Re: [asterisk-users] Sipura ATA's Forwarding PSTN Calls to Asterisk
For the OP, do you have an entry against "Display Name" on the PSTN tab, whilst logged in as admin/advanced? If I have an entry in this, what you describe happens for me. If the field is empty, CLID is sent correctly to my Asterisk box. On 21/07/06, Rich Adamson <[EMAIL PROTECTED]> wrote: I just ran into a problem with the spa3k and spa942's that I could not diagnose. It "appears" as though the sipura boxes have a problem with calls that include a CallerID with "-" in it. I can't say with 100% certainty yet, but that's my story and I'm sticking to it (for now). ;) Douglas Garstang wrote: >> -Original Message- >> From: Brian Capouch [mailto:[EMAIL PROTECTED] >> Sent: Friday, July 21, 2006 11:20 AM >> To: Asterisk Users Mailing List - Non-Commercial Discussion >> Subject: Re: [asterisk-users] Sipura ATA's Forwarding PSTN Calls to >> Asterisk >> >> >> Douglas Garstang wrote: >>> I'm working with a Sipura 3000 ATA here. I'm trying to get >> incoming PSTN calls on the FXO port to go automatically to >> Asterisk. I have it working, but I had to configure the ATA >> to register with Asterisk, which means that all calls are >> being sent to Asterisk with a caller id of the username used >> to register with Asterisk. >>> I want the real caller ID to be sent to Asterisk, which >> means I don't want the ATA to register. The badly written >> Sipura docs aren't clear about how to do this. Anyone set this up? >> That's not correct. >> >> My SPA-3000 FXO port registers with my Asterisk server, and when the >> PSTN calls come in, it uses the incoming caller's CallerID >> for the call. >> >> Sounds like you have something misconfigured. > > Here's my invite Brian. The From: is always going to contain the auth id the ATA used to register with Asterisk. > > INVITE sip:[EMAIL PROTECTED] SIP/2.0 > Via: SIP/2.0/UDP xxx.187.142.203:5060;branch=z9hG4bK208dd2be;rport > From: "Cody XXX-527-7107" ;tag=as3a94778b > To: > Contact: > Call-ID: [EMAIL PROTECTED] > CSeq: 102 INVITE > User-Agent: Asterisk PBX > Max-Forwards: 70 > Remote-Party-ID: "Cody XXX-527-7107" ;privacy=off;screen=no > Date: Fri, 21 Jul 2006 17:44:20 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Content-Type: application/sdp > Content-Length: 269 > > v=0 > o=root 28771 28771 IN IP4 xxx.187.142.203 > s=session > c=IN IP4 xxx.187.142.203 > t=0 0 > m=audio 21652 RTP/AVP 0 18 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > ___ > --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] Can I get caller id passed to a phone connected to a Supura 2100?
Does the Sipura web interface on the info page reveal that the spa2100 is successfully receiving CLID? My SPA2100 passes CLID from asterisk to the connected phone without problem. On 23/06/06, Jim Lynch <[EMAIL PROTECTED]> wrote: I have a Uniden wireless phone connected into Linksys/Supura 2100. It works well, except I never see any caller ID information displayed on the phone. Is that a setting in the 2100 that I'm missing, or is it an Asterisk setting or isn't it possible? Thanks, Jim. ___ --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] No incoming sip calls
Could your register line require attention ? (2001?)
7960xxx:[EMAIL PROTECTED]/2001 - I thought your target was 2201?
On 13/06/06, Russell Horn <[EMAIL PROTECTED]> wrote:
Hi folks - I've recently returned to asterisk after an eighteen month break.
I've two sip providers - gradwell in the UK (inbound and outbound)
and talklite in the US (outbound only).
I've managed to get outbound dialing working but am not receiving any
calls from gradwell.
I've included my sip.conf and extensions.conf as well as the output
from tethereal. When a call is placed to rgadwell I'm seeing no sip
traffic whatsoever on asterisk. My aim is to have inbound calls ring
SIP extension 2201
I'm guessing this is something pretty straightforward, but any help
would be much appreciated.
