Re: [Asterisk-Users] VoIP service through Asterisk?
why nobody use sipphone.com to connect to asterisk ? Best Regards Zhao Zigang Alcatel Shanghai Bell Co., LTD *:388,NingQiao Rd.,Shanghai 201206 *:086-21-50554550-7762 *:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] (no subject)
I don't know what's means about register in sip.conf such as: register = user:secret:[EMAIL PROTECTED]:port/extension even if I registe a sip proxy , but how use it ? I think : when incoming from sip proxy to asterisk : user a -- sip proxy -- asterisk -- pstn sip proxy : SER in ser.cfg farword ( 192.168.0.10 , 5060 ) this will forward a call to asterisk , and asterisk will deal with this call by extensions.conf. when call from asterisk to SER: only config extensions.conf: pstn -- asterisk -- sip proxy -- user a exten = _,1,Dial(SIP/[EMAIL PROTECTED],30) so this will slove question. but I don't know why I need register for sip.conf. who can give a info , thank you. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] I look for some copartner
I will build a voip network by use SER+asterisk in China. but I don't connect my sip network to pstn , this is don't be permission in China. if you have a asterisk and have pstn access, then you can contact me about copartner. I will let my user that he want to dial to pstn will use your pstn . and I need your rate to pstn and I only hope can get a little part. I hope negotiate about it. Thank you. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] limit about asterisk pstn out
I have a system include asterisk + ser.
when I want to limit a dial out to pstn , I will do that :
extensions.conf
exten = _9NXXNX/[EMAIL PROTECTED],Congestion
exten = _9NXXNX, 1,Dial(ZAP/g2/{EXTEN:1},30,t)
exten = _9NXXNX, 2,Hungup
but I don't confirm is it right.
I have no env to test it.
who can help me?
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[Asterisk-Users] how buy digium card such as TDM400.
I am in China , I cann't buy digium card. I want to resales asterisk in China for chinese enterprise. who can give a card for test ? I only hope COD. I hope buy a TDM400 and a FXO . Thank u. Best Regards Zhao Zigang Alcatel Shanghai Bell Co., LTD *:388,NingQiao Rd.,Shanghai 201206 *:086-21-50554550-7762 *:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] what is best free softphone.
I use xlite , but it isn't support video when it is free. who used better softphone ? Thank u. Best Regards Zhao Zigang Alcatel Shanghai Bell Co., LTD *:388,NingQiao Rd.,Shanghai 201206 *:086-21-50554550-7762 *:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How work by Asterisk and SER ?
my means , how could use asterisk and ser in same box. my ser support mysql database , so whether asterisk don't config user in sip.conf ? and how to do I should ? thanks a lot. and I want to agent asterisk product in China Mainland, who can contect me. Best Regards Zhao Zigang Alcatel Shanghai Bell Co., LTD *:388,NingQiao Rd.,Shanghai 201206 *:086-21-50554550-7762 *:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] why I don't do this test ?
I do this test :
exten = _[123456789],1,Dial(SIP/${EXTEN},20,L(0))
exten = _[123456789],2,Hangup
when I use 12345 dial 12346 , it should be hangup.
but it don't link I think. why?
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[Asterisk-Users] about dial parameter L
quesion 1:
If I write extension.conf
exten = 100,1,dial(sip/100,20,L(0))
I will listen ring but I don't access when other one dial 100.
who have this experience ?
quesion 2:
and I look at this extension.conf in voip-info.org
exten = _908.,1,Dial(Modem/ttyI0:${EXTEN:1})
Could this modem dial to PSTN?
and will I install dial program for the modem?
and when I write :
exten = _908.,1,dial(modem/ttyl0:${EXTEN:1},20,L(0))
I will dial to PSTN, but dialee will only ring.
then for me,if dialee access,will I pay phone rate?
who have experience? help me.
Thanks.
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[Asterisk-Users] what is phone: Linux Telephony channel
need hardware ? can dial to PSTN? help me. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] help me : about dial to PSTN
I want to use asterisk dial to PSTN,but only dial,don't connect. when you hear ring,you only can hungup,don't connect. when you connect , asterisk will disconnect . who can tell me what write extension.conf? Thanks. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hope cooperate
I want to build a SER in China, but I don't build a PSTN gateway in China. because goverment don't give permission in China. I want to look for a cooperater, let's my user can dial PSTN to world. if you are interested in my idea,please mail to me. Best Regards Zhao Zigang Alcatel Shanghai Bell Co., LTD *:388,NingQiao Rd.,Shanghai 201206 *:086-21-50554550-7762 *:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] about call out : a strange question.
