[asterisk-users] Connecting 2 asterisk servers
hi all, actually i have partially connected the 2 servers but there is a problem. 2 servers A and B server A forwards call to server B without any problem but when i try to forward call from server B to A, server shows the following error on the cli WARNING[7751]: app_dial.c:1081 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) == Everyone is busy/congested at this time (1:0/0/1) here is my configuration SERVER A sip.conf ~~ [general] register=fromtrixbox:[EMAIL PROTECTED] [receive] ;this should receive call from server B type=friend ;host=192.168.0.7 host = dynamic secret=mysecret context=rizwan ;trunk=yes insecure = very [backtorazi];this is used to throw call to server B (works fine) type=friend host=dynamic secret=mysecret context=rizwan SERVER A extensions ~~~ [rizwan] exten= 5123,1,Dial(SIP/backtorazi/1234) ;this dials to server B exten= 5123,2,Hangup exten= 1234,1,NoOp(Got connected successfully!); ;this receives from server B SERVER B sip.conf ~ [general] register=backtorazi:[EMAIL PROTECTED] [sender] ;used tro call server A (not working) type=friend ;host=192.168.0.100 host=dynamic secret=mysecret context=rizwan [fromtrixbox] ;used to receive call from server A (working fine) type=friend host=192.168.0.100 secret=mysecret context=rizwan ;trunk=yes insecure = very SERVER B extensions ~~~ [rizwan] exten = 5123,1,Dial(SIP/sender/5123) ;this dials server A exten= 1234,1,Dial(SIP/1234) ;this receives from server A -- Regards Rizwan Hisham Software Engineer ___ --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-users Digest, Vol 30, Issue 79
Hi,
what do you now get in the way of error messages?
Robert Jenkins.
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Vidura
Senadeera
Sent: 19 January 2007 23:03
To: asterisk-users@lists.digium.com
Cc: [EMAIL PROTECTED]
Subject: [asterisk-users] Re: asterisk-users Digest, Vol 30, Issue 79
Hi,
I checked by changing to from-zaptel, but no luck yet. Pls guide me on
this.
Regards,
vudura senadeera
--
Message: 9
Date: Fri, 19 Jan 2007 16:47:18 -
From: Robert Jenkins [EMAIL PROTECTED]
Subject: RE: [asterisk-users] Integrating asterisk with Toshiba
Astrata DK380
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=us-ascii
Hi,
your zapata.con has 'context=from-pstn'
Try changing this to 'context=from-zaptel'
Robert Jenkins.
_
From: [EMAIL PROTECTED]
[mailto: mailto:[EMAIL PROTECTED]
[EMAIL PROTECTED] On Behalf Of Vidura
Senadeera
Sent: 19 January 2007 15:19
To: asterisk-users@lists.digium.com
Cc: [EMAIL PROTECTED]
Subject: [asterisk-users] Integrating asterisk with Toshiba Astrata DK380
Deat all,
I am in middle of integrate Asterisk with Toshiba astrata legacy pbx.
Following is my setup
Asterisk - Digium TE110P - E1 card in toshiba pbx - Toshiba PBX
A = B
C D
Asterisk PBX and strata PBX connected using back to back E1 cross cable.
Physicall connectivity is OK. The digium TE110p
LED state green. zttool also OK.
Toshiba stata configured to make outbound call via E1 link with pressing 9
and then the out side number.
I was able to make call from soft phone to analog extension at toshiba pbx.
A== B way as shown above. But when trying to dial from
Toshiba PBX analog extension to asterisk extension, by pressing 9 the call
rejected.
In the asterisk command prompt I'm having following error message.
-- Extension '' in context 'from-pstn' from '' does not exist. Rejecting
call on channel 0/31, span 1
Is there any wrong in my setup. dial plan??, additional configuration if i
required to put please guide me.
I will be greately appreciated your feedback on this regard.
configuration details
/etc/zaptel.conf
# Span 1: WCT1/0 Digium Wildcard TE110P T1/E1 Card 0
span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16
/etc/asterisk/zapata.conf
signalling=pri_net ; pri_cpe = PRI slave ; pri_net = PRI master
switchtype=euroisdn
;switchtype=national
echocancel=yes ; You can set this to 32, 64, or 128, tweak to your needs.
echocancelwhenbridged=yes
echotraining=400 ; Asterisk trains to the beginning of the call, number is
in milliseconds
callerid=asreceived
overlapdial=no
pridialplan=unknown
immediate=no
;rxwink=300
callprogress=no
loadzone=au
context=from-pstn ; Points to the default context of your extensions.conf
group=2
channel=1-15
channel=17-31 ;PRI/E1 link
[trunkgroups]
trunkgroup=2,16
spanmap=1,2,1
/etc/asterisk/extension.conf
[from-zaptel]
exten = _X.,1,Set(DID=${EXTEN})
exten = _X.,n,Goto(s,1)
exten = s,1,NoOp(Entering from-zaptel with DID == ${DID})
; If ($did == ) { $did = s; }
exten = s,n,Set(DID=${IF($[${DID}= ]?s:${DID})})
exten = s,n,NoOp(DID is now ${DID})
exten = s,n,GotoIf($[${CHANNEL:0:3}=Zap]?zapok:notzap)
exten = s,n(notzap),Goto(ext-did,${DID},1)
; If there's no ext-did,s,1, that means there's not a no did/no cid route.
Hangup.
exten = s,n,Macro(hangup)
exten = s,n(zapok),NoOp(Is a Zaptel Channel)
exten = s,n,Set(CHAN=${CHANNEL:4})
exten = s,n,Set(CHAN=${CUT(CHAN,-,1)})
exten = s,n,Macro(from-zaptel-${CHAN},${DID},1)
; If nothing there, then treat it as a DID
exten = s,n,NoOp(Returned from Macro from-zaptel-${CHAN})
exten = s,n,Goto(ext-did,${DID},1)
--
Thanks Regards,
Vidura B. Senadeera.
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Message: 10
Date: Fri, 19 Jan 2007 11:46:57 -0500
From: Chris Earle \(CBL\) [EMAIL PROTECTED]
Subject: [asterisk-users] Disconnect Supervision UK / BT solution?
To: mailto:asterisk-users@lists.digium.com
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=iso-8859-1
Hi all
I'm using sangoma a200 cards in the UK and have the ongoing, often noted
problem of disconnect supervision with BT POTS lines.
