Re: [Asterisk-Users] X-Lite & Asterisk: Speex & iLBC not working?
If this may be of any use: I'm not an expert but I did the test with the FWD soft phone from X-ten and iLBC & SPX don't work. Asterisk wasn't between the connections. Just x-lite and fwd (who is an Asterisk Server?) The soft phone makes the connection but I can't hear any sound. Wim - Original Message - From: "Fran Boon" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Monday, January 26, 2004 3:01 PM Subject: [Asterisk-Users] X-Lite & Asterisk: Speex & iLBC not working? > This seems to have been reported before, but I've seen no resolution: > http://lists.digium.com/pipermail/asterisk-dev/2003-August/001470.html > http://lists.digium.com/pipermail/asterisk-users/2003-August/019091.html > http://lists.digium.com/pipermail/asterisk-users/2003-October/024472.html > > When forcing use of Speex, no sound comes out at all (Speex-1.0.3 on the > Asterisk server) > When forcing iLBC, there is some very garbled noise, but nothing > intelligible. > > Sniffing the packets, I can see that X-Lite & Asterisk have chosen > differing 'Payload type' numbers: > X-Lite: > a=rtpmap:97 speex/8000 > a=rtpmap:98 iLBC/8000 > Asterisk: > a=rtpmap:97 iLBC/8000 > a=rtpmap:110 SPEEX/8000 > > According to the Speex RFC, this is acceptable: > http://speex.org/drafts/draft-herlein-speex-rtp-profile-00.txt > "Dynamic payload type codes MUST be negotiated 'out-of-band' for the > assignment of a dynamic payload type from the range of 96-127." > > I'm wondering whether the system is at all case sensitive? > From the RFC: > "When conveying information by SDP [4], the encoding name SHALL be "speex"." > > NB Ethereal shows payload-type as being 97 when X-Lite reports iLBC & > 110 when X-Lite reports Speex, so the Asterisk numbers seem to 'win'. > > Any light shed on this would be great. > Whilst GSM is ok, it would be great to leverage the power of Speex :) > > Thanks a lot, > Fran. > ___ > 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] Unable to create channem of type 'Zap'
Can anyone help me, (after a two day search, also on the mailing list) I have the following situation: Asterisk works fine, until I added a FXO card. (Digium) When I tried to call to the pstn I have the following error Executing Dial("SIP/Phone2-fc49", "Zap/1/2355") in new stack NOTIVE[16401]: FILE APP_DIAL.C, LINE 516 (DIAL_EXEC): UNABLE TO CREATE CHANNEL OF TYPE 'ZAP' == Everyone is busy at this time When I start Asterisk I have no error Only the following isn't right: ZAP SHOW CHANNELS = No channels modprobe wcfxo = ok (no errors) I have following config. ZAPATA [channels]language=encontext=incomingsignalling=fxs_ksusecallerid=yeshidecallerid=nocallwaiting=yescallwaitingcallerid=yesthreewaycalling=yestransfer=yescancallforward=yescallreturn=yesechocancel=yesechocancelwhenbridged=yesrxgain=0.0txgain=0.0group=1pickupgroup=1immediate=yesmusiconhold=default channel => 1 ZAPTEL loadzone = usdefaultzone = usfxsks = 1 EXTENSION [incoming]exten => s,1,Dial(SIP/Phone1&SIP/Phone3,20,tr) [outgoing]exten => _0X.,1,Dial,Zap/1/${EXTEN:1} IN [SIP] include => outgoing I'm don't know what I can change to the config. Anyone an idea Thanks, Wim
Re: [Asterisk-Users] Unable to create channem of type 'Zap'
