Re: [asterisk-users] doorphone?
could anybody suggest a usable doorphone and magnetic door opener hardphone system for me, please? Of course should be connectable to asterisk. I am in the EU, should be available here. I would recommend using a normal doorphone, and connecting it to a SIP gateway like the PAP2T. Otherwise, you need a network connection directly into the doorphone unit, and some people don't like that because it can give a hacker/burglar, direct access to your internal network. Hope that helps. Dan Journo Kesher Communications (UK) Business Phone Systemshttp://www.keshercommunications.com/ | Hosted PBXhttp://www.keshercommunications.com/hostedpbx.html -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIPAddHeader not working
Hello list, I notice that the dialplan method SIPAddHeader is not working : in dialplan : /exten = s,n,SIPAddHeader(Privacy: id)/ in SIP invite no trace of this header : /INVITE sip:0...@sip.domain.be SIP/2.0 Via: SIP/2.0/UDP 192.168.1.106:5063;branch=z9hG4bK-5b2b1b97 From: VC sip:vo...@sip.domain.be;tag=729476652f511c67o2 To: sip:0...@sip.domain.be Remote-Party-ID: VC sip:vo...@sip.domain.be;screen=yes;party=calling Call-ID: 307124bd-f6881ef@192.168.1.106 CSeq: 101 INVITE Max-Forwards: 70 Contact: VC sip:voip2@192.168.1.106:5063 Expires: 240 User-Agent: Linksys/SPA941-5.1.8 Content-Length: 401 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp/ Using Asterisk 1.6.2.16.1 How do I correctly add the Privacy header ?! Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] doorphone?
Hi, I've been used a Alphatech doorphone (SIP) with asterisk and works fine. Cumps Com os melhores cumprimentos, Best regards, CÉSAR SEQUEIRA IT Expert M: +351 961 355 772 @: cesar-seque...@justbit.pt skype: cesar.sequeira.justbit msn: cesar-seque...@justbit.pt -Mensagem original- De: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Em nome de Tóth Csaba Enviada: quarta-feira, 9 de Março de 2011 05:36 Para: Asterisk Users Mailing List - Non-Commercial Discussion Assunto: [asterisk-users] doorphone? Hi, could anybody suggest a usable doorphone and magnetic door opener hardphone system for me, please? Of course should be connectable to asterisk. I am in the EU, should be available here. thank you, Csaba -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [1.4] Reading phone number the French way?
On Tue, Mar 08, 2011 at 02:58:02PM -0600, Tilghman Lesher wrote: On Tuesday 08 March 2011 06:49:55 Faisal Hanif wrote: You can also set it in dialplan using Set(LANGUAGE=FR) Actually, the right way to do this is: Set(CHANNEL(language)=fr) The LANGUAGE pseudo-variable is read-only. Also note that Asterisk will use French number saying rules as long as the language (the LANGUAGE pseudo variable) is fr or its prefix (anything up to the first '_') is fr. If you want to test that, try on a system with no French prompts but with the standard English ones: ln -s en /var/lib/asterisk/sounds/fr Or alternatively: ln -s en /var/lib/asterisk/sounds/fr_WhatEver Now set the language (as described above) to 'fr_WhatEver' and try using SayNumber. e.g.: channel originate SIP/yourphone Application SayNumber 245 I figure a prompt or two will be missing. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] trunk not working if I register a phone at the same IP as the trunk peer's IP
1.8 supports static peers along with realtime peers. I have tested. On Mon, Feb 21, 2011 at 11:04 PM, Ricardo Carvalho rjcarvalho.li...@gmail.com wrote: Thanks Faisal, in fact I made a test that confirmed that in realtime asterisk doesn’t supported static peers, like you told me. Do you know if newer versions of asterisk, like 1.8, have this issue already solved? Regards, Ricardo. On Wed, Feb 16, 2011 at 6:26 PM, Faisal Hanif fai...@vopium.com wrote: I have played a lot on this issue with asterisk config but in realtime it doesn’t supported static peers with version 1.6.2.14. *From:* Ricardo Carvalho [mailto:rjcarvalho.li...@gmail.com] *Sent:* Wednesday, February 16, 2011 10:21 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Cc:* Faisal Hanif *Subject:* Re: [asterisk-users] trunk not working if I register a phone at the same IP as the trunk peer's IP Isn't this a limitation that can be surpassed with some configuration that I'm lacking in my sip.conf or extensions.conf of my asterisk? Ricardo. On Wed, Feb 16, 2011 at 4:54 PM, Faisal Hanif fai...@vopium.com wrote: Well a quick n easy fix for you is you can configure you call sending peers to use username secret in INVITE. As far as I know it possible in almost all CISCO, Avaya and all other standard Gateway and SBCs which follows full SIP RFCs. If you can’t do it then you need to use curl as realtime engine instead of MySQL. It will call a URL for each SIP request which you can handle with flexibility in your CGI scripts with apache. But be careful as per my experience asterisk 1.6 with curl as realtime engine can handle a max of 120 registration in parallel if registration refresh time is 120 seconds. Faisal Hanif *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Ricardo Carvalho *Sent:* Wednesday, February 16, 2011 9:41 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] trunk not working if I register a phone at the same IP as the trunk peer's IP How should I configure my asterisk server so that I can receive calls from an unregistered peer from whom I also receive registrations of sip phones? I'm asking you this, because with my actual configuration, when I register a contact from that peer's IP, no more inbound calls are accepted from that peer, as my asterisk rejects those INVITEs with 407 Proxy Authentication Required, I assume because they don't carry the registered contact registration!!! My SIP contacts have type=friend and all inbound calls not coming from my registered phones fall in the default context without authentication, so that someone in the Internet be able to call freely through the Internet anyone in my server's dial plan. Some ideas? Regards, Ricardo Carvalho. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 6767 26 E: rizwanhas...@gmail.com W: www.axvoice.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [Opinion Request] SIP phones that work well with Asterisk
Hi, Would you recommend some standalone SIP phones that work well with Asterisk? Personal experience preferred. Thanks, -- Raj -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Multiple SIP endpoint registrations
Hi, With Asterisk 1.8 is it now possible to register the same SIP account at multiple endpoints and for both to ring when the associated extension is dialed ? -- Thanks, Phil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Opinion Request] SIP phones that work well with Asterisk
I highly recommend Yealink phones. On Wed, Mar 9, 2011 at 7:01 PM, Raj Mathur r...@linux-delhi.org wrote: Hi, Would you recommend some standalone SIP phones that work well with Asterisk? Personal experience preferred. Thanks, -- Raj -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Opinion Request] SIP phones that work well with Asterisk
Grandstream GXV3140 Multimedia IP Phones. Fully SIP capable. work well with Asterisk. On Wed, 2011-03-09 at 16:31 +0530, Raj Mathur wrote: Hi, Would you recommend some standalone SIP phones that work well with Asterisk? Personal experience preferred. Thanks, -- Raj -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Opinion Request] SIP phones that work well with Asterisk