Just noticed this post on
http://www.voip-info.org/wiki/view/UK+Asterisk+Details
stating that potentially someone's got a solution :
TDM400P amp; Not Detecting Hangups:
Got a TDM400P installed and having problems with Asterisk not detecting
hangups? Using BT? If so, contact BT and ask what the Disconnect Clear
Time setting is for your phone line. Odds are it's probably 100. Increasing
it to 800 fixed the issue for me.
Re: [asterisk-users] Disconnect Supervision UK / BT solution?
Well, I have just phoned BT today who said they can increase the CPC value on the line - however it needs to be done at the exchange - and has been booked for Tues. I suppose I will know wether this worked on Tues :-) - I shall post my findings. Regards -- Matt Brown On 19 Jan 2007, at 16:46, Chris Earle ((CBL)) wrote: Hi all I'm using sangoma a200 cards in the UK and have the ongoing, often noted problem of disconnect supervision with BT POTS lines. Just noticed this post on http://www.voip-info.org/wiki/view/UK+Asterisk+Details stating that potentially someone's got a solution : TDM400P amp; Not Detecting Hangups: Got a TDM400P installed and having problems with Asterisk not detecting hangups? Using BT? If so, contact BT and ask what the Disconnect Clear Time setting is for your phone line. Odds are it's probably 100. Increasing it to 800 fixed the issue for me. Disconnect Clear Time is BT's name for CPC. Does anyone have any thoughts/confirmation about this finally being a viable solution? This disconnect supervision problem has plagued TDM and Sangoma cards for a long time! Comments appreciated before I get on the phone with BT -- Chris Earle System Solutions Specialist -- This message has been scanned for viruses and dangerous content and is believed to be clean. ___ --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 7970 Unprovisioned
Hi! I did manage to load phone with SIP image : SIP70.8-0-3S, made SEP-MAC.cnf.xml, but phone never read the configuration from it. On the screen it's written Unprovisioned, and phone is not trying to register with asterisk. Please help!! MihaelaMJ ___ --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] About BRI / ISDN hardware. What to buy?
Hi, http://store.%50honiceq.com has the quadbri card for $400. We can also offer free shipping for this card. The card has 4 BRI ports and is based on the same main chipset (HFC-4S) as Digium/Beronet/Junghanns cards. It does not have EC onboard. best regards On 1/18/07, Cosmin Prund [EMAIL PROTECTED] wrote: I finally found a price tag for the darn thing, at around 500 euros I can handle it. Qustion: Do they behave properly if I've got an other Digium TDM400 card in the system? How about installing two cards in the same server? At the moment I've only got 1 ISDN line plus a few analog lines going into the TDM but in the very near future we might want to get a second ISDN. Alberto Pastore wrote: Jens Vagelpohl ha scritto: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On 18 Jan 2007, at 18:31, Patrick wrote: I think http://www.melware.de carries the Eicon Server ISDN cards which have hardware echo cancellation. They are also the author of the chan_capi driver for Asterisk. I use the Eicon Server BRI cards with Asterisk myself and they work very well. I concur, I have a Eicon DIVA single port BRI card and it works very well. Cosmin, if you want to use it for Fax traffic as well make sure you do *not* get a V-BRI card. Those will not do Fax. jens Tried almost all cards (Junghanns, Sangoma, Beronet, some hfc-based oem cards, Eicon Diva Server). Eicon is expensive but is *REALLY* worth it. The other cards are just a waste of money (even if little money). If you want a reliable PBX (who doesn't want it?), Diva Server cards are the definitive choice. The best card ever. Zero echo problems, superb hardware echo cancellation. Top reliability. Excellent FAX support with Hylafax (only cards with builtin DSPs, that is, NOT the V-series, as pointed out by Jens). Easy driver installation and powerful utilities/configuration tools. I tested BRI-2M, 4BRI-8M, PRI-30M on several installations, even older 1.0 version cards (PCI 5v only) just work great. I use diva server drivers software source rpm from Eicon, chan_capi from www.melware.org (0.7.1) on asterisk 1.2.14 (kernel 2.6.17.3). We've deployed more than 40 PBX (from 1 bri to 8 bri) without a flaw. I'm only a little bit annoyed about not being able to take advantage of the onboard DSPs to perform audio transcoding, because of the lack of a suitable asterisk driver (the cards themselves support hardware gsm/g726 codecs, for instance). Alberto. ___ --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 -- regards Martin * Visit http://pbxhardware.com for alternative T1/E1 Voice/Data cards * ___ --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] chanskype
Hi, I tried the try version of chanskype, however, everytime that I make a call asterisk generate an error So you think is easy to us guess wich error you are getting? Seriously, I think you should read this: http://www.catb.org/~esr/faqs/smart-questions.html Anyone has experience with this? Since I tried to contact the support but they never replied. Please provide more information about the error, and search in voip-info.org how to raise the verbosity level of asterisk in the console. Also sometimes help searching the error in google. Regards -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --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] error message
Recently, I got the following error messages in CLI periodically. Jan 20 17:43:18 ERROR[8641]: chan_sip.c:11002 handle_request_subscribe: Got SUBSCRIBE for extension [EMAIL PROTECTED] from 192.168.0.123, but there is no hint for that extension I have no idea what the error message tell me. I am sure I haven't that account XXX in my database and there is no hint extensions in dial plan as well. Anyone can help tell me why there is such an error message? asterisk version: 1.2.10 using realtime with mysql 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] On what distribution is www.asterisknow.com based on ?
Hello Asteriskies, Has someone tried www.asterisknow.com ? What is the package manager used? And what is the added value compared to the well maintained debian based asterisk ? Thanks, -- Cheers, Maxim Veksler Free as in Freedom - Do u GNU ? ___ --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 Suggestion for 2 PRI with call recording
We have a similar system up and runing for 6 months, wiith 60 channels, and average of simultaneas recorded calls us between 20 and 30. We make test for recording 60 calls without any problems We use a PIV Dual core with 3.2 Ghz with 2 mb of cache and 1Gb Ram. regards Mehdi On 12/13/06, A.R. Nasir Qureshi [EMAIL PROTECTED] wrote: Hi everybody, I intend to setup an Asterisk Box to handle 2 ISDN PRI (60) channels, for incoming calls. Callers will first get an IVR, and 20 of them can talk to Agents over IP Phones. The Agent Calls will be recorded. Thus at max, 20 calls will be recorded at a time. Please suggest the hardware I should use. -- Regards, Nasir. ___ --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] Disconnect Supervision UK / BT solution?
