Re: [asterisk-users] fax problem
Thank you francois!
Where could you find that info ?
-Message d'origine-
De : asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] De la part de F6HQZ
Envoyé : mercredi 23 décembre 2009 22:46
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [asterisk-users] fax problem
Oops !
The sendmail macro was missing, sorry !
[macro-Sendmail]
;===
; ARG1 = Address To
; ARG2 = Address From
; ARG3 = File attachment
; ARG4 = Pages Qty
; ARG5 = Rate
; ARG6 = HeaderInfo
; ARG7 = RemoteID
; ARG8 = Resolution
;===
exten = s,1,NoOp( SENDMAIL )
exten = s,n,NoOp(To:${ARG1} From:${ARG2} Subject:Fax de ${ARG6}
Attach:${ARG3} Pg:${ARG4} Rate:${ARG5} HeaderInfo:${ARG6}
RemoteID:${ARG7} Res:${ARG8})
exten = s,n,System(echo Entete FAX : ${ARG6} - ${ARG4} pages -
Rate:${ARG5} - CID:${ARG7}, Resolution : ${ARG8}|/bin/mailx -s
FAX de : ${ARG6} - CID : ${ARG7} -a ${ARG3} -r ${ARG2} ${ARG1})
exten = s,n,NoOp( SENT )
exten = s,n,System(rm ${ARG3})
-Message d'origine-
De : F6HQZ [mailto:f6hq...@hamwlan.net]
Envoyé : mercredi 23 décembre 2009 22:44
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : RE: [asterisk-users] fax problem
Hi Francois,
here is Francois too ;-)
Check that :
[fax-outbound-calls]
exten = _X.,1,Dial(${ACROPOLIS}/${EXTEN},,G(fax-tx^send^1))
[fax-tx]
exten = send,1,NoOp( SENDING FAX )
exten = send,n,Set(FaxTxDir=/var/spool/fax/tx/)
exten = send,n,Set(FAXFILEPDF=fax-${FAXCOUNT}-tx.pdf)
exten = send,n,Wait(6)
exten = send,n,Set(GLOBAL(FAXCOUNT)=$[ ${GLOBAL(FAXCOUNT)} + 1 ])
exten = send,n,Set(FAXCOUNT=${GLOBAL(FAXCOUNT)})
exten = send,n,Set(FAXFILE=test.tif)
; Set FAXOPTs
exten = send,n,NoOp( SETTING FAXOPT )
exten = send,n,Set(FAXOPT(filename)=${FAXFILE})
exten = send,n,Set(FAXOPT(ecm)=yes)
exten = send,n,Set(FAXOPT(headerinfo)=Fax from ${GLOBAL(LASTFAXCALLERNAME)}
at ${GLOBAL(LASTFAXCALLERNUM)} was received.)
exten = send,n,Set(FAXOPT(localstationid)=0170619058)
exten = send,n,Set(FAXOPT(maxrate)=14400)
exten = send,n,Set(FAXOPT(minrate)=2400)
; Send the fax
exten = send,n,NoOp( SENDING FAX )
exten = send,n,SendFAX(${FaxTxDir}${FAXFILE}|d)
; Hangup! Print FAXOPTs
exten = h,1,NoOp(FAXOPT(ecm) : ${FAXOPT(ecm)})
exten = h,n,NoOp(FAXOPT(filename) : ${FAXOPT(filename)})
exten = h,n,NoOp(FAXOPT(headerinfo) : ${FAXOPT(headerinfo)})
exten = h,n,NoOp(FAXOPT(localstationid) : ${FAXOPT(localstationid)})
exten = h,n,NoOp(FAXOPT(maxrate) : ${FAXOPT(maxrate)})
exten = h,n,NoOp(FAXOPT(minrate) : ${FAXOPT(minrate)})
exten = h,n,NoOp(FAXOPT(pages) : ${FAXOPT(pages)})
exten = h,n,NoOp(FAXOPT(rate) : ${FAXOPT(rate)})
exten = h,n,NoOp(FAXOPT(remotestationid) : ${FAXOPT(remotestationid)})
exten = h,n,NoOp(FAXOPT(resolution) : ${FAXOPT(resolution)})
exten = h,n,NoOp(FAXOPT(status) : ${FAXOPT(status)})
exten = h,n,NoOp(FAXOPT(statusstr) : ${FAXOPT(statusstr)})
exten = h,n,NoOp(FAXOPT(error) : ${FAXOPT(error)})
; Sendmail options for reports by email :
exten = h,n,System(/usr/bin/tiff2pdf -o ${FaxTxDir}${FAXFILEPDF}
${FaxRxDir}${FAXFILE})
exten =
h,n,macro(Sendmail,postmas...@acropolis.fr,aster...@acropolis.fr,${FaxRxDir}
${FAXFILEPDF},${FAXOPT(pages)},${FAXOPT(rate)},${FAXOPT(
headerinfo)},${FAXOPT(remotestationid)},${FAXOPT(resolution)})
Mainly extracted from the Digium FFA manual.
I hope this can help you.
Best Regards,
Francois
On Wed, Dec 23, 2009 at 10:49 AM, BERGANZ François
franc...@acropolistelecom.net wrote:
Hello,
I need to send a tiff via fax with my asterisk 1.6.1.0.
I tried in the dialplan
[default]
exten = _X.,1,SendFax(/root/test.tiff)
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[asterisk-users] fax problem
Hello, I need to send a tiff via fax with my asterisk 1.6.1.0. I tried in the dialplan [default] exten = _X.,1,SendFax(/root/test.tiff) but I have: salledeconf1*CLI console dial 1...@default [Dec 23 16:24:22] WARNING[31739]: chan_oss.c:492 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory -- Executing [...@default:1] SendFAX(Console/dsp, /root/test.tiff) in new stack Console call has been answered [Dec 23 16:24:22] NOTICE[31739]: console_video.c:133 console_video_start: voice only, console video support not present [Dec 23 16:24:23] WARNING[31745]: chan_oss.c:492 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory [Dec 23 16:24:24] WARNING[31745]: chan_oss.c:492 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory [Dec 23 16:24:25] WARNING[31745]: chan_oss.c:492 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory [Dec 23 16:24:26] WARNING[31745]: chan_oss.c:492 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory Know you why? Please help me ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fax problem
The problem isnt in my tiff images, I could use it with a paton. I still stryed to stand on one foot offcourse J De : asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] De la part de Danny Nicholas Envoyé : mercredi 23 décembre 2009 17:58 À : 'Asterisk Users Mailing List - Non-Commercial Discussion' Objet : Re: [asterisk-users] fax problem Did you stand on one foot and hold out your tongue when you made the tiff?? J The tiff has to be a very specific format I spent days making my output tiffs match the format of a received tiff I was able to send. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of BERGANZ François Sent: Wednesday, December 23, 2009 10:50 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] fax problem Hello, I need to send a tiff via fax with my asterisk 1.6.1.0. I tried in the dialplan [default] exten = _X.,1,SendFax(/root/test.tiff) but I have: salledeconf1*CLI console dial 1...@default [Dec 23 16:24:22] WARNING[31739]: chan_oss.c:492 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory -- Executing [...@default:1] SendFAX(Console/dsp, /root/test.tiff) in new stack Console call has been answered [Dec 23 16:24:22] NOTICE[31739]: console_video.c:133 console_video_start: voice only, console video support not present [Dec 23 16:24:23] WARNING[31745]: chan_oss.c:492 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory [Dec 23 16:24:24] WARNING[31745]: chan_oss.c:492 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory [Dec 23 16:24:25] WARNING[31745]: chan_oss.c:492 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory [Dec 23 16:24:26] WARNING[31745]: chan_oss.c:492 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory Know you why? Please help me ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk x-lite
Try tcpdump to see where RTP go from asterisk.