Made changes: 1) musiconhold= default channel => 1 2) reboot modprobe wcfxo = ok ztcfg -v result = 1 channel configured Try to dial, still the same problem. (error) Wim - Original Message - From: "Brent Franks" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Monday, February 23, 2004 9:19 PM Subject: RE: [Asterisk-Users] Unable to create channem of type 'Zap' > Make sure you run a ztcfg after you do a modprobe. > > ztcfg will configure (or bring up) the zap channels on zaptel interface > cards. Do this before starting * and after the modprobe. > > (You may also do a ztcfg -v to see whats configured) > > - Brent > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Wim Venneman > Sent: Monday, February 23, 2004 3:10 PM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] Unable to create channem of type 'Zap' > > Can anyone help me, (after a two day search, also on the mailing list) > I have the following situation: > Asterisk works fine, until I added a FXO card. (Digium) > When I tried to call to the pstn I have the following error > Executing Dial("SIP/Phone2-fc49", "Zap/1/2355") in new stack > NOTIVE[16401]: FILE APP_DIAL.C, LINE 516 (DIAL_EXEC): UNABLE TO CREATE > CHANNEL OF TYPE 'ZAP' > == Everyone is busy at this time > When I start Asterisk I have no error > Only the following isn't right: > ZAP SHOW CHANNELS = No channels > modprobe wcfxo = ok (no errors) > I have following config. > ZAPATA > [channels] > language=en > context=incoming > signalling=fxs_ks > usecallerid=yes > hidecallerid=no > callwaiting=yes > callwaitingcallerid=yes > threewaycalling=yes > transfer=yes > cancallforward=yes > callreturn=yes > echocancel=yes > echocancelwhenbridged=yes > rxgain=0.0 > txgain=0.0 > group=1 > pickupgroup=1 > immediate=yes > musiconhold=default channel => 1 > ZAPTEL > loadzone = us > defaultzone = us > fxsks = 1 > EXTENSION > [incoming] > exten => s,1,Dial(SIP/Phone1&SIP/Phone3,20,tr) > [outgoing] > exten => _0X.,1,Dial,Zap/1/${EXTEN:1} > IN [SIP] > include => outgoing > I'm don't know what I can change to the config. > Anyone an idea > Thanks, > Wim > > ___ > 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] Unable to create channem of type 'Zap'
Thanks for the help ! Made changes, still the same message. I have two NIC's with IRQ 11 The FXO card has IRQ10 (and no other card has IRQ10) Wim - Original Message - From: "Brent Franks" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Monday, February 23, 2004 10:21 PM Subject: RE: [Asterisk-Users] Unable to create channem of type 'Zap' > Wim, I made some changes to your Zapata.conf and zaptel.conf config > files below. > > Hope this helps. > > Also, do a less /proc/interrupts and see if the card is on it's own IRQ. > > - Brent > > -Original Message----- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Wim Venneman > Sent: Monday, February 23, 2004 3:10 PM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] Unable to create channem of type 'Zap' > > Can anyone help me, (after a two day search, also on the mailing list) > I have the following situation: > Asterisk works fine, until I added a FXO card. (Digium) > When I tried to call to the pstn I have the following error > Executing Dial("SIP/Phone2-fc49", "Zap/1/2355") in new stack > NOTIVE[16401]: FILE APP_DIAL.C, LINE 516 (DIAL_EXEC): UNABLE TO CREATE > CHANNEL OF TYPE 'ZAP' > == Everyone is busy at this time > When I start Asterisk I have no error > Only the following isn't right: > ZAP SHOW CHANNELS = No channels > modprobe wcfxo = ok (no errors) > I have following config. > ZAPATA > [channels] > language=en > group=1 > pickupgroup=1 > context=incoming > signalling=fxs_ks > usecallerid=yes > hidecallerid=no > callwaiting=yes > callwaitingcallerid=yes > threewaycalling=yes > transfer=yes > cancallforward=yes > callreturn=yes > echocancel=yes > echocancelwhenbridged=yes > rxgain=0.0 > txgain=0.0 > immediate=yes > musiconhold=default > channel = 1 > > ZAPTEL > loadzone = us > defaultzone = us > fxsks = 1 > > EXTENSION > [incoming] > exten => s,1,Dial(SIP/Phone1&SIP/Phone3,20,tr) > [outgoing] > exten => _0X.,1,Dial,Zap/1/${EXTEN:1} > > IN [SIP] > include => outgoing > I'm don't know what I can change to the config. > Anyone an idea > Thanks, > Wim > > ___ > 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] Unable to create channem of type 'Zap'