Hello, What's your zapata.conf and zaptel.conf? On 1/20/07, Matt Brown [EMAIL PROTECTED] wrote: Well, I have just phoned BT today who said they can increase the CPC value on the line - however it needs to be done at the exchange - and has been booked for Tues. I suppose I will know wether this worked on Tues :-) - I shall post my findings. Regards -- Matt Brown On 19 Jan 2007, at 16:46, Chris Earle ((CBL)) wrote: Hi all I'm using sangoma a200 cards in the UK and have the ongoing, often noted problem of disconnect supervision with BT POTS lines. Just noticed this post on http://www.voip-info.org/wiki/view/UK+Asterisk+Details stating that potentially someone's got a solution : TDM400P amp; Not Detecting Hangups: Got a TDM400P installed and having problems with Asterisk not detecting hangups? Using BT? If so, contact BT and ask what the Disconnect Clear Time setting is for your phone line. Odds are it's probably 100. Increasing it to 800 fixed the issue for me. Disconnect Clear Time is BT's name for CPC. Does anyone have any thoughts/confirmation about this finally being a viable solution? This disconnect supervision problem has plagued TDM and Sangoma cards for a long time! Comments appreciated before I get on the phone with BT -- Chris Earle System Solutions Specialist -- This message has been scanned for viruses and dangerous content and is believed to be clean. ___ --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] Cisco 7970 Unprovisioned
Are you setting the TFTP server address in the DHCP? Are you checking the TFTP log to see what files the phone is requesting and not finding? Regards Jon Jon Farmer Telford, Shropshire, UK - Original Message From: Token PBX [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Saturday, 20 January, 2007 1:01:25 PM Subject: [asterisk-users] Cisco 7970 Unprovisioned Hi! I did manage to load phone with SIP image : SIP70.8-0-3S, made SEP-MAC.cnf.xml, but phone never read the configuration from it. On the screen it's written Unprovisioned, and phone is not trying to register with asterisk. Please help!! MihaelaMJ ___ --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 ___ New Yahoo! Mail is the ultimate force in competitive emailing. Find out more at the Yahoo! Mail Championships. Plus: play games and win prizes. http://uk.rd.yahoo.com/evt=44106/*http://mail.yahoo.net/uk ___ --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] FAX using T38
Hello Asterisk implement only passtrough T.38, so you cant terminate calls with asterisk using T.38. You need T.38 gateways. Regards On 11/13/06, Ricardo Carvalho [EMAIL PROTECTED] wrote: Dear all, I'm trying to enable Asterisk to work with FAX using T38. I've tried Asterisk 1.2.4 with the available patch found at URL http://bugs.digium.com/view.php?id=5090 and also with the new 1.4 Beta3 that is announced to support it too. With both Asterisk versions, I've sent with success FAXes between two FAX machines each one attached to an ATA interface, both registered in Asterisk. Although I can't send FAXes to PSTN using T38. The problem, as far as I know, might be assigned with the Content-Length shown in the message header of every SIP message negotiating T38 parameters. I've observed that after leaving Asterisk, the Content-Length of every message carrying T38 parameters gets shorter than truly is, and contrarily to my ATAs that seem to don't care about this, my Telco analyses the packet length written in this messages and truncates them, aborting the call. Does anyone experienced this too? Any ideas besides editing the chan_sip.c code to fix this problem? Thanks, Ricardo. ___ --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] SIP registration problem w/ SBC
Hi, I'm trying to get my * server connected to a softswitch through an SBC. I get the following error when * trys to register. Got SIP response 423 Interval Too Brief back from xxx.xxx.xxx.xxx Jan 20 12:43:54 NOTICE[2138]: chan_sip.c:5473 sip_reg_timeout:-- Registration for '[EMAIL PROTECTED] ' timed out, trying again (Attempt #9) Is there something I can tweak on my end to fix this? TIA, -Tom ___ --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] FAX using T38
OpenPBX.org has better support, due to license issues and politial bullshit I don't see Asterisk getting T.38 support that isnt a joke (codec pass-thru?? LOL) for a long time. OpenPBX should have a stable release within the month, if I am not mistaken they have a Release Candiate #2 right now On 1/20/07, Mehdi chouikh [EMAIL PROTECTED] wrote: Hello Asterisk implement only passtrough T.38, so you cant terminate calls with asterisk using T.38. You need T.38 gateways. Regards On 11/13/06, Ricardo Carvalho [EMAIL PROTECTED] wrote: Dear all, I'm trying to enable Asterisk to work with FAX using T38. I've tried Asterisk 1.2.4 with the available patch found at URL http://bugs.digium.com/view.php?id=5090 and also with the new 1.4 Beta3 that is announced to support it too. With both Asterisk versions, I've sent with success FAXes between two FAX machines each one attached to an ATA interface, both registered in Asterisk. Although I can't send FAXes to PSTN using T38. The problem, as far as I know, might be assigned with the Content-Length shown in the message header of every SIP message negotiating T38 parameters. I've observed that after leaving Asterisk, the Content-Length of every message carrying T38 parameters gets shorter than truly is, and contrarily to my ATAs that seem to don't care about this, my Telco analyses the packet length written in this messages and truncates them, aborting the call. Does anyone experienced this too? Any ideas besides editing the chan_sip.c code to fix this problem? Thanks, Ricardo. ___ --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] Asterisk 1.4 and g723
What G723 codec do you have on Asterisk? What is your SIP.CONF? What ATA/Phone is being used and what are the exact settings, especially for the codec? On 1/19/07, Phil French [EMAIL PROTECTED] wrote: I am setting up Asterisk for use in a low bandwidth environment. As bandwidth is precious and our ATA's support it, the decision was made to use the g723 codec. I have been working on this for a few days and have not been successful. The issue that I am having is garbled noise at the client on calls whose RTP streams are terminated by Asterisk system. This is the case for all the dialplan applications I have tested except for Echo. The critical application for us is Voicemail. When a call to voicemail extension is initiated the Asterisk console does not indicate any error. Packet captures indicate the call is active and streaming g723 data. Everything seems well but is not. The audio at the client is unrecognizable. One thing that I have noticed is that the bitrates in the upstream and downstream direction differ. From Asterisk to ATA the g723 bitrate switches between 5.3 kb/s and 6.3 kb/s. From ATA to Asterisk the bitrate is a constant 6.3 kb/s. I don't think this is a problem but seems odd. As a comparison I captured packets