Configure your x-lite
Use stun server ?
P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.
De : asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] De la part de zehra yildiz
Envoyé : mardi 22 décembre 2009 10:26
À : asterisk-users@lists.digium.com
Objet : [asterisk-users] asterisk x-lite
Hello All,
I installed Asterisk 1.6.1.11 on Redhat 5.1. I use X-lite SoftPhone. The
softphone can call the other one but I can' t hear any voice. My
configuration files are below:
[r...@localhost asterisk]# cat sip.conf
[general]
canreinvite=yes
[1001]
username=1001
password=1001
type=friend
context=phones
host=dynamic
[1002]
callerid=1002
username=1002
password=1002
type=friend
context=phones
host=dynamic
[r...@localhost asterisk]# cat extensions.conf
[globals]
[general]
autofallthrough=yes
[default]
[incoming_calls]
[phones]
exten = _1XXX,1,NoOp()
exten = _1XXX,n,Dial(SIP/${EXTEN},30)
exten = _1XXX,n,Playback(the-party-you-are-callingis-curntly-unavail)
exten = _1XXX,n,Hangup()
PS: My sip server and softphones are in the same network subnet. There are
not any firewall or iptables rules. I tried the nat=yes parameter but no
changes.
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Re: [asterisk-users] asterisk x-lite
It is a nat problem
François BERGANZ
P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.
De : asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] De la part de zehra yildiz
Envoyé : mardi 22 décembre 2009 10:26
À : asterisk-users@lists.digium.com
Objet : [asterisk-users] asterisk x-lite
Hello All,
I installed Asterisk 1.6.1.11 on Redhat 5.1. I use X-lite SoftPhone. The
softphone can call the other one but I can' t hear any voice. My
configuration files are below:
[r...@localhost asterisk]# cat sip.conf
[general]
canreinvite=yes
[1001]
username=1001
password=1001
type=friend
context=phones
host=dynamic
[1002]
callerid=1002
username=1002
password=1002
type=friend
context=phones
host=dynamic
[r...@localhost asterisk]# cat extensions.conf
[globals]
[general]
autofallthrough=yes
[default]
[incoming_calls]
[phones]
exten = _1XXX,1,NoOp()
exten = _1XXX,n,Dial(SIP/${EXTEN},30)
exten = _1XXX,n,Playback(the-party-you-are-callingis-curntly-unavail)
exten = _1XXX,n,Hangup()
PS: My sip server and softphones are in the same network subnet. There are
not any firewall or iptables rules. I tried the nat=yes parameter but no
changes.
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[asterisk-users] possible to list a conference in the database?
Hello all, I try to do a web interface for my conferences. Is it possible to export in realtime all user in conferences as we could do with a CLImeetme list all ? François BERGANZ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MeetMe unactive pin access
Hello, I have conferences in my database. I need at some moments, to access the database without asking pin access, or with using cdr(accountcode). Is it possible? Thank you Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MeetMe unactive pin access
Sorry, there are some errors, here the right question: Hello, I have conferences in my database. I need at some moments, to access the CONFEERENCE without asking pin access, or with using cdr(accountcode). Is it possible? Thank you Cordialement, BERGANZ François cid:image001.gif@01C8F7CD.6BC1D2C0 http://www.acropolistelecom.net/ http://www.acropolistelecom.net P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. De : asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] De la part de BERGANZ François Envoyé : jeudi 3 septembre 2009 11:37 À : 'Asterisk Users Mailing List - Non-Commercial Discussion' Objet : [asterisk-users] MeetMe unactive pin access Hello, I have conferences in my database. I need at some moments, to access the database without asking pin access, or with using cdr(accountcode). Is it possible? Thank you Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. image001.gif___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MeetMe unactive pin access
I found ! If I need to enter in a conference (without pinacces) which is in the database (and have a pin access), Just add ,thepinacces at the end of meetme! Cordialement, P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. De : asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] De la part de BERGANZ François Envoyé : jeudi 3 septembre 2009 11:42 À : 'Asterisk Users Mailing List - Non-Commercial Discussion' Objet : Re: [asterisk-users] MeetMe unactive pin access Sorry, there are some errors, here the right question: Hello, I have conferences in my database. I need at some moments, to access the CONFEERENCE without asking pin access, or with using cdr(accountcode). Is it possible? Thank you Cordialement, BERGANZ François cid:image001.gif@01C8F7CD.6BC1D2C0 http://www.acropolistelecom.net/ http://www.acropolistelecom.net P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. De : asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] De la part de BERGANZ François Envoyé : jeudi 3 septembre 2009 11:37 À : 'Asterisk Users Mailing List - Non-Commercial Discussion' Objet : [asterisk-users] MeetMe unactive pin access Hello, I have conferences in my database. I need at some moments, to access the database without asking pin access, or with using cdr(accountcode). Is it possible? Thank you Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. image001.gif___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] set language in asterisk-1.6.x
Hello, How can Set(language()) in asterisk-1.6.x ? Cordialement, P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] set language in asterisk-1.6.x
Sorry, I just found the solution Set(CHANNEL(language)=en) Cordialement, P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. De : asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] De la part de BERGANZ François Envoyé : mardi 1 septembre 2009 16:15 À : asterisk-users@lists.digium.com Objet : [asterisk-users] set language in asterisk-1.6.x Hello, How can Set(language()) in asterisk-1.6.x ? Cordialement, P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Regular expression to validate any phonenumber
Use the pattern matching P137 in Asterisk: the future of telephony For example Exten = 919X,n, Cordialement, P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. De : asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] De la part de DHAVAL INDRODIYA Envoyé : lundi 31 août 2009 11:53 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : [asterisk-users] Asterisk Regular expression to validate any phonenumber Hi I am using asterisk version 1.6.0.5 I have build up one utility that will fire Originate Action on Manager... In which, i have define number to call eg. 919912312345 (MobileNumber) How can i know that this number format is true for Indian Number... In originate action, user can enter any international number.. How can I came to know this number format is right for that country...?? IS there any regular expression to validate this number . regards Dhaval ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Regular expression to validate any phonenumber
