[asterisk-users] H323 no audio
Hi, My configuration is SipPhone-asterisk1 -asterisk2. My asterisk version is 1.2.10. I installed chan_h323 according to 'http://astrecipes.net/?n=102'. When i call from asterisk1 to asterisk2, there is no audio. Using 'rtp debug', I can see that rtp packets are being received. Regards, Jason. #--h323.conf for both [general] port = 1720 bindaddr = 0.0.0.0 disallow=all allow=ulaw context=default #--dial plan of asterisk1 exten = *59,1,Wait(1) exten = *59,2,Dial(H323/[EMAIL PROTECTED]) #--dial plan of asterisk2 exten = 3500,1,Playback(hello) exten = 3500,2,Hangup() #--'rtp debug' messages-- Raw PDU: 08 02 55 13 62 1c 00 7e 00 0f 05 28 10 01 00 04 ..U.b..~...( c0 01 80 05 01 03 28 00 01 ..(.. 2:15:36.845 H225 Caller:89bf340 h323.cxx(4301) H323 InternalEstablishedConnectionCheck: connectionState=EstablishedConnection fastStartState=FastStartAcknowledged Got RTP packet from 192.168.1.232:16426 (type 0, seq 1540, ts 161645797, len 240) Got RTP packet from 192.168.1.232:16426 (type 0, seq 1541, ts 161646037, len 240) Got RTP packet from 192.168.1.232:16426 (type 0, seq 1542, ts 161646277, len 240) Got RTP packet from 192.168.1.232:16426 (type 0, seq 1543, ts 161646517, len 240) Got RTP packet from 192.168.1.232:16426 (type 0, seq 1544, ts 161646757, len 240) Got RTP packet from 192.168.1.232:16426 (type 0, seq 1545, ts 161646997, len 240) Got RTP packet from 192.168.1.232:16426 (type 0, seq 1546, ts 161647237, len 240) Got RTP packet from 192.168.1.232:16426 (type 0, seq 1547, ts 161647477, len 240) Got RTP packet from 192.168.1.232:16426 (type 0, seq 1548, ts 161647717, len 240) Sponsored Link Don't quit your job - take classes online www.Classesusa.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: strip + sign from incoming ${EXTEN} var?
EW == Eric \ManxPower\ Wieling Eric writes:
EW exten = _+NXXNXX,1,Goto(${EXTEN:1},1)
You are assuming a fixed-length number here...
The basic problem is to ensure that the extension gets run first,
since asterisk has its very own ideas about what goes first. So put
the incoming calls in context removeplus instead of incoming, and do:
[removeplus]
_+!,1,Goto(incoming,${EXTEN:1},1)
_X!,1,Goto(incoming,${EXTEN},1)
This assumes that ${EXTEN} starts with either a number or a plus. If
it can start with other things, you can do
_!,1,Goto(incoming,${EXTEN},1)
for the second line, but then you are relying on asterisks extension
ordering again.
/Benny
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RE: [asterisk-users] Ringing a group of phones but not if they arebusy
I need to ring a group of 8 phones, but not if they are already on another call. How can I determine which of those 8 phones are busy so I only ring the others? I've done this in the past by disabling call waiting on the phones and put all 8 phones into a ringall queue. Then, when you call that queue, the phones already on calls return SIP BUSY,whilst the others ring as normal. It's not perfect, but for most of our users the call waiting noise in the earpiece is an annoyance anyway. Hope that helps. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited This email is made from 100% recycled electrons ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ringing a group of phones but not if they are busy
Chris Bagnall ha scritto: I need to ring a group of 8 phones, but not if they are already on another call. How can I determine which of those 8 phones are busy so I only ring the others? I've done this in the past by disabling call waiting on the phones and put all 8 phones into a ringall queue. Then, when you call that queue, the phones already on calls return SIP BUSY,whilst the others ring as normal. It's not perfect, but for most of our users the call waiting noise in the earpiece is an annoyance anyway. Hope that helps. Regards, Chris If you disable call waiting, then you don't need a queue. With grandstream gxp-2000 phones, calling Dial(SIP/phone1SIP/phone2SIP/phone3) rings only off-hook phones. However, I have also SPA-941 phones. Is it possible to disable the call waiting feature on Linksys SPA-941? I haven't succeeded so far... and the multiple Dial() method or the Queue are not working either. I had to change my extensions.conf macro to do ChanIsAvail sequentially, that is, for each phone I call ChanIsAvail and then check the results to see if the phone is busy. If not, I add it to the dialstring to pass to Dial() eventually. Since there are 10 phones to check, and the process is not atomic, it can (very rarely) occur that a phone is included in the dialstring but has just become busy, and the user gets the annoying call waiting tone. Any clue? -- -- Alberto Pastore B-Press Srl - Gruppo MSoft P.IVA 01697420030 P.le Lombardia, 4 - 28100 Novara - Italy Tel. 0321-499508 Fax 0321-492974 http://www.msoft.it ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] spc.exe
Sipura Profile Compiler is only for ITSPs and agreements does not permit that Regards Andrew Joakimsen escribió: Does anyone have a copy of spc.exe they could send me? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] PortSip and Astericks new install
Thanks for you reply, it does not output anything on the console when i make the call. However i have turned on SIP Debug and get the following: -- SIP read from 192.168.2.3:8099: REGISTER sip:192.168.1.1:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.3:8099;rport;branch=z9hG4bK32333 From: sip:[EMAIL PROTECTED]:5060;tag=9948 To: sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER Contact: sip:[EMAIL PROTECTED]:8099 Max-Forwards: 70 User-Agent: PortSIP softphone 2.0 Expires: 150 Content-Length: 0 --- (11 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.2.3 : 8099 (NAT) Transmitting (NAT) to 192.168.2.3:8099: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.2.3:8099;branch=z9hG4bK32333;received=192.168.2.3;rport=8099 From: sip:[EMAIL PROTECTED]:5060;tag=9948 To: sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- Transmitting (NAT) to 192.168.2.3:8099: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.2.3:8099;branch=z9hG4bK32333;received=192.168.2.3;rport=8099 From: sip:[EMAIL PROTECTED]:5060;tag=9948 To: sip:[EMAIL PROTECTED]:5060;tag=as3d203e7d Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=5d41ffa1 Content-Length: 0 --- Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms [Kserver1*CLI -- SIP read from 192.168.2.3:8099: REGISTER sip:192.168.1.1:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.3:8099;rport;branch=z9hG4bK41 From: sip:[EMAIL PROTECTED]:5060;tag=9948 To: sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] CSeq: 2 REGISTER Contact: sip:[EMAIL PROTECTED]:8099 Authorization: Digest username=4289, realm=asterisk, nonce=5d41ffa1, uri=sip:192.168.1.1:5060, response=c3f43fd747f5ef51168bbfa2401b680b, algorithm=MD5 Max-Forwards: 70 User-Agent: PortSIP softphone 2.0 [Kserver1*CLI Expires: 150 Content-Length: 0 --- (12 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.2.3 : 8099 (NAT) Transmitting (NAT) to 192.168.2.3:8099: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.2.3:8099;branch=z9hG4bK41;received=192.168.2.3;rport=8099 From: sip:[EMAIL PROTECTED]:5060;tag=9948 To: sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] CSeq: 2 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- [Kserver1*CLI 12 headers, 0 lines Reliably Transmitting (NAT) to 192.168.2.3:8099: OPTIONS sip:[EMAIL PROTECTED]:8099 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.1:5060;branch=z9hG4bK60c2e2eb;rport From: asterisk sip:[EMAIL PROTECTED];tag=as6aa7255c To: sip:[EMAIL PROTECTED]:8099 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 17 Nov 2006 20:47:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- Transmitting (NAT) to 192.168.2.3:8099: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.3:8099;branch=z9hG4bK41;received=192.168.2.3;rport=8099 From: sip:[EMAIL PROTECTED]:5060;tag=9948 To: sip:[EMAIL PROTECTED]:5060;tag=as3d203e7d Call-ID: [EMAIL PROTECTED] CSeq: 2 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Expires: 150 Contact: sip:[EMAIL PROTECTED]:8099;expires=150 Date: Fri, 17 Nov 2006 20:47:18 GMT Content-Length: 0 --- Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms [Kserver1*CLI -- SIP read from 192.168.2.3:8099: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.1:5060;branch=z9hG4bK60c2e2eb;rport=5060 From: asterisk sip:[EMAIL PROTECTED];tag=as6aa7255c To: sip:[EMAIL PROTECTED]:8099;tag=18467 Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: PortSIP softphone 2.0 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, MESSAGE, INFO, REFER, UPDATE Content-Length: 0 --- (9 headers 0 lines) --- Destroying call '[EMAIL PROTECTED]' [Kserver1*CLI -- SIP read from 192.168.2.3:8099: INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.3:8099;rport;branch=z9hG4bK31747 From: sip:[EMAIL PROTECTED]:5060;tag=30024 To: sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] CSeq: 20 INVITE Contact: sip:[EMAIL PROTECTED]:8099 Max-Forwards: 70 User-Agent: PortSIP softphone 2.0 Subject: call Expires: 120 Allow: INVITE, ACK, UPDATE, INFO, CANCEL, BYE, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE Content-Type: application/sdp Content-Length: 325 v=0 o=- 2177823 2177823 IN IP4 192.168.2.3 s=PortSIP VOIP SDK 2.0 c=IN IP4 192.168.2.3 t=0 0 m=audio 51636 RTP/AVP 0 3 8 97 4 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:4 G723/8000
Re: [asterisk-users] Re: zaptel, bristuff zaphfc, and florz question
On Thursday 16 November 2006 18:40, Steve Davies wrote: Replying to myself with one further piece of information - In the most recent log, the otherwise stable florz version of the driver seems to have died with this message: Nov 16 15:48:34 pabx kernel: zaphfc[0]: empty HDLC frame received. Might this mean something to someone? Thanks again, Steve Steve I assume that you have disabled all unnecessary devices, unloaded unnecessary modules,moved cards so as to not share interupts, enabled apic and tweaked pci latency using 'setpci' ? when things are this bad with HFC cards, I have always ended up changing the motherboard... Paul ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] If of external small box supply fxs Isdn and E1 ?