My personal experience : Corded : Snom 3xx and 8xx, Aastra 6731i, 6755i and 6757i and Polycom IP330, IP650. DECT : Siemens C470, Polycom Kirk KWS300 and 600v3 Work well AB2L +33 (0)367100783 sebast...@ab2l.eu Le 09/03/2011 13:09, John Kosmas a écrit : Grandstream GXV3140 Multimedia IP Phones. Fully SIP capable. work well with Asterisk. On Wed, 2011-03-09 at 16:31 +0530, Raj Mathur wrote: Hi, Would you recommend some standalone SIP phones that work well with Asterisk? Personal experience preferred. Thanks, -- Raj -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Opinion Request] SIP phones that work well with Asterisk
Arstan, Very Nice phone indeed. Yealink and the GXV3140's Grandstream's are quite nice! On Wed, 2011-03-09 at 20:04 +0800, Arstan Jusupov wrote: Re: [asterisk-users] [Opinion Request] SIP phones that work well with Asterisk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Opinion Request] SIP phones that work well with Asterisk
Siemens IP A580 works fairly well. 2011/3/9 Sébastien BERGER sebast...@ab2l.eu My personal experience : Corded : Snom 3xx and 8xx, Aastra 6731i, 6755i and 6757i and Polycom IP330, IP650. DECT : Siemens C470, Polycom Kirk KWS300 and 600v3 Work well AB2L +33 (0)367100783 sebast...@ab2l.eu Le 09/03/2011 13:09, John Kosmas a écrit : Grandstream GXV3140 Multimedia IP Phones. Fully SIP capable. work well with Asterisk. On Wed, 2011-03-09 at 16:31 +0530, Raj Mathur wrote: Hi, Would you recommend some standalone SIP phones that work well with Asterisk? Personal experience preferred. Thanks, -- Raj -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIPAddHeader not working
From: Jonas Kellens jonas.kell...@telenet.be
Sent: Wednesday, March 09, 2011 4:18 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: [asterisk-users] SIPAddHeader not working
Hello list,
I notice that the dialplan method SIPAddHeader is not working :
in dialplan :
exten = s,n,SIPAddHeader(Privacy: id)
in SIP invite no trace of this header :
Using Asterisk 1.6.2.16.1
How do I correctly add the Privacy header ?!
Kind regards,
Jonas.
Jonas
Here is the way we add the rfc-3325 privacey header so our vendors pick it
up correctly. This is what we use in 1.6.x and 1.8.x
When I check on my versions the privacy header appears to be there.
exten = rfc-3325-CPN,1,NoOp(Set Call Privacy)
exten = rfc-3325-CPN,n,NoOp(From ${SIP_HEADER(From)})
exten = rfc-3325-CPN,n,NoOp(To ${SIP_HEADER(To)})
exten =
rfc-3325-CPN,n,Set(l_sipheaderfromip=${CUT(SIP_HEADER(From),@,2)})
exten = rfc-3325-CPN,n,GotoIf($[${l_sipheaderfromip} != ]?hasat)
exten =
rfc-3325-CPN,n,Set(l_sipheaderfromip=${CUT(CUT(SIP_HEADER(From),,1),:,2)})
exten = rfc-3325-CPN,n,Goto(gotip)
exten =
rfc-3325-CPN,n(hasat),Set(FROM_IP=${CUT(CUT(CUT(SIP_HEADER(From),@,2),,1),:
,1)})
exten = rfc-3325-CPN,n(gotip),NoOp(Gateway IP is ${FROM_IP})
exten =
rfc-3325-CPN,n,SIPAddHeader(P-Preferred-Identity:${CALLERID(name)}
sip:+1${CALLERID(num)}@${FROM_IP}\;user=phone)
exten = rfc-3325-CPN,n,SIPAddHeader(Privacy: id)
exten = rfc-3325-CPN,n,Set(CALLERPRES()=prohib_not_screened)
exten = rfc-3325-CPN,n,Set(CALLERID(num)=Anonymous)
exten = rfc-3325-CPN,n,Set(CALLERID(name)=Anonymous)
exten = rfc-3325-CPN,n,Return()
--
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[asterisk-users] Asterisk pri card replecement
Hey guys, Currently we have non HWEC sangoma pri card but now we are planing to replace card with HWEC support card for echo cancellation. So in this case do I need to re-install everything? Like zaptel, asterisk etc.. Or just replace the card? -- Sent from my iPhone -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Opinion Request] SIP phones that work well with Asterisk
Phones what I'm using: D-Link: 150s, 300S Nokia E60, E65, E71, E52 Cisco: 7940, 7960 All above works perfect :) On Wed, Mar 9, 2011 at 2:33 PM, asterisk asterisk aster...@ck-lee.com wrote: Siemens IP A580 works fairly well. 2011/3/9 Sébastien BERGER sebast...@ab2l.eu My personal experience : Corded : Snom 3xx and 8xx, Aastra 6731i, 6755i and 6757i and Polycom IP330, IP650. DECT : Siemens C470, Polycom Kirk KWS300 and 600v3 Work well AB2L +33 (0)367100783 sebast...@ab2l.eu Le 09/03/2011 13:09, John Kosmas a écrit : Grandstream GXV3140 Multimedia IP Phones. Fully SIP capable. work well with Asterisk. On Wed, 2011-03-09 at 16:31 +0530, Raj Mathur wrote: Hi, Would you recommend some standalone SIP phones that work well with Asterisk? Personal experience preferred. Thanks, -- Raj -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- -- With regards, Eugene Sudyr -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple SIP endpoint registrations
From: --[ UxBoD ]-- ux...@splatnix.net Sent: Wednesday, March 09, 2011 6:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] Multiple SIP endpoint registrations Hi, With Asterisk 1.8 is it now possible to register the same SIP account at multiple endpoints and for both to ring when the associated extension is dialed ? -- Thanks, Phil Phil Based on what we have seen you must have a sip account per end point. If you want to ring multiple endpoints you can specify them in the dial command exten = exp,n,Dial(SIP/Account1SIP/Account2SIP/Account3, options). This is the only way we know of to do this as you must have an IP and port number to send traffic to and we have seen no method of having two IP's and Ports per account. The only other way I could think of is some outside the box multicast method and the endpoints would need to be set to receive any SIP traffice without registration. This would not be secure and to my knowledge would be beyond basic asterisk at this time. Thanks Bryant -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple SIP endpoint registrations
You can register multiple end users with only one sip account but asterisk does not support ringing all the registered phones on single account. Whenever a new registration comes, asterisk updates its contact info in memory. So if the registration is coming from multiple end users (multiple ip address and port) then the call will be placed to the phone who sent latest registration request. Asterisk does not keep track of all the ip addresses for single account registration. What we have done to ring all the end users with same account is that we listen to registration requests thru manager api in order to detect multiple registration. If we have detected multiple registration then we store the contact information of all the end user phones which are related to single account. And when asterisk receives a dial request for that user, we create a temporary/fake users (as many as needed) in memory and dial all of them in the code not thru Dial application as it does not support thsi scenario. We are still working on this scenario. It is in working condition but in testing phase. On Wed, Mar 9, 2011 at 4:14 PM, --[ UxBoD ]-- ux...@splatnix.net wrote: Hi, With Asterisk 1.8 is it now possible to register the same SIP account at multiple endpoints and for both to ring when the associated extension is dialed ? -- Thanks, Phil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 6767 26 E: rizwanhas...@gmail.com W: www.axvoice.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [1.4] Reading phone number the French way?