Use the pattern matching P137 in Asterisk: the future of telephony For example Exten = _919X,n, Cordialement, P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. De : asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] De la part de DHAVAL INDRODIYA Envoyé : lundi 31 août 2009 11:53 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : [asterisk-users] Asterisk Regular expression to validate any phonenumber Hi I am using asterisk version 1.6.0.5 I have build up one utility that will fire Originate Action on Manager... In which, i have define number to call eg. 919912312345 (MobileNumber) How can i know that this number format is true for Indian Number... In originate action, user can enter any international number.. How can I came to know this number format is right for that country...?? IS there any regular expression to validate this number . regards Dhaval ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 t 38 passthrough
Some news: Fax-asterisk-gatewayt38 Fax-* invite(g711) ... *-faxringing+200OK Fax-* invite(T38) * accept the T38 and reply trying *-* invite (t38) * find chan_sip.c:6737 get_ip_and_port_from_sdp: Failed to read an alternate host or port in SDP. Expect audio problems The problem is that asterisk generate an invite (t38) for himself with others parameters in the SDP Why? Cordialement, Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. -Message d'origine- De : asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] De la part de Kevin P. Fleming Envoyé : vendredi 21 août 2009 17:11 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [asterisk-users] Asterisk 1.6 t 38 passthrough BERGANZ François wrote: I have that problem: [Aug 21 15:57:35] WARNING[5198]: chan_sip.c:6737 get_ip_and_port_from_sdp: Failed to read an alternate host or port in SDP. Expect audio problems [Aug 21 15:57:35] WARNING[5198]: chan_sip.c:17425 handle_request_invite: Failed to set an alternate media source on glared reinvite. Audio may not work properly on this call. [Aug 21 15:57:37] WARNING[5198]: chan_sip.c:6737 get_ip_and_port_from_sdp: Failed to read an alternate host or port in SDP. Expect audio problems [Aug 21 15:57:37] WARNING[5198]: chan_sip.c:17425 handle_request_invite: Failed to set an alternate media source on glared reinvite. Audio may not work properly on this call. Why? It is at the second invite to do T38 That would mean that the second INVITE happened at an improper time; please open an issue on issues.asterisk.org, and include a complete console log include 'core set verbose 10', 'core set debug 10' and 'sip set debug on'. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] looking for user:Zoa for T38...
Zoa, I could see that you could do T38 passthrough with asterisk ans Zoiper. I have some problems to do it, can you help me? Cordialement, P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] install the digium card TC400P howto?
Ok. Now, I follow the digium documents. I have the TC400B-user-manual-1.pdf Where have I to insert the mode=g729 ? Chapter 3 Configuration At this time no zaptel.conf or zapata.conf changes are necessary to utilize this card. The mode module parameter may be used to specify which complex codes are allowed. mode = mixed: This default option will enable 92 calls of G.729a or G.723.1 (5.3Kbit) mode = g729: This option will enable 96 calls of G.729a mode = g723: This default option will enable 92 calls of G.723.1 (5.3Kbit) Cordialement, BERGANZ François cid:image001.gif@01C8F7CD.6BC1D2C0 http://www.acropolistelecom.net/ http://www.acropolistelecom.net P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. De : asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] De la part de Steve Totaro Envoyé : lundi 24 août 2009 19:17 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [asterisk-users] install the digium card TC400P howto? On Mon, Aug 24, 2009 at 1:06 PM, Sean Bright sean.bri...@gmail.com wrote: Steve Totaro wrote: No hardware timing source found in /proc/dahdi, loading dahdi_dummy would make me think it is not loading correctly. The TC400P is a transcoder card. It is not a timing source. -- Sean Bright sean.bri...@gmail.com Silly me. I forgot about those overpriced transcoder cards. Dollar for dollar, I will go for bogomips! -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) image001.gif___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Set up SER as a SIP extension on asterisk server
The authentication is in the SER and asterisk trust the ser (insecure=invite) Just do some iptables to be sure to don't receive sip from others... Cordialement, Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. -Message d'origine- De : asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] De la part de hh174 Envoyé : mardi 25 août 2009 12:28 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [asterisk-users] Set up SER as a SIP extension on asterisk server No way to authenticate from Ser, it's a proxy... Just make an entry in your asterisk sip.conf with the IP of SER and no authentification. Do the necesssary stuff in your extension.conf to identify and bill your client. Olivier Ishfaq Malik a écrit : Hi People We have a client who want to route their outbound calls through our asterisk server. We need them to authenticate as a sip extension so we know which calls are coming through them but the people over their side seem to be a bit clueless and claim they can't authenticate. Does anyone have any pointers for me for setting up a sip extension in a SER that I can pass on as I've never looks at one? Cheers Ish ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Authenticating SIP peer on IP address only
In the sip.conf, just insert the host= Cordialement, Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. -Message d'origine- De : asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] De la part de Ishfaq Malik Envoyé : mardi 25 août 2009 12:55 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : [asterisk-users] Authenticating SIP peer on IP address only Hi I know this is far from best practice but is it possible to authenticate a sip peer on the IP address it's coming from so that it doesn't need to use a UN and Pass? Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Set up SER as a SIP extension on asterisk server
Try us...@lists.kamailio.org http://www.acropolistelecom.net Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. -Message d'origine- De : asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] De la part de Ishfaq Malik Envoyé : mardi 25 août 2009 14:47 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [asterisk-users] Set up SER as a SIP extension on asterisk server Hi I've created a peer that only uses IP Address for authentication but now I'm being told that SER can't generate a register command, is this true? Ish hh174 wrote: No way to authenticate from Ser, it's a proxy... Just make an entry in your asterisk sip.conf with the IP of SER and no authentification. Do the necesssary stuff in your extension.conf to identify and bill your client. Olivier Ishfaq Malik a écrit : Hi People We have a client who want to route their outbound calls through our asterisk server. We need them to authenticate as a sip extension so we know which calls are coming through them but the people over their side seem to be a bit clueless and claim they can't authenticate. Does anyone have any pointers for me for setting up a sip extension in a SER that I can pass on as I've never looks at one? Cheers Ish ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Set up SER as a SIP extension on asterisk server