Hi anyone know a list of external hardware supported by asterisk for connect old Pbx to VoIP line ? For supply Isdn BRI and PRI to my clients thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AdvancedVoIP Billing ?
Hi after 2 mounth of search, i don't have see a billing solution for my small business.. i see only AdvancedVoIPBilling but i don't know if he can work's with Asterisk. I am search a billing software for the invoice of my custumers, no Calling Card. but i don't see a small and simple product for this. thanks bye ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Freepbx changes dont reflect in asterisk
Alex was right. The problem is that when i make changes in freepbx, those changes are not written in the config files. I only made modifications in files_custom.conf. The version of freePbx that i use is 2.1.1 (not beta) and Asterisk 1.2.12.1. Thanks by your help, Ps. 2006/11/18, Alex Robar [EMAIL PROTECTED]: I think you guys are all misunderstanding the problem here. Unless I'm misunderstanding, Pedro's issue is that when he makes changes in FreePBX, those changes are not written out to the config files. Alex On 11/17/06, Zeeshan Zakaria [EMAIL PROTECTED] wrote: You can't do any modifications in extensions_additional.conf and sip_additional.conf. Same is true for extensions.conf and sip.conf, and other original trixbox files. As soon as you press the red bar, they are returned to their original state. For modifications, you create your own files or use sip_customs.conf and extensions_custom.conf. Please don't mix trixbox with asterisk just because its based on asterisk. Its a completely customized solution of various software packages configured to make it work according to its own requirements. For help, post on trixbox.org forums. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AdvancedVoIP Billing ?
Did you look at a2billing? Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Are you outside of the US? Do you need to call US Toll Free Numbers? We can help you save money on calling US toll free numbers. Email for information: [EMAIL PROTECTED] (VOIP PBX) 1-866-638-1254 (Cellular) 1-712-432-5401 (Voip PBX) Free World DialUp: 780-217 EXT: 250 WebSite: http://www.freeworlddialup.com/ BUY and sell Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security GOLD PLATING SERVICES http://www.bochterservices.com/?j=platingt=email Noc Phibee wrote: Hi after 2 mounth of search, i don't have see a billing solution for my small business.. i see only AdvancedVoIPBilling but i don't know if he can work's with Asterisk. I am search a billing software for the invoice of my custumers, no Calling Card. but i don't see a small and simple product for this. thanks bye ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Inbound (clean). Database: 0649-0, 11/15/2006 - 11/18/2006 7:13:37 AM ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: zaptel, bristuff zaphfc, and florz question
On 11/18/06, Paul Hewlett [EMAIL PROTECTED] wrote: On Thursday 16 November 2006 18:40, Steve Davies wrote: Replying to myself with one further piece of information - In the most recent log, the otherwise stable florz version of the driver seems to have died with this message: Nov 16 15:48:34 pabx kernel: zaphfc[0]: empty HDLC frame received. Might this mean something to someone? Thanks again, Steve Steve I assume that you have disabled all unnecessary devices, unloaded unnecessary modules,moved cards so as to not share interupts, enabled apic and tweaked pci latency using 'setpci' ? ... setpci... I knew there was something I had missed. Thanks! Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AdvancedVoIP Billing ?
Yes ;=) but a2billing it's for calling card ;) Al Bochter a écrit : Did you look at a2billing? Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Are you outside of the US? Do you need to call US Toll Free Numbers? We can help you save money on calling US toll free numbers. Email for information: [EMAIL PROTECTED] (VOIP PBX) 1-866-638-1254 (Cellular) 1-712-432-5401 (Voip PBX) Free World DialUp: 780-217 EXT: 250 WebSite: http://www.freeworlddialup.com/ BUY and sell Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security GOLD PLATING SERVICES http://www.bochterservices.com/?j=platingt=email Noc Phibee wrote: Hi after 2 mounth of search, i don't have see a billing solution for my small business.. i see only AdvancedVoIPBilling but i don't know if he can work's with Asterisk. I am search a billing software for the invoice of my custumers, no Calling Card. but i don't see a small and simple product for this. thanks bye ___ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AdvancedVoIP Billing ?
Noc Phibee wrote: after 2 mounth of search, i don't have see a billing solution for my small business.. Not quite sure as I didn't research very much their product, but did you check Aradial? http://www.aradial.com/voip-billing-radius.html ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] wget from within asterisk?
I believe asterisk requires the source/devel package. I'm afraid I'm not going to be much help on that. We use Gentoo and it was a simple 'emerge curl' and I was done. -Original Message- From: Damon Estep [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Fri, 17 Nov 2006 16:18:07 -0700 Subject: RE: [asterisk-users] wget from within asterisk? On version 1.2.12.1 running on FC4 with curl.i386 installed the asterisk CURL function is not registered, perhaps in need something else (curl-devel.i386 ?) From:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf OfRuss Beaupre Sent: Friday, November 17, 2006 1:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] wget from within asterisk? Make sure the curl library/package is installed, then re-compile asterisk. We're using it on 1.2. -Original Message- From: Damon Estep [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Fri, 17 Nov 2006 13:40:40 -0700 Subject: RE: [asterisk-users] wget from within asterisk? I saw CURL, but it does not register appear in show functions or show applications, deprecated or add-on? From:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf OfRuss Beaupre Sent: Friday, November 17, 2006 1:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] wget from within asterisk? -Original Message- From: Damon Estep [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Date: Fri, 17 Nov 2006 13:32:53 -0700 Subject: [asterisk-users] wget from within asterisk? What would be the simplest way to retrieve information form a CNAM database that provides http based query responses? Does an application or script already exist that does this? Basically, I want to do a wget of a URL that contains the callerID number as a variable, and assign the returned text to another variable which can be used to set the caller ID name. Any suggestions? Look at the CURL function. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Using ChanSpy for spying voicemail
Hello list,
I currently try to get ChanSpy working to listen in to what people leave
on my voicemail. The problem seems to be though that ChanSpy only sends
the voicemail part of the conversation, namely the announcements,
which is not really helpful.
Is this a feature, or is spying into voicemail just not possible?
How do I do it correcty?
For those interested, my current setup looks like (removing anything not
having to do with this)
...
exten = 301,1,Set(DB(spy/301)=${CHANNEL})
exten = 301,2,AGI(voicemailspy)
exten = 301,3,VoiceMail(u01)
...
[voicespy]
exten = 301,1,NoOp(Spying for ${EXTEN} on Channel ${DB(spy/301)})
exten = 301,2,ChanSpy(${DB(spy/301)})
exten = 301,3,Hangup
...
There is an AGI called voicemailspy which basically generates call
files appropriately. This definitively works - else I would not get the
call and listen in to the VoiceMail announcement.
Hints, anyone?
Thanks
Anselm
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[asterisk-users] Hardware Echo cancelation
Hi ! I have a Digium TE411P with echo cancelation board. I'm not sure if i need to enable the echocancelation=yes in the zapata.conf file. Or since the card hast the echo cancel board, asterisk does not need that option. Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] spc.exe
I didn't ask what SPC.exe was, I asked kindly if someone who had it could e-mail it to me. On 11/18/06, Alberto Sagredo [EMAIL PROTECTED] wrote: Sipura Profile Compiler is only for ITSPs and agreements does not permit that Regards Andrew Joakimsen escribió: Does anyone have a copy of spc.exe they could send me? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AdvancedVoIP Billing ?