Thanks Derek, Changed the channel = 1 to channel => 1, makes no difference. Wim - Original Message - From: "Derek Samford" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Cc: <[EMAIL PROTECTED]> Sent: Tuesday, February 24, 2004 6:38 PM Subject: RE: [Asterisk-Users] Unable to create channem of type 'Zap' > Wim, > Made one more change below in Zapata.conf > It should be channel => 1 > > -Original Message- > From: Wim Venneman [mailto:[EMAIL PROTECTED] > Sent: Monday, February 23, 2004 4:46 PM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] Unable to create channem of type 'Zap' > > Thanks for the help ! > > Made changes, still the same message. > I have two NIC's with IRQ 11 > The FXO card has IRQ10 (and no other card has IRQ10) > > Wim > > > - Original Message - > From: "Brent Franks" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Sent: Monday, February 23, 2004 10:21 PM > Subject: RE: [Asterisk-Users] Unable to create channem of type 'Zap' > > > > Wim, I made some changes to your Zapata.conf and zaptel.conf config > > files below. > > > > Hope this helps. > > > > Also, do a less /proc/interrupts and see if the card is on it's own > IRQ. > > > > - Brent > > > > -Original Message- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] On Behalf Of Wim > Venneman > > Sent: Monday, February 23, 2004 3:10 PM > > To: [EMAIL PROTECTED] > > Subject: [Asterisk-Users] Unable to create channem of type 'Zap' > > > > Can anyone help me, (after a two day search, also on the mailing list) > > I have the following situation: > > Asterisk works fine, until I added a FXO card. (Digium) > > When I tried to call to the pstn I have the following error > > Executing Dial("SIP/Phone2-fc49", "Zap/1/2355") in new stack > > NOTIVE[16401]: FILE APP_DIAL.C, LINE 516 (DIAL_EXEC): UNABLE TO CREATE > > CHANNEL OF TYPE 'ZAP' > > == Everyone is busy at this time > > When I start Asterisk I have no error > > Only the following isn't right: > > ZAP SHOW CHANNELS = No channels > > modprobe wcfxo = ok (no errors) > > I have following config. > > ZAPATA > > [channels] > > language=en > > group=1 > > pickupgroup=1 > > context=incoming > > signalling=fxs_ks > > usecallerid=yes > > hidecallerid=no > > callwaiting=yes > > callwaitingcallerid=yes > > threewaycalling=yes > > transfer=yes > > cancallforward=yes > > callreturn=yes > > echocancel=yes > > echocancelwhenbridged=yes > > rxgain=0.0 > > txgain=0.0 > > immediate=yes > > musiconhold=default > > channel => 1 > > > > ZAPTEL > > loadzone = us > > defaultzone = us > > fxsks = 1 > > > > EXTENSION > > [incoming] > > exten => s,1,Dial(SIP/Phone1&SIP/Phone3,20,tr) > > [outgoing] > > exten => _0X.,1,Dial,Zap/1/${EXTEN:1} > > > > IN [SIP] > > include => outgoing > > I'm don't know what I can change to the config. > > Anyone an idea > > Thanks, > > Wim > > > > ___ > > 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 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] Unable to create channem of type 'Zap'
Brent, Zaptel configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channel configured. Registered tone zone 0 (United States / North America) Wim - Original Message - From: "Brent Franks" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Tuesday, February 24, 2004 8:13 PM Subject: Re: [Asterisk-Users] Unable to create channem of type 'Zap' > Wim, > > What happens when you do a ztcfg -vv > > - Brent > > On Tue, 24 Feb 2004, Wim Venneman wrote: > > > Thanks Derek, > > > > Changed the channel = 1 to channel => 1, makes no difference. > > > > Wim > > > > - Original Message - > > From: "Derek Samford" <[EMAIL PROTECTED]> > > To: <[EMAIL PROTECTED]> > > Cc: <[EMAIL PROTECTED]> > > Sent: Tuesday, February 24, 2004 6:38 PM > > Subject: RE: [Asterisk-Users] Unable to create channem of type 'Zap' > > > > > > > Wim, > > > Made one more change below in Zapata.conf > > > It should be channel => 1 > > > > > > -Original Message- > > > From: Wim Venneman [mailto:[EMAIL PROTECTED] > > > Sent: Monday, February 23, 2004 4:46 PM > > > To: [EMAIL PROTECTED] > > > Subject: Re: [Asterisk-Users] Unable to create channem of type 'Zap' > > > > > > Thanks for the help ! > > > > > > Made changes, still the same message. > > > I have two NIC's with IRQ 11 > > > The FXO card has IRQ10 (and no other card has IRQ10) > > > > > > Wim > > > > > > > > > - Original Message - > > > From: "Brent Franks" <[EMAIL PROTECTED]> > > > To: <[EMAIL PROTECTED]> > > > Sent: Monday, February 23, 2004 10:21 PM > > > Subject: RE: [Asterisk-Users] Unable to create channem of type 'Zap' > > > > > > > > > > Wim, I made some changes to your Zapata.conf and zaptel.conf config > > > > files below. > > > > > > > > Hope this helps. > > > > > > > > Also, do a less /proc/interrupts and see if the card is on it's own > > > IRQ. > > > > > > > > - Brent > > > > > > > > -Original Message- > > > > From: [EMAIL PROTECTED] > > > > [mailto:[EMAIL PROTECTED] On Behalf Of Wim > > > Venneman > > > > Sent: Monday, February 23, 2004 3:10 PM > > > > To: [EMAIL PROTECTED] > > > > Subject: [Asterisk-Users] Unable to create channem of type 'Zap' > > > > > > > > Can anyone help me, (after a two day search, also on the mailing list) > > > > I have the following situation: > > > > Asterisk works fine, until I added a FXO card. (Digium) > > > > When I tried to call to the pstn I have the following error > > > > Executing Dial("SIP/Phone2-fc49", "Zap/1/2355") in new stack > > > > NOTIVE[16401]: FILE APP_DIAL.C, LINE 516 (DIAL_EXEC): UNABLE TO CREATE > > > > CHANNEL OF TYPE 'ZAP' > > > > == Everyone is busy at this time > > > > When I start Asterisk I have no error > > > > Only the following isn't right: > > > > ZAP SHOW CHANNELS = No channels > > > > modprobe wcfxo = ok (no errors) > > > > I have following config. > > > > ZAPATA > > > > [channels] > > > > language=en > > > > group=1 > > > > pickupgroup=1 > > > > context=incoming > > > > signalling=fxs_ks > > > > usecallerid=yes > > > > hidecallerid=no > > > > callwaiting=yes > > > > callwaitingcallerid=yes > > > > threewaycalling=yes > > > > transfer=yes > > > > cancallforward=yes > > > > callreturn=yes > > > > echocancel=yes > > > > echocancelwhenbridged=yes > > > > rxgain=0.0 > > > > txgain=0.0 > > > > immediate=yes > > > > musiconhold=default > > > > channel => 1 > > > > > > > > ZAPTEL > > > > loadzone = us > > > > defaultzone = us > > > > fxsks = 1 > > > > > > > > EXTENSION > > > > [incoming] > > > > exten => s,1,Dial(SIP/Phone1&SIP/Phone3,20,tr) > > > > [outgoing] > > > > exten => _0X.,1,Dial,Zap/1/${EXTEN:1} > > > > > > > > IN [SIP] > &
Re: [Asterisk-Users] Unable to create channel of type 'Zap'
Dave, I have fxsks = 1 (fxsls is for loopstart prot. and it doesn't work for me) Wim - Original Message - From: "David J Carter" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Tuesday, February 24, 2004 8:33 PM Subject: RE: [Asterisk-Users] Unable to create channel of type 'Zap' > I had this after my last CVS update. > > A line in Zaptel.conf was set to fxsls=1 instaead of fxsks=1 > > Dave > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Wim Venneman > Sent: 24 February 2004 19:17 > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] Unable to create channem of type 'Zap' > > > Thanks Derek, > > Changed the channel = 1 to channel => 1, makes no difference. > > Wim > > - Original Message - > From: "Derek