SER can't generate REGISTER You can play to control if it is yours with adding headers if you want, we can imagine that... Cordialement, Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. -Message d'origine- De : asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] De la part de Ishfaq Malik Envoyé : mardi 25 août 2009 14:47 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [asterisk-users] Set up SER as a SIP extension on asterisk server Hi I've created a peer that only uses IP Address for authentication but now I'm being told that SER can't generate a register command, is this true? Ish hh174 wrote: No way to authenticate from Ser, it's a proxy... Just make an entry in your asterisk sip.conf with the IP of SER and no authentification. Do the necesssary stuff in your extension.conf to identify and bill your client. Olivier Ishfaq Malik a écrit : Hi People We have a client who want to route their outbound calls through our asterisk server. We need them to authenticate as a sip extension so we know which calls are coming through them but the people over their side seem to be a bit clueless and claim they can't authenticate. Does anyone have any pointers for me for setting up a sip extension in a SER that I can pass on as I've never looks at one? Cheers Ish ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] problem on compiling asterisk-addons-1.6.2.0-rc1
hello, I tried to compil asterisk-addons-1.6.2.0-rc1, and I have that error: [CC] res_config_mysql.c - res_config_mysql.o res_config_mysql.c:1367: error: unknown field âupdate2_funcâ specified in initializer res_config_mysql.c: In function âparse_configâ: res_config_mysql.c:1432: error: âCONFIG_STATUS_FILEMISSINGâ undeclared (first use in this function) res_config_mysql.c:1432: error: (Each undeclared identifier is reported only once res_config_mysql.c:1432: error: for each function it appears in.) res_config_mysql.c:1436: error: âCONFIG_STATUS_FILEINVALIDâ undeclared (first use in this function) Here, all my commands to do it: apt-get install curl doxygen libnewt-dev mysql-client php5 php5-cli libmysqlclient15-dev libncurses5 libncurses5-dev openssl libssl-dev libssl0.9.8 mpg123 make g++ subversion subversion-tools newt-tcl linux-headers-`uname -r` php5-memcache php-pear DAHDI #cd /usr/src #wget http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete/dahdi-linu x-complete-2.2.0.2+2.2.0.tar.gz http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete/dahdi-linux -complete-2.2.0.2+2.2.0.tar.gz [ http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete/dahdi-linu x-complete-2.2.0.2+2.2.0.tar.gz ^] #tar zxfv dahdi-linux-complete-2.2.0.2+2.2.0.tar.gz #cd dahdi-linux-complete-2.2.0.2+2.2.0/ #make all make install make config LIBPRI #wget http://downloads.asterisk.org/pub/telephony/libpri/releases/libpri-1.4.10.t ar.gz http://downloads.asterisk.org/pub/telephony/libpri/releases/libpri-1.4.10.ta r.gz [ http://downloads.asterisk.org/pub/telephony/libpri/releases/libpri-1.4.10.t ar.gz ^] #tar zxfv libpri-1.4.10.tar.gz #cd libpri-1.4.10 #make make install ASTERISK #wget http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-1.6. 1.5-rc1.tar.gz http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-1.6.1 .5-rc1.tar.gz [ http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-1.6. 1.5-rc1.tar.gz ^] #tar zxfv asterisk-1.6.1.5-rc1.tar.gz #ln -s asterisk-1.6.1.5-rc1 asterisk #cd asterisk #./configure make menuselect -- attention à vérifier que chan_dahdi soit sélectionné #make make install make samples #cp /usr/src/asterisk/contrib/init.d/rc.debian.asterisk /etc/init.d/asterisk #/usr/sbin/update-rc.d asterisk defaults 99 99 #groupadd asterisk #useradd asterisk -g asterisk #vim /etc/init.d/asterisk Décommenter : AST_USER=asterisk AST_GROUP=asterisk #vim /etc/asterisk/asterisk.conf Effacer le (!) Changer: astrundir = /var/run/asterisk #mkdir /var/run/asterisk #chown asterisk /var/run/asterisk/ #chown asterisk /etc/asterisk -R #chown asterisk /var/lib/asterisk -R #chmod 777 /var/log/asterisk #chown asterisk /usr/lib/asterisk/ -R #chown asterisk /var/spool/asterisk/ -R #chown asterisk /etc/dahdi -R #chown asterisk /lib/modules/2.6.26-2-686/dahdi -R #chown asterisk /usr/share/dahdi ADDONS #wget http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-addo ns-1.6.2.0-rc1.tar.gz http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-addon s-1.6.2.0-rc1.tar.gz [ http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-addo ns-1.6.2.0-rc1.tar.gz ^] #tar zxfv asterisk-addons-1.6.2.0-rc1.tar.gz #cd asterisk-addons-1.6.2.0-rc1 #./configure make menuselect make make install make samples thank you for help Cordialement, BERGANZ François cid:image001.gif@01C8F7CD.6BC1D2C0 http://www.acropolistelecom.net/ http://www.acropolistelecom.net P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. image001.gif___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] install the digium card TC400P howto?
Hello, I need help to install a digium card TC400P. I compiled the dahdi source, but dahdi dont find the card! debian:/usr/src/dahdi-linux-complete-2.2.0.2+2.2.0# lspci 11:03.0 Ethernet controller: Digium, Inc. Wildcard TC400P transcoder base card (rev 11) debian:/usr/src/dahdi-linux-complete-2.2.0.2+2.2.0# /etc/init.d/dahdi restart Unloading DAHDI hardware modules: done Loading DAHDI hardware modules: wct4xxp: done wcte12xp: done wct1xxp: done wcte11xp: done wctdm24xxp: done wcfxo: done wctdm: done wcb4xxp: done wctc4xxp: done xpp_usb: done No hardware timing source found in /proc/dahdi, loading dahdi_dummy Running dahdi_cfg: done. Have you an idea? Thank you P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] install the digium card TC400P howto?
What have I to do? Cordialement, Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. -Message d'origine- De : asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] De la part de Sean Bright Envoyé : lundi 24 août 2009 18:21 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [asterisk-users] install the digium card TC400P howto? BERGANZ François wrote: debian:/usr/src/dahdi-linux-complete-2.2.0.2+2.2.0# /etc/init.d/dahdi restart Unloading DAHDI hardware modules: done Loading DAHDI hardware modules: wct4xxp: done wcte12xp: done wct1xxp: done wcte11xp: done wctdm24xxp: done wcfxo: done wctdm: done wcb4xxp: done wctc4xxp: done xpp_usb: done wctc4xxp: done -- That's the module for the card. So it is loading. -- Sean Bright sean.bri...@gmail.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to install asterisk
If I do #dmesg, I have it :
[ 101.994189] Unregistered codec translator 'DTE Decoder' with 92
transcoders (srcs=0101, dsts=000c)
[ 101.994189] Unregistered codec translator 'DTE Encoder' with 92
transcoders (srcs=000c, dsts=0101)
[ 101.999162] dahdi_transcode: Unloaded.
[ 102.011417] dahdi_transcode: Loaded.
[ 102.011418] wctc4xxp: tc400b0: Attached to device at :11:03.0.
[ 102.011418] firmware: requesting dahdi-fw-tc400m.bin
[ 107.427805] wctc4xxp: tc400b0: (G.729a / G.723.1) Transcoder support
LOADED (firm ver = 6.12)
[ 107.427805] wctc4xxp: tc400b0: Installed a Wildcard TC: Wildcard
TC400P+TC400M
[ 107.427805] dahdi_transcode: Registered codec translator 'DTE Encoder'
with 92 transcoders (srcs=000c, dsts=0101)
[ 107.427805] dahdi_transcode: Registered codec translator 'DTE Decoder'
with 92 transcoders (srcs=0101, dsts=000c)
[ 107.574569] dahdi_dummy: Trying to load High Resolution Timer
[ 107.574569] dahdi_dummy: Initialized High Resolution Timer
[ 107.574569] dahdi_dummy: Starting High Resolution Timer
[ 107.574569] dahdi_dummy: High Resolution Timer started, good to go
It know my card but why it dont load it!?
