[asterisk-users] How can we pickup a call that is not going to a real extension?
Hello,
We have a situation where a call comes in, users are notified via an external
process (curl request to web service), and we can't answer the call until a
callee can call in and pickup the call.
How can we implement this functionality?
We tried using :
[caller-inbound-leg]
; code to send the CALL_UUID information to users.
exten =>
_[+0-9a-zA-Z*#_].,n(r203),Dial(LOCAL/${call_uu...@inbound-wait-loop,,r)
; wait for call pickup from callee's inbound leg
[inbound-wait-loop]
exten => _[+0-9a-zA-Z*#_].,1,Wait(30)
[callee-inbound-leg]
; code to figure out the CALL_UUID used for the callee-leg
exten => _[+0-9a-zA-Z*#_].,n,Pickup(${call_uu...@inbound-wait-loop)
We thought Pickup() would work, but it only seems to work if the call is in the
Dial state.
The logs have results like:
-- Executing [5552...@caller-inbound-leg:8]
Pickup("SIP/20678350-5cd1-11de-bcf8-123139006632-004e",
"14f0dff4-9a34-11dd-93fd-0015588ab...@inbound-wait-loop") in new stack
[2010-02-25 19:20:28.936] NOTICE[18759]: app_directed_pickup.c:294 pickup_exec:
No target channel found for 14f0dff4-9a34-11dd-93fd-0015588ab9f3.
Is there a way for a callee to pickup a call in the Wait state?
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Eric Chamberlain
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[asterisk-users] Anyone have provisioning documentation for LeadTek devices?
Hi, A friend has a few hundred deployed LeadTek BVA8055's and needs to bulk re-provision them. There isn't much documentation on the web. Anyone have documentation explaining the LeadTek provisioning process and the provisioning file format? -- Eric Chamberlain -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk trunk CURL hangs in the dialplan
On Nov 24, 2009, at 6:17 AM, Tilghman Lesher wrote: > > Sounds like your local DNS resolver isn't answering queries promptly. > Thanks, I'll look into it. Our CURL function only calls one hostname over and over. Would setting CURLOPT dnstimeout be of use in this situation? -- Eric Chamberlain, Founder RF.com - http://RF.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk trunk CURL hangs in the dialplan
We've encountered a strange issue with the trunk version of asterisk.
Our dialplan makes CURL calls and occasionally CURL stops working.
The dialplan looks something like this:
[macro-curl]
; ${ARG1} CURL URL
; ${ARG2} CURL POST
exten => s,1,NoOp(CURL)
...
exten =>
s,n(post),Set(RF_CURL_POST=userID=${RF_DIALER_USERID}&password=${RF_PASSWORD}&${ARG2})
exten => s,n,Set(CURLOPT(httptimeout)=5)
exten => s,n,Set(CURLOPT(conntimeout)=5)
exten => s,n,NoOp(CURL(${RF_URL}/${ARG1}?${RF_CURL_POST}))
exten => s,n,Set(RF_CURL_RESPONSE=${CURL(${RF_URL}/${ARG1},${RF_CURL_POST})})
At this point, CURL either works or it will occasionally hang for a few
minutes. tcpdump doesn't show any traffic from the asterisk box to the web
server.
Something seems to be causing CURL to hang, before it sends out the http
request and the CURLOPT timeouts have no effect on the behavior.
Once CURL hangs, any additional calls to CURL also hang.
After a few minutes, tcpdump will show the CURL traffic going to the web server.
And CURL begins functioning normally for a while.
Has anyone else seen this? Or have any suggestions on how to debug this?