Samford" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Cc: <[EMAIL PROTECTED]> > Sent: Tuesday, February 24, 2004 6:38 PM > Subject: RE: [Asterisk-Users] Unable to create channem of type 'Zap' > > > > Wim, > > Made one more change below in Zapata.conf > > It should be channel => 1 > > > > -Original Message- > > From: Wim Venneman [mailto:[EMAIL PROTECTED] > > Sent: Monday, February 23, 2004 4:46 PM > > To: [EMAIL PROTECTED] > > Subject: Re: [Asterisk-Users] Unable to create channem of type 'Zap' > > > > Thanks for the help ! > > > > Made changes, still the same message. > > I have two NIC's with IRQ 11 > > The FXO card has IRQ10 (and no other card has IRQ10) > > > > Wim > > > > > > - Original Message - > > From: "Brent Franks" <[EMAIL PROTECTED]> > > To: <[EMAIL PROTECTED]> > > Sent: Monday, February 23, 2004 10:21 PM > > Subject: RE: [Asterisk-Users] Unable to create channem of type 'Zap' > > > > > > > Wim, I made some changes to your Zapata.conf and zaptel.conf config > > > files below. > > > > > > Hope this helps. > > > > > > Also, do a less /proc/interrupts and see if the card is on it's own > > IRQ. > > > > > > - Brent > > > > > > -Original Message- > > > From: [EMAIL PROTECTED] > > > [mailto:[EMAIL PROTECTED] On Behalf Of Wim > > Venneman > > > Sent: Monday, February 23, 2004 3:10 PM > > > To: [EMAIL PROTECTED] > > > Subject: [Asterisk-Users] Unable to create channem of type 'Zap' > > > > > > Can anyone help me, (after a two day search, also on the mailing list) > > > I have the following situation: > > > Asterisk works fine, until I added a FXO card. (Digium) > > > When I tried to call to the pstn I have the following error > > > Executing Dial("SIP/Phone2-fc49", "Zap/1/2355") in new stack > > > NOTIVE[16401]: FILE APP_DIAL.C, LINE 516 (DIAL_EXEC): UNABLE TO CREATE > > > CHANNEL OF TYPE 'ZAP' > > > == Everyone is busy at this time > > > When I start Asterisk I have no error > > > Only the following isn't right: > > > ZAP SHOW CHANNELS = No channels > > > modprobe wcfxo = ok (no errors) > > > I have following config. > > > ZAPATA > > > [channels] > > > language=en > > > group=1 > > > pickupgroup=1 > > > context=incoming > > > signalling=fxs_ks > > > usecallerid=yes > > > hidecallerid=no > > > callwaiting=yes > > > callwaitingcallerid=yes > > > threewaycalling=yes > > > transfer=yes > > > cancallforward=yes > > > callreturn=yes > > > echocancel=yes > > > echocancelwhenbridged=yes > > > rxgain=0.0 > > > txgain=0.0 > > > immediate=yes > > > musiconhold=default > > > channel => 1 > > > > > > ZAPTEL > > > loadzone = us > > > defaultzone = us > > > fxsks = 1 > > > > > > EXTENSION > > > [incoming] > > > exten => s,1,Dial(SIP/Phone1&SIP/Phone3,20,tr) > > > [outgoing] > > > exten => _0X.,1,Dial,Zap/1/${EXTEN:1} > > > > > > IN [SIP] > > > include => outgoing > > > I'm don't know what I can change to the config. > > > Anyone an idea > > > Thanks, > > > Wim > > > > > > ___ > > > Asterisk-Users mailing list > > > [EMAIL PROTECTED] > > > http://lists.digium.c
Re: [Asterisk-Users] Unable to create channel of type 'Zap'
Thanks Brent, You will never believe, it works. I copied a config from a friend (ZAPATA.CONF) and now it works fine. I can call to * from pstn and can call from * to pstn. I will take a look tomorrow what was wrong in my config and let you know. Wim - Original Message - From: "Brent Franks" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Tuesday, February 24, 2004 10:21 PM Subject: RE: [Asterisk-Users] Unable to create channel of type 'Zap' > The best thing I can recommend is giving Digium a call. That's part of > the reason to buy the cards from Digium, as the support is included free > with the hardware you purchase from them. Well worth the money. They > have an excellent support staff that should be able to see where the > problem is cropping up. > > - B > > > -Original Message- > > From: [EMAIL PROTECTED] [mailto:asterisk-users- > > [EMAIL PROTECTED] On Behalf Of Wim Venneman > > Sent: Tuesday, February 24, 2004 2:47 PM > > To: [EMAIL PROTECTED] > > Subject: Re: [Asterisk-Users] Unable to create channel of type 'Zap' > > > > Dave, > > > > I have fxsks = 1 > > (fxsls is for loopstart prot. and it doesn't work for me) > > > > Wim > > > > - Original Message - > > From: "David J Carter" <[EMAIL PROTECTED]> > > To: <[EMAIL PROTECTED]> > > Sent: Tuesday, February 24, 2004 8:33 PM > > Subject: RE: [Asterisk-Users] Unable to create channel of type 'Zap' > > > > > > > I had this after my last CVS update. > > > > > > A line in Zaptel.conf was set to fxsls=1 instaead of fxsks=1 > > > > > > Dave > > > > > > -Original Message- > > > From: [EMAIL PROTECTED] > > > [mailto:[EMAIL PROTECTED] Behalf Of Wim > Venneman > > > Sent: 24 February 2004 19:17 > > > To: [EMAIL PROTECTED] > > > Subject: Re: [Asterisk-Users] Unable to create channem of type 'Zap' > > > > > > > > > Thanks Derek, > > > > > > Changed the channel = 1 to channel => 1, makes no difference. > > > > > > Wim > > > > > > - Original Message - > > > From: "Derek Samford" <[EMAIL PROTECTED]> > > > To: <[EMAIL PROTECTED]> > > > Cc: <[EMAIL PROTECTED]> > > > Sent: Tuesday, February 24, 2004 6:38 PM > > > Subject: RE: [Asterisk-Users] Unable to create channem of type 'Zap' > > > > > > > > > > Wim, > > > > Made one more change below in Zapata.conf > > > > It should be channel => 1 > > > > > > > > -Original Message- > > > > From: Wim Venneman [mailto:[EMAIL PROTECTED] > > > > Sent: Monday, February 23, 2004 4:46 PM > > > > To: [EMAIL PROTECTED] > > > > Subject: Re: [Asterisk-Users] Unable to create channem of type > 'Zap' > > > > > > > > Thanks for the help ! > > > > > > > > Made changes, still the same message. > > > > I have two NIC's with IRQ 11 > > > > The FXO card has IRQ10 (and no other card has IRQ10) > > > > > > > > Wim > > > > > > > > > > > > - Original Message - > > > > From: "Brent Franks" <[EMAIL PROTECTED]> > > > > To: <[EMAIL PROTECTED]> > > > > Sent: Monday, February 23, 2004 10:21 PM > > > > Subject: RE: [Asterisk-Users] Unable to create channem of type > 'Zap' > > > > > > > > > > > > > Wim, I made some changes to your Zapata.conf and zaptel.conf > config > > > > > files below. > > > > > > > > > > Hope this helps. > > > > > > > > > > Also, do a less /proc/interrupts and see if the card is on it's > own > > > > IRQ. > > > > > > > > > > - Brent > > > > > > > > > > -Original Message- > > > > > From: [EMAIL PROTECTED] > > > > > [mailto:[EMAIL PROTECTED] On Behalf Of Wim > > > > Venneman > > > > > Sent: Monday, February 23, 2004 3:10 PM > > > > > To: [EMAIL PROTECTED] > > > > > Subject: [Asterisk-Users] Unable to create channem of type 'Zap' > > > > > > > > > > Can anyone help me, (after a two day search, also on the mailing > > list) > > > > > I have the f
Re: [Asterisk-Users] Dial up adapter
Maybe this??: http://www.grandstream.com/y-product.htm - Original Message - From: "Jason Miller" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Monday, March 01, 2004 10:53 PM Subject: [Asterisk-Users] Dial up adapter I was wondering if anyone has used an adapter to dial up to a local internet service then used the VOIP phone instead of needing a computer. If so what product do you suggest? An idea of what I am looking for, configure the device which has a analog port to dial said ISP and authenticate Has an ethernet port to hook up to the phone Or am I just dreaming up a new product to market? Jason ___ 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] Host unspecified ??