Cordialement,
P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.
De : asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] De la part de Valter
Nogueira
Envoyé : lundi 24 août 2009 18:30
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [asterisk-users] how to install asterisk
I will consider it and to change it to DAHDI too.
It woul be great if repository had files named CURRENT like
asterisk_1.4.CURRENT so would have no need to change any script.
Thanks,
Valter
2009/8/24 Steve Edwards asterisk@sedwards.com
2009/8/21 aster...@opensourcesolution.in
i have to configures asterisk n my hardware details are
Is it just me, or would you think someone from a domain named like
Open Source Solution should be able to figure this one out...
On Mon, 24 Aug 2009, Valter Nogueira wrote:
I have a small script that do the trick for you.
http://www.fastway.com.br/wp-content/uploads/2009/02/install_asterisk.sh
Just a suggestion...
If you define the version numbers as variables your script will be
easier to maintain. For example:
ADDONS_VERSION=1.4.7
ASTERISK_VERSION=1.4.23.1
LIBPRI_VERSION=1.4.9
ZAPTEL_VERSION=1.4.12
cd /usr/src
wget
http://downloads.digium.com/pub/libpri/releases/libpri-${LIBPRI_VERSION}.tar
.gz
http://downloads.digium.com/pub/libpri/releases/libpri-$%7BLIBPRI_VERSION%7
D.tar.gz
wget
http://downloads.digium.com/pub/zaptel/releases/zaptel-${ZAPTEL_VERSION}.tar
.gz
http://downloads.digium.com/pub/zaptel/releases/zaptel-$%7BZAPTEL_VERSION%7
D.tar.gz
wget
http://downloads.digium.com/pub/asterisk/releases/asterisk-${ASTERISK_VERSIO
N}.tar.gz
http://downloads.digium.com/pub/asterisk/releases/asterisk-$%7BASTERISK_VER
SION%7D.tar.gz
wget
http://downloads.digium.com/pub/asterisk/releases/asterisk-addons-${ADDONS_V
ERSION}.tar.gz
http://downloads.digium.com/pub/asterisk/releases/asterisk-addons-$%7BADDON
S_VERSION%7D.tar.gz
tar -zxvf libpri-${LIBPRI_VERSION}.tar.gz
tar -zxvf zaptel-${ZAPTEL_VERSION}.tar.gz
tar -zxvf asterisk-${ASTERISK_VERSION}.tar.gz
tar -zxvf asterisk-addons-${ADDONS_VERSION}.tar.gz
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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[asterisk-users] Asterisk 1.6 t 38 passthrough
Hello, How can I do t-38 passthrough with asterisk 1.6 ? I know how to do with 1.4 but not with 1.6 Thank you Cordialement, P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 t 38 passthrough
When I receive a fax it is in g711 After pickup, the fax invite again with T38 in the SDP. Have I something to insert in the dialplan or other to let the T38 passthrough ? Cordialement, Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. -Message d'origine- De : asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] De la part de Kevin P. Fleming Envoyé : vendredi 21 août 2009 15:05 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [asterisk-users] Asterisk 1.6 t 38 passthrough BERGANZ François wrote: How can I do t-38 passthrough with asterisk 1.6 ? I know how to do with 1.4 but not with 1.6… There is no difference, the identical configuration should work. I would recommend using the 1.6.0.14 or 1.6.1.5 release candidates (or any later releases) as they contain a couple of months worth of T.38-related bug fixes and improvements that aren't in 1.6.1.4 and 1.6.0.13. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 t 38 passthrough
I have that problem: [Aug 21 15:57:35] WARNING[5198]: chan_sip.c:6737 get_ip_and_port_from_sdp: Failed to read an alternate host or port in SDP. Expect audio problems [Aug 21 15:57:35] WARNING[5198]: chan_sip.c:17425 handle_request_invite: Failed to set an alternate media source on glared reinvite. Audio may not work properly on this call. [Aug 21 15:57:37] WARNING[5198]: chan_sip.c:6737 get_ip_and_port_from_sdp: Failed to read an alternate host or port in SDP. Expect audio problems [Aug 21 15:57:37] WARNING[5198]: chan_sip.c:17425 handle_request_invite: Failed to set an alternate media source on glared reinvite. Audio may not work properly on this call. Why? It is at the second invite to do T38 Cordialement, BERGANZ François http://www.acropolistelecom.net Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. -Message d'origine- De : asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] De la part de Kevin P. Fleming Envoyé : vendredi 21 août 2009 15:31 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [asterisk-users] Asterisk 1.6 t 38 passthrough BERGANZ François wrote: When I receive a fax it is in g711 After pickup, the fax invite again with T38 in the SDP. Have I something to insert in the dialplan or other to let the T38 passthrough ? No. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MEETME how to lock the conference if no admin are connected
hello is it possible to lock a conference IF no admin are connected ? or how to do to have a conference offline? thank you Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] problem with sangoma card a108d
Hello, I need help to use my sangoma card a108d. I need that another server give me an E1 with a clock. The server with the sangoma reseive the E1 clock on port1 and is MASTER E1 on port2. But, I cant receive the clock (I am connected). Anyone can help me? Thank you Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] cant use h,1 at cancel!
Hello, In my dialplan, I do s,n,DIAL( ) If my called phone response and after hangup, asterisk execute the h,1, But, if I the caller hangup at ringing (cancel), it dont execute the h,1, Know you why? Thank you Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cant use h,1 at cancel!