Hi thanks for your answer, no i don't have see this software because i don't see screenshot or demo ;) Hermann Wecke a écrit : Noc Phibee wrote: after 2 mounth of search, i don't have see a billing solution for my small business.. Not quite sure as I didn't research very much their product, but did you check Aradial? http://www.aradial.com/voip-billing-radius.html ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] spc.exe
I have (2) of the ST2030 and (2) of the sidecar modules in stock. $125/ea with sidecar. Cory Andrews Executive Vice President ++ VoIPSupply.com PBXSelect.com ++ 454 Sonwil Drive Buffalo, NY 14225 voice direct - 716.250.3402 fax - 716.630.1548 e - [EMAIL PROTECTED] m - 716.907.4059 aim - B2Cory From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Joakimsen Sent: Saturday, November 18, 2006 11:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] spc.exe I didn't ask what SPC.exe was, I asked kindly if someone who had it could e-mail it to me. On 11/18/06, Alberto Sagredo [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Sipura Profile Compiler is only for ITSPs and agreements does not permit that Regards Andrew Joakimsen escribió: Does anyone have a copy of spc.exe they could send me? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware Echo cancelation
Carlos Prieto wrote: Hi ! I have a Digium TE411P with echo cancelation board. I'm not sure if i need to enable the echocancelation=yes in the zapata.conf file. Or since the card hast the echo cancel board, asterisk does not need that option. Yes, this needs to be enabled. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Manager: equivalent of 'show channels'?
I'm interested in knowing if anyone else has worked around this issue: I have an application that needs to check the status of the calls going through Asterisk about every 5 seconds or so. I don't want to do asterisk -rx 'show channels verbose' at the Linux command line 12 times per minute so I am looking at the AMI. I see that there isn't a manager command for 'show channels.' Has anyone come up with an equivalent of 'show channels' using the extant manager commands? If so, could you post how you did it? Thanks! -MC ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Manager: equivalent of 'show channels'?
take a look at Flash Operator Panel, as far as i know they use AMI , and they also provide real time channel status. On 11/18/06, Michael Collins [EMAIL PROTECTED] wrote: I'm interested in knowing if anyone else has worked around this issue: I have an application that needs to check the status of the calls going through Asterisk about every 5 seconds or so. I don't want to do asterisk –rx 'show channels verbose' at the Linux command line 12 times per minute so I am looking at the AMI. I see that there isn't a manager command for 'show channels.' Has anyone come up with an equivalent of 'show channels' using the extant manager commands? If so, could you post how you did it? Thanks! -MC ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Com os melhores cumprimentos, Marco Mouta ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Manager: equivalent of 'show channels'?
As far as I know, you can do something like this, i.e. in PHP: fputs($socket, Action: Command\r\n); fputs($socket, Command: show channels concise\r\n); I hope that helps, but perhaps there is a better possibility. NB Michael Collins schrieb: I’m interested in knowing if anyone else has worked around this issue: I have an application that needs to check the status of the calls going through Asterisk about every 5 seconds or so. I don’t want to do “asterisk –rx ‘show channels verbose’” at the Linux command line 12 times per minute so I am looking at the AMI. I see that there isn’t a manager command for ‘show channels.’ Has anyone come up with an equivalent of ‘show channels’ using the extant manager commands? If so, could you post how you did it? Thanks! -MC ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TDD/TTY device for the deaf
I have a client who would like to install a TTY device in their office. They have a Asterisk PBX with a CBeyond T1 to the PSTN, and the system uses the CBeyond SIPConnect facility for PSTN voice calls. The TTY device would connect with a TDM10B (FXS) card and have its own DID number. I believe the protocol between the calling TTY and the local TTY is FSK. In any case, would two TTY devices be able to communicate over an RTP stream using g711? Or should the TTY device be attached to a POTS circuit away from the PBX? Thanks, -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AdvancedVoIP Billing ?
I am also searching one for post-paid billing .. but most like astpp wants to eat whole system themselves managing extensions and all . I need a type of solution that can just bill people based on mysql cdr using accountcode and amagflags .. I am thinking to make some myself now but it will take me time to learn php so i am still searching :( On 18/11/06, Noc Phibee [EMAIL PROTECTED] wrote: Hi thanks for your answer, no i don't have see this software because i don't see screenshot or demo ;) Hermann Wecke a écrit : Noc Phibee wrote: after 2 mounth of search, i don't have see a billing solution for my small business.. Not quite sure as I didn't research very much their product, but did you check Aradial? http://www.aradial.com/voip-billing-radius.html ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AdvancedVoIP Billing ?
I'll jump in here. As the author of ASTPP, I have gone to considerable effort to make it so that ASTPP does NOT need to eat a whole system. All that you really need on the asterisk box to get ASTPP working in terms of asterisk requirements, is to make sure that the cdrs in the database have an accountcode set. You do not need to use it to manage your dids and extensions, etc. Darren Wiebe [EMAIL PROTECTED] Vicky wrote: I am also searching one for post-paid billing .. but most like astpp wants to eat whole system themselves managing extensions and all . I need a type of solution that can just bill people based on mysql cdr using accountcode and amagflags .. I am thinking to make some myself now but it will take me time to learn php so i am still searching :( On 18/11/06, *Noc Phibee* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi thanks for your answer, no i don't have see this software because i don't see screenshot or demo ;) Hermann Wecke a écrit : Noc Phibee wrote: after 2 mounth of search, i don't have see a billing solution for my small business.. Not quite sure as I didn't research very much their product, but did you check Aradial? http://www.aradial.com/voip-billing-radius.html http://www.aradial.com/voip-billing-radius.html ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Manager: equivalent of 'show channels'?
On Saturday 18 November 2006 13:28, Michael Collins wrote: I'm interested in knowing if anyone else has worked around this issue: I have an application that needs to check the status of the calls going through Asterisk about every 5 seconds or so. I don't want to do asterisk -rx 'show channels verbose' at the Linux command line 12 times per minute so I am looking at the AMI. I see that there isn't a manager command for 'show channels.' Has anyone come up with an equivalent of 'show channels' using the extant manager commands? If so, could you post how you did it? Thanks! -MC An approach i take with my event monitor is to have a manager application register for all events, then when something in asterisk happens you get an event record. No need to 'poll' to check. Then you can easily determine what is going on. http://micpc.com/eventmonitor uses the event records to display the state of phones, channels, etc. 'realtime' with AJAX without polling. earl ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Manager: equivalent of 'show channels'?
Michael Collins wrote: I’m interested in knowing if anyone else has worked around this issue: I have an application that needs to check the status of the calls going through Asterisk about every 5 seconds or so. I don’t want to do “asterisk –rx ‘show channels verbose’” at the Linux command line 12 times per minute so I am looking at the AMI. I see that there isn’t a manager command for ‘show channels.’ Has anyone come up with an equivalent of ‘show channels’ using the extant manager commands? If so, could you post how you did it? Thanks! -MC you need to look again maybe read, http://www.dynx.net/ASTERISK/DOCS/RTF/MANAGER.RTF action: command command: show channels Response: Follows Privilege: Command Channel Location State Application(Data) 0 active channels 0 active calls --END COMMAND-- goodluck ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AdvancedVoIP Billing ?
I will definitely give it a try again to astpp then . I actually saw its online demo and was bit confused . I thought its managing extensions and all and i will have to start from scratch so i didnt gave it a try .It is a great software but only thing holding me back was thought that i will have to start from scratch :P . On 18/11/06, Darren Wiebe [EMAIL PROTECTED] wrote: I'll jump in here. As the author of ASTPP, I have gone to considerable effort to make it so that ASTPP does NOT need to eat a whole system. All that you really need on the asterisk box to get ASTPP working in terms of asterisk requirements, is to make sure that the cdrs in the database have an accountcode set. You do not need to use it to manage your dids and extensions, etc. Darren Wiebe [EMAIL PROTECTED] Vicky wrote: I am also searching one for post-paid billing .. but most like astpp wants to eat whole system themselves managing extensions and all . I need a type of solution that can just bill people based on mysql cdr using accountcode and amagflags .. I am thinking to make some myself now but it will take me time to learn php so i am still searching :( On 18/11/06, *Noc Phibee* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi thanks for your answer, no i don't have see this software because i don't see screenshot or demo ;) Hermann Wecke a écrit : Noc Phibee wrote: after 2 mounth of search, i don't have see a billing solution for my small business.. Not quite sure as I didn't research very much their product, but did you check Aradial? http://www.aradial.com/voip-billing-radius.html http://www.aradial.com/voip-billing-radius.html ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco 2801 and asterisk
Hello, Is anybody using a Cisco 2801 to connect to the PSTN? I am having some issues regarding SIP responses from the router to asterisk (too general e.g. SIP 404 errors when they should be more specific). Regards, Bas pgpj8PZe7U89B.pgp Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AdvancedVoIP Billing ?