On Tue, 08 Mar 2011 07:47:39 EST, ken...@gnat.com (Richard Kenner)
wrote:
Maybe something like:
exten = s,n,SayDigits(${NBR2CALL:0:1})
exten = s,n,SayNumber(${NBR2CALL:2:2})
exten = s,n,SayNumber(${NBR2CALL:4:2})
exten = s,n,SayNumber(${NBR2CALL:6:2})
exten = s,n,SayNumber(${NBR2CALL:8:2})
Or make changes in say.conf.
Thanks for the great tip.
Adding the FR sound files and say.conf, and making sure zapata.conf
and sip.conf set the right language sort of worked: Using the trick
above, Asterisk does read the number correctly, expect that in case a
tuple is 00, it only says zero instead of zero-zero, and when
using 01 it just says one instead of zero-one. I guess Asterisk
ignores leading zeros:
;... eighty-one,zero
exten = ,1,Set(NBR2CALL=0142928100)
exten = ,n,SayDigits(${NBR2CALL:0:1})
exten = ,n,SayDigits(${NBR2CALL:1:1})
exten = ,n,SayNumber(${NBR2CALL:2:2})
exten = ,n,SayNumber(${NBR2CALL:4:2})
exten = ,n,SayNumber(${NBR2CALL:6:2})
exten = ,n,SayNumber(${NBR2CALL:8:2})
;... eighty-one,one
;exten = ,n,Set(NBR2CALL=0142928101)
exten = ,n,SayDigits(${NBR2CALL:0:1})
exten = ,n,SayDigits(${NBR2CALL:1:1})
exten = ,n,SayNumber(${NBR2CALL:2:2})
exten = ,n,SayNumber(${NBR2CALL:4:2})
exten = ,n,SayNumber(${NBR2CALL:6:2})
exten = ,n,SayNumber(${NBR2CALL:8:2})
exten = ,n,Playback(demo-thanks)
exten = ,n,Hangup
Actually, removing say.conf and restarting Asterisk doesn't seem to
have an impact: Does Asterisk really use say.conf, and does it add
features that could solve this issue?
Thank you.
--
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Re: [asterisk-users] Asterisk pri card replecement
Only by replacing it.should not be a problem. Juan. Linux User #441131 On Wed, Mar 9, 2011 at 8:13 AM, Satish Patel satish...@hotmail.com wrote: Hey guys, Currently we have non HWEC sangoma pri card but now we are planing to replace card with HWEC support card for echo cancellation. So in this case do I need to re-install everything? Like zaptel, asterisk etc.. Or just replace the card? -- Sent from my iPhone -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [1.4] Reading phone number the French way?
On Wed, 9 Mar 2011 12:43:37 +0200, Tzafrir Cohen
tzafrir.co...@xorcom.com wrote:
On Tue, Mar 08, 2011 at 02:58:02PM -0600, Tilghman Lesher wrote:
On Tuesday 08 March 2011 06:49:55 Faisal Hanif wrote:
You can also set it in dialplan using Set(LANGUAGE=FR)
Actually, the right way to do this is:
Set(CHANNEL(language)=fr)
The LANGUAGE pseudo-variable is read-only.
Also note that Asterisk will use French number saying rules as long as
the language (the LANGUAGE pseudo variable) is fr or its prefix
(anything up to the first '_') is fr.
Thanks for the information. With say.conf in /etc/asterisk, I used the
following dialplan:
==
exten = ,1,NoOp(Pseudo LANGUAGE is ${LANGUAGE})
exten = ,n,Set(CHANNEL(language)=fr)
exten = ,n,Set(NBR2CALL=0142928100)
exten = ,n,SayNumber(${NBR2CALL})
exten = ,Hangup
==
Unfortunately, the LANGUAGE variable is empty, and Asterisk still
reads the number as 142 million, 928 thousand, one hundred:
==
-- Executing [@internal:1] NoOp(SIP/xlite-03640004, Pseudo
LANGUAGE is ) in new stack
-- Executing [@internal:2] Set(SIP/xlite-03640004,
CHANNEL(language)=fr) in new stack
-- Executing [@internal:3] Set(SIP/xlite-03640004,
NBR2CALL=0142928100) in new stack
-- Executing [@internal:4] SayNumber(SIP/xlite-03640004,
0142928100) in new stack
-- SIP/xlite-03640004 Playing 'digits/hundred' (language 'fr')
-- SIP/xlite-03640004 Playing 'digits/40' (language 'fr')
-- SIP/xlite-03640004 Playing 'digits/2' (language 'fr')
-- SIP/xlite-03640004 Playing 'digits/million' (language 'fr')
-- SIP/xlite-03640004 Playing 'digits/9' (language 'fr')
-- SIP/xlite-03640004 Playing 'digits/hundred' (language 'fr')
-- SIP/xlite-03640004 Playing 'digits/20' (language 'fr')
-- SIP/xlite-03640004 Playing 'digits/8' (language 'fr')
-- SIP/xlite-03640004 Playing 'digits/thousand' (language 'fr')
==
FYI, I put language=fr in zapata.conf and sip.conf. Do I need to use
another function than SayNumber() in the diaplan?