Thanks,
Russell.
sip.conf
[general]
context=incoming; Default context for incoming calls
register => 7960xxx:[EMAIL PROTECTED]/2001
register => 9479xxx:[EMAIL PROTECTED]
port=5060 ; UDP Port to bind to (SIP standard
port is 5060)
bindaddr=0.0.0.0; address to bind to (0.0.0.0 binds to all)
nat=yes ; NAT settings
allow=all
[Gradwell]
type=peer
username=796
fromuser=796
secret=
host=sip.gradwell.net
context=flat
fromdomain=sip.gradwell.net
nat=yes
allow=all
canreinvite=no
dtmfmode=inband
qualify=yes
[talklite]
type=peer
username=9479
qualify=yes
secret=
host=sip.talklite.net
canreinvite=yes
disallow=all
allow=ulaw
[2201]
type=friend
context=flat
username=albanach
secret=
defaultip=192.168.1.100
qualify=yes
type=friend
callerid="Russell Horn" <>
host=dynamic
nat=no ; X-Lite is behind a NAT router
canreinvite=yes ; Typically set to NO if behind NAT
allow=all
=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-
extensions.conf
[general]
static=yes
writeprotect=no
[globals]
TRUNK=Gradwell
TRUNKMSD=1 ; MSD digits to strip
(usually 1 or 0)
PHONES1=SIP/2201
[flat]
include => home
include => outgoing
[home]
exten => 2201,1,Dial(${PHONES1},20,Ttm)
exten => 2201,2,Macro(vmessage,${PHONES1VM})
exten => 2201,3,Hangup
[outgoing]
ignorepat => 9
ignorepat => 8
exten => _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten => _8.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-
linux:/etc/asterisk # tethereal -R "sip"
Capturing on eth0
0.00 207.44.248.78 -> 192.168.1.102 SIP Request: OPTIONS
sip:[EMAIL PROTECTED]
0.000831 192.168.1.102 -> 207.44.248.78 SIP Status: 404 Not Found
1.350584 192.168.1.102 -> 192.168.1.100 SIP Request: OPTIONS
sip:[EMAIL PROTECTED]:5060
1.350730 192.168.1.102 -> 207.44.248.78 SIP Request: OPTIONS
sip:sip.talklite.net
1.350887 192.168.1.102 -> 193.111.200.56 SIP Request: OPTIONS
sip:sip.gradwell.net
1.369388 192.168.1.100 -> 192.168.1.102 SIP Status: 200 OK
1.455492 207.44.248.78 -> 192.168.1.102 SIP Status: 404 Not Found
1.502618 193.111.200.56 -> 192.168.1.102 SIP Status: 404 Invalid
account for voicemail
1.552845 192.168.1.102 -> 207.44.248.78 SIP Request: REGISTER
sip:sip.talklite.net
1.654933 207.44.248.78 -> 192.168.1.102 SIP Status: 100 Trying(1 bindings)
1.655832 192.168.1.102 -> 193.111.200.56 SIP Request: REGISTER
sip:sip.gradwell.net
1.657951 207.44.248.78 -> 192.168.1.102 SIP Status: 401 Unauthorized
(1 bindings)
1.658229 192.168.1.102 -> 207.44.248.78 SIP Request: REGISTER
sip:sip.talklite.net
1.770875 207.44.248.78 -> 192.168.1.102 SIP Status: 100 Trying(1 bindings)
1.773894 207.44.248.78 -> 192.168.1.102 SIP Status: 200 OK(1 bindings)
1.792718 193.111.200.56 -> 192.168.1.102 SIP Status: 401
Unauthorized(0 bindings)
1.793529 192.168.1.102 -> 193.111.200.56 SIP Request: REGISTER
sip:sip.gradwell.net
1.937253 193.111.200.56 -> 192.168.1.102 SIP Status: 200 OK(1 bindings)
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Re: [Asterisk-Users] How can I use my regular phones with Asterisk running on my Linksys WRT54G router?