Dear, When I start asterisk -vvgrc and I ask 'sip show peers', I don't get the ip adress in the 'Host" field. Name = phone1 and phone2 Host=unspecified mask 255.255.255.255 port = 0 status = unmonitored I can ping the two phone's and get a reply (also from the laptop) phone ip adres 192.168.10.12 and 192.168.10.13 (server 192.168.10.11and laptop 192.168.10.14) hardware config: server - phone1 - phone2 - laptop configurations used SIP.CONF [phone1] type=friend host=dynamic defaultip=192.168.10.12 dtmfmode=info mailbox=1000 context=sip callerid="phone1"<100> [phone2] type=friend host=dynamic defaultip=192.168.10.13 dtmfmode=info mailbox=1000 context=sip callerid="phone2"<200> EXTENSIONS.CONF exten => 100,1,dial(sip/phone1,20,tr) exten => 200,1,dial(sip/phone2,20,tr) Wim
Re: [Asterisk-Users] Host unspecified ??
Dear, I changed the host to a fixed ip address (host1=192.168.10.12 and host2=192.168.10.13) now the ip address shows up in the 'host' field = ok. Try to call, no succes, nothing happens! What's wrong? Wim - Original Message - From: "Florian Overkamp" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Thursday, October 30, 2003 9:25 AM Subject: Re: [Asterisk-Users] Host unspecified ?? > Hi Wim, > > It doesnt show the host (at least) until the phones have registered with > asterisk, because you've set the host to dynamic in your config. Either > verify if the phones register with asterisk, or set the host to their > static IP-adresses. > > Best regards, > Florian > > At 19:51 29-10-2003 +0100, you wrote: > >When I start asterisk -vvgrc and I ask 'sip show peers', I don't get > >the ip adress in the 'Host" field. > > > >Name = phone1 and phone2 > >Host=unspecified > >mask 255.255.255.255 > >port = 0 > >status = unmonitored > > > >I can ping the two phone's and get a reply (also from the laptop) > >phone ip adres 192.168.10.12 and 192.168.10.13 (server 192.168.10.11and > >laptop 192.168.10.14) > >hardware config: server - phone1 - phone2 - laptop > > > >configurations used > > > > SIP.CONF > > > >[phone1] > >type=friend > >host=dynamic > >defaultip=192.168.10.12 > >dtmfmode=info > >mailbox=1000 > >context=sip > >callerid="phone1"<100> > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Host unspecified ??
Hi, Here is wath happens: Asterisk*CLI>sip debug SIP Debugging Enabled Asterisk*CLI> Nothing happens when I use 'sip debug'. It seems that sip doesn't work. Wim - Original Message - From: "Florian Overkamp" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Thursday, October 30, 2003 9:02 PM Subject: Re: [Asterisk-Users] Host unspecified ?? > Hi Wim, > > Citeren Wim Venneman <[EMAIL PROTECTED]>: > > > I changed the host to a fixed ip address (host1=192.168.10.12 and > > host2=192.168.10.13) now the ip address shows up in the 'host' field = ok. > > Try to call, no succes, nothing happens! > > > > What's wrong? > > That's a bit difficult to determine without more info. Could you enter the > command 'sip debug', try calling with the phones and then copy what the > console says ? Feel free to send off-list :-) > > -- > Met vriendelijke groet, > Florian Overkamp > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Grandstream problem
Hi, I installed Asterisk an all works fine exept for Grandstream. When I call with a softphone (ex X-ten) to a Grandstream (BudgetTone-100), I can make a conversation. = ok When I call to a softphone with a Grandstream I can pich up the call with the softphone but the Grandstream keeps ringing like on the other site you didn't pick up the phone.(even if you do so) It's the same when I call between two Grandstream phone's. Call from phone1 to phone 2, I pick up phone2 and afther 3 seconds I get congestion tone from both phone's. Info from command *CLI> -- Executing Dial("SIP/phone2-a030a", "sip/phone1") in new stack -- Called phone1 -- SIP/phone1-663a is ringing -- SIP/phone1-663a answered SIP/phone2-a030a -- Attempting native bridge of SIP/phone2-a030a and SIP/phone1-663a == Spawn extension (sip, 1,1) exited non-zero on 'SIP/phone2-a030a' and I get congestion Can anyone give me a direction to solve my problem? Thanks in advance, Wim