I found a bug in asterisk 1.6: http://lists.digium.com/pipermail/asterisk-dev/2009-April/037684.html in fact, the h,1 in the macro don’t work with cancel! Cordialement, BERGANZ François Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. -Message d'origine- De : asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] De la part de Steve Howes Envoyé : jeudi 11 juin 2009 10:59 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [asterisk-users] cant use h,1 at cancel! On 11 Jun 2009, at 08:59, BERGANZ François wrote: In my dialplan, I do s,n,DIAL(…) If my called phone response and after hangup, asterisk execute the h,1,… But, if I the caller hangup at ringing (cancel), it don’t execute the h,1,… Know you why? Because the call was cancelled and not actually hung up? Generally the hangup context is used to 'clean up' or provide info about the call. If it didn't happen its a bit irrelevant. S ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] problem install Asterisk-FastAGI
Hello all, I have a problem when I try to install FastAGI. I try to do #perl Makefile.PEL And it return : Can't locate inc/Module/Install.pm in @INC (@INC contains: /etc/perl /usr/local/lib/perl/5.8.8 /usr/local/share/perl/5.8.8 /usr/lib/perl5 /usr/share/perl5 /usr/lib/perl/5.8 /usr/share/perl/5.8 /usr/local/lib/site_perl .) at Makefile.PL line 1. BEGIN failed--compilation aborted at Makefile.PL line 1. Why? What have I to do? Thank you Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] error with dial timeout
Hello, I am trying to do : Exten =_X.,n,Dial(SIP/ser_sei0/1130,L(10208400:61000:1)) But it return that error: [Jun 2 10:04:44] WARNING[18920]: app_dial.c:1623 dial_exec_full: Invalid timeout specified: 'L(10208400:61000:1)' Why? I forgot something ? Thank you Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] error with dial timeout
Thank you! I did understood that i twas THAT timeout :-) I thought that it speak about my 'limit call' Thank you Cordialement, BERGANZ François Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. -Message d'origine- De : asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] De la part de Philipp Kempgen Envoyé : mardi 2 juin 2009 10:37 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [asterisk-users] error with dial timeout BERGANZ François schrieb: Exten =_X.,n,Dial(SIP/ser_sei0/1130,L(10208400:61000:1)) But it return that error: [Jun 2 10:04:44] WARNING[18920]: app_dial.c:1623 dial_exec_full: Invalid timeout specified: 'L(10208400:61000:1)' Syntax: Dial(Technology/resource[Tech2/resource2...][,timeout][,options][,URL]) You have to pass L() as the options argument. Exten =_X.,n,Dial(SIP/ser_sei0/1130,,L(10208400:61000:1)) ^ Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] play with varibles
Dont work
I need that it suppr the
Thank you
Cordialement,
BERGANZ François
De : asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] De la part de Danny
Nicholas
Envoyé : mercredi 20 mai 2009 17:53
À : 'Asterisk Users Mailing List - Non-Commercial Discussion'
Objet : Re: [asterisk-users] play with varibles
Cut should do this for you
Exten = x,x,Set(var2=cut(var1,\)
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of BERGANZ
François
Sent: Wednesday, May 20, 2009 10:33 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] play with varibles
Hello,
I have a var like blabla with the
I need to suppr the
Is it possible with the ${var:x:y} ?
Thank you
Cordialement,
BERGANZ François
P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.
___
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[asterisk-users] play with varibles
Hello,
I have a var like blabla with the
I need to suppr the
Is it possible with the ${var:x:y} ?
Thank you
Cordialement,
BERGANZ François
P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.
___
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Re: [asterisk-users] play with varibles
I found !
exten = _X.,n,Set(var2=${CUT(var,',2)})
Cordialement,
BERGANZ François
P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.
De : asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] De la part de BERGANZ
François
Envoyé : mercredi 20 mai 2009 18:21
À : 'Asterisk Users Mailing List - Non-Commercial Discussion'
Objet : Re: [asterisk-users] play with varibles
Dont work
I need that it suppr the
Thank you
Cordialement,
BERGANZ François
De : asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] De la part de Danny
Nicholas
Envoyé : mercredi 20 mai 2009 17:53
À : 'Asterisk Users Mailing List - Non-Commercial Discussion'
Objet : Re: [asterisk-users] play with varibles
Cut should do this for you
Exten = x,x,Set(var2=cut(var1,\)
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of BERGANZ
François
Sent: Wednesday, May 20, 2009 10:33 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] play with varibles
Hello,
I have a var like blabla with the
I need to suppr the
Is it possible with the ${var:x:y} ?
Thank you
Cordialement,
BERGANZ François
P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.
___
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[asterisk-users] VM_DATE in french?
Hello,
I work on voicemail.conf and I need that ${VM_DATE} is in french!
How can I do it?
Thank you
Cordialement,
BERGANZ François
P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.
___
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[asterisk-users] problem with an agi in PHP
Hello, I need to execute an agi in php. I have that: == Using SIP RTP CoS mark 5 -- Executing [0170725...@mnupprx1:1] Answer(SIP/33179977999-b6c18478, ) in new stack -- Executing [0170725...@mnupprx1:2] GotoIf(SIP/33179977999-b6c18478, 0?6:3)) in new stack -- Goto (mnupprx1,0170725000,3) -- Executing [0170725...@mnupprx1:3] DeadAGI(SIP/33179977999-b6c18478, a2billing.php) in new stack [Mar 9 14:25:08] WARNING[7518]: res_agi.c:3064 deadagi_exec: DeadAGI has been deprecated, please use AGI in all cases! -- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php -- SIP/33179977999-b6c18478AGI Script a2billing.php completed, returning 0 -- Executing [0170725...@mnupprx1:4] Wait(SIP/33179977999-b6c18478, 2) in new stack -- Executing [0170725...@mnupprx1:5] Hangup(SIP/33179977999-b6c18478, ) in new stack == Spawn extension (mnupprx1, 0170725000, 5) exited non-zero on 'SIP/33179977999-b6c18478' I dont know why it dont exec it ! Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problem with an agi in PHP
I have the same thing with AGI in the dialplan And php is install Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. De : asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] De la part de Danny Nicholas Envoyé : lundi 9 mars 2009 14:36 À : 'Asterisk Users Mailing List - Non-Commercial Discussion' Objet : Re: [asterisk-users] problem with an agi in PHP The message indicates that DEADAGI will not work and that you should use AGI instead. Are you sure PHP is installed on your machine and functioning properly (from $, php a2billing.php works)? _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of BERGANZ François Sent: Monday, March 09, 2009 8:30 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] problem with an agi in PHP Hello, I need to execute an agi in php. I have that: == Using SIP RTP CoS mark 5 -- Executing [0170725...@mnupprx1:1] Answer(SIP/33179977999-b6c18478, ) in new stack -- Executing [0170725...@mnupprx1:2] GotoIf(SIP/33179977999-b6c18478, 0?6:3)) in new stack -- Goto (mnupprx1,0170725000,3) -- Executing [0170725...@mnupprx1:3] DeadAGI(SIP/33179977999-b6c18478, a2billing.php) in new stack [Mar 9 14:25:08] WARNING[7518]: res_agi.c:3064 deadagi_exec: DeadAGI has been deprecated, please use AGI in all cases! -- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php -- SIP/33179977999-b6c18478AGI Script a2billing.php completed, returning 0 -- Executing [0170725...@mnupprx1:4] Wait(SIP/33179977999-b6c18478, 2) in new stack -- Executing [0170725...@mnupprx1:5] Hangup(SIP/33179977999-b6c18478, ) in new stack == Spawn extension (mnupprx1, 0170725000, 5) exited non-zero on 'SIP/33179977999-b6c18478' I dont know why it dont exec it ! Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problem with an agi in PHP