Ok, cool. If you run into problems, please post at forums.astpp.org on the the astpp mailling list. Good luck, Darren Wiebe [EMAIL PROTECTED] Vicky wrote: I will definitely give it a try again to astpp then . I actually saw its online demo and was bit confused . I thought its managing extensions and all and i will have to start from scratch so i didnt gave it a try .It is a great software but only thing holding me back was thought that i will have to start from scratch :P . On 18/11/06, *Darren Wiebe* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I'll jump in here. As the author of ASTPP, I have gone to considerable effort to make it so that ASTPP does NOT need to eat a whole system. All that you really need on the asterisk box to get ASTPP working in terms of asterisk requirements, is to make sure that the cdrs in the database have an accountcode set. You do not need to use it to manage your dids and extensions, etc. Darren Wiebe [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Vicky wrote: I am also searching one for post-paid billing .. but most like astpp wants to eat whole system themselves managing extensions and all . I need a type of solution that can just bill people based on mysql cdr using accountcode and amagflags .. I am thinking to make some myself now but it will take me time to learn php so i am still searching :( On 18/11/06, *Noc Phibee* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi thanks for your answer, no i don't have see this software because i don't see screenshot or demo ;) Hermann Wecke a écrit : Noc Phibee wrote: after 2 mounth of search, i don't have see a billing solution for my small business.. Not quite sure as I didn't research very much their product, but did you check Aradial? http://www.aradial.com/voip-billing-radius.html http://www.aradial.com/voip-billing-radius.html ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AdvancedVoIP Billing ?
Really? I am using a2billing to bill customers for Per min DID inbound to there IVR's, Voice Mail Box tracking, Billing users for outbound from Softphones And Calling Cards :() there is alot more but I don't want type the much... My god it's asterisk think outside of the box Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VOIP PBX) 1-866-638-1254 (Voip PBX) Free World DialUp: 780-217 WebSite: http://www.freeworlddialup.com/ BUY Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security Noc Phibee wrote: Yes ;=) but a2billing it's for calling card ;) Al Bochter a écrit : Did you look at a2billing? Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Are you outside of the US? Do you need to call US Toll Free Numbers? We can help you save money on calling US toll free numbers. Email for information: [EMAIL PROTECTED] (VOIP PBX) 1-866-638-1254 (Cellular) 1-712-432-5401 (Voip PBX) Free World DialUp: 780-217 EXT: 250 WebSite: http://www.freeworlddialup.com/ BUY and sell Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security GOLD PLATING SERVICES http://www.bochterservices.com/?j=platingt=email Noc Phibee wrote: Hi after 2 mounth of search, i don't have see a billing solution for my small business.. i see only AdvancedVoIPBilling but i don't know if he can work's with Asterisk. I am search a billing software for the invoice of my custumers, no Calling Card. but i don't see a small and simple product for this. thanks bye ___ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Inbound (clean). Database: 0649-0, 11/15/2006 - 11/18/2006 1:04:03 PM ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dialout Conferences?
How do I set up an existing call to dial out to a new terminal which is included in a conference with the two existing legs of the call? When the dialplan executes the Dial(terminal) command, control does not return to the dialplan until the terminal disconnects, after which it's obviously too late to conference it. Is there a conference command or option that lets the dialplan dial out to a terminal to a new conference member? Is there a dial command or option that returns control to the dialplan for then calling a conference command on the dialed terminal, or executes another command inside the process cycle of the dial command that adds the dialed terminal to a conference, after the terminal answers but before it hangs up? -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Need help with a function
Hi all, I want to do the following: 1. A person dials an extension where he/she can record a message. 2. After recording it, he/she can enter the extensions the message should be plaied for. 3. Hang up and Asterisk will dial each extension to play the message. Many thanks for all your help, Christian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] spc.exe
I want to buy one of those ST2030 then, do you think you can get me the spc.exe also? Its for a phone I bought from you guys. Also if you could get my the Polycom 2.x firmware. On 11/18/06, Cory Andrews [EMAIL PROTECTED] wrote: I have (2) of the ST2030 and (2) of the sidecar modules in stock. $125/ea with sidecar. Cory Andrews Executive Vice President ++ VoIPSupply.com PBXSelect.com ++ 454 Sonwil Drive Buffalo, NY 14225 voice direct - 716.250.3402 fax - 716.630.1548 e - [EMAIL PROTECTED] m - 716.907.4059 aim - B2Cory -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Andrew Joakimsen *Sent:* Saturday, November 18, 2006 11:09 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] spc.exe I didn't ask what SPC.exe was, I asked kindly if someone who had it could e-mail it to me. On 11/18/06, Alberto Sagredo [EMAIL PROTECTED] wrote: Sipura Profile Compiler is only for ITSPs and agreements does not permit that Regards Andrew Joakimsen escribió: Does anyone have a copy of spc.exe they could send me? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to use Sipura SPA3k POTS line to dial Asterisk SIP phones?
Doug Crompton wrote:
Doug, please forgive me but I'm still having trouble understanding two
points from your last response.
Can you please post your extension 405 (analog extension on spa3k) in
sip.conf
and your [sipurafxs1] ?
I finally understand that INRINGSDEV is meant to specify which analog
and SIP phones to ring at extension INRINGSEXT = 405 and would like to
see just how you do it.
Larry
On Wed, 15 Nov 2006, Larry Alkoff wrote:
Thank you very much Doug for your detailed response to my question.
I'm working on a new sip.conf and extensions.conf using your code as a
guide.
Questions:
In INRINGSDEV what does sipurafxs1 and grandstream406 refer to?
The comment says ring analog phones on spa3k fxs but grandstream406
seems to refer a Grandstream sip phone, not an analog one.
Does INRINGSDEV mean ring a specific sip phone and the analog ones?
INRINGSDEV is a list of the devices you want to ring when you use this
variable in the dial statement. sipurafxs1 is the fxs side of the spa3k
and I have one grandstream 200, at extension 406, named grandstream406.
The analog extension, fxs on the spa3k, is 405.
How would I ring all the _sip_ phones when a pstn call comes in?
My macro 'ring-all' ?
You just add them all together in the ring statement with the as in my
INRINGSDEV variable. Actually the use of the variable was taken from
sample code given to me when I started out. It is probably a good idea
though. you could just put them all in the dial statement but if you use
it in more than one place it is handy to just change it in one place and
use the variable.
SIP/sipurafxs1SIP/grandstream406thirdfourth.
Notes:
Your sipurafxo1 is my spa3k-pstn-in defined in both Sipura and sip.conf.
My extension to ring incoming calls is 120 vs your 405. All ok on these
two.
I'm nearly there thanks to you.
OK glad it helped. If you have any other questions let me know. The spa3k
has a million settings.
Larry
Doug Crompton wrote:
Below is my config for spa3k fxo. I do not show the settings in the spa3k
which must reflect settings here, port, username, secret, etc. I have
DTMF set to inband here and in spa3k to fix a problem with DTMF not
working for menus from PSTN. This was discussed earlier and is a problem
in asterisk that may (or may not) be solved in 1.4. I am using earlier
version. Inband must also be specifed in spa3k pstn.
[sipurafxo1]
type=peer
username=sipurafxo1
secret=x
canreinvite=no
context=from-pstn
host=dynamic
nat=no
port=5061
disallow=all
allow=alaw
allow=ulaw
allow=gsm
allow=g723.1
dtmfmode=inband
In extensions.conf. This is a little fancy but the bottom line is that it
ends up in either a day or night mode. Only day shown. The spa3k fxo in
sip calls the from-pstn but the pstn-day-time (below) could be relabeled
from-pstn to always go to phones. The night mode basically goes to VM.
INRINGSEXT and INRINGSDEV are just variables defined to -
INRINGSDEV=SIP/sipurafxs1SIP/grandstream406 ; ring analog phones on spa3k
fxs
INRINGSEXT=405 ; the extension to ring for incomming calls
The stdexten macro is just the standard one in sample extension file.
[from-pstn]
exten = s,1,GotoIf($[ ${day-night} = 0 ]?2:10
exten = s,2,GotoIfTime(9:30-23:59,*,*,*?pstn-day-time,s,1
exten = s,3,GotoIfTime(0:00-09:29,*,*,*?night-time,s,1
exten = s,10,GotoIf($[ ${day-night} = 1 ]?pstn-day-time,s,1
exten = s,11,GotoIf($[ ${day-night} = 2 ]?night-time,s,1
[pstn-day-time]
exten = s,1,SetGlobalVar(RingTimeout=35)
exten = s,2,NoOp(${CALLERID})
exten = s,3,Macro(stdexten,${INRINGSEXT},${INRINGSDEV},)
On Tue, 14 Nov 2006, Larry Alkoff wrote:
My SIP phones can dial out through Sipura SPA3k to POTS for local and
911 calls _but_ incoming POTS calls are being swallowup somehow.