have a look here <http://voxilla.com/name-News-article-sid-12.html> for a decent write up, from a Noth American perspective, about wirng VoIP into your home. Beware, the pages at voxilla can take a while to load On 12/06/06, John Klimek <[EMAIL PROTECTED]> wrote: Ahh, thanks! That's what I thought but I wasn't sure because I thought ATA boxes were only for specific VOIP providers. Which ATA with an FXS port would you recommend for around (or under) $50? Also, can I simple plug the ATA into an existing RJ-11 jack so that ALL of the phone jacks in my house have a dial tone? On 6/12/06, Jonathan Attwood <[EMAIL PROTECTED]> wrote: > Analogue Telephone Adapter(s) > Linksys/Sipura range SPA-3102; SPA2100; SPA1001 to name but 3 > > On 12/06/06, John Klimek <[EMAIL PROTECTED]> wrote: > > Ok, I've done some more research and I don't think I want an FXO box... > > > > What I'd like to do is use BroadVoice (with their BYOD plan) and then > > run Asterisk on my WRT54G router. I'd also like to use my regular home > > phones without having to use a special "SIP" phone... (eg. I like my > > Vtech normal cordless phones) > > > > What do I need to buy to get this working? It sounds like I need to > > purchase a "Zaptel" interface card, but of course I can't use those > > with a router... > > ___ > > --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 can I use my regular phones with Asterisk running on my Linksys WRT54G router?
Analogue Telephone Adapter(s) Linksys/Sipura range SPA-3102; SPA2100; SPA1001 to name but 3 On 12/06/06, John Klimek <[EMAIL PROTECTED]> wrote: Ok, I've done some more research and I don't think I want an FXO box... What I'd like to do is use BroadVoice (with their BYOD plan) and then run Asterisk on my WRT54G router. I'd also like to use my regular home phones without having to use a special "SIP" phone... (eg. I like my Vtech normal cordless phones) What do I need to buy to get this working? It sounds like I need to purchase a "Zaptel" interface card, but of course I can't use those with a router... ___ --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: DISA & SPA3000 issues
INFO is the way to go for DTMF at least on the PSTN tab of your SPA3K I have dtmfmode=auto in sip.conf & I use DISA daily On 5/17/06, Philippe Lindheimer <[EMAIL PROTECTED]> wrote: Just tried it on mine, worked fine: Cellphone Call -> POTS -> SPA3000 -> Asterisk -> DISA -> Telasip As an FYI, I have my SPA3000 setup with INFO for the DTMF. When I originally installed it, I couldn't get the DTMF digits to work coming in using AUTO, which is why I have it using INFO (needs to be set on both the SPA and in Asterisk). I'm running the SPA300 with Software Version: 2.0.13(GWg), Hardware Version 2.0.1(4e16). philippe > > From: Dave Hawkes <[EMAIL PROTECTED]> > To: asterisk-users@lists.digium.com > Date: Wed, 17 May 2006 13:44:43 -0400 > Subject: [Asterisk-Users] Re: DISA & SPA3000 issues I have this exact same issue with the SPA3000, I'm assuming it must be a SPA3000 bug? Dave Hawkes Alchaemist wrote: > Hi, > > These days I run into something quite odd. > I have an [EMAIL PROTECTED] that was modified to meet our requirements. > We have a completely funtional DISA which we use pretty much all the > time. > I works flawlessly with incomming SIP calls from several providers, > IAX calls from FWD and with ZAP. > > Recently we came out with a situation where it doesn't work... with > a SPA3000 PSTN Line. > You can call, navigate de IVR, log in into our app, and then when > you go to DISA, and DISA plays the dialtone... whatever you dial is not > recognized... > > This was REALLY odd... so I made a network capture with Ethereal, > and... the SPA actually STOPS sending the RTP Events after the second > dialtone... > > To verify this, I created an IVR which played the dialtone, and > verified that it was true no RTP DTMF events (RFC2833) are sent after > the SPA listens the second dialtone. > > I just reviewed the 87 pages PDF of the SPA3000... and didn't find > anything about such "feature". > Now I am going to try to figure out if it has something to do with > the tones recognition of the SPA. > I the meanwhile I had to write a little DISA-like app, based on > something I found on this forum, without the dialtone. > > Did anyone find out anything about this issue before? > > REGARDS!!! > Alchaemist > > > > > > ___ > --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] Need to Install Fax to Email feature