Thank you.
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Re: [asterisk-users] OT: OpenSIPS vs Kamailio -- which do you use and why?
On Fri, 4 Mar 2011, Steve Edwards wrote: I'm starting a new project similar to a previous project where I used OpenSER to front a bunch of Asterisk servers. Now that OpenSER is gone, OpenSIPS and Kamailio seem the likely candidates. I'm leaning towards OpenSIPS because it's in EPEL so I can install it with yum. Also, because I think the name sounds more 'professional' when discussing architecture with clients :) Which do you use and why? So I got 1 'vote' for each. Surely more than 2 users use OpenSIPS or Kamailio. I guess Friday afternoon is not the best time to post an open question :) -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help on incoming
...or for DAHDI channnels - the same thing in chan_dahdi.conf -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bakko Sent: 07 March 2011 19:12 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Help on incoming Hi, for sip channels, look at faxdetect options on the sip.conf file BR - Andrea If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIPAddHeader not working
On 03/09/2011 02:09 PM, Bryant Zimmerman wrote:
*From*: Jonas Kellens jonas.kell...@telenet.be
*Sent*: Wednesday, March 09, 2011 4:18 AM
*To*: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
*Subject*: [asterisk-users] SIPAddHeader not working
Hello list,
I notice that the dialplan method SIPAddHeader is not working :
in dialplan :
/exten = s,n,SIPAddHeader(Privacy: id)/
in SIP invite no trace of this header :
Using Asterisk 1.6.2.16.1
How do I correctly add the Privacy header ?!
Kind regards,
Jonas.
Jonas
Here is the way we add the rfc-3325 privacey header so our vendors
pick it up correctly. This is what we use in 1.6.x and 1.8.x
When I check on my versions the privacy header appears to be there.
exten = rfc-3325-CPN,1,NoOp(Set Call Privacy)
exten = rfc-3325-CPN,n,NoOp(From ${SIP_HEADER(From)})
exten = rfc-3325-CPN,n,NoOp(To ${SIP_HEADER(To)})
exten =
rfc-3325-CPN,n,Set(l_sipheaderfromip=${CUT(SIP_HEADER(From),@,2)})
exten = rfc-3325-CPN,n,GotoIf($[${l_sipheaderfromip} != ]?hasat)
exten =
rfc-3325-CPN,n,Set(l_sipheaderfromip=${CUT(CUT(SIP_HEADER(From),,1),:,2)})
exten = rfc-3325-CPN,n,Goto(gotip)
exten =
rfc-3325-CPN,n(hasat),Set(FROM_IP=${CUT(CUT(CUT(SIP_HEADER(From),@,2),,1),:,1)})
exten = rfc-3325-CPN,n(gotip),NoOp(Gateway IP is ${FROM_IP})
exten =
rfc-3325-CPN,n,SIPAddHeader(P-Preferred-Identity:${CALLERID(name)}
sip:+1${CALLERID(num)}@${FROM_IP}\;user=phone)
exten = rfc-3325-CPN,n,SIPAddHeader(Privacy: id)
exten = rfc-3325-CPN,n,Set(CALLERPRES()=prohib_not_screened)
exten = rfc-3325-CPN,n,Set(CALLERID(num)=Anonymous)
exten = rfc-3325-CPN,n,Set(CALLERID(name)=Anonymous)
exten = rfc-3325-CPN,n,Return()
I see no great difference. What does
/Set(CALLERPRES()=prohib_not_screened)/ do ?
How does your INVITE look like ? Does the header /Privacy: id/ appears
? Because it does not in my INVITE.
Kind regards,
Jonas.
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Re: [asterisk-users] [Opinion Request] SIP phones that work well with Asterisk
Would you recommend some standalone SIP phones that work well with Asterisk? Personal experience preferred. We use Polycom phones. Mainly the IP321. We chose them because they can be easily provisioned using an FTP server which allows us to configure settings without visiting the phone, and the phone can be rebooted through Asterisk to update the settings instantly. It also stores the phone's directory on the FTP server so users don't lose any contacts if the phone needs replacing. Hope that helps. Dan Journo Kesher Communications (UK) Business Phone Systemshttp://www.keshercommunications.com/ | Hosted PBXhttp://www.keshercommunications.com/hostedpbx.html -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: OpenSIPS vs Kamailio -- which do you use and why?
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Wednesday, March 09, 2011 9:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] OT: OpenSIPS vs Kamailio -- which do you use and why? On Fri, 4 Mar 2011, Steve Edwards wrote: I'm starting a new project similar to a previous project where I used OpenSER to front a bunch of Asterisk servers. Now that OpenSER is gone, OpenSIPS and Kamailio seem the likely candidates. I'm leaning towards OpenSIPS because it's in EPEL so I can install it with yum. Also, because I think the name sounds more 'professional' when discussing architecture with clients :) Which do you use and why? So I got 1 'vote' for each. Surely more than 2 users use OpenSIPS or Kamailio. I guess Friday afternoon is not the best time to post an open question :) Probably not, no. :) I'll throw my vote in for Kamailio. I've been using it (and OpenSER before the fork/rename) for about 5 years now, and have never had an issue that wasn't my own fault (misconfiguration, etc.). - Brad -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Opinion Request] SIP phones that work well with Asterisk
I use Yealink, Linksys and Grandstream phones. Com os melhores cumprimentos, Best regards, CÉSAR SEQUEIRA IT Expert M: +351 961 355 772 @: cesar-seque...@justbit.pt skype: cesar.sequeira.justbit msn: cesar-seque...@justbit.pt -Mensagem original- De: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Em nome de Raj Mathur Enviada: quarta-feira, 9 de Março de 2011 11:02 Para: asterisk-users@lists.digium.com Assunto: [asterisk-users] [Opinion Request] SIP phones that work well with Asterisk Hi, Would you recommend some standalone SIP phones that work well with Asterisk? Personal experience preferred. Thanks, -- Raj -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] doorphone?
On 03/09/2011 02:57 AM, Dan Journo wrote: could anybody suggest a usable doorphone and magnetic door opener hardphone system for me, please? Of course should be connectable to asterisk. I am in the EU, should be available here. I would recommend using a normal doorphone, and connecting it to a SIP gateway like the PAP2T. Otherwise, you need a network connection directly into the doorphone unit, and some people don't like that because it can give a hacker/burglar, direct access to your internal network. Hope that helps. Dan Journo That's not always true. Some door phones have a remote unit that connects to the network and a local device at the door, giving some better security. I've used the Valcom VIP-172 phones. They are simple and work well. Very good support if you need to call them. http://www.valcom.com/Home_links/sipdoorintercom.htm Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] BLF, Directed pickup and Polycom 601 with SIP 3.1.6
Hi, Latest SIP firmware for Polycom 601 is 3.1.6. With this, is Directed Pickup supported ? At the moment, when an extension is ringing, I can see BLF turning to solid Red but I can't see it turning to Blinking Red. Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] doorphone?