Am I on the right track with the code snippit below?
sip.conf:
-
In sip.conf the following code is _supposed_ to ring the SIP phones when
a POTS line call comes in through Sipuara to Asterisk.
[spa3k-pstn-in] ; Pots-line-in from Sipura
; If you're using Asterisk, this goes into the Incoming settings
; For your Trunk
host=dynamic
type=friend ; should be peer if incoming only ??
context=[macro-ringall] ;ring all the sip phones
secret=x
dtmfmode=rfc2833
disallow=all
allow=ulaw
insecure=very
extensions.conf
context to ring all SIP phones when a POTS call comes into SPA3k:
[macro-ringall] ; ring all SIP phones
exten = s,1,Dial(SIP/120SIP/121SIP/122SIP/124SIP/125SIP/126SIP/127)
exten = s,2,hangup
--
Larry Alkoff N2LA - Austin TX
--
Larry Alkoff N2LA - Austin TX
Using Thunderbird on Linux
___
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[asterisk-users] Statistics on Number of Minutes
Does anyone know where to get infomation on the number of minutes used from US to another country? I tried the FCC but the infomation was not good enough. Who keeps those statitics?___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Freepbx changes dont reflect in asterisk
i also used to have this problem, for instance we use the pin code functionality of FreePBX and whenever i add or modify a pin number, it is not effected or changed in the config files. i dont know what causes this error but i have noticed that restarting FreePBX or re-installing the application stops this. Just restart the box On 11/18/06, Pedro Silva [EMAIL PROTECTED] wrote: Alex was right. The problem is that when i make changes in freepbx, those changes are not written in the config files. I only made modifications in files_custom.conf. The version of freePbx that i use is 2.1.1 (not beta) and Asterisk 1.2.12.1. Thanks by your help, Ps. 2006/11/18, Alex Robar [EMAIL PROTECTED]: I think you guys are all misunderstanding the problem here. Unless I'm misunderstanding, Pedro's issue is that when he makes changes in FreePBX, those changes are not written out to the config files. Alex On 11/17/06, Zeeshan Zakaria [EMAIL PROTECTED] wrote: You can't do any modifications in extensions_additional.conf and sip_additional.conf. Same is true for extensions.conf and sip.conf, and other original trixbox files. As soon as you press the red bar, they are returned to their original state. For modifications, you create your own files or use sip_customs.conf and extensions_custom.conf. Please don't mix trixbox with asterisk just because its based on asterisk. Its a completely customized solution of various software packages configured to make it work according to its own requirements. For help, post on trixbox.org forums. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to use Sipura SPA3k POTS line to dial Asterisk SIP phones?
This is my spa3k fxs port sip.conf params. This uses the default context
in my extensions.conf
What are you having trouble doing? Can you make calls out to PSTN? Is it
just incoming call that are not ringing?
Doug
[sipurafxs1]
type=friend
regexten=405
username=sipurafxs1
secret=
context=default
context=from-pstn
callerid=Doug Crompton 405
host=dynamic
nat=no
port=5061
canreinvite=no
disallow=all
allow=alaw
allow=ulaw^M
allow=gsm
allow=g723.1^M
[EMAIL PROTECTED]
dtmfmode=rfc2833
On Sat, 18 Nov 2006, Larry Alkoff wrote:
Doug Crompton wrote:
Doug, please forgive me but I'm still having trouble understanding two
points from your last response.
Can you please post your extension 405 (analog extension on spa3k) in
sip.conf
and your [sipurafxs1] ?
I finally understand that INRINGSDEV is meant to specify which analog
and SIP phones to ring at extension INRINGSEXT = 405 and would like to
see just how you do it.
Larry
On Wed, 15 Nov 2006, Larry Alkoff wrote:
Thank you very much Doug for your detailed response to my question.
I'm working on a new sip.conf and extensions.conf using your code as a
guide.
Questions:
In INRINGSDEV what does sipurafxs1 and grandstream406 refer to?
The comment says ring analog phones on spa3k fxs but grandstream406
seems to refer a Grandstream sip phone, not an analog one.
Does INRINGSDEV mean ring a specific sip phone and the analog ones?
INRINGSDEV is a list of the devices you want to ring when you use this
variable in the dial statement. sipurafxs1 is the fxs side of the spa3k
and I have one grandstream 200, at extension 406, named grandstream406.
The analog extension, fxs on the spa3k, is 405.
How would I ring all the _sip_ phones when a pstn call comes in?
My macro 'ring-all' ?
You just add them all together in the ring statement with the as in my
INRINGSDEV variable. Actually the use of the variable was taken from
sample code given to me when I started out. It is probably a good idea
though. you could just put them all in the dial statement but if you use
it in more than one place it is handy to just change it in one place and
use the variable.
SIP/sipurafxs1SIP/grandstream406thirdfourth.
Notes:
Your sipurafxo1 is my spa3k-pstn-in defined in both Sipura and sip.conf.
My extension to ring incoming calls is 120 vs your 405. All ok on these
two.
I'm nearly there thanks to you.
OK glad it helped. If you have any other questions let me know. The spa3k
has a million settings.
Larry
Doug Crompton wrote:
Below is my config for spa3k fxo. I do not show the settings in the spa3k
which must reflect settings here, port, username, secret, etc. I have
DTMF set to inband here and in spa3k to fix a problem with DTMF not
working for menus from PSTN. This was discussed earlier and is a problem
in asterisk that may (or may not) be solved in 1.4. I am using earlier
version. Inband must also be specifed in spa3k pstn.
[sipurafxo1]
type=peer
username=sipurafxo1
secret=x
canreinvite=no
context=from-pstn
host=dynamic
nat=no
port=5061
disallow=all
allow=alaw
allow=ulaw
allow=gsm
allow=g723.1
dtmfmode=inband
In extensions.conf. This is a little fancy but the bottom line is that it
ends up in either a day or night mode. Only day shown. The spa3k fxo in
sip calls the from-pstn but the pstn-day-time (below) could be relabeled
from-pstn to always go to phones. The night mode basically goes to VM.
INRINGSEXT and INRINGSDEV are just variables defined to -
INRINGSDEV=SIP/sipurafxs1SIP/grandstream406 ; ring analog phones on spa3k
fxs
INRINGSEXT=405 ; the extension to ring for incomming calls
The stdexten macro is just the standard one in sample extension file.
[from-pstn]
exten = s,1,GotoIf($[ ${day-night} = 0 ]?2:10
exten = s,2,GotoIfTime(9:30-23:59,*,*,*?pstn-day-time,s,1
exten = s,3,GotoIfTime(0:00-09:29,*,*,*?night-time,s,1
exten = s,10,GotoIf($[ ${day-night} = 1 ]?pstn-day-time,s,1
exten = s,11,GotoIf($[ ${day-night} = 2 ]?night-time,s,1
[pstn-day-time]
exten = s,1,SetGlobalVar(RingTimeout=35)
exten = s,2,NoOp(${CALLERID})
exten = s,3,Macro(stdexten,${INRINGSEXT},${INRINGSDEV},)
On Tue, 14 Nov 2006, Larry Alkoff wrote:
My SIP phones can dial out through Sipura SPA3k to POTS for local and
911 calls _but_ incoming POTS calls are being swallowup somehow.
Am I on the right track with the code snippit below?
sip.conf:
-
In sip.conf the following code is _supposed_ to ring the SIP phones when
a POTS line call comes in through Sipuara to Asterisk.
[spa3k-pstn-in] ; Pots-line-in from Sipura
; If you're using Asterisk, this goes into the Incoming settings
; For your Trunk
host=dynamic
type=friend ; should be peer if incoming only ??
context=[macro-ringall] ;ring all the sip phones
secret=x
RE: [asterisk-users] Asterisk Manager: equivalent of 'show channels'?
you need to look again maybe read, http://www.dynx.net/ASTERISK/DOCS/RTF/MANAGER.RTF Mea culpa! I did not realize it was so easy! I totally missed the command action when looking over the docs. Thanks for pointing out the mistake and thanks too for the link. It has helped a lot. -MC ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to use Sipura SPA3k POTS line to dial Asterisk SIP phones?
Thanks very much for your sipurafxs1.
The problem has been that incoming POTS calls are swallowed up after the
first ring or so if the pstn line is connected to Sipura.
I'll try this and let you know.
Larry
Doug Crompton wrote:
This is my spa3k fxs port sip.conf params. This uses the default context
in my extensions.conf
What are you having trouble doing? Can you make calls out to PSTN? Is it
just incoming call that are not ringing?
Doug
[sipurafxs1]
type=friend
regexten=405
username=sipurafxs1
secret=
context=default
context=from-pstn
callerid=Doug Crompton 405
host=dynamic
nat=no
port=5061
canreinvite=no
disallow=all
allow=alaw
allow=ulaw^M
allow=gsm
allow=g723.1^M
[EMAIL PROTECTED]
dtmfmode=rfc2833
On Sat, 18 Nov 2006, Larry Alkoff wrote:
Doug Crompton wrote:
Doug, please forgive me but I'm still having trouble understanding two
points from your last response.