have a look here: http://nerdvittles.com/index.php?p=88 On 4/4/06, Wasif <[EMAIL PROTECTED]> wrote: > Hi, > > I need to receive FAX over DID and forward that FAX in email to particular > person. I read some articles about www.voip-info.org but I am confused in > HylaFax, IAXmodem & spandsp. > > Can anyone guide me what is what and how can I achieve my goal. > > 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 > ___ --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 Out IVR
http://nerdvittles.com/index.php?p=122 On 3/10/06, Sharath Chandra <[EMAIL PROTECTED]> wrote: > > How can i configure the following scenario, > > - User 'A' dials into Asterisk, > - Asterisk puts user 'A' on hold > - Dials Out to User 'B' > - Consults user B' if he wants to take the call (Press 1) or divert to > voicemail (press 2) > - Depending on the option chosen, either user A' call is bridged with the > out call or transfered to voicemail. > > Thanks, > Sharath Chandra > ___ > --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] UK Provider
On 1/24/06, Peter Bowyer <[EMAIL PROTECTED]> wrote: > On 24/01/06, scott <[EMAIL PROTECTED]> wrote: > > Hi > > > > Does anyone know a UK Voip Proivder that will give me more than 1 telephone > > number and point it to my sip account. > > > > www.SipGate.co.uk are great but they only allow 1 telephone number per > > user, you can register another telephone number by registering as another > > user but Asterisk doesn't allow multiple registrations. > > Which part of Asterisk? > > register => nn:[EMAIL PROTECTED]/m > register => oo:[EMAIL PROTECTED]/ > > Works fine for me > > Peter > -- > Peter Bowyer > Email: [EMAIL PROTECTED] > Tel: +44 1296 768003 > VoIP: sip:[EMAIL PROTECTED] > VoIP: [EMAIL PROTECTED] > FWD: **275*5048707000 > VoipTalk: **473*5048707000 > ___ > --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 > I too have more than one Sipgate number - they all coexist perfectly well. However, to answer the OP's question. www.voip.co.uk would also be worth a look. ___ --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] Draytek Vigor 2900 & Asterisk
It certainly does. How many rules can you create in the port forwarding section of the V2900? I was told that the V2900 has SIP_ALG. Is this something you've activated? On 1/7/06, Faris Raouf <[EMAIL PROTECTED]> wrote: > Jonathan Attwood wrote: > > I'm in conversation with Draytek's pre-sales dept.. > > > > Here's the most recent reply: > > > > > > > We really don't know of anyone who has run an Asterisk server on > > a Vigor2900. There are doubtless people around, but it's relatively > > rare. Most people don't run SIP servers. > > > > Regards,> > > > > All I want to know is, if I buy one of these routers, will it break my setup > > or not - ie. assuming I set up the relevant port-forwarding, can I > > expect any one-way audio issues. Can't get a definitive answer from > > suppliers or the manufacturer, so I hope someone here uses this model > > with Asterisk.? > > ___ > > I have a 2900G behind a Cisco 1720 with dual ADSL WICs in one office, > and a 2600VGi standalone in another (I don't use the 2600's built-in FXS > ports -- they aren't very good - seem noisy). > > I have Asterisk servers in both offices, linked via IAX. I have incoming > voip services going independently to both Asterisk servers. > > I've had no problems whatsoever -- everything has worked perfectly. The > QoS facility in both routers allows you to reserve a certain amount of > bandwidth (in or out) for IAX and SIP and this seems to work fine though > it isn't necessary on our networks. > > I'm using port forwarding on both routers to route IAX and SIP to the > private IPs of the Asterisk boxes. > > But you will need to open the appropriate ports on the firewall in the > router, or firewall the Asterisk boxes and DMZ the Asterisk boxes. > > However, the new Dreytek 3300 series of routers is even more > interesting. Multiple WAN ports for backup/load balancing, and optional > hardware FXO/FXS ports. > > I hope this helps. > > Faris. > > ___ > --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] Draytek Vigor 2900 & Asterisk