You can Try: Helios from 2N http://www.2n.cz/en/products/communicators/doors/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom Do Not Disturb button and asterisk hints
2011/2/17 Mike l...@net-wall.com Hi, Is there ANY way for me to see the status of the Polycom DND buttons in the Asterisk hints? I`m using the BLF buttons to see the status of other people`s lines, and DND should logically be somehow reflected (I don`t care as much about Polycom showing the BLF button as DND, but I do care about Asterisk hints showing it in the CLI). Right now, a Polycom phone on DND shows up as being idle. Which is it, but it doesn’t help reception say “Sorry he`s not available right now”. I don’t mind paying a reasonable bounty for it, or working closely with anyone to make this feature available, but I`ve been asked this by many customers in the past and it’s starting to be one of those “why the heck not” question. Mike Hi, I tried to work around this by centralizing DND requests in Asterisk and sending back a short (You're in DND mode) text to Polycom's screen (using sipsak for this). This was rather disappointing as Poycoms redirect text messages to an Instant Messaging mailbox and do not keep them visible on screen. Maybe, some king of XML magic would be a better mean to return current DND status to users. Any suggestion ? regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HK DIDs
Sorry, just realised I posted this to the wrong mailing list. Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom Do Not Disturb button and asterisk hints
2011/3/9 Russell Bryant russ...@digium.com - Original Message - I tried to work around this by centralizing DND requests in Asterisk and sending back a short (You're in DND mode) text to Polycom's screen (using sipsak for this). This was rather disappointing as Poycoms redirect text messages to an Instant Messaging mailbox and do not keep them visible on screen. Maybe, some king of XML magic would be a better mean to return current DND status to users. Any suggestion ? One solution that I had come up with for this situation was to use a softkey and use custom device state to have the LED on or off based on whether DND was on or off. I documented it here: http://ofps.oreilly.com/titles/9780596517342/ch14.html#usingCustomDeviceStates -- Russell Bryant Digium, Inc. | Engineering Manager, Open Source Software 445 Jan Davis Drive NW -Huntsville, AL 35806 - USA jabber: rbry...@digium.com-=-skype: russell-bryant www.digium.com -=- www.asterisk.org -=- blogs.asterisk.org - This is interesting but though a LED is perfect binary status such as DND, in fact, I'm more after a status line also showing Screening or Forwarding destination (for instance, Fwd = 12345, Fwd = Cellphone ...). Polycom phones have a custom Status window with which you can pick Forwarding settings but, to my knowledge, it can't used to let Asterisk manage those settings (I would be very happy to be proven wrong). -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] doorphone?
- Original Message - Hi, could anybody suggest a usable doorphone and magnetic door opener hardphone system for me, please? Of course should be connectable to asterisk. I am in the EU, should be available here. thank you, Csaba -- http://www.cyberdata.net/products/voip/digitalanalog/intercom/index.html -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] doorphone?
On Wed, Mar 9, 2011 at 12:53 PM, Darrick Hartman (lists) dhart...@djhsolutions.com wrote: On 03/09/2011 02:57 AM, Dan Journo wrote: could anybody suggest a usable doorphone and magnetic door opener hardphone system for me, please? Of course should be connectable to asterisk. I am in the EU, should be available here. I would recommend using a normal doorphone, and connecting it to a SIP gateway like the PAP2T. Otherwise, you need a network connection directly into the doorphone unit, and some people don't like that because it can give a hacker/burglar, direct access to your internal network. Hope that helps. Dan Journo That's not always true. Some door phones have a remote unit that connects to the network and a local device at the door, giving some better security. I've used the Valcom VIP-172 phones. They are simple and work well. Very good support if you need to call them. http://www.valcom.com/Home_links/sipdoorintercom.htm Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com To repeat and support Darrick's point. Using a doorphone that is analog and or coax for the last 3+ meters will save some headaches down the road. I have used Valcom, Viking and others. With a Xorcom appliance you can also have the contact closure I/O to open doors or ring phones. ~~~ Andrew lathama Latham lath...@gmail.com ~~~ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] doorphone?
On 3/9/11 6:35 AM, Tóth Csaba wrote: could anybody suggest a usable doorphone and magnetic door opener hardphone system for me, please? Of course should be connectable to asterisk. I am in the EU, should be available here. I don't have direct Asterisk exerience, but when I tested http://robin.nl/en/products/robin-compact-sip/ it worked flawlessly; I don't have a doubt it will work with Asterisk. -- Andreas Sikkema -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asked to transmit frame type slin, while native formats is 0x8 (alaw) read/write = 0x4 (ulaw)/0x8 (alaw)
Hello! Client is using ulaw, however server sometimes fills the log with following: [2011-03-09 21:23:07] WARNING[27204] chan_sip.c: Asked to transmit frame type slin, while native formats is 0x8 (alaw) read/write = 0x4 (ulaw)/0x8 (alaw) [2011-03-09 21:23:07] WARNING[27204] chan_sip.c: Asked to transmit frame type slin, while native formats is 0x8 (alaw) read/write = 0x4 (ulaw)/0x8 (alaw) [2011-03-09 21:23:07] WARNING[27204] chan_sip.c: Asked to transmit frame type slin, while native formats is 0x8 (alaw) read/write = 0x4 (ulaw)/0x8 (alaw) [2011-03-09 21:23:07] WARNING[27204] chan_sip.c: Asked to transmit frame type slin, while native formats is 0x8 (alaw) read/write = 0x4 (ulaw)/0x8 (alaw) [2011-03-09 21:23:07] WARNING[27204] chan_sip.c: Asked to transmit frame type slin, while native formats is 0x8 (alaw) read/write = 0x4 (ulaw)/0x8 (alaw) Using asterisk 1.8.4-rc2 What could be the cause? Nick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom Do Not Disturb button and asterisk hints
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: Wednesday, March 09, 2011 12:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom Do Not Disturb button and asterisk hints 2011/3/9 Russell Bryant russ...@digium.com - Original Message - I tried to work around this by centralizing DND requests in Asterisk and sending back a short (You're in DND mode) text to Polycom's screen (using sipsak for this). This was rather disappointing as Poycoms redirect text messages to an Instant Messaging mailbox and do not keep them visible on screen. Maybe, some king of XML magic would be a better mean to return current DND status to users. Any suggestion ? One solution that I had come up with for this situation was to use a softkey and use custom device state to have the LED on or off based on whether DND was on or off. I documented it here: http://ofps.oreilly.com/titles/9780596517342/ch14.html#usingCustomDeviceStates -- Russell Bryant Digium, Inc. | Engineering Manager, Open Source Software 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA jabber: rbry...@digium.com -=- skype: russell-bryant www.digium.com -=- www.asterisk.org -=- blogs.asterisk.org - This is interesting but though a LED is perfect binary status such as DND, in fact, I'm more after a status line also showing Screening or Forwarding destination (for instance, Fwd = 12345, Fwd = Cellphone ...). Polycom phones have a custom Status window with which you can pick Forwarding settings but, to my knowledge, it can't used to let Asterisk manage those settings (I would be very happy to be proven wrong). Another option would be to use Custom: device state like Russell suggest, but instead of a softkey remap the Do Not Disturb button to a speed dial that is configured to be an Enhanced Feature Key macro that includes toggling of DND as well as dialing the extension that changes the Custom: device state. Off the cuff, assuming the number to dial for the device state mojo is 1234, it would probably be something like: $FDoNotDisturb$1234$Tinvite$ Regards, - Brad -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 1.8 and no alsa input
Is there a way to configure asterisk 1.8 and ALSA so I dont read
anything in for the input port.