Can you please post your extension 405 (analog extension on spa3k) in
sip.conf
and your [sipurafxs1] ?
I finally understand that INRINGSDEV is meant to specify which analog
and SIP phones to ring at extension INRINGSEXT = 405 and would like to
see just how you do it.
Larry
On Wed, 15 Nov 2006, Larry Alkoff wrote:
Thank you very much Doug for your detailed response to my question.
I'm working on a new sip.conf and extensions.conf using your code as a
guide.
Questions:
In INRINGSDEV what does sipurafxs1 and grandstream406 refer to?
The comment says ring analog phones on spa3k fxs but grandstream406
seems to refer a Grandstream sip phone, not an analog one.
Does INRINGSDEV mean ring a specific sip phone and the analog ones?
INRINGSDEV is a list of the devices you want to ring when you use this
variable in the dial statement. sipurafxs1 is the fxs side of the spa3k
and I have one grandstream 200, at extension 406, named grandstream406.
The analog extension, fxs on the spa3k, is 405.
How would I ring all the _sip_ phones when a pstn call comes in?
My macro 'ring-all' ?
You just add them all together in the ring statement with the as in my
INRINGSDEV variable. Actually the use of the variable was taken from
sample code given to me when I started out. It is probably a good idea
though. you could just put them all in the dial statement but if you use
it in more than one place it is handy to just change it in one place and
use the variable.
SIP/sipurafxs1SIP/grandstream406thirdfourth.
Notes:
Your sipurafxo1 is my spa3k-pstn-in defined in both Sipura and sip.conf.
My extension to ring incoming calls is 120 vs your 405. All ok on these
two.
I'm nearly there thanks to you.
OK glad it helped. If you have any other questions let me know. The spa3k
has a million settings.
Larry
Doug Crompton wrote:
Below is my config for spa3k fxo. I do not show the settings in the spa3k
which must reflect settings here, port, username, secret, etc. I have
DTMF set to inband here and in spa3k to fix a problem with DTMF not
working for menus from PSTN. This was discussed earlier and is a problem
in asterisk that may (or may not) be solved in 1.4. I am using earlier
version. Inband must also be specifed in spa3k pstn.
[sipurafxo1]
type=peer
username=sipurafxo1
secret=x
canreinvite=no
context=from-pstn
host=dynamic
nat=no
port=5061
disallow=all
allow=alaw
allow=ulaw
allow=gsm
allow=g723.1
dtmfmode=inband
In extensions.conf. This is a little fancy but the bottom line is that it
ends up in either a day or night mode. Only day shown. The spa3k fxo in
sip calls the from-pstn but the pstn-day-time (below) could be relabeled
from-pstn to always go to phones. The night mode basically goes to VM.
INRINGSEXT and INRINGSDEV are just variables defined to -
INRINGSDEV=SIP/sipurafxs1SIP/grandstream406 ; ring analog phones on spa3k
fxs
INRINGSEXT=405 ; the extension to ring for incomming calls
The stdexten macro is just the standard one in sample extension file.
[from-pstn]
exten = s,1,GotoIf($[ ${day-night} = 0 ]?2:10
exten = s,2,GotoIfTime(9:30-23:59,*,*,*?pstn-day-time,s,1
exten = s,3,GotoIfTime(0:00-09:29,*,*,*?night-time,s,1
exten = s,10,GotoIf($[ ${day-night} = 1 ]?pstn-day-time,s,1
exten = s,11,GotoIf($[ ${day-night} = 2 ]?night-time,s,1
[pstn-day-time]
exten = s,1,SetGlobalVar(RingTimeout=35)
exten = s,2,NoOp(${CALLERID})
exten = s,3,Macro(stdexten,${INRINGSEXT},${INRINGSDEV},)
On Tue, 14 Nov 2006, Larry Alkoff wrote:
My SIP phones can dial out through Sipura SPA3k to POTS for local and
911 calls _but_ incoming POTS calls are being swallowup somehow.
Am I on the right track with the code snippit below?
sip.conf:
-
In sip.conf the following code is _supposed_ to ring the SIP phones when
a POTS line call comes in through Sipuara to Asterisk.
[spa3k-pstn-in] ; Pots-line-in from Sipura
; If you're using Asterisk, this goes into the Incoming settings
; For your Trunk
host=dynamic
type=friend ; should be peer if incoming only ??
context=[macro-ringall] ;ring all the sip
Re: [asterisk-users] Freepbx changes dont reflect in asterisk
I also restarted the box and the problem is not solved :( PS 2006/11/18, Dumpolid Exeplish [EMAIL PROTECTED]: i also used to have this problem, for instance we use the pin code functionality of FreePBX and whenever i add or modify a pin number, it is not effected or changed in the config files. i dont know what causes this error but i have noticed that restarting FreePBX or re-installing the application stops this. Just restart the box On 11/18/06, Pedro Silva [EMAIL PROTECTED] wrote: Alex was right. The problem is that when i make changes in freepbx, those changes are not written in the config files. I only made modifications in files_custom.conf. The version of freePbx that i use is 2.1.1 (not beta) and Asterisk 1.2.12.1. Thanks by your help, Ps. 2006/11/18, Alex Robar [EMAIL PROTECTED]: I think you guys are all misunderstanding the problem here. Unless I'm misunderstanding, Pedro's issue is that when he makes changes in FreePBX, those changes are not written out to the config files. Alex On 11/17/06, Zeeshan Zakaria [EMAIL PROTECTED] wrote: You can't do any modifications in extensions_additional.conf and sip_additional.conf. Same is true for extensions.conf and sip.conf, and other original trixbox files. As soon as you press the red bar, they are returned to their original state. For modifications, you create your own files or use sip_customs.conf and extensions_custom.conf. Please don't mix trixbox with asterisk just because its based on asterisk. Its a completely customized solution of various software packages configured to make it work according to its own requirements. For help, post on trixbox.org forums. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to use Sipura SPA3k POTS line to dial Asterisk SIP phones?
I'm gotten confused trying to figure out which extension in my sip.conf corresponds to your sipurafsx1. Below are the configurations from the Voxilla Configuration Wizard and your sipurafxs1 at the bottom. I've snipped the rest as it's pretty long and this is the meat of my present problem. My guess is that I should modify username=spa3k-pstn-in to conform to your sipurafxs1 but I'd like a reality check as I've lost the overall picture. Larry Doug Crompton wrote: This is my spa3k fxs port sip.conf params. This uses the default context in my extensions.conf What are you having trouble doing? Can you make calls out to PSTN? Is it just incoming call that are not ringing? Doug ;; --- ;; Sipura sip.conf entries 11/6/06 ;; from Voxilla SPA-3000 Configuration Wizard: ;; --- [spa3k-pots-dial] ; was 6664 POTS phone connected to Sipura dials ;this, then pstn-spa3k: ;There is no connection made to Telasip ; type=friend host=dynamic context=pstn-out secret=x mailbox=spa3k-pots-dial dtmfmode=rfc2833 disallow=all allow=ulaw [spa3k-pstn-in] ; was 6667 in Wizard. Pots-line-in from Sipura ; Sipura fxo (right hand port) from Doug ; Crompton [sipurafxo1] ; connect POTS line1 to line to CO type=peer username=spa3k-pstn-in secret=x canreinvite=no context=from-pstn host=dynamic nat=no port=5061 disallow=all allow=alaw allow=ulaw allow=gsm allow=g723.1 dtmfmode=inband [spa3k-pstn-out]; was pstn-spa3k ; If you're using Asterisk, this section goes ; into the Outgoing Settings for your trunk. type=peer ; outgoing auth=md5 host=10.22.44.41 port=5061 secret=xx username=asterisk fromuser=asterisk dtmfmode=rfc2833 ; If using [EMAIL PROTECTED], change the below line to context=from-internal context=pstn-out insecure=very [sipurafxs1] type=friend regexten=405 username=sipurafxs1 secret= context=default context=from-pstn ; callerid=Doug Crompton 405 callerid=Larryy Alkoff 405 host=dynamic nat=no port=5061 canreinvite=no disallow=all allow=alaw allow=ulaw allow=gsm allow=g723.1 [EMAIL PROTECTED] dtmfmode=rfc2833 -- Larry Alkoff N2LA - Austin TX Using Thunderbird on Linux ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Asterisk to listen for sip traffic on 80 and 5060
I have Asterisk listening for sip traffic on port 5060. I want to allow users to use either port 80 or 5060 if they want. Hopefully this will avoid some firewall issues. Is this a sensible/crazy thing to do? I have done a bunch of searching and believe iptables can help but haven't been able to find an example to forward something from 80 to 5060 inbound and outbound where iptables is running on the same machine as Asterisk. Is iptables the best way to do it (without other hardware) or is there an alternative? If anyone has used iptables to do this would you be willing to share the setup? Would something like ths work for inbound?: iptables -t nat -A PREROUTING -p udp --dport 80 --sport 1024:65535 -j DNAT --to 127.0.0.1:5060 iptables -A FORWARD -p udp -d 1270.0.1 \ --dport 5060 -m state --state NEW -j ACCEPT iptables -A FORWARD -t filter -m state \ --state NEW,ESTABLISHED,RELATED -j ACCEPT What about outbound? Alternatively is there a better option? Any suggestions appreciated. Regards Cameron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Asterisk to listen for sip traffic on 80 and 5060
kjcsb wrote: I have Asterisk listening for sip traffic on port 5060. I want to allow users to use either port 80 or 5060 if they want. Hopefully this will avoid some firewall issues. If you're think that by sending SIP on port 80 will fool the firewall into thinking it's HTTP traffic, then I'd suggest you look elsewhere. For a start, most firewalls only allow HTTP on TCP/80 not UDP/80. Leo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialout Conferences?