Re: [Asterisk-Users] Grandstream problem
Thanks William, Works fine now. Wim - Original Message - From: William Carlson To: [EMAIL PROTECTED] Sent: Thursday, November 06, 2003 9:43 PM Subject: Re: [Asterisk-Users] Grandstream problem try disallow=all allow=ulaw under the general section of sip.conf that half fixes it for me calls between phones work but talking to asterisk has some problems. - Original Message - From: Wim Venneman To: [EMAIL PROTECTED] Sent: Thursday, November 06, 2003 2:29 PM Subject: [Asterisk-Users] Grandstream problem Hi, I installed Asterisk an all works fine exept for Grandstream. When I call with a softphone (ex X-ten) to a Grandstream (BudgetTone-100), I can make a conversation. = ok When I call to a softphone with a Grandstream I can pich up the call with the softphone but the Grandstream keeps ringing like on the other site you didn't pick up the phone.(even if you do so) It's the same when I call between two Grandstream phone's. Call from phone1 to phone 2, I pick up phone2 and afther 3 seconds I get congestion tone from both phone's. Info from command *CLI> -- Executing Dial("SIP/phone2-a030a", "sip/phone1") in new stack -- Called phone1 -- SIP/phone1-663a is ringing -- SIP/phone1-663a answered SIP/phone2-a030a -- Attempting native bridge of SIP/phone2-a030a and SIP/phone1-663a == Spawn extension (sip, 1,1) exited non-zero on 'SIP/phone2-a030a' and I get congestion Can anyone give me a direction to solve my problem? Thanks in advance, Wim
[Asterisk-Users] SIP (peer to peer?)
Hi all,Has anyone have an idea why, if you capture the files on a Asterisk network (ex with Ethereal) you always see the communication between the two sip phones( hard or soft) passing through the asterisk server (on UDP layer) Isn't SIP a protocol that (after that it has established the call) , he connects the two users with each other? Maybe a stupid question, but I'm not a SIP expert. Thank you for your help. Wim
[Asterisk-Users] Dial plan
Hi, For our study we are searching for a (working) dial plan that is used in a Cisco call manager or other PBX, to compare it with Asterisk. If anyone has a (simple) working example, can I have a copy? Thanks
Re: [Asterisk-Users] Re: Analogue telephone cards for the UK
Paul, http://www.voip-info.org/wiki-Asterisk Wim - Original Message - From: Paul Tyreman To: [EMAIL PROTECTED] Sent: Saturday, April 10, 2004 6:47 PM Subject: Re: [Asterisk-Users] Re: Analogue telephone cards for the UK Sorry to sound stupid, but where can I get copied of the Asterisk manual ? What is the VoIP wiki and where can I get that too ? Thanks, Paul. -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Iain StevensonPosted At: 10 April 2004 17:20Posted To: Asterisk-UsersConversation: [Asterisk-Users] Re: Analogue telephone cards for the UKSubject: Re: [Asterisk-Users] Re: Analogue telephone cards for the UK ... Most is explained in the asterisk manual or on the VoIP wiki. Iain
Re: [Asterisk-Users] sip software
www.xten.com http://www.sjlabs.com/ - Original Message - From: "James Moran" <[EMAIL PROTECTED]> To: "Asterisk" <[EMAIL PROTECTED]> Sent: Wednesday, April 14, 2004 8:52 PM Subject: [Asterisk-Users] sip software > Anyone have any suggestions on free sip phone software for windows?? > Only have one IP phone and want to have one other computer hooked up to > my Asterisk box for testing. > > ___ > 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