from a call to the echo application and found that the bitrate was 6.3 kb/s in both upstream and downstream packets. Additionally, all prompts are g723 format. Voicemail is saved as g723sf. As a parrallel task a co-worker has gotten 1.2 to work with g723. However we require 1.4 for t.38 pass-through. The end-to-end system is illustrated below. [Asterisk] / \ ip ip / \ [PSTN]--pri--[GATEWAY]--ip--[ATA]--2pr--[Phone] System details -Asterisk server version 1.4 - compiled from source - Fedora Core 6 -Gateway - Cisco 2811 -ATA - Linksys 2102 I would appreciate any advice or suggestions. It should be noted that the calls to the PSTN through the gateway and calls between ATA's are working fine. Regards, Phil French Phil French Systems Engineer --- CapRock Communications 4400 S. Sam Houston Parkway E. Houston, Texas 77048 Office: 832 668 2643 [EMAIL PROTECTED] www.caprock.com NOTICE OF CONFIDENTIALITY: This e-mail message may contain confidential information and is intended only for the person(s) named above. Any review, use, disclosure or distribution by any other person is prohibited. If you are not the intended recipient, please contact the sender by e-mail and destroy all copies of this message. ___ --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] Cisco 7970 Unprovisioned
On 1/20/07, Jon Farmer [EMAIL PROTECTED] wrote: Are you setting the TFTP server address in the DHCP? Are you checking the TFTP log to see what files the phone is requesting and not finding? Regards Jon Jon Farmer Telford, Shropshire, UK Hi Jon! Yes I checked log, and phone requested and loaded all required files and then some: It also requested file: CTLSEP-MAC.tlv, that has something to do with license. Since it couldn't find it returned error and continued to load SEP- MAC.cnf.xml . Phone booted with SIP firmware but did not load any of the settings from SEP-MAC.cnf.xml. I checked that from phone's display. None of the settings were loaded, no sip proxy address, phone label, SIP lines etc.. All was blank. Just some dynamically assigned settings were set like DHCP address, phone's IP and such. I followed instructions from wiki voip-info when building SEP- MAC.cnf.xml. Please help and thanks. Mihaela MJ - Original Message From: Token PBX [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Saturday, 20 January, 2007 1:01:25 PM Subject: [asterisk-users] Cisco 7970 Unprovisioned Hi! I did manage to load phone with SIP image : SIP70.8-0-3S, made SEP-MAC.cnf.xml, but phone never read the configuration from it. On the screen it's written Unprovisioned, and phone is not trying to register with asterisk. Please help!! MihaelaMJ ___ --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 -- To help you stay safe and secure online, we've developed the all new Yahoo! Security Centrehttp://us.rd.yahoo.com/mail/uk/taglines/default/security_centre/*http://uk.security.yahoo.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 ___ --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] chanskype
Hi, I was wondering if someone had problems with chanskype. Since I am wondering if they are a credible company or not. See you On 1/20/07, Moises Silva [EMAIL PROTECTED] wrote: Hi, I tried the try version of chanskype, however, everytime that I make a call asterisk generate an error So you think is easy to us guess wich error you are getting? Seriously, I think you should read this: http://www.catb.org/~esr/faqs/smart-questions.html Anyone has experience with this? Since I tried to contact the support but they never replied. Please provide more information about the error, and search in voip-info.org how to raise the verbosity level of asterisk in the console. Also sometimes help searching the error in google. Regards -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --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] ISDN30 and TDM400P + FAXing ...
I have a system I'll be installing soon which has an ISDN30 (E1, UK) feed and they want to hook up a fax machine and use existing analogue conference phones (expensive polycom units) This is something I've seen and used on legacy PBXs and would seem to be a fairly standard offering, but an earlier post seemd to suggest that the cards would need some sort of timing synchronisation between them to make faxing work - can anyone confirm or deny this? Would there be any issues in the dialplan when seeing a call on the PRI DDI associated with the fax line simply Dials the corresponding Zap line that the FAX machine is connected to? Gordon ___ --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] Cisco 7970 Unprovisioned
Sounds like you need to dig into the documentation for the 7970 and perhaps even contact Cisco TAC if that doesn't help. It doesn't sound like your problem is related to Asterisk. The Cisco IP phone won't register with asterisk until it's been provisioned. Those 7900 series cisco phones are very finicky. Best of luck! From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Token PBX Sent: Saturday, January 20, 2007 6:01 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Cisco 7970 Unprovisioned Hi! I did manage to load phone with SIP image : SIP70.8-0-3S, made SEP-MAC.cnf.xml, but phone never read the configuration from it. On the screen it's written Unprovisioned, and phone is not trying to register with asterisk. Please help!! Mihaela ___ --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] Cisco 7970 Unprovisioned
you have probably something wron in config file and phone refuses to configure, here is my minimalistic file for 7941/61, you can try... device deviceProtocolSIP/deviceProtocol sshUserIdadmin/sshUserId sshPasswordadmin/sshPassword devicePool dateTimeSetting dateTemplateD-M-Y/dateTemplate timeZoneCentral Europe Standard/Daylight Time/timeZone ntps ntp namentpserver/name /ntp /ntps /dateTimeSetting callManagerGroup members member priority=0 callManager ports ethernetPhonePort2000/ethernetPhonePort sipPort5060/sipPort securedSipPort5061/securedSipPort /ports processNodeNameasteriskserver/processNodeName /callManager /member /members /callManagerGroup /devicePool sipProfile sipProxies registerWithProxytrue/registerWithProxy /sipProxies enableVadfalse/enableVad preferredCodecg729a/preferredCodec natEnabled0/natEnabled phoneLabelSIP/phoneLabel sipLines line button=1 featureID9/featureID featureLabelSIP 999/featureLabel proxyasteriskserver/proxy name999/name displayNameyourname/displayName authName999/authName authPasswordxxx/authPassword messagesNumber/messagesNumber /line line button=2 featureID21/featureID featureLabelHelpdesk/featureLabel speedDialNumber5880/speedDialNumber /line /sipLines dialTemplateDRdialplan.xml/dialTemplate /sipProfile commonProfile phonePasswordadmin/phonePassword /commonProfile loadInformationSIP41.8-2-1S/loadInformation versionStamp1143565489-a3cbf294-7526-4c29-8791-c4fce4ce4c37/versionStamp directoryURL/directoryURL servicesURL/servicesURL /device ___ --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] Disconnect Supervision UK / BT solution?