I tried this in asound.conf
pcm.nullpcm {
type null
{
then in the alsa.conf file
input_device=plug:nullpcm
This did not seem to work as I still get feedback.
Is there a way to do this?
Thanks,
Jerry
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[asterisk-users] Anyone have BRI working with Asterisk 1.8, Latest DAHDI, LibPRI?
Hi, We have a site where we'd like to move from mISDN/chan_lcr to DAHDI with a b410p card. We've tried everything we can think of to get it working but we never seem to receive any calls etc - even though the card has no alarms. We've tried replacing the card, changing the jumpers etc but no go. The cards both work with mISDN and chan_lcr, but we get reasonably frequent crashes. Does anyone have BRI working at all with the latest Asterisk, DAHDI, LibPRI? -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 1.8 still need dahdi
On 11-03-09 12:16 PM, Jerry Geis wrote: Does asterisk 1.8 still need dahdi installed if you only doing SIP and ALSA/console. Only if you plan to use MeetMe(). -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] One Way Audio
I am having trouble with no return audio on inbound calls. I have been working on this for 18 hours and even built a fresh server and moved everything over and I am getting the same results. I need someone that can help get this resolved tonight. I can provide access to all machines involved. Please email me at tim.compnetw...@gmail.com if you can help. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One Way Audio
So that suggests audio is flowing as follows in a unidirectional manner 199.173.66.22.53102 74.204.4.5.11732 IP 74.204.4.5.11732 209.216.2.203.60362 Somewhere near the end this pops up which is slightly different, I am guessing 74.204.4.5 is your asterisk box 19:18:36.389548 IP 74.204.4.5.11732 174.133.195.194.18364: UDP, length 172 I am not sure why this is happening or if its still part of the same conversation Overall it looks a bit like the asterisk box thinks it needs to send rtp to a different location than perhaps its meant to i.e. its asymmetric - you can check the sdp in the sip invite to see where media is expected to be sent to There is no rtp coming back from 209.216.2.203 so possibly this is device that isn't meant to be part of the conversation and either doesn't exist or is not expecting anything and thus not responding What are the addresses of the devices in this conversation? so that you can match the traffic to device Cheers Duncan On 10/03/2011, at 1:20 PM, Tim King wrote: It looks like this: 19:18:34.782016 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.789527 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.802064 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.809757 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.821855 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.829598 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.842015 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.849764 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.861902 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.869568 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.881882 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.889739 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.901882 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.909612 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.921984 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.929664 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.941855 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.949589 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.962003 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.969592 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.981851 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.989543 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.002006 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.009973 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.022008 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.029539 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.042071 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.049561 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.062008 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.069612 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.081986 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.089519 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.101918 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.109722 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.122021 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.129590 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.141878 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.149709 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.161886 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.169561 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.181879 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.189710 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.201965 IP 199.173.66.22.53103 74.204.4.5.11733: UDP, length 60 19:18:35.201974 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.209552 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.221898 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.229625 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.241894 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.249566 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.261999 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.269701 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.281873 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.289521 IP 74.204.4.5.11732
Re: [asterisk-users] One Way Audio
209.216.2.203 is sip signaling server and 199.173.66.22 is media servers. BTW Did you try config_1 option. Please send us your configuration and we will help you configure it properly. Either you can post them here or you can send them directly to contact-supp...@didforsale.com Jai www.didforsale.com. On Wed, Mar 9, 2011 at 4:43 PM, Duncan Turnbull dun...@e-simple.co.nzwrote: So that suggests audio is flowing as follows in a unidirectional manner 199.173.66.22.53102 74.204.4.5.11732 IP 74.204.4.5.11732 209.216.2.203.60362 Somewhere near the end this pops up which is slightly different, I am guessing 74.204.4.5 is your asterisk box 19:18:36.389548 IP 74.204.4.5.11732 174.133.195.194.18364: UDP, length 172 I am not sure why this is happening or if its still part of the same conversation Overall it looks a bit like the asterisk box thinks it needs to send rtp to a different location than perhaps its meant to i.e. its asymmetric - you can check the sdp in the sip invite to see where media is expected to be sent to There is no rtp coming back from 209.216.2.203 so possibly this is device that isn't meant to be part of the conversation and either doesn't exist or is not expecting anything and thus not responding What are the addresses of the devices in this conversation? so that you can match the traffic to device Cheers Duncan On 10/03/2011, at 1:20 PM, Tim King wrote: It looks like this: 19:18:34.782016 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.789527 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.802064 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.809757 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.821855 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.829598 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.842015 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.849764 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.861902 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.869568 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.881882 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.889739 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.901882 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.909612 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.921984 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.929664 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.941855 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.949589 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.962003 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.969592 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.981851 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.989543 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.002006 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.009973 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.022008 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.029539 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.042071 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.049561 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.062008 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.069612 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.081986 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.089519 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.101918 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.109722 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.122021 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.129590 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.141878 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.149709 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.161886 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.169561 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.181879 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.189710 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.201965 IP 199.173.66.22.53103 74.204.4.5.11733: UDP, length 60 19:18:35.201974 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.209552 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.221898 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.229625 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.241894 IP
