devices (UK)
Hi I can see what seems to be both devices in use, so I guess it must be down to the capi.conf (below), does this look correct ??? [interfaces] msn=292880 incomingmsn=292880, 292881, 292882, 292883, 292884, 292885, 292886, 292887, 292888, 292889 outgoingmsn=292880 controller=1 softdtmf=1 ;accountcode= context=demo ;echosquelch=1 ;echocancel=yes ;echotail=64 ;callgroup=1 ;deflect=12345678 devices=2 msn=292xxx incomingmsn=292880, 292881, 292882, 292883, 292884, 292885, 292886, 292887, 292888, 292889 outgoingmsn=292880 controller=2 softdtmf=1 ;accountcode= context=demo ;echosquelch=1 ;echocancel=yes ;echotail=64 ;callgroup=1 ;deflect=12345678 devices=2 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Armin Schindler Sent: 18 May 2005 12:50 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] 2 x Eicon BRI ISDN devices (UK) On Wed, 18 May 2005, Lee Norvall wrote: Hi I have installed 2 x Eicon BRI ISDN cards into a Suse 9.2 server. We can use all 4 lines for out going calls fine, but on incoming we can only use 2. On calling in using the main msn, the 3rd line gives a an engaged signal. I have unplugged 1 of the cards, and the other card takes the 2 calls. I then swapped this around, and this also works fine. But when using both cards, we can only use 2 line in. There are two possibilities: 1) your Telco doesn't send the 3rd call to your other line. You can verify that by using divactrl mlog -c 1 -o (diva_idi module must be leaded) and see if an incoming call is shown. (use -c 2 for the second card) 2) your configuration of chan_capi is not correct and the 3rd call is ignored/rejected. If you don't use DIVA Server cards with CAPI, forget this mail. 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] X101P on a UK BT line ---- txgain issue
I am having the same, some people can just about hear me while others do not say a thing or it is fine. I can hear them fine. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Stenton Sent: 24 June 2004 17:11 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] X101P on a UK BT line txgain issue I am finding that I have to increase the txgain in zapata.conf to 8 when my X101P is connected to my BT phone line, otherwise people can hardly hear me. This then gives echo issues. Do other people have the same problem on BT lines. I was wondering if I should give BT a call and get them to increase the gain on the line. Strange though as the rxgain is OK and I don't have this problem with an ordinary phone. Chris ___ 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 ___ 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] X100P Noise
Hi I remembered that I had disabled USB 2.0 on the motherboard last week. I rebooted the server, enabled USB 2.0 and all seems a lot better. I guess this may have cleared any conflicts.. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ryan Courtnage Sent: 23 June 2004 16:13 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] X100P Noise On Wednesday 23 June 2004 08:17, Lee Norvall wrote: I have 2 x X100P on UK BT, both have been working fine for a while, but now I have started to get a beeping sound my end every 8/10 sec, and break-up in the voice call inbound/outbound. Any ideas??? Sounds like your x100p cards are sharing interrupts with another device. Check /proc/interrupts. ___ 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 ___ 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
[Asterisk-Users] Capture user input
Title: Message Hi I just wanted to know if anyone has done the following, or knows how to. When a customer dials into *, we would then ask them to for an account number (which they would type in with the key pad), we would then ask them to select and option (1 to 9). We would then like to capture this information and send it in an email. All hints 'n tips welcome...
RE: [Asterisk-Users] DTMF and SIP
Title: Message Hi I am using the latest cvs version. I can call other remote systems via PSTN and navigate menu systems with key presses ok ! I am using IP2006 SIP phone. Rgds -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Santiago AguiarSent: 04 June 2004 21:03To: [EMAIL PROTECTED]Subject: Re: [Asterisk-Users] DTMF and SIPhi!I'm having the same problem, I'm connecting through a Planet VIP-450 ITG, and when I send a DTMF code I get a:WARNING: codec_ilbc.c:141 ilbctolin_framein: Huh? An ilbc frame that isn't a multiple of 50 bytes long from RTP (4)?I tried using different dtmf settings in sip.conf, but the message is still there. I don't have problems using a softphone...any ideas???saludos! santiago.Lee Norvall wrote: Hi Just tried that, and still the same with the same error! The spec for the phones includes rfc2833, so I don't think that is it. Rgds -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Justin Carlson Sent: 02 June 2004 19:23 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] DTMF and SIP have you tried commenting out the dtmf lines in your sip.conf we had similar problems with our snom 200's and after commenting out the dtmf lines in sip.conf asterisk reload they worked great :-) On Wed, 2004-06-02 at 11:36, Lee Norvall wrote: Hi I have 2 x SIP hand phones. I have set the DTMF to rfc2833 on the phones and tried both dtmfmode=rfc2833 and sipdtmfmode=rcf2833 (also tried inband) and I get the following error: june 2 17:21:10 WARNING[213006]: codec_ilbc.c:145 ilbctolin_framein: Huh? An ilbc frame that isn't a multiple of 50 bytes long from RTP (4)? This means that I cannot get access to voicemail from the handsets !!! Any clues??? ___ 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 ___ 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] DTMF and SIP
Hi Just tried that, and still the same with the same error! The spec for the phones includes rfc2833, so I don't think that is it. Rgds -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Justin Carlson Sent: 02 June 2004 19:23 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] DTMF and SIP have you tried commenting out the dtmf lines in your sip.conf we had similar problems with our snom 200's and after commenting out the dtmf lines in sip.conf asterisk reload they worked great :-) On Wed, 2004-06-02 at 11:36, Lee Norvall wrote: Hi I have 2 x SIP hand phones. I have set the DTMF to rfc2833 on the phones and tried both dtmfmode=rfc2833 and sipdtmfmode=rcf2833 (also tried inband) and I get the following error: june 2 17:21:10 WARNING[213006]: codec_ilbc.c:145 ilbctolin_framein: Huh? An ilbc frame that isn't a multiple of 50 bytes long from RTP (4)? This means that I cannot get access to voicemail from the handsets !!! Any clues??? ___ 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 ___ 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