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Re: [asterisk-users] AstriCon videos: a question of method
Something that didn't require flash (works on the iPhone) would be nice. blip.tv may be an option. On Oct 22, 2009, at 3:34 PM, John Todd wrote: > > I'm doing some quick research on how to get our videos from AstriCon > available in a "reasonable" format that allows easy viewing, reduces > our bandwidth costs, and allows good tracking for who/where/what is > viewing the videos. > > YouTube seems to have a very nice set of tools and statistics > collection methods, and might be perfect EXCEPT Their main > limitation right now seems to be that they limit videos to 10 minutes, > which clearly doesn't work for our longer presentations. I could > "patch" them together in multiple 10-minute sessions, but... ugh. > UGH. > > There are other video sites out there - lots, actually. I could spend > hours digging through them all, or hopefully ask here on the list and > have some people give me prior experiences based on their expertise > with hosted video solutions. > > Requirements (not exhaustive list): > - free or very close to free (we'll pay, but not a lot) > - good statistical collection (who is linking? how many views? how > much video watched each view? where do people stop?) > - reasonably easy interaction (good upload tools, good UI) > - good viewing experience from North America, Europe, Asia > > Before anyone suggests it, I'm not interested in Torrent-based > distribution for various political reasons. I've started to look at > Flowplayer, which is appealing due to it's OSS nature and > customization capability, but it leaves us holding the bandwidth bill > (which may not be horrible, but it's a concern.) > > What are your experiences? I can't say we'll end up actually using > what you might think is best, but I'm very interested to hear what > everyone might suggest for distributing Asterisk-focused video > material. > > In the interests of keeping this thread from getting out of control, > please limit yourself to factual, content-rich posts. "I hate > YouTube" or "Why didn't you film blah" is something we can discuss > off- > list. > > JT > > --- > John Todd email:jt...@digium.com > Digium, Inc. | Asterisk Open Source Community Director > 445 Jan Davis Drive NW - Huntsville AL 35806 - USA > direct: +1-256-428-6083 http://www.digium.com/ > > > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Is there a way to force a codec on an incoming sip uri call?
Hello, I'd like to implement some public sip uri's that poeple can call into and get an echo test. Is there a way to force a codec so that users can test various codecs? Something like: echo-t...@example.com (negotiates whatever codec, is there a way to figure out what codec was negotiated and tell the user) echo-test-g...@example.com (forces g711) echo-test-g...@example.com (forces g729) echo-test-...@example.com (forces gsm) ... echo-test-i...@example.com (forces ilbc) -- Eric Chamberlain, Founder RF.com - http://RF.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk and FreePBX Amazon EC2 instances are now available in Europe
Based on interest expressed at AstriCon, we've published Asterisk and FreePBX Amazon EC2 instances in Europe (previously they were only available in the U.S. region). More information is available at: http://voxilla.com/2009/10/15/asterisk-on-the-cloud-with-a-click-1405 http://voxilla.com/2009/10/15/freepbx-in-a-cloud-with-a-click-1436 -- Eric Chamberlain, Founder RF.com - http://RF.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk in the Cloud
On Oct 14, 2009, at 1:04 AM, Dan Journo wrote: > Thanks Eric, > > I'd love to be able to make it to an Astricon one day. At the > moment, its a bit out of my price range. > > Do you happen to know whether RackspaceCloud.com offers a Kernel > with a timing device enabled? > > Many thanks and good luck with the presentation. > Dan > Dan, I'm not sure what Rackspace Cloud offers kernel wise. We didn't go with them because of their higher bandwidth costs and all the other services Amazon offers. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk in the Cloud
On Oct 13, 2009, at 5:22 PM, Dan Journo wrote: Hi, I was wondering if anyone is successfully running Asterisk in a cloud environment. If you could state which cloud you are using, I’d appreciate it. Many thanks Dan Journo Dan, I'm giving a presentation at AstriCon on this very topic. Here's a writeup of our experience with Asterisk on Amazon EC2 <http://voxilla.com/2009/02/18/asterisk-on-the-cloud-with-a-click-1405 >. -- Eric Chamberlain, Founder RF.com - http://RF.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Is anyone doing real time updates to where asterisk registers?
Hello, We need Asterisk to register with a variable and changing number (hundreds) of VoIP providers, is there a way to do this in a database and without reloading the entire sip config? Where Asterisk needs to register is determined by downstream users, so we need to do it real time and with minimal impact on the server. If Asterisk can't do this, is anyone using anything else to handle the registrations for Asterisk? -- Eric Chamberlain ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to deal with PayPal frauds?