Matt, I'm not exactly sure what you are trying to do, but you should just be able to throw that call into the conference with a context... something like [throw-into-conference] exten = s,1,MeetMe(1234) Whatever you are using to start the dialplan dialing should throw the call into this context when it completes. On 11/18/06, Matthew Rubenstein [EMAIL PROTECTED] wrote: How do I set up an existing call to dial out to a new terminal which is included in a conference with the two existing legs of the call? When the dialplan executes the Dial(terminal) command, control does not return to the dialplan until the terminal disconnects, after which it's obviously too late to conference it. Is there a conference command or option that lets the dialplan dial out to a terminal to a new conference member? Is there a dial command or option that returns control to the dialplan for then calling a conference command on the dialed terminal, or executes another command inside the process cycle of the dial command that adds the dialed terminal to a conference, after the terminal answers but before it hangs up? -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] spc.exe
Why do you need it? You can just use XML config files. On 11/18/06, Andrew Joakimsen [EMAIL PROTECTED] wrote: Does anyone have a copy of spc.exe they could send me? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial/Continue/Announce
Try the background() command to play a sound file or say digits to the user, while not holding up the dial-plan. On 11/13/06, Matthew Rubenstein [EMAIL PROTECTED] wrote: I initiate a call with a callfile, specifying the From phone# as the channel Dial(), and the To phone# as the Extension Dial(). I announce the To phone# to the From listener with the A() option to the Dial() command. It seems that the A() app plays audio while blocking return from the From Dial(), so the From person has to wait more time while listening to the announcement before the To Dial() can even begin, which then must start ringing. How can I connect to the From person, then immediately connect to the To person, while playing an announcement to the From person? -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] odd issue with IP tables
I put iptables on my asterisk box and an odd thing occurs. I allow 5060 and 1-2. As soon as I start iptables and make a call it literally takes 60-90 seconds before the call even starts to ring. As soon as I shut iptables off, the call goes through immediately again. Its quite odd. The call does eventually go through and talks fine but it takes sooo long to connect. Anyone have some suggestions? Thanks Curt ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] odd issue with IP tables
Hi Curt, I would try to find out what it's doing that's slowing your system down so much. Try turning up logging (or examining the logs) to see what's going on there, or what it's waiting for. Off the top of my head, I'm wondering if you've blocked everything else, or have you allowed through the standard fare? Ie. Have you allowed existing and related connections? Also, where is iptables sitting? Is it on the local Asterisk box, or is it on a firewall/router box in front of the Asterisk box? Alex On 11/18/06, Curt Shaffer [EMAIL PROTECTED] wrote: I put iptables on my asterisk box and an odd thing occurs. I allow 5060 and 1-2. As soon as I start iptables and make a call it literally takes 60-90 seconds before the call even starts to ring. As soon as I shut iptables off, the call goes through immediately again. Its quite odd. The call does eventually go through and talks fine but it takes sooo long to connect. Anyone have some suggestions? Thanks Curt ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] odd issue with IP tables
Post your IP tables configuration here if it isn't too big. Ron -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Curt Shaffer Sent: Saturday, November 18, 2006 5:05 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] odd issue with IP tables I put iptables on my asterisk box and an odd thing occurs. I allow 5060 and 1-2. As soon as I start iptables and make a call it literally takes 60-90 seconds before the call even starts to ring. As soon as I shut iptables off, the call goes through immediately again. Its quite odd. The call does eventually go through and talks fine but it takes sooo long to connect. Anyone have some suggestions? Thanks Curt ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] odd issue with IP tables
-A INPUT # Accept traffic with the ACK flag set -A INPUT -p tcp -m tcp --tcp-flags ACK ACK -j ACCEPT # Allow incoming data that is part of a connection we established -A INPUT -m state --state ESTABLISHED -j ACCEPT # Allow data that is related to existing connections -A INPUT -m state --state RELATED -j ACCEPT -A INPUT -p tcp -m tcp --dport ssh -j ACCEPT -A INPUT -p tcp -m tcp --dport 80 -j ACCEPT -A INPUT -p udp -m udp --dport 5060:5061 -j ACCEPT -A INPUT -p udp -m udp --dport 1:2 -j ACCEPT -A INPUT -p udp -m udp --dport 4569 -j ACCEPT And to respond to Alex, The box is only doing Asterisk. 2.8Ghz proc with 1GB of RAM. The iptables is on the server itself. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ron McLeod Sent: Saturday, November 18, 2006 7:36 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] odd issue with IP tables Post your IP tables configuration here if it isn't too big. Ron -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Curt Shaffer Sent: Saturday, November 18, 2006 5:05 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] odd issue with IP tables I put iptables on my asterisk box and an odd thing occurs. I allow 5060 and 1-2. As soon as I start iptables and make a call it literally takes 60-90 seconds before the call even starts to ring. As soon as I shut iptables off, the call goes through immediately again. Its quite odd. The call does eventually go through and talks fine but it takes sooo long to connect. Anyone have some suggestions? Thanks Curt ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] odd issue with IP tables
Do your user agents use some services from the server such as DNS? Ron -Original Message- From: Curt Shaffer [mailto:[EMAIL PROTECTED] Sent: Saturday, November 18, 2006 5:41 PM To: [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non- Commercial Discussion' Subject: RE: [asterisk-users] odd issue with IP tables -A INPUT # Accept traffic with the ACK flag set -A INPUT -p tcp -m tcp --tcp-flags ACK ACK -j ACCEPT # Allow incoming data that is part of a connection we established -A INPUT -m state --state ESTABLISHED -j ACCEPT # Allow data that is related to existing connections -A INPUT -m state --state RELATED -j ACCEPT -A INPUT -p tcp -m tcp --dport ssh -j ACCEPT -A INPUT -p tcp -m tcp --dport 80 -j ACCEPT -A INPUT -p udp -m udp --dport 5060:5061 -j ACCEPT -A INPUT -p udp -m udp --dport 1:2 -j ACCEPT -A INPUT -p udp -m udp --dport 4569 -j ACCEPT And to respond to Alex, The box is only doing Asterisk. 2.8Ghz proc with 1GB of RAM. The iptables is on the server itself. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ron McLeod Sent: Saturday, November 18, 2006 7:36 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] odd issue with IP tables Post your IP tables configuration here if it isn't too big. Ron -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Curt Shaffer Sent: Saturday, November 18, 2006 5:05 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] odd issue with IP tables I put iptables on my asterisk box and an odd thing occurs. I allow 5060 and 1-2. As soon as I start iptables and make a call it literally takes 60-90 seconds before the call even starts to ring. As soon as I shut iptables off, the call goes through immediately again. Its quite odd. The call does eventually go through and talks fine but it takes sooo long to connect. Anyone have some suggestions? Thanks Curt ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ringing a group of phones but not if they are busy
Try something like this:
exten = s,1,Dial(Local/[EMAIL PROTECTED]Local/[EMAIL PROTECTED]Local/[EMAIL
PROTECTED])
[callphones]
exten = _X.,1,ChanIsAvail(Sip/${EXTEN},js)
exten = _X.,2,Dial(Sip/${EXTEN})
exten = _X.,102,Noop(${EXTEN} is on a call)
On 11/17/06, Carlos Chavez [EMAIL PROTECTED] wrote:
I need to ring a group of 8 phones, but not if they are already on
another call. How can I determine which of those 8 phones are busy so I
only ring the others?
--
Telecomunicaciones Abiertas de Mexico S.A. de C.V.