I'm in conversation with Draytek's pre-sales dept.. Here's the most recent reply: All I want to know is, if I buy one of these routers, will it break my setup or not - ie. assuming I set up the relevant port-forwarding, can I expect any one-way audio issues. Can't get a definitive answer from suppliers or the manufacturer, so I hope someone here uses this model with Asterisk.? ___ --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: Cell phone dock/switch as Asterisk FXO source
On 1/5/06, Brian McEntire <[EMAIL PROTECTED]> wrote: > Wow! Thanks for all the responses! Very informative. > > Erik: I'm just looking for simple dial-out and pass-along incoming cell > calls to *. Looks like the doc-n-talk should do it, except I checked with > them and, silly me, the new Samsung t309 phone I just got is not supported > yet. Hopefully it will be in a few months. > Is it not supported, even with the Bluetooth module? (That's assuming the phone's BTth) ___ --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] Announcing a call transfer
. > > > There is a feature called attended transfer which does what you want. > Receptionist dials the attended transfer code, followed by your > extension. The caller hears hold music while the receptionist announces > the call to you. When she hangs up you get the call. If you hang up > before she does, the call goes back to her. > > It can be enabled in the features.conf file. Under the [featuremap] > section add > atxfer => code > on my system it's > atxfer =>*2 > so I dial *2 followed by the extension to do attended transfer. > > However, I don't know anything specific to [EMAIL PROTECTED], so if it's > different than a stock asterisk setup then I don't know. It's called "Call Transfer - Managed" in [EMAIL PROTECTED] All explained here: http://members.optusnet.com.au/~bsharif/asterisk/AsteriskDumbMeGuide.htm#_Toc123532542 I use this feature if I'm on a phone without a flash key (softphone) however, on phones with a flash/recall key, I can use flash in the same way as the OP does on his PBX. Watch out for wrap. ___ --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] GSM Gateway / Terminal for sale
Couple of ways I use mine: My mobile operator has just started charging for calls to Freephone numbers. Therefore, I call into my GSM terminal, free of charge then hop back out on VoIP or PSTN to make the Freephone call. My wife has a pay-as-you-go mobile. She can ring a DID on my Asterisk, which will never answer, so costs her (me) nothing. Then Asterisk checks the CLI. If it's a CLI from an allowed list, Asterisk will call her back, using the GSM terminal, out of my inclusive minutes & give her a dial tone. She can then ring anywhere she likes, as if she were still at home. On 1/6/06, JCC <[EMAIL PROTECTED]> wrote: > I don't get it. What is the advantage of using a GSM gateway? VOIP calls are > pretty inexpensive as they are now. Is the use of a gateway intended as a > backup incase a wired network connection goes down? I have being looking > around the net for information on this. Anyone out there using it and if so > you can please share with me how you use this technology? Any information > will be appreciated. > > Thanks, > > Jay > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Chris Mason > (Lists) > Sent: Friday, January 06, 2006 5:52 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] GSM Gateway / Terminal for sale > > Remco Barende wrote: > > Not really, their suggested retail price is USD 300 for the analog > > unit, probably because of the intelligent stuff in the box (which we > > do not need when using *). > > > > At USD 300 you can find SIP capable devices, for an analog unit the > > SIPCE is 3x more expensive than the unit we were discussing. > > > Where can I find the $300 SIP capable units? > > -- > Chris Mason > NetConcepts > (264) 497-5670 Fax: (264) 497-8463 > Int: (305) 704-7249 Fax: (815)301-9759 > Cell: 264-235-5670 > Yahoo IM: [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] Dialer