Hello all,
I want to use asterisk pbx to give a ring for sip user.when A call B ,
user B 's mobile will ring.(B always register his sip number and his mobile
number first.)
ignorepat = 9
exten = _9NXX,1,Dial(Zap/g2/${EXTEN:1})
exten = _9NXX,2,Congestion
but I want only let B's mobile ring,B can't access. or when B
access,the phone auto hang up.
what I should do?
who can told me?
thank you.
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[Asterisk-Users] who used ser and asterisk?
I install ser and found my ser don't support mysql. my ser version : ser-0.8.14_src.tar.gz and ser-0.8.14_linux_i386.tar.gz who can help me? thanks. Best Regards Zhao Zigang Alcatel Shanghai Bell Co., LTD *:388,NingQiao Rd.,Shanghai 201206 *:086-21-50554550-7762 *:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Help:could asterisk work with other sip proxy?
Help:could asterisk work with other sip proxy? Different Enterprise asterisk PBX want to contact each other. could I use other manufacturer sip proxy contact different enterprise asterisk PBX ? B.R. Zhao Zigang. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Qestion about TDM over enthernet
who can tell me how to do TDM over enthernet ? pc a connect pc b only use TDM card? thank you John. -- : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] : 20041223 11:47 : asterisk-users@lists.digium.com : Asterisk-Users Digest, Vol 5, Issue 336 Send Asterisk-Users mailing list submissions to asterisk-users@lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than Re: Contents of Asterisk-Users digest... Today's Topics: 1. Re: Still unable to use g729 codec... please HELP (Rodolfo Grave) 2. Re: Can't Receive/Send Calls (Norman Zhang) 3. RE: Zaptel/Zapata config from T410p to Brooktrout T1 (Jason Kawakami) 4. Re: Still unable to use g729 codec... please HELP (Kristian Kielhofner) 5. Re: Still unable to use g729 codec... please HELP (Rodolfo Grave) 6. Re: Still unable to use g729 codec... please HELP (Kristian Kielhofner) 7. Re: hint extension and Snom phones - CVS or stable? (Karl Brose) 8. WARNING Maximum retries exceeded on call for seqno102 (Kevin) 9. Re: Re: 'I'nvalid extension handling problems,even with workaround ([EMAIL PROTECTED]) 10. Re: Still unable to use g729 codec... please HELP (Rodolfo Grave) 11. Re: Still unable to use g729 codec... please HELP (Kristian Kielhofner) 12. RE: polycom and cdp (Richard) 13. Re: RE: Zaptel/Zapata config from T410p toBrooktroutT1 (jbebeau) -- Message: 1 Date: Thu, 23 Dec 2004 03:32:22 +0100 From: Rodolfo Grave [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Still unable to use g729 codec... please HELP To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed Yeap... :( Kristian Kielhofner wrote: Rodolfo Grave wrote: Hi. Has anyone accomplished to use the g729 codec? I have the license installed, and I have tried with X-Pro and a Grandstream Budgetone configured to use g729 only. This is what I get from Asterisk: Dec 23 02:38:07 WARNING[21176]: chan_sip.c:2764 process_sdp: No compatible codecs! Dec 23 02:38:09 NOTICE[21176]: chan_sip.c:7295 handle_request: Unable to create/find channel My sip.conf contains: disallow=all allow=g729 for both devices!! I dont know what to do, I need to use the g729 codec. Please help. If I enable GSM in the device and add allow=gsm everything works, so it is a codec problem. The license and the codec seems to be correctly installed: *CLI show g729 0/0 encoders/decoders of 1 licensed channels are currently in use *CLI Please, help me. RODOLFO Rodolfo, I assume that you tried the obvious stuff, re-reading the docs, running the register program, getting the proper g729 binary from Digium, etc? -- Kristian Kielhofner ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Message: 2 Date: Wed, 22 Dec 2004 18:40:03 -0800 From: Norman Zhang [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Can't Receive/Send Calls To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed With this trimmed down versions of sip.conf and extensions.conf. I can now receive calls from outside. But audio will not traverse out to the internet. I can hear the caller no problem. Also I still cannot dial-out. Any ideas? Regards, Norman Zhang I removed all the PSTN stuffs. As I'm only trying to make SIP work. Would someone kindly give me a few pointers? [general] disallow=all allow=ulaw port=5060 bindaddr=0.0.0.0 externip=207.34.136.26 localnet=192.168.22.0/255.255.255.0 context=inbound-sip maxexpirey=180 defaultexpirey=160 tos=reliability srvlookup=yes register = 533990:[EMAIL PROTECTED]/533990 [fwd] type=friend secret=normanzhang username=533990 fromuser=533990 fromdomain=fwd.pulver.com host=fwd.pulver.com dtmfmode=inband nat=yes canreinvite=no [101] disallow=all allow=ulaw type=friend host=dynamic dtmfmode=inband username=101 secret=testing123 context=home nat=no ; extensions.conf [general] static=yes writeprotect=no [globals] MAINPHONE=SIP/101 FWDUSERID=533990 FWDUSERNAME=Norman Zhang FWDPREFIX=* ; Macros
[Asterisk-Users] where I can find some learning book about asterisk?