I have all permissioned Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. De : asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] De la part de Danny Nicholas Envoyé : lundi 9 mars 2009 15:07 À : 'Asterisk Users Mailing List - Non-Commercial Discussion' Objet : Re: [asterisk-users] problem with an agi in PHP You didnt say whether a2billing.php works from the shell. Is it 755 permissioned? _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of BERGANZ François Sent: Monday, March 09, 2009 8:53 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] problem with an agi in PHP I have the same thing with AGI in the dialplan And php is install Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. De : asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] De la part de Danny Nicholas Envoyé : lundi 9 mars 2009 14:36 À : 'Asterisk Users Mailing List - Non-Commercial Discussion' Objet : Re: [asterisk-users] problem with an agi in PHP The message indicates that DEADAGI will not work and that you should use AGI instead. Are you sure PHP is installed on your machine and functioning properly (from $, php a2billing.php works)? _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of BERGANZ François Sent: Monday, March 09, 2009 8:30 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] problem with an agi in PHP Hello, I need to execute an agi in php. I have that: == Using SIP RTP CoS mark 5 -- Executing [0170725...@mnupprx1:1] Answer(SIP/33179977999-b6c18478, ) in new stack -- Executing [0170725...@mnupprx1:2] GotoIf(SIP/33179977999-b6c18478, 0?6:3)) in new stack -- Goto (mnupprx1,0170725000,3) -- Executing [0170725...@mnupprx1:3] DeadAGI(SIP/33179977999-b6c18478, a2billing.php) in new stack [Mar 9 14:25:08] WARNING[7518]: res_agi.c:3064 deadagi_exec: DeadAGI has been deprecated, please use AGI in all cases! -- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php -- SIP/33179977999-b6c18478AGI Script a2billing.php completed, returning 0 -- Executing [0170725...@mnupprx1:4] Wait(SIP/33179977999-b6c18478, 2) in new stack -- Executing [0170725...@mnupprx1:5] Hangup(SIP/33179977999-b6c18478, ) in new stack == Spawn extension (mnupprx1, 0170725000, 5) exited non-zero on 'SIP/33179977999-b6c18478' I dont know why it dont exec it ! Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problem with an agi in PHP
I am sorry it work ! In fact, I had mistakes in my config Sorry And thank you for answering Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. De : asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] De la part de Danny Nicholas Envoyé : lundi 9 mars 2009 15:07 À : 'Asterisk Users Mailing List - Non-Commercial Discussion' Objet : Re: [asterisk-users] problem with an agi in PHP You didnt say whether a2billing.php works from the shell. Is it 755 permissioned? _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of BERGANZ François Sent: Monday, March 09, 2009 8:53 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] problem with an agi in PHP I have the same thing with AGI in the dialplan And php is install Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. De : asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] De la part de Danny Nicholas Envoyé : lundi 9 mars 2009 14:36 À : 'Asterisk Users Mailing List - Non-Commercial Discussion' Objet : Re: [asterisk-users] problem with an agi in PHP The message indicates that DEADAGI will not work and that you should use AGI instead. Are you sure PHP is installed on your machine and functioning properly (from $, php a2billing.php works)? _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of BERGANZ François Sent: Monday, March 09, 2009 8:30 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] problem with an agi in PHP Hello, I need to execute an agi in php. I have that: == Using SIP RTP CoS mark 5 -- Executing [0170725...@mnupprx1:1] Answer(SIP/33179977999-b6c18478, ) in new stack -- Executing [0170725...@mnupprx1:2] GotoIf(SIP/33179977999-b6c18478, 0?6:3)) in new stack -- Goto (mnupprx1,0170725000,3) -- Executing [0170725...@mnupprx1:3] DeadAGI(SIP/33179977999-b6c18478, a2billing.php) in new stack [Mar 9 14:25:08] WARNING[7518]: res_agi.c:3064 deadagi_exec: DeadAGI has been deprecated, please use AGI in all cases! -- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php -- SIP/33179977999-b6c18478AGI Script a2billing.php completed, returning 0 -- Executing [0170725...@mnupprx1:4] Wait(SIP/33179977999-b6c18478, 2) in new stack -- Executing [0170725...@mnupprx1:5] Hangup(SIP/33179977999-b6c18478, ) in new stack == Spawn extension (mnupprx1, 0170725000, 5) exited non-zero on 'SIP/33179977999-b6c18478' I dont know why it dont exec it ! Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] question about MeetMe performance.
hello, I will do a server to do a lots of conferences (MeetMe). I want to know that if I dont use a digum card, the limit of simultaneous calls is harder without a card than with a card ?if, yes, how harder is the limit? thank you Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] canreinvite question
In the sip.conf [2001] ... Canreinvite=yes [2002] ... Canreinvite=no Cordialement, BERGANZ François http://www.acropolistelecom.net Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. -Message d'origine- De : asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] De la part de Tim Johnson Envoyé : jeudi 18 décembre 2008 19:49 À : asterisk-users@lists.digium.com Objet : [asterisk-users] canreinvite question Is it possible to allow reinvites to/from specific devices? For example; exten 2001 and 2002 can reinvite to each other, but not 2003 and 2004 exten 2003 and 2004 can reinvite to each other, but not 2001 and 2002 Can that be done? Devices 2001 2002 are behind one firewall, and 2003 2004 are behind another. Tim ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] prepaid solution
Hello, I am looking for a good prepaid solution. What is the best ? Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Friday, Asterisk is 9 years old!
Happy birthday asterisk! -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de SIP Envoyé : vendredi 5 décembre 2008 06:14 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [asterisk-users] Friday, Asterisk is 9 years old! randulo wrote: Hi, December 5th, 1999 was the initial release of Asterisk by Mark Spencer. We'll be celebrating this by gathering as usual at 12 Noon Eastern (9AM Pacific, 10 MST, 11 Central, 5PM UK and Western EU) for the VoIP Users Conference. You can get all the dial in information at http://VoipUsersConference.org including info on a SipAddHeader() kludge to avoid DTMF problems. IRC is Freenode.net #voip-users-conference join this even if you can't call in. Call via SIP: [EMAIL PROTECTED] (thanks to OnSip.com) Call via PSTN (724) 444-7444 DTMF 22622# 1# or try this: [EMAIL PROTECTED] (thanks to IdeaSIP.com) or to just look up talkshoe server IP: ts.x2z.eu (thanks top me for the DNS record) We start about 15 minutes to the hour with an informal chat. Join us anytime, but especially, grab a virtual beer and join us Friday the 5th. /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users December 5th, 1879 is also the date when the first automatic telephone switch was patented. A good day for telecom all-round. N. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] call loop in the network