Here, this may be of use: http://mundy.org/blog/index.php?p=95 On 1/6/06, Wiley Siler <[EMAIL PROTECTED]> wrote: If this or any other example is available, I would be most thankful tohave it.I got the go ahead on this project to day so now I have to start seeing how to do this.Thanks,Wiley-Original Message-From: [EMAIL PROTECTED][mailto: [EMAIL PROTECTED]] On Behalf Of DarrenWiebeSent: Tuesday, January 03, 2006 5:00 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Dialer I'm supposed to have a "mostly" canned script that will do this donealready. It will pull the list of people to call out of a db and playthem the file specified in the db table. Contact me offlist if you're interested. It will be done real soon but I'm not done testing yet.Darren Wiebe[EMAIL PROTECTED]Kerry Garrison wrote:> You actually aren't far from it. If the system only needs to play the > same file to each person, a simple script can be used to pull from a> database and create call files. Asterisk will use the call files to> place the calls and play a sound. A few minutes of searching on that > should get you started. I haven't seen anyone else have a canned> script ready to go, but would like to know if anyone does.> -Kerry>>> > *From:* [EMAIL PROTECTED]> [mailto:[EMAIL PROTECTED] ] *On Behalf Of> *Wiley Siler> *Sent:* Tuesday, January 03, 2006 3:32 PM> *To:* Asterisk Users Mailing List - Non-Commercial Discussion> *Subject:* [Asterisk-Users] Dialer >> Hello All,>> I am having trouble finding a specific * piece of software so I> thought I would see If you guys can help me get my terminologyclear.>> First off let me premise this with "no, this is absolutely not for > doing call marketing".> I need to make my Asterisk box call a group of people and play> them a message.> My company deals with education so we need to do follow ups if > students are not logging on.> We do this manually now but it would be easier and cheaper to just> play them a message.>> What is the term I should be looking for? I keep thinking "auto > dialer" or something like that but I am not quite getting there.>> Any help would be appreciated. I have been learning a bit of Perl> so I was thinking I could auto generate and AGI file and then just > do a Play() of the mp3 when they pick up at the other end? Seems> a little kludge though.>>> Thanks,> Wiley>>>--- >->>___>--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>>--Darren Wiebe [EMAIL PROTECTED]Aleph CommunicationsASTPP - Open Source Voip Billing & Calling Cards www.aleph-com.net/astpp___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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] Cell phone dock/switch as Asterisk FXO source
The unit dials whatever asterisk tells it to, although it seems to take a second or two for the mobile to start dialling. It passes the Caller Name from the cellular phone's directory, together with the CLID on inbound calls. On 1/3/06, Paul Dugas <[EMAIL PROTECTED]> wrote: On Mon, 2006-01-02 at 16:06 +0000, Jonathan Attwood wrote:> I use a Dock-n-Talk in conjuction with a Sipura SPA3000 & Asterisk. Does this unit require any funky dialing when placing outbound callsfrom * through the phone? Do the docs indicate operation is anydifferent between CDMA, TDMS, AMPS, or GSM phones? I'd guess not or, if so, it was simple to handle it in the dialplan but I'm curious anyway.I've been considering this as a way to have "work" calls that come to mycell appear different to the server. At the moment, I have my GSM phone forward calls to the house when it's off so I can't really tell betweenthem.--Paul Dugas, Computer EngineerDugas Enterprises, LLC[EMAIL PROTECTED]phone: 404-932-1355 522 Black Canyon Park http://dugas.ccfax: 866-751-6494 Canton, GA 30114 USA--On site at GDOT's W.Annex, 404-463-2860 x199--This e-mail and any attachments are confidential. If you receive this message in error or are not the intended recipient, you shouldnot retain, distribute, disclose or use any of this information andyou should destroy the e-mail and any attachments or copies.___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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] Cell phone dock/switch as Asterisk FXO source
I use a Dock-n-Talk in conjuction with a Sipura SPA3000 & Asterisk. Because I'm using Asterisk, I cannot use voice dialling, however inbound & outbound calls work extremely well. I have Asterisk outbound routes set up to make a calls to cell phones go through the Dock-n-Talk. On 1/1/06, Brian McEntire <[EMAIL PROTECTED]> wrote: Is anyone familiar with cell phone switches that allow routing cell phone calls through in-home wiring? One example of these devices is the Phone Labs Dock-N-Talk. It says it keeps your cell charged when you are home and connects your cell (for incoming and outgoing calls) to your home wiring or cordless phones. But it also has features such as allowing speed dialing and voice dialing from extensions if your cell phone has those features. So I'm not sure if the device offers a fully compatible FXO signalling.I'm currently running Asterisk with 1 POTS and 1 VOIP (via Sipura 3000) lines coming into Zaptel FXS modules, and then I have two FXO modules for two extensions. I'm thinking of doing away with the land line. Should something like the Dock-N-Talk allow substituting a cell phone line for the POTS line?___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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] UK, Disconnect supervision