Matt Brown wrote: Well, I have just phoned BT today who said they can increase the CPC value on the line - however it needs to be done at the exchange - and has been booked for Tues. I suppose I will know wether this worked on Tues :-) - I shall post my findings. I would be keen to hear your findings - however, I'm still not clear exactly what the problem is in your case. There are numerous kinds of disconnect problems - which one are you having (so we know which one the CPC fixes...) Cheers Ed W ___ --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] func_odbc still working in trunk?
Is func_odbc still working in trunk? I've recently (in the last few weeks) started having a problem where my custom functions don't work. The module loads, the configuration file is parsed fine, and the functions are even created and visible in core show functions, but when executed from the dialplan I get a no such function error. Here's a bit of console log: [Jan 20 12:10:42] ERROR[19482]: func_odbc.c:218 acf_odbc_read: No such function 'ODBC_CIDName(314321)' And that, after... *CLI core show functions like ODBC Matching Custom Functions: ODBC_CIDName ODBC_CIDName(arg1[...[,argN]]) Runs the referenced query with the specified arguments ODBC_LookupCIDNameODBC_LookupCIDName(arg1[...[,arg Runs the referenced query with the specified arguments ODBC_TEST ODBC_TEST(arg1[...[,argN]]) Runs the referenced query with the specified arguments 3 matching custom functions installed. I'd swear I haven't changed anything in the odbc or func_odbc setup, but that just gets people in trouble. Any help, greatly appreciated. Rob ___ --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] Cisco 7970 Unprovisioned
On 1/20/07, Pavel Jezek [EMAIL PROTECTED] wrote: you have probably something wron in config file and phone refuses to configure, here is my minimalistic file for 7941/61, you can try... device deviceProtocolSIP/deviceProtocol sshUserIdadmin/sshUserId sshPasswordadmin/sshPassword devicePool dateTimeSetting dateTemplateD-M-Y/dateTemplate timeZoneCentral Europe Standard/Daylight Time/timeZone ntps ntp namentpserver/name /ntp /ntps /dateTimeSetting callManagerGroup members member priority=0 callManager ports ethernetPhonePort2000/ethernetPhonePort sipPort5060/sipPort securedSipPort5061/securedSipPort /ports processNodeNameasteriskserver/processNodeName /callManager /member /members /callManagerGroup /devicePool sipProfile sipProxies registerWithProxytrue/registerWithProxy /sipProxies enableVadfalse/enableVad preferredCodecg729a/preferredCodec natEnabled0/natEnabled phoneLabelSIP/phoneLabel sipLines line button=1 featureID9/featureID featureLabelSIP 999/featureLabel proxyasteriskserver/proxy name999/name displayNameyourname/displayName authName999/authName authPasswordxxx/authPassword messagesNumber/messagesNumber /line line button=2 featureID21/featureID featureLabelHelpdesk/featureLabel speedDialNumber5880/speedDialNumber /line /sipLines dialTemplateDRdialplan.xml/dialTemplate /sipProfile commonProfile phonePasswordadmin/phonePassword /commonProfile loadInformationSIP41.8-2-1S/loadInformation versionStamp1143565489-a3cbf294-7526-4c29-8791-c4fce4ce4c37/versionStamp directoryURL/directoryURL servicesURL/servicesURL /device ___ --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 Hi! Here's my configuration file: device xsi:type=axl:XIPPhone fullConfigtrue/fullConfig deviceProtocolSIP/deviceProtocol sshUserIduser/sshUserId sshPasswordpass/sshPassword devicePool nameDefault/name dateTimeSetting nameCMLocal/name dateTemplateD.M.Y/dateTemplate timeZoneW. Europe Standard/Daylight Time/timeZone /dateTimeSetting callManagerGroup members member priority=0 callManager ports ethernetPhonePort2000/ethernetPhonePort /ports processNodeNameMy Asterisk IP/processNodeName /callManager /member /members /callManagerGroup srstInfo nameEnable/name srstOptionEnable/srstOption userModifiabletrue/userModifiable ipAddr1My Asterisk IP/ipAddr1 port12000/port1 ipAddr2/ipAddr2 port22000/port2 ipAddr3/ipAddr3 port32000/port3 /srstInfo mlppDomainId-1/mlppDomainId mlppIndicationStatusDefault/mlppIndicationStatus preemptionDefault/preemption /devicePool commonProfile phonePassword/phonePassword backgroundImageAccesstrue/backgroundImageAccess callLogBlfEnabled2/callLogBlfEnabled /commonProfile loadInformation/loadInformation vendorConfig disableSpeakerfalse/disableSpeaker disableSpeakerAndHeadsetfalse/disableSpeakerAndHeadset forwardingDelay1/forwardingDelay pcPort0/pcPort settingsAccess1/settingsAccess garp0/garp voiceVlanAccess0/voiceVlanAccess videoCapability1/videoCapability autoSelectLineEnable0/autoSelectLineEnable webAccess1/webAccess daysDisplayNotActive1,7/daysDisplayNotActive displayOnTime08:30/displayOnTime displayOnDuration11:30/displayOnDuration displayIdleTimeout01:00/displayIdleTimeout spanToPCPort1/spanToPCPort loggingDisplay1/loggingDisplay /vendorConfig versionStamp1136931633-57191cee-5ffc-4342-b286-4246b4991890/versionStamp userLocale nameEnglish_United_States/name uid1/uid langCodeen_US/langCode version1.0.0.0-1/version winCharSetiso-8859-1/winCharSet /userLocale networkLocaleUnited_States/networkLocale networkLocaleInfo nameUnited_States/name uid64/uid version1.0.0.0-1/version /networkLocaleInfo deviceSecurityMode1/deviceSecurityMode idleTimeout120/idleTimeout authenticationURL/authenticationURL directoryURL/directoryURL idleURL/idleURL informationURL/informationURL messagesURL/messagesURL proxyServerURL/proxyServerURL servicesURL/servicesURL dscpForSCCPPhoneConfig96/dscpForSCCPPhoneConfig dscpForSCCPPhoneServices0/dscpForSCCPPhoneServices dscpForCm2Dvce96/dscpForCm2Dvce capfAuthMode0/capfAuthMode capfList capf phonePort3804/phonePort processNodeNameccm-beta-5-1/processNodeName /capf /capfList certHash/certHash encrConfigfalse/encrConfig sipProfile
Re: [asterisk-users] Question about FXO/FXS device.