Carlos Chàvez Prats
Director de Tecnologìa
+52-55-91169161 ext 2001
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RE: [asterisk-users] odd issue with IP tables
Nope the server is for Asterisk only. I have SSH on it for management, FreePBX for configuration, SIP clients and IAX termination. -Original Message- From: Ron McLeod [mailto:[EMAIL PROTECTED] Sent: Saturday, November 18, 2006 8:06 PM To: 'Curt Shaffer'; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] odd issue with IP tables Do your user agents use some services from the server such as DNS? Ron -Original Message- From: Curt Shaffer [mailto:[EMAIL PROTECTED] Sent: Saturday, November 18, 2006 5:41 PM To: [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non- Commercial Discussion' Subject: RE: [asterisk-users] odd issue with IP tables -A INPUT # Accept traffic with the ACK flag set -A INPUT -p tcp -m tcp --tcp-flags ACK ACK -j ACCEPT # Allow incoming data that is part of a connection we established -A INPUT -m state --state ESTABLISHED -j ACCEPT # Allow data that is related to existing connections -A INPUT -m state --state RELATED -j ACCEPT -A INPUT -p tcp -m tcp --dport ssh -j ACCEPT -A INPUT -p tcp -m tcp --dport 80 -j ACCEPT -A INPUT -p udp -m udp --dport 5060:5061 -j ACCEPT -A INPUT -p udp -m udp --dport 1:2 -j ACCEPT -A INPUT -p udp -m udp --dport 4569 -j ACCEPT And to respond to Alex, The box is only doing Asterisk. 2.8Ghz proc with 1GB of RAM. The iptables is on the server itself. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ron McLeod Sent: Saturday, November 18, 2006 7:36 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] odd issue with IP tables Post your IP tables configuration here if it isn't too big. Ron -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Curt Shaffer Sent: Saturday, November 18, 2006 5:05 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] odd issue with IP tables I put iptables on my asterisk box and an odd thing occurs. I allow 5060 and 1-2. As soon as I start iptables and make a call it literally takes 60-90 seconds before the call even starts to ring. As soon as I shut iptables off, the call goes through immediately again. Its quite odd. The call does eventually go through and talks fine but it takes sooo long to connect. Anyone have some suggestions? Thanks Curt ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cant record phone calls
Hi all, I tried enabling the automon feature, but for some reason it doesn't start recording when I or the caller press *1 when the call is iniciated. exten = 200,1,set(DYNAMIC_FEATURE=automon exten = 200,2dial(SIP/phone1,30,rWw) Any thoughts? Many thanks, Christian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] spc.exe
The phone downloads the XML file but does not apply it. On 11/18/06, Matt [EMAIL PROTECTED] wrote: Why do you need it? You can just use XML config files. On 11/18/06, Andrew Joakimsen [EMAIL PROTECTED] wrote: Does anyone have a copy of spc.exe they could send me? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to use Sipura SPA3k POTS line to dial Asterisk SIP phones?
Doug tThis is my latest sip.conf and extensions.conf with your
sipuraxfs1 renamed to spa3k-pstn-in as it's called in my Sipura.
The non-Sipura parts have been mostly snipped.
I still have the problem that an incoming POTS call is swallowed up
after the 2nd ring.
Larry
[EMAIL PROTECTED] work # cat extensions.conf
-
[general]
;
static=yes ; These two lines prevent the command-line interface
writeprotect=yes; from overwriting the config file. Leave them here.
autofallthrough=yes ; terminate call with BUSY, CONGESTION or HANGUP
if runs out of things to do.
clearglobalvars=yes ; clear and reparse global variables extensions
or Asterisk reload
prioityjumping=no
; INRINGSDEV=SIP/sipurafxs1SIP/grandstream406 ; Compton: ring analog
phones on spa3k fxs and selected SIP
INRINGSDEV=SIP/sipurafxs1SIP/120 ; ring spa3k fxs and selected SIP phones
; INRINGSEXT=405; Compton: the extension to ring for incomming calls
INRINGSEXT=120 ; the extension to ring for incomming calls
; INRINGSDEV: is a list of the devices you want to ring when you use this
; variable in the dial statement.
; sipurafxs1 is the fxs side of the spa3k presently 405
; and I have one grandstream 200, at extension 406, named grandstream406.
; INRINGSEXT: The analog extension, fxs on the spa3k, is 405.
[sipura-in]
;==
;exten = _3011414,1,macro(calllarry,s,1)
;; exten = _5123011414,1,macro(calllarry,s,1)
exten = s,1,macro(calllarry,s,1)
exten = s,2,Hangup()
[telasip-in]
;===
exten = _512879677[67],1,macro(ringall,s,1)
exten = _879677[67],1,macro(ringall,s,1)
[macro-ringall] ; ring all stations
exten = s,1,Dial(SIP/120SIP/121SIP/122SIP/124SIP/125SIP/126SIP/127)
exten = s,2,hangup
[telasip-out]
;=
;; WORKS CID 1415 for local calls from a SIP phone:
exten = _NXX,1,SetCallerID(512-301-1415)
exten = _NXX,2,Dial(SIP/[EMAIL PROTECTED])
;; WORKS CID 1416 for long distance calls without dialing 1:
exten = _NXXNXX,1,SetCallerID(512-301-1416)
exten = _NXXNXX,2,Dial(SIP/[EMAIL PROTECTED])
;; WORKS CID 1417 for long distance calls with a dialed 1:
exten = _1NXXNXX,1,SetCallerID(512-301-1417)
exten = _1NXXNXX,2,Dial(SIP/1${EXTEN:[EMAIL PROTECTED])
;;[local-lba]
;; ?? exten = _NXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
;; ---
;; Sipura extensions.conf entries 11/6/06
;; from Voxilla SPA-3000 Configuration Wizard:
;; ---
;; In extensions.conf (assume calls dialed as 7 digits or 800 calls
;; go out the PSTN line as well as 911 calls):
[pstn-out]
exten = spa3k-pots-dial,1,Ringing
exten = spa3k-pots-dial,2,Dial(SIP/spa3k-pots-dial,20,T)
exten = spa3k-pots-dial,3,Voicemail(uspa3k-pots-dial)
exten = spa3k-pots-dial,4,Hangup
exten = 911,1,Dial(SIP/[EMAIL PROTECTED],60,)
exten = 911,2,Congestion
exten = _XXX,1,Dial(SIP/[EMAIL PROTECTED],60,)
exten = _XXX,2,Congestion
exten = _1800XXX,1,Dial(SIP/[EMAIL PROTECTED],60,)
exten = _1800XXX,2,Congestion
exten = _1888XXX,1,Dial(SIP/[EMAIL PROTECTED],60,)
exten = _1888XXX,2,Congestion
exten = _1877XXX,1,Dial(SIP/[EMAIL PROTECTED],60,)
exten = _1877XXX,2,Congestion
exten = _1866XXX,1,Dial(SIP/[EMAIL PROTECTED],60,)
exten = _1866XXX,2,Congestion
[from-pstn] ; from Doug Crompton
exten = s,1,SetGlobalVar(RingTimeout=35)
exten = s,2,NoOp(${CALLERID})
; exten = s,3,Macro(stdexten,${INRINGSEXT},${INRINGSDEV},) ; Compton
exten = s,3,Macro(stdexten,,${INRINGSDEV},)
; The stdexten macro is just the standard one in sample extension file.
[EMAIL PROTECTED] work # cat sip.conf
---
[general]
;
port = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
allow=all ; Allow all codecs
;; context = default; default context for incoming calls
context = bogon-calls ; Send SIP callers that we don't know about here
register = :[EMAIL PROTECTED]/5128796776
;
[telasip-gw]; INBOUND sip trunk to Telasip:
;===
context=telasip-in
type=friend
qualify=200 ; Mark peer unavailable if ping more than 200ms
host=gw3.telasip.com; set in Sipura setup Line 1 Gateway
Accounts - currently GW1 and GW1 Auth ID = blank
; PSTN = gw0 (by default)
username=xxx
secret=
insecure=very
canreinvite=no
callerid=x xx
nat=yes
; Next 2 lines 11/6/06 per Telasip instructions.
fromuser=xxx
sendrpid=yes
[
[120] ; Larry Office extension
;===
context=default
username=120
type=friend
qualify=yes
insecure=very
host=dynamic
canreinvite=no
nat=no
;; ---
;; Sipura sip.conf entries 11/6/06
Re: [asterisk-users] odd issue with IP tables
On Sat, Nov 18, 2006 at 07:04:55PM -0600, Curt Shaffer wrote: I put iptables on my asterisk box and an odd thing occurs. I allow 5060 and 1-2. As soon as I start iptables and make a call it literally takes 60-90 seconds before the call even starts to ring. As soon as I shut iptables off, the call goes through immediately again. Its quite odd. The call does eventually go through and talks fine but it takes sooo long to connect. Anyone have some suggestions? You're almost certainly blocking DNS lookups, somewhere. Cheers, -- jra -- Jay R. Ashworth[EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 That's women for you; you divorce them, and 10 years later, they stop having sex with you. -- Jennifer Crusie; _Fast_Women_ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