Re: [asterisk-users] One Way Audio
You can use this link too. http://www.didforsale.com/blog/how-to-setup-your-asterisk-server-with-didforsale Keep the context as context=from-trunk. -Jai On Wed, Mar 9, 2011 at 5:01 PM, Jai Rangi jpra...@didforsale.com wrote: 209.216.2.203 is sip signaling server and 199.173.66.22 is media servers. BTW Did you try config_1 option. Please send us your configuration and we will help you configure it properly. Either you can post them here or you can send them directly to contact-supp...@didforsale.com Jai www.didforsale.com. On Wed, Mar 9, 2011 at 4:43 PM, Duncan Turnbull dun...@e-simple.co.nzwrote: So that suggests audio is flowing as follows in a unidirectional manner 199.173.66.22.53102 74.204.4.5.11732 IP 74.204.4.5.11732 209.216.2.203.60362 Somewhere near the end this pops up which is slightly different, I am guessing 74.204.4.5 is your asterisk box 19:18:36.389548 IP 74.204.4.5.11732 174.133.195.194.18364: UDP, length 172 I am not sure why this is happening or if its still part of the same conversation Overall it looks a bit like the asterisk box thinks it needs to send rtp to a different location than perhaps its meant to i.e. its asymmetric - you can check the sdp in the sip invite to see where media is expected to be sent to There is no rtp coming back from 209.216.2.203209.216.2.203 so possibly this is device that isn't meant to be part of the conversation and either doesn't exist or is not expecting anything and thus not responding What are the addresses of the devices in this conversation? so that you can match the traffic to device Cheers Duncan On 10/03/2011, at 1:20 PM, Tim King wrote: It looks like this: 19:18:34.782016 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.789527 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.802064 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.809757 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.821855 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.829598 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.842015 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.849764 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.861902 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.869568 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.881882 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.889739 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.901882 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.909612 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.921984 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.929664 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.941855 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.949589 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.962003 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.969592 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:34.981851 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:34.989543 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.002006 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.009973 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.022008 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.029539 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.042071 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.049561 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.062008 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.069612 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.081986 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.089519 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.101918 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.109722 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.122021 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.129590 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.141878 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.149709 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.161886 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.169561 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.181879 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172 19:18:35.189710 IP 74.204.4.5.11732 209.216.2.203.60362: UDP, length 172 19:18:35.201965 IP 199.173.66.22.53103 74.204.4.5.11733: UDP, length 60 19:18:35.201974 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172
[asterisk-users] [1.8] Unable to Register: Registration denied because of contact ACL
Hello All, Some new security stuff is going on I suppose in 1.8 that I am not familiar with and would appreciate your help In a scenario such as the following: Internet -- SBC -- Asterisk upon trying to register an endpoint, the following is being observed on the Asterisk Console. Have Googled this but haven't come up with anything that helped much. [Mar 10 11:53:59] ERROR[21272]: netsock2.c:94 ast_sockaddr_stringify_fmt: getnameinfo(): ai_family not supported [Mar 10 11:53:59] WARNING[21272]: chan_sip.c:13120 parse_register_contact: Domain '172.16.16.6:5060' disallowed by contact ACL (violating IP ) [Mar 10 11:53:59] WARNING[21272]: chan_sip.c:13837 register_verify: Registration denied because of contact ACL Note, that the server IP is 172.16.16.11 and the SBC internal Interface IP is 172.16.16.6 the following lines have been added to sip.conf dynamic_exclude_static = yes autodomain=yes domain=172.16.16.6 allowexternaldomains=no In addition, in the general endpoint template in sip.conf, I have the lines contactdeny=0.0.0.0/0.0.0.0 contactpermit=172.16.16.0/255.255.255.0 host=dynamic What else am I missing? Thanks \RR -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Opinion Request] SIP phones that work well with Asterisk
On Wednesday 09 Mar 2011, Raj Mathur wrote: Would you recommend some standalone SIP phones that work well with Asterisk? Personal experience preferred. Thanks to all who replied. Regards, -- Raj -- Raj Mathurr...@kandalaya.org http://kandalaya.org/ GPG: 78D4 FC67 367F 40E2 0DD5 0FEF C968 D0EF CC68 D17F PsyTrance Chill: http://schizoid.in/ || It is the mind that moves -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ChanSpy with alphanumeric SIP channels [1.6.2]
Hi, I'm using SIP users of the form 'ab_12345' (two letters, underscore, 5 digits). ChanSpy is working fine for listening in to conversations initiated by these channels, and I can use '*' to randomly switch channels. However, is there any way in this scenario to be able to switch ChanSpy to a specific channel by typing in a ...# key sequence during a spy session? Regards, -- Raj -- Raj Mathurr...@kandalaya.org http://kandalaya.org/ GPG: 78D4 FC67 367F 40E2 0DD5 0FEF C968 D0EF CC68 D17F PsyTrance Chill: http://schizoid.in/ || It is the mind that moves -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ChanSpy with alphanumeric SIP channels [1.6.2]
I think in the chanspy application you can give it a template to prepend to
what is entered. If you do chanspy(ab_) you might be able to enter the
remaining digits.
Short of that you can set up a loop that reads the digits, calls
chanspy(ab_${digits}), if the version you are using has my S option then * will
exit the chanspy app and you can loop back to the top.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Mar 9, 2011, at 6:28 PM, Raj Mathur (राज माथुर) wrote:
Hi,
I'm using SIP users of the form 'ab_12345' (two letters, underscore, 5
digits). ChanSpy is working fine for listening in to conversations
initiated by these channels, and I can use '*' to randomly switch
channels. However, is there any way in this scenario to be able to
switch ChanSpy to a specific channel by typing in a ...# key sequence
during a spy session?