Hi! I have several SPA3000 devices (older versions of SPA3102) and they are working OK, sound quality is good. It is very configurable to the slightest details. I use it whenever I need just one or two FXO ports, like for small scale PSTN integration, or for connecting some other equipment that requires FXO like GSM Gateway. Best regards Mihaela MJ On 1/17/07, Jonson Player [EMAIL PROTECTED] wrote: Hello, I intend to buy a FXO/FXS device from Linksys. I'm thinking about SPA3102. What you guys think about it. Is ok, is working with asterisk, can i use it like voip peer. Thank you for your advice. Jonson. ___ --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] extra sounds description file?
The asterisk core sounds includes a text file which gives the filename and description of what the audio file says. Is there a similar file for the extra sounds? I can't seem to find one. -A. ___ --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] 1.4 svn voicemail broken?
Ever since upgrading to 1.4 SVN, the advanced options on voicemail have disappeared. When I press 3 for advanced options, it just reviews the message. It used to present me with a menu to 1 = reply, 2 = call the person back, 3 = play message envelope. What gives? ___ --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] Attention all Aastra IP phone users...
If you own Aastra phones, here's a group dedicated to your specific needs. BTW - The Asterisk-users mailing list is great but it has way too much volume to be useful for specific problems. It needs to be broken up into smaller more manageable lists. Homepage: http://groups.google.com/group/aastra-asterisk-users Group email:[EMAIL PROTECTED] Description: Aastra-asterisk-users is a group where owners of Aastra IP telephones can discuss tips and issues with Asterisk-based PBX systems. ___ --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] ISDN30 and TDM400P + FAXing ...
Assuming your PRI supports timing from the remote end (CO) which I highly suspect is the case, then you should set the asterisk machine to be a slave to the CO timing and then set any other interfaces you have to NOT be masters, so that the CO timing is always used. Assuming you do this and disable echo cancelation then there should be no issue if the entire path is low latency (IE not over the internet, not over a VPN gateway and not over a WAN unless you have end-to-end QoS and even then you could have latency issues, of course T.38 would solve almost 100% of any issues that these would induce in your system, but Asterisk does not and I higly doubt it will support anything besides T.38 pass-through which wouldn't work with any sort of PSTN interface you could use with Asterisk) On 1/20/07, Gordon Henderson [EMAIL PROTECTED] wrote: I have a system I'll be installing soon which has an ISDN30 (E1, UK) feed and they want to hook up a fax machine and use existing analogue conference phones (expensive polycom units) This is something I've seen and used on legacy PBXs and would seem to be a fairly standard offering, but an earlier post seemd to suggest that the cards would need some sort of timing synchronisation between them to make faxing work - can anyone confirm or deny this? Would there be any issues in the dialplan when seeing a call on the PRI DDI associated with the fax line simply Dials the corresponding Zap line that the FAX machine is connected to? Gordon ___ --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] Anyone know what this warning is about? Nothing in list history about it either..
Perhaps someone could help you... if they actually had any knowledge as to what your configuration is, which I doubt they do. On 1/19/07, Eric Bishop [EMAIL PROTECTED] wrote: On inbound calls from my SIP provider I get multiple warnings as follows: WARNING[5351]: chan_sip.c:7086 check_via: 'MODH6QD' is not a valid host Everything else works but these warnings are a pain and I don't know what they are about Nothing on previos lists or Google explains... ___ --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] SIP registration problem w/ SBC
http://www.google.com/search?q=423+%22Interval+Too+Brief%22start=0ie=utf-8oe=utf-8client=firefox-arls=org.mozilla:en-US:official Hint: Who develops Asterisk? On 1/20/07, Thomas Madler [EMAIL PROTECTED] wrote: Hi, I'm trying to get my * server connected to a softswitch through an SBC. I get the following error when * trys to register. Got SIP response 423 Interval Too Brief back from xxx.xxx.xxx.xxx Jan 20 12:43:54 NOTICE[2138]: chan_sip.c:5473 sip_reg_timeout:-- Registration for '[EMAIL PROTECTED] ' timed out, trying again (Attempt #9) Is there something I can tweak on my end to fix this? TIA, -Tom ___ --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] chanskype
That's a great site! Perhaps it should be auto-sent to every poster for the first 30 days of their membership: Despite this, hackers have a reputation for meeting simple questions with what looks like hostility or arrogance. It sometimes looks like we're reflexively rude to newbies and the ignorant. But this isn't really true. What we are, unapologetically, is hostile to people who seem to be unwilling to think or to do their own homework before asking questions. People like that are time sinks — they take without giving back, and they waste time we could have spent on another question more interesting and another person more worthy of an answer. We call people like this losers (and for historical reasons we sometimes spell it lusers). On 1/20/07, Moises Silva [EMAIL PROTECTED] wrote: Hi, I tried the try version of chanskype, however, everytime that I make a call asterisk generate an error So you think is easy to us guess wich error you are getting? Seriously, I think you should read this: http://www.catb.org/~esr/faqs/smart-questions.html Anyone has experience with this? Since I tried to contact the support but they never replied. Please provide more information about the error, and search in voip-info.org how to raise the verbosity level of asterisk in the console. Also sometimes help searching the error in google. Regards -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --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] Re: One way choppy sound
I've actually found in many cases a lower bandwidth codec doesn't improve at all and however it oftentimes makes the issue worse. On 1/19/07, Martin Joseph [EMAIL PROTECTED] wrote: On 2007-01-17 10:29:43 -0800, Yelson Vivas [EMAIL PROTECTED] said: Hi Guys I'm conecting 2 astersk servers using this arquitecture (Ext softphone)==sip==(asterisk 1)iax2 trunk(asterisk 2) ===alaw==(pstn) If i call from the Ext to the asterisk 2 the sound is perfect, but if i call from Ext to the pstn, i can hear perfect but they tell me that sound really choppy, i tried using several codecs (same problem) but i don't understand why the sound is bad in only one way. Any sugestions to solve it more than welcome Usually sounds can be choppy one way due to constrained upstream bandwidth. There might be plenty of room for the audio to get to you, but that doesn't mean the reverse is at all true. Jitter buffering can help this, or using a more compact format (like GSM or g729) is also a potential helper. Good luck, hope this helps, Marty ___ --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] func_odbc still working in trunk?