Hello , I learn handbook-draft.but I think I don't understand asterisk. where I can find some learning book about asterisk? thank u. B.R. John. -- : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] : 20041224 7:51 : asterisk-users@lists.digium.com : Asterisk-Users Digest, Vol 5, Issue 350 Send Asterisk-Users mailing list submissions to asterisk-users@lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than Re: Contents of Asterisk-Users digest... Today's Topics: 1. RE: rtp channels not through asterisk (Brian West) 2. Turning * Hangup off in queues ([EMAIL PROTECTED]) 3. Re: Voicemail email notification (Rich Adamson) 4. Can't Make Outgoing Call (Norman Zhang) 5. Re: Voicemail email notification (Dorn Hetzel) 6. Re: Asterisk in parallel with PSTN [OT] (Rich Adamson) 7. Re: rtp channels not through asterisk (Rich Adamson) 8. Re: Realtime sipbuddies table structure why? (Greg - Cirelle Enterprises) 9. RE: Polycom Buddies (Paul Hales) 10. Re: Queue - roundrobin member order (Adam Goryachev) 11. Re: Voicemail email notification (Rich Adamson) 12. Re: Can't Make Outgoing Call (Norman Zhang) 13. Re: Recommended IAX softphone. (Bruno Hertz) 14. Re: sip seeding vs registration (Greg - Cirelle Enterprises) 15. Asterisk 1.0.3 no RedHat zaptel script? (Jerry Geis) 16. Re: Recommended IAX softphone. (Erik Espinoza) -- Message: 1 Date: Thu, 23 Dec 2004 16:51:22 -0600 From: Brian West [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] rtp channels not through asterisk To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=US-ASCII canreinvite=yes Aterisk stays in the signaling path so unless you're running tcpdump or the like you'll never notice this. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of bijan Sent: Thursday, December 23, 2004 4:46 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] rtp channels not through asterisk In wiki pages it is stated that The audio channels (RTP) may go directly from phone to phone or may go through Asterisk's media bridge. Currently with my settings, I notice that all rtp's are passing through my asterisk. How could I achieve that they go directly from phone to phone? I assume this way, my machine will have less load and therefore could handle more calls. regards Bijan Karimi -- Message: 2 Date: Thu, 23 Dec 2004 19:16:19 -0600 (CST) From: [EMAIL PROTECTED] Subject: [Asterisk-Users] Turning * Hangup off in queues To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: TEXT/PLAIN; charset=US-ASCII Hi ! Can somebody tell me how to turn the * Hangup option utrned off in queues. I have not used any H option but still as an agent if I press * key the user gets disconnected. Somehow it is turned on by default. Can I turn this option off In my extensions.conf I have written : exten = 8000,3,Queue(supportq|t) plz help me inthis regard ... Thanks ! Usman. -- Message: 3 Date: Thu, 23 Dec 2004 16:51:34 -0600 From: Rich Adamson [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Voicemail email notification To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: TEXT/PLAIN; CHARSET=ISO-8859-1 Are there any common silent failure modes for email notification from the Voicemail module. I put the email and pager email addresses in my entry in voicemail.conf but no mail gets sent when I leave a voicemail. No obvious error messages either, unless I'm just not looking in the right place. Thanks for any clues :) Nop, that's it other then you have to have sendmail configured and running on the system (or have a substitute mail handler). Rich -- Message: 4 Date: Thu, 23 Dec 2004 14:58:04 -0800 From: Norman Zhang [EMAIL PROTECTED] Subject: [Asterisk-Users] Can't Make Outgoing Call To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed Hi, I can't get dial-out working. I'm trying to call 523936. Is there something wrong with my setup here? Could someone please give me a few pointers? Regards, Norman Zhang [fwd-out] exten =
[Asterisk-Users] help:could asterisk be used such as sip proxy?
I means that asterisk can be used such as PBX,and streams across PBX. and if asterisk used by sip proxy,I can use asterisk in internet network. and could I let enterprise asterisk PBX behind nat network connect each other by internet asterisk sip proxy ? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