Regards,
-- Raj
--
Raj Mathurr...@kandalaya.org http://kandalaya.org/
GPG: 78D4 FC67 367F 40E2 0DD5 0FEF C968 D0EF CC68 D17F
PsyTrance Chill: http://schizoid.in/ || It is the mind that moves
--
_
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Re: [asterisk-users] [1.8] Unable to Register: Registration denied because of contact ACL
It just have ACL concept. You can add permitted IPs List to any peer then only from that IPs user can register. If you want to permit all you can add 0.0.0.0 to ACL From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of RR Sent: Thursday, March 10, 2011 7:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] [1.8] Unable to Register: Registration denied because of contact ACL Hello All, Some new security stuff is going on I suppose in 1.8 that I am not familiar with and would appreciate your help In a scenario such as the following: Internet -- SBC -- Asterisk upon trying to register an endpoint, the following is being observed on the Asterisk Console. Have Googled this but haven't come up with anything that helped much. [Mar 10 11:53:59] ERROR[21272]: netsock2.c:94 ast_sockaddr_stringify_fmt: getnameinfo(): ai_family not supported [Mar 10 11:53:59] WARNING[21272]: chan_sip.c:13120 parse_register_contact: Domain '172.16.16.6:5060' disallowed by contact ACL (violating IP ) [Mar 10 11:53:59] WARNING[21272]: chan_sip.c:13837 register_verify: Registration denied because of contact ACL Note, that the server IP is 172.16.16.11 and the SBC internal Interface IP is 172.16.16.6 the following lines have been added to sip.conf dynamic_exclude_static = yes autodomain=yes domain=172.16.16.6 allowexternaldomains=no In addition, in the general endpoint template in sip.conf, I have the lines contactdeny=0.0.0.0/0.0.0.0 contactpermit=172.16.16.0/255.255.255.0 host=dynamic What else am I missing? Thanks \RR -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [1.8] Unable to Register: Registration denied because of contact ACL
On Thu, Mar 10, 2011 at 12:20 AM, Faisal Hanif fai...@vopium.com wrote: It just have ACL concept. You can add permitted IPs List to any peer then only from that IPs user can register. If you want to permit all you can add 0.0.0.0 to ACL Thanks. but could you be a little more specific? I have added the local net 172.16.16.0/24 almost everywhere I can think of, but it keeps giving that error. Even in sip.conf in the template for company IP phones, I've added contactpermit as well as just permit=172.16.16.0/24 but it still complains about that *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *RR *Sent:* Thursday, March 10, 2011 7:04 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] [1.8] Unable to Register: Registration denied because of contact ACL Hello All, Some new security stuff is going on I suppose in 1.8 that I am not familiar with and would appreciate your help In a scenario such as the following: Internet -- SBC -- Asterisk upon trying to register an endpoint, the following is being observed on the Asterisk Console. Have Googled this but haven't come up with anything that helped much. [Mar 10 11:53:59] ERROR[21272]: netsock2.c:94 ast_sockaddr_stringify_fmt: getnameinfo(): ai_family not supported [Mar 10 11:53:59] WARNING[21272]: chan_sip.c:13120 parse_register_contact: Domain '172.16.16.6:5060' disallowed by contact ACL (violating IP ) [Mar 10 11:53:59] WARNING[21272]: chan_sip.c:13837 register_verify: Registration denied because of contact ACL Note, that the server IP is 172.16.16.11 and the SBC internal Interface IP is 172.16.16.6 the following lines have been added to sip.conf dynamic_exclude_static = yes autodomain=yes domain=172.16.16.6 allowexternaldomains=no In addition, in the general endpoint template in sip.conf, I have the lines contactdeny=0.0.0.0/0.0.0.0 contactpermit=172.16.16.0/255.255.255.0 host=dynamic What else am I missing? Thanks \RR -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [1.8] Unable to Register: Registration denied because of contact ACL
You can add following line to your peers configuration permit=0.0.0.0/0.0.0.0 It will allow to use that peer's account from any IP From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of RR Sent: Thursday, March 10, 2011 11:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] [1.8] Unable to Register: Registration denied because of contact ACL On Thu, Mar 10, 2011 at 12:20 AM, Faisal Hanif fai...@vopium.com wrote: It just have ACL concept. You can add permitted IPs List to any peer then only from that IPs user can register. If you want to permit all you can add 0.0.0.0 to ACL Thanks. but could you be a little more specific? I have added the local net 172.16.16.0/24 almost everywhere I can think of, but it keeps giving that error. Even in sip.conf in the template for company IP phones, I've added contactpermit as well as just permit=172.16.16.0/24 but it still complains about that From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of RR Sent: Thursday, March 10, 2011 7:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] [1.8] Unable to Register: Registration denied because of contact ACL Hello All, Some new security stuff is going on I suppose in 1.8 that I am not familiar with and would appreciate your help In a scenario such as the following: Internet -- SBC -- Asterisk upon trying to register an endpoint, the following is being observed on the Asterisk Console. Have Googled this but haven't come up with anything that helped much. [Mar 10 11:53:59] ERROR[21272]: netsock2.c:94 ast_sockaddr_stringify_fmt: getnameinfo(): ai_family not supported [Mar 10 11:53:59] WARNING[21272]: chan_sip.c:13120 parse_register_contact: Domain '172.16.16.6:5060' disallowed by contact ACL (violating IP ) [Mar 10 11:53:59] WARNING[21272]: chan_sip.c:13837 register_verify: Registration denied because of contact ACL Note, that the server IP is 172.16.16.11 and the SBC internal Interface IP is 172.16.16.6 the following lines have been added to sip.conf dynamic_exclude_static = yes autodomain=yes domain=172.16.16.6 allowexternaldomains=no In addition, in the general endpoint template in sip.conf, I have the lines contactdeny=0.0.0.0/0.0.0.0 contactpermit=172.16.16.0/255.255.255.0 host=dynamic What else am I missing? Thanks \RR -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [1.8] Unable to Register: Registration denied because of contact ACL
On Thu, Mar 10, 2011 at 2:12 AM, Faisal Hanif fai...@vopium.com wrote: You can add following line to your peers configuration permit=0.0.0.0/0.0.0.0 It will allow to use that peer’s account from any IP Thanks. But Like I said, that's all done. Here's the Endpoint config: [authentication] [basic-options](!); a template dtmfmode=rfc2833 context=Phones type=friend contactdeny=0.0.0.0/0.0.0.0 contactpermit=172.16.16.0/255.255.255.0 deny=0.0.0.0/0.0.0.0 permit=172.16.16.0/24 host=dynamic qualify=no insecure=port,invite [natted-phone](!,basic-options) ; another template inheriting basic-options nat=yes directmedia=no [555](natted-phone) secret=$$ecret$$ disallow=all allow=ulaw allow=gsm no deal! The irony is that we have a similar configuration at another place, but we didn't need to put anything there and the phones register regardless! Is this broken -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Display something on the top line of Polycom SPIP 3.1 screen
Hi, I'm discovering Polycom phones and specifically SPIP 3.1.6 powered ones. Default display is showing : - a blank line at the top of the screen - then the date (2nd line) - then the time (3rd line) Is there a way to display something on the first line (the one above the date line) (? I saw this line used in MGCP-powered phones. Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