On Saturday 20 January 2007 6:21 pm, Rob Fugina wrote: Is func_odbc still working in trunk? I've recently (in the last few weeks) started having a problem where my custom functions don't work. The module loads, the configuration file is parsed fine, and the functions are even created and visible in core show functions, but when executed from the dialplan I get a no such function error. Here's a bit of console log: [Jan 20 12:10:42] ERROR[19482]: func_odbc.c:218 acf_odbc_read: No such function 'ODBC_CIDName(314321)' Reported by me, actually fixed by Corydon in rev 51353. Kevin Fleming's constification went a little far. :-) -A. ___ --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 1.4 and g723
I appreciate the response. The ATA is the linksys SPA-2102 and some of its configured settings are below. After the ATA information I have included the sip.conf file and packet summary of a call with garbled audio. Regarding the G723 codec, we have compiled a g723.1 codec. This same source is working for Asterisk 1.2. ---start ata settings--- SIP Settings SIP Transport: UDP SIP Port: 5060 SIP 100REL Enable: noEXT SIP Port: Auth Resync-Reboot: yes SIP Proxy-Require: SIP Remote-Party-ID: yes SIP GUID: no SIP Debug Option:none RTP Log Intvl: 0 Restrict Source IP: noReferor Bye Delay: 4 Refer Target Bye Delay: 0 Referee Bye Delay: 0 Refer-To Target Contact: noSticky 183:no Auth INVITE: no Audio Configuration Preferred Codec: G723 Silence Supp Enable: no Use Pref Codec Only: yes Silence Threshold: medium G729a Enable:yes Echo Canc Enable:yes G723 Enable: yes Echo Canc Adapt Enable: yes G726-16 Enable: noEcho Supp Enable:yes G726-24 Enable: noFAX CED Detect Enable: yes G726-32 Enable: noFAX CNG Detect Enable: yes G726-40 Enable: yes FAX Passthru Codec: G711u DTMF Process INFO: yes FAX Codec Symmetric: yes DTMF Process AVT:yes FAX Passthru Method: None DTMF Tx Method: InBandDTMF Tx Mode: Strict FAX Process NSE: yes Hook Flash Tx Method:None FAX Disable ECAN:yes Release Unused Codec:yes FAX Enable T38: yes FAX T38 Redundancy: 0 FAX Tone Detect Mode:caller or callee ---end ata settings--- ---start sip.conf--- [general] context=default allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=yes tos_sip=cs3 tos_audio=ef tos_video=af41 minexpiry=60 t1min=1000 t38pt_udptl = yes canreinvite=yes [authentication] [pstn_gw1] type=friend host=172.17.17.20 disallow=all allow=g723 context=us qualify=500 insecure=port [sip_ata_01] type=friend secret=password host=dynamic disallow=all allow=g723 context=us dtmfmode=RFC2833 qualify=2000 [sip_ata_02] type=friend secret=password host=dynamic disallow=all allow=g723 context=us dtmfmode=RFC2833 qualify=2000 ---end sip.conf- ---start packet summary- SRCDST Protocol Info ATAAST SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session desc. ASTATA SIP Status: 407 Proxy Authentication Required ATAAST SIP Request: ACK sip:[EMAIL PROTECTED] ATAAST SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session desc. ASTATA SIP Status: 100 Trying ASTATA SIP/SDP Status: 200 OK, with session description ATAAST SIP Request: ACK sip:[EMAIL PROTECTED]:5060 ATAAST G.723.1 Payload type=ITU-T G.723, SSRC=2925445169, Seq=56662, ASTATA G.723.1 Payload type=ITU-T G.723, SSRC=1606418004, Seq=59141, ASTATA G.723.1 Payload type=ITU-T G.723, SSRC=1606418004, Seq=59142, ATAAST G.723.1 Payload type=ITU-T G.723, SSRC=2925445169, Seq=56663, ASTATA G.723.1 Payload type=ITU-T G.723, SSRC=1606418004, Seq=59143, ATAAST G.723.1 Payload type=ITU-T G.723, SSRC=2925445169, Seq=56664, AST ATA G.723.1 Payload type=ITU-T G.723, SSRC=1606418004, Seq=59144, ATA AST G.723.1 Payload type=ITU-T G.723, SSRC=2925445169, Seq=56665, ATA AST G.723.1 Payload type=ITU-T G.723, SSRC=2925445169, Seq=5, AST ATA G.723.1 Payload type=ITU-T G.723, SSRC=1606418004, Seq=59145, #truncated for the sake of brevity# ATAAST SIP Request: BYE sip:[EMAIL PROTECTED]:5060 ASTATA G.723.1 Payload type=ITU-T G.723, SSRC=1606418004, Seq=59174, ASTATA G.723.1 Payload type=ITU-T G.723, SSRC=1606418004, Seq=59175, ASTATA G.723.1 Payload type=ITU-T G.723, SSRC=1606418004, Seq=59176, ASTATA SIP Status: 200 O ---end packet summary--- Thanks, Phil Phil French Systems Engineer --- CapRock Communications 4400 S. Sam Houston Parkway E. Houston, Texas 77048 Office: 832 668 2643 [EMAIL PROTECTED] www.caprock.com NOTICE OF CONFIDENTIALITY: This e-mail message may contain confidential information and is intended only for the person(s) named above. Any review, use, disclosure or distribution by any other person is
RE: [asterisk-users] IAX call limit
Hi Philipp, Thanks for the tip, but that is not what I initially meant. I'm using IDEfisk, and I would like it when a call comes Into IDEfisk to generate a BUSY signal, if there is already a call in the client. Any ideas ? Nir S -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philipp Kempgen Sent: Thursday, January 18, 2007 12:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] IAX call limit Nir Simionovich wrote: Stupid and silly question - is there a way to limit the number of concurrent calls an IAX client can make? something in the similar sense of incominglimit and outgoing limit on SIP? It can be done in the dial plan: http://www.voip-info.org/wiki/view/Asterisk+Queue+with+limited+calls+per+I AX+agent Best regards, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk - http://www.das-asterisk-buch.de ___ --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: Fwd: [asterisk-users] Some queries on g729 license.