Hello, I think, that I find a bug for a specific asterisk use. My network test is: Client 1 --asterisk0(client)---asterisk1(ss7)--Asterisk2(media) Client 2 I need that a call from Client1 go through asterisk0-asterisk1-asterisk2-asterisk1-asterisk0-Client2 When I do the call from Client1, · Asterisk0 forward to asterisk1 · Asterisk1 forward to asterisk2 · Asterisk2 forward to asterisk1 · Asterisk1 forward to asterisk0 When Asterisk1 forward to asterisk0, asterisk0 reply SIP/2.0 401 Unauthorized I think that the user have to be authorize because the first call come from himself! And I insert insecure very for everyone I think that if asterisk receive a call from an user which isnt from the IP registered, asterisk refuse the call no? Can you help me ? Need you debug? Thank you FRANCOIS ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] canreinvite=yes problem
Now, I have : Client 1 -Asterisk1--Asterisk2 Client 2 I need that sip sign go to Asterisk2 But RTP go to Asterisk1 and no more. Where have I to insert canreinvite ? Thank you -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Eric ManxPower Wieling Envoyé : mercredi 3 décembre 2008 19:25 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [asterisk-users] canreinvite=yes problem canreinvite=yes should work as long as 1) there is no NAT involved anywhere in the call path, 2) All legs of the call are using the same codec, 3) you do not have the t/T/w/W (and maybe a few other) options to the Dial line. Remember the only way you can really tell if a reinvite happens is by looking at the RTP audio. The SIP signaling will not and has never had a reinvite feature for signaling. Why did you post the same message at :23, :28, and :35 mins past the hour? If you need immediate support you should contact Digium support and pay for a service contract. BERGANZ François wrote: I need to test canreinvite=yes with 2softphones and 1 asterisk. I want to have that : http://www.panoramisk.com/wp-content/uploads/2007/05/asterisk-call-flow-outb ridge.png But I have that http://www.zimagez.com/zimage/canreinvite.php Canreinvite=yes work for all phones or just asterisk?... -- Consulting and design services for LAN, WAN, voice and data. Based near Birmingham, AL. Now accepting clients worldwide. Contact me for Tellabs echo canceling systems. Also see http://www.fnords.org/skillslist.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] canreinvite=yes problem
I still have: Client 1 -Asterisk1--Asterisk2 Client 2 When client1 do a call, asterisk1 forward to asterisk2, asterisk2 forward to Asterisk1 At this moment, asterisk1 say : 404Not found But I have insecure=very This is the sip debug at that moment: - --- (11 headers 0 lines) --- --- SIP read from UDP://192.168.1.151:5060 --- INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.151:5060;branch=z9hG4bK2b4a242e;rport Max-Forwards: 70 From: 103 sip:[EMAIL PROTECTED];tag=as636875d3 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.0.1 Date: Thu, 04 Dec 2008 14:55:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Type: application/sdp Content-Length: 289 v=0 o=root 1545198644 1545198644 IN IP4 192.168.1.151 s=Asterisk PBX 1.6.0.1 c=IN IP4 192.168.1.151 t=0 0 m=audio 12272 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv - --- (14 headers 13 lines) --- == Using SIP RTP CoS mark 5 Sending to 192.168.1.151 : 5060 (NAT) Using INVITE request as basis request - [EMAIL PROTECTED] No user '103' in SIP users list Found peer 'media' for '103' from 192.168.1.151:5060 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 192.168.1.151:12272 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.1.151:12272 Looking for 33170725012 in media (domain 192.168.1.153) --- Reliably Transmitting (no NAT) to 192.168.1.151:5060 --- SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.1.151:5060;branch=z9hG4bK2b4a242e;received=192.168.1.151;rport=5060 From: 103 sip:[EMAIL PROTECTED];tag=as636875d3 To: sip:[EMAIL PROTECTED];tag=as242de969 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.0.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 Have you an idea why ? -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de BERGANZ François Envoyé : jeudi 4 décembre 2008 09:15 À : 'Asterisk Users Mailing List - Non-Commercial Discussion' Objet : Re: [asterisk-users] canreinvite=yes problem Now, I have : Client 1 -Asterisk1--Asterisk2 Client 2 I need that sip sign go to Asterisk2 But RTP go to Asterisk1 and no more. Where have I to insert canreinvite ? Thank you -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Eric ManxPower Wieling Envoyé : mercredi 3 décembre 2008 19:25 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [asterisk-users] canreinvite=yes problem canreinvite=yes should work as long as 1) there is no NAT involved anywhere in the call path, 2) All legs of the call are using the same codec, 3) you do not have the t/T/w/W (and maybe a few other) options to the Dial line. Remember the only way you can really tell if a reinvite happens is by looking at the RTP audio. The SIP signaling will not and has never had a reinvite feature for signaling. Why did you post the same message at :23, :28, and :35 mins past the hour? If you need immediate support you should contact Digium support and pay for a service contract. BERGANZ François wrote: I need to test canreinvite=yes with 2softphones and 1 asterisk. I want to have that : http://www.panoramisk.com/wp-content/uploads/2007/05/asterisk-call-flow-outb ridge.png But I have that http://www.zimagez.com/zimage/canreinvite.php Canreinvite=yes work for all phones or just asterisk?... -- Consulting and design services for LAN, WAN, voice and data. Based near Birmingham, AL. Now accepting clients worldwide. Contact me for Tellabs echo canceling systems. Also see http://www.fnords.org/skillslist.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided
[asterisk-users] chan_sip.c:14889 handle_response_invite: Failed to authenticate on INVITE to
Hello, I need help for that error message: chan_sip.c:14889 handle_response_invite: Failed to authenticate on INVITE to My network is: Client1-- ---asterisk1--asterisk2 Client2-- · With client1, I do a call · Asterisk1 forward the call to asterisk2 · Asterisk2 forward the call to asterisk1 · Asterisk1 forward the call to client2 But, in the asterisk2 CLI, I have the error, and with a tcpdump capture, I see that asterisk1 send to asterisk2 unauthorized Have you an idea? Thank you ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] problem with RTP
Hello, My network is: Client_SS7_1-- ---asterisk1--asterisk2 Client_SS7_2-- · I receive a fax from Client_SS7_1 · Asterisk1 forward the call to asterisk2 · Asterisk2 forward the call to asterisk1 · Then, asterisk2 forward the fax to Client_SS7_2 I want that the SIP signaling go to asterisk2, But, I need that the RTP dont go more than asterisk1. Have you an idea? Thank you FRANCOIS ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] canreinvite=yes problem
Hello, I need to test canreinvite=yes with 2softphones and 1 asterisk. I want to have that : http://www.panoramisk.com/wp-content/uploads/2007/05/asterisk-call-flow-outb ridge.png But I have that http://www.zimagez.com/zimage/canreinvite.php Canreinvite=yes work for all phones or just asterisk?... Can you help me? Thank you ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] canreinvite=yes --problems
Hello, I need to test canreinvite=yes with 2softphones and 1 asterisk.. I want to have that : http://www.panoramisk.com/wp-content/uploads/2007/05/asterisk-call-flow-outb ridge.png But I have that http://www.zimagez.com/zimage/canreinvite.php Canreinvite=yes work for all phones or just asterisk?... Can you help me? Thank you ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] canreinvite=yes problem
Someone have a solution for me ? De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de BERGANZ François Envoyé : mercredi 3 décembre 2008 18:24 À : asterisk-users@lists.digium.com Objet : [asterisk-users] canreinvite=yes problem Hello, I need to test canreinvite=yes with 2softphones and 1 asterisk. I want to have that : http://www.panoramisk.com/wp-content/uploads/2007/05/asterisk-call-flow-outb ridge.png But I have that http://www.zimagez.com/zimage/canreinvite.php Canreinvite=yes work for all phones or just asterisk?... Can you help me? Thank you ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