Peter, I'm using the firmware 3.1.5(GWb) and was wondering if your suggestions would be of any benefit to me. Incidentally, I've never had an issue upgrading or downgrading the firmware in 2 spa-3000s, I just had to make sure the unit had only just been powered up when initiating the upgrade. (YMMV) Anyway, if you're wanting somewhere else to read & ask questions have a look at http://voxilla.com/forum-viewforum-f-14.html for Sipura/Linksys adapters or http://voxilla.com/PNphpBB2-viewforum-f-17.html for asterisk. Fantastic resources with helpful & knowledgeable respondents. Peter Hoppe wrote: || Jonathan, || || many thanks for your reply. The adapter has firmware version || 3.1.3(GWa). || || Does that version have problems with diconnect tones? Would you || recommend I should upgrade? Can you give me some reasons or point me to || resources (apart from google) where I can research further? What would || you say are the risks of upgrading? || || I am usually a bit anxious about firmware upgrade because I have that || fixed idea that EITHER the new firmware may break other features || (like - registration problems with SIP provider, connectivity issues || and so on) OR there may be some problem during firmware upgrade || which damages the device in question. For example, for the Grandstream || Budgetone 100 phone, power outage during firmware upgrade from TFTP || will damage the device(1). And I can't fix it once it's broken; it's || not like a computer where I simply reinstall the OS / put in a new || component etc. Once it's gone, it's gone. || My fears are probably totally unfounded, but better safe than sorry. || So I wouldn't upgrade unless there are good reasons to do so (if it || ain't broke, don't fix it). || || But thanks very much for that hint. I actually have two other || adapters, and they may be way out of date: 2.0.13(GWg) - so they may || really need updating. || || Peter || || -- || (1)BudgeTone-100 User Manual, version 1.0.5.11, section 6.1: || "Upgrade with TFTP", warning: The device WILL get damaged if there is a || power outage during firmware upgrade. Grandstream STRONGLY recommend || customer maintain UNINTERRUPTED POWER SUPPLY during firmware upgrade. || This damage is NOT covered by the manufacture warranty. Grandstream || will NOT take any responsibility for this kind of damage. Please be || very CAREFUL when doing firmware upgrade. || || || ||| Which firmware version are you using on your spa3000? ||| ||| Peter Hoppe wrote: | Hello! | | This is actually less a question than some information, if anyone | else struggles with the same issue. | | I am located in the UK and use a Sipura-3000 adapter to connect to | a BT line (via fxo port). One problem I had was that disconnect | supervision didn't work: | | Some caller phones me (my adapter) | adapter goes off-hook (answers call) | caller hangs up | adapter doesn't realize and stays off hook. || || || -- || dyslexics of the world - untie ! || ___ || --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] UK, Disconnect supervision
Which firmware version are you using on your spa3000? Peter Hoppe wrote: || Hello! || || This is actually less a question than some information, if anyone else || struggles with the same issue. || || I am located in the UK and use a Sipura-3000 adapter to connect to a BT || line (via fxo port). One problem I had was that disconnect supervision || didn't work: || || Some caller phones me (my adapter) || adapter goes off-hook (answers call) || caller hangs up || adapter doesn't realize and stays off hook. || || I researched into it and found that the BT exchange delivers a CPC (0.1 || sec) upon the caller's hangup and a disconnect tone about 3 seconds || later. The tone lasts 6 seconds and remains constant during that time. || I recorded the disconnect tone with Cool Edit and did a frequency || analysis on it and got the following components: || || 400Hz/-56dB + || 1200Hz/-69dB + || 2000Hz/-65dB + || 2800Hz/-59dB + || 3600Hz/-59dB || || If normalized with -56 dB reference level, I get || || 400Hz/0dB + || 1200Hz/-13dB + || 2000Hz/-9dB + || 2800Hz/-3dB + || 3600Hz/-3dB || || I suspect that the harmonics (1200, 2000, 2800, 3600)Hz may not have || come from the Exchange, but were distortions. || || After the 6 seconds the tone stopped and there was succession of two || 'clicks' on the line (another CPC? haven't looked into that); both were || 0.3 secs apart. || || I tried to use both, the CPC and the disconnect tone in the Sipura-3000 || settings. Unfortunately I couldn't get disconnect supervision via the || disconnect tone to work. In a post on the voxilla forum || || http://voxilla.com/PNphpBB2-viewtopic-t-2904.html || || I found some further info on disconnect tone in the UK - unfortunately || I couldn't get those settings to work. I also looked into BT's SIN || notes (on http://www.sinet.bt.com , notes 350, 351) but failed to see || further information on what tone they exactly deliver for a disconnect || event. || || However, the CPC did work. I set the adapter to a CPC minimum time of || 0.05 seconds, and from then on it recognized remote disconnection in || every test phone call. If I set the time too short I got problems - for || incoming calls the adapter started to mistake the incoming ringing for || disconnect events and simply wouldn't go off hook anymore. || || Now, these are the settings I configured my adapter with: || || Settings: || spa-3000-setup-web-page/PSTN-line-tab/PSTN-disconnect-detection-section: || || 'Detect CPC' => 'yes' || 'Min CPC Duration' => '0.05' || || I also have || 'Detect Disconnect Tone' => 'no' [as it didn't work] || 'Disconnect Tone' => '[EMAIL PROTECTED],[EMAIL PROTECTED];5(5/5/1+2)' || 'Detect Polarity Reversal' => 'Yes' || 'Detect PSTN Long Silence' => 'yes' || 'PSTN Long Silence Duration' => '150' || 'Detect VoIP Long Silence' => 'no' || 'VoIP Long Silence Duration' => '30' || || but the relevant values are 'Detect CPC' => 'yes' and 'Min CPC || Duration' => '0.05' || || || I hope this helps, if anyone struggles with unrecognized disconnects. || || God bless, Peter || || || CPC: Calling Party Control: A short break in the line current in the || called party's phone line when the calling party hangs up. || (def. from http://www.vikingelectronics.com/glossary/telecom-term.php) || -- || dyslexics of the world - untie ! || ___ || --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] Can Asterisk do This?
Asterisk can authenticate by CLID - it's not a good idea, though as CLID can be spoofed - Original Message - From: Goran Donev To: asterisk-users@lists.digium.com Sent: Thursday, December 01, 2005 10:36 PM Subject: [Asterisk-Users] Can Asterisk do This? I have a client who is looking for the proposed solution and was wondering if any asterisk professionals know if this can be done by asterisk. Calling card platform. Users calling in through local access numbers, they dial local access numbers and make calls through the system to make affordable long distance lines. The lines would be coming to a PRI gateway probably MediaTrix or asterisks directly via a PRI card. They want the calling card platform to identify the users pin through Caller ID. Either if they call from home or they call phone. If they call from a 3rd party location to give them choice to enter their pin to be authorized by the system for them to make a outbound calling. These calls would be registered to their account and would be bill accordingly to the rates given to them. They want easy administration of this software, I saw A2Billing but I didnt see a part to identify the Pin through caller id. They want this software to be GUI driven and to be easy to administer. 2nd part they want is a VOIP Platform for VOIP ATAs for internet clients. They want to be able to attach ATA clients with DID numbers to they can make calls from their homes and receive incoming calls through this system. This part I know Asterisk can do, but I want to know if this is possible with the system they are looking to implement to have the complete package. They want the system to have a nice GUI like AMP to make the changes. If anyone knows how this can be done affordably with a small startup pilot system. Please let me know if this can be done it would be greatly appreciated. Thanks. ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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