Leo Ann Boon wrote: Andrew Joakimsen wrote: Most of the Cisco phones sold cheap are UNLICENSED (global spare) thus you would not be able to purchase (or at least aren't supposed to) the smartnet contracts, you need to buy the license ($100+) and the contract ($10 or so) I'm always surprised by by the number of people who don't read the fine print :). Even if you have a new licensed unit, it's only licensed to run Skinny out of the box. SIP requires additional licensing. Back to the G.729A licensing, I just received a new 'low-volume' quote from Sipro. For 1,000 channels - it's US$6 per channel (US$4 for 5,000) just for the right to use G.729A. You'll still have to fork out money to separately licensed a working codec - unless you're happy with the suboptimal ITU implementation or Intel's IPP sample. One vendor we spoke to asked for US$2,000/year to license their G.729A implementation on top of the Sipro licensing. That works out to a total of US$8/channel if we use 1,000 channel in a year. Throw in the cost of license administration (because Sipro will require audit), the US$10 charged by Digium looks very reasonable. That is only for Sipro's part. You also need to pay Nokia and NEC (maybe others). That should be a similar sized bill to the Sipro one. 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] Announce option for meetme - is it used?
BerkHolz, Steven wrote: Announce option for meetme - is it used? It makes a caller record their name, but I do not see where this name recording is ever used. Thank You, Steven BerkHolz - MCSA - MCSE - Manager of Information Systems HIROTEC AMERICA Fax. 248-836-5101 www.hirotecamerica.com Board member of www.glimasoutheast.org Our company name has changed to HIROTEC AMERICA www.hirotecamerica.com Please update any contact info with my new email address [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 It used when you have enabled the option to announce joining new participants your meetme room. It would play recorded name to others who are connected to that room. ___ --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] IAX call limit
IDEfisk is based on iaxclient (iaxclient.sourceforge.net) which will reject a call with the BUSY signal if there is no available line in the softphone to take the call. This means you need to configure IDEfisk to use only one line (call context). I don't know if this is possible. Somewhere in IDEfisk is this call that initialised iax client: iaxclient.h:EXPORT int iaxc_initialize(int audType, int nCalls); what you want is nCalls to be 1. Hope this helps, Cristi -- Cristian Draghici http://www.loudhush.ro On 1/21/07, Nir Simionovich [EMAIL PROTECTED] wrote: Hi Philipp, Thanks for the tip, but that is not what I initially meant. I'm using IDEfisk, and I would like it when a call comes Into IDEfisk to generate a BUSY signal, if there is already a call in the client. Any ideas ? Nir S -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philipp Kempgen Sent: Thursday, January 18, 2007 12:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] IAX call limit Nir Simionovich wrote: Stupid and silly question - is there a way to limit the number of concurrent calls an IAX client can make? something in the similar sense of incominglimit and outgoing limit on SIP? It can be done in the dial plan: http://www.voip-info.org/wiki/view/Asterisk+Queue+with+limited+calls+per+IAX+agent Best regards, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk - http://www.das-asterisk-buch.de ___ --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] Connect a Skype adapter to TDM400P
Hi, I was wondering if it is possible to connect a skype phone adapter, for example: http://zonetusa.com/DispProduct.asp?ProductID=191 http://www.actiontec.com/products/communications/ipw_usb/index.php http://www.eradian.com/ERadianUS/staticpages/SkytoneRST301Details.htm http://www.dlink.com/products/?pid=466 http://www.usr.com/products/voip/voip-product.asp?sku=USR9620 to a TDM400P, so that the Line port of the adapter connects to the FXS port of the TDM400P and the Phone port of the adapter connects to the FXO port of the TDM400P Appreciate any help Regards. Be a PS3 game guru. Get your game face on with the latest PS3 news and previews at Yahoo! Games. http://videogames.yahoo.com/platform?platform=120121 ___ --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] NAT solutions
Some NAT problems you can solve, some you never will. Many modern phones have NAT support in them, via STUN, or a static external IP address. Most NATs also offer port forwarding, so you can open a hole for the SIP port in the NAT so all outside can reach it. (With port forwarding, you need a constant address for each SIP phone, so that means either static IP for the phone, or a DHCP server with the ability to always bind a device to the same address - the latter is preferable because you can move your phone to other networks more easily.) Many devices also feature NAT keep alive on the SIP port. That is a must if you can't open ports, but it sure generates a lot of annoying debug output when you turn on sip debug. Nothing beats a permanent NAT entry point though. Some devices, notably Ciscos, just don't support NAT as well. They don't have STUN, and while they may have a static external IP mapping, that's no good if your NAT itself has a dynamic address, as most home broadband NATs do. Asterisk, if you set nat=yes (or often even without that) will take incoming packets from a natted phone, and look at the incoming address, and send back to it regardless of what the phone says in its SIP headers. That's handy, but unfortunately it does not do the same thing for the SDP, so if the phone hands out an SDP with an unreachable address, Asterisk handles it badly. Some SIP gateways are smarter, and if they see an unreachable address in the SDP, ignore it and send to whatever address they get incoming RTP from. You'll have better luck connecting to such endpoints. Many termination providers do this, so you may find your phones can talk to the term provider, but not to other phones on the same * box. Many consumer nats will not hairpin audio. That means if you do all this work to rewrite the addresses in your SIP headers/SDP via STUN so you look like an externally routable device, and Asterisk hooks you up with another device behind your same NAT, you will get one way audio. I get this problem -- I have a * box at one location, with most of the phones (no problem for those) and some other phones at another location behind NAT. These phones can talk to the main location, but not to one another, due to the hairpin. What fun. A new method, called ICE, was drafted a while ago but is getting slow adoption. In ICE, devices are given a list of possible ways they could reach one another (directly, through nats, via RTP forwarders etc.) They try them all and pick the best. In the end it will always work through the RTP forwarders, but that costs bandwidth and latency. So far, however, support is limited. ___ --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