On Aug 30, 2009, at 7:45 PM, Zeeshan Zakaria wrote: > I charge my customers through PayPal, but recently faced a fraud > which previously had only heard about. Somebody registered a few > accounts, paid online with paypal (as my service is only prepaid) > and started making expensive long distance calls. In fact the IP > registering the accounts was from Florida, and IPs making calls were > from Africa. After about 20 minutes the first payment was reversed. > Then a few times more payments were made, and every payment was > reversed almost as soon as it was made. Payments were made from > different PayPal accounts. And then I started getting emails from > PayPal resolution center that some payments were made by users who > didn't authorize them. > > Obviously either somebody was using stolen paypal accounts, or > somebody knows that he can pay and reverse the payment and in the > meanwhile make enough long distance calls. What is really fishy that > reversals were made almost as soon as the payments were made, one > after another. > > Those who are more experienced in this business, please advise how > to avoid this type of fraud, and which service to use in place of > PayPal, because PayPal doesn't seem the right payment solution for a > prepaid VoIP service. Also now that they have all the payments put > on hold and asking for a resolution, their resolution center is good > only for shipped merchendise, not for online services. How would I > prove to them that the buyer who is asking his money back has > already utilized my service by making lot of international calls, > which I now have to pay for to the carrier. Despite what PayPal and any of the other processors tell you in their marketing material, there is very little protection for online merchants. The only way to be mostly sure, is to accept cash or wire transfers. Having said that, you might want to look into MasterCard's SecureCard program (http://www.mastercard.com/us/merchant/solutions/mastercard_securecode.html ). I don't remember the exact details when a physical product is not involved, but the general idea is that if you enroll in the securecard program, you will be covered from cardholder unauthorized chargebacks, Visa has something similar. AmEx has a number you can call and they will verify transactions over $250 with the card holder. You might also want to consider shipping a welcome packet to the customer, that may cover you under PayPal's physical goods terms. -- Eric Chamberlain, Founder RF.com - http://RF.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone had any luck with SIP clients on the iPhone platform?
On Aug 6, 2009, at 9:43 PM, randulo wrote: > Hi, > > I've tried two SIP clients so far and both have unusable outgoing > audio quality. Skype app sounds fine, and recording the same mic > sounds fine, so I can only assume there is an issue with the clients > themselves. > > Both clients allow you to register and make calls via SIP with any > abitrary provider and credentials, so they'll work with Asterisk. I've > tried them with two good providers and one has unrecognizable audio > and the other has noises as if the cable was badly soldered. I've > never experienced such troubles with "regular" SIP clients. > > Anyone have any recommendations? > Contact me off-list and I can get you a beta version of RF Dial. -- Eric Chamberlain, Founder RF.com - http://RF.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TLS Manager
The pem file should contain both the private key and the certificate. On Jul 24, 2009, at 4:08 PM, John A. Sullivan III wrote: > Hello, all. After many pages of googling and testing in the lab, I'm > still a bit perplexed about how to implement tls protection for the > asterisk manager. manager.conf allows one to specify the cert file > but > one normally must also specify the private key file. If I simply > enter > the cert file: > > sslenable=yes > sslbindport=5038 > sslbindaddr=172.x.x.8 > sslcert=/etc/pki/tls/certs/pbxc.pem ; path to the certificate. > ; sslcipher= > > It errors as I expect it would: > > pbx*CLI> manager reload > == Parsing '/etc/asterisk/manager.conf': == Found > SSL cert error > > How does one specify the private key for the manager.conf file? > Thanks - > John > -- > John A. Sullivan III > Open Source Development Corporation > +1 207-985-7880 > jsulli...@opensourcedevel.com > > http://www.spiritualoutreach.com > Making Christianity intelligible to secular society > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Help - create_addr_from_peer: 'UDP' is not a valid transport for 'exten1'. we only use 'TLS'! ending call.
Hello,
I'm running into an issue with TLS transport and I am probably missing
something obvious.
We are trying to configure an extension to use TLS for the transport.
The extension can make outbound calls using TLS, but inbound calls fail.
The extension configuration in sip.conf is set to transport=tls.
To keep things simple for testing, the dialplan looks like:
[default]
exten => 100,1,NoOp("Inbound call for Eric!")
exten => 100,n,Dial(SIP/exten1)
Asterisk generates the following log when a call comes in for this
extension:
-- Executing [...@default:1] NoOp("SIP/rf.com-b69da2f8",
""Inbound call for Eric!"") in new stack
-- Executing [...@default:2] Dial("SIP/rf.com-b69da2f8", "SIP/
exten1") in new stack
== Using SIP RTP CoS mark 5
[2009-06-09 15:31:39.107] ERROR[19542]: chan_sip.c:4003
create_addr_from_peer: 'UDP' is not a valid transport for 'exten1'. we
only use 'TLS'! ending call.
Really destroying SIP dialog '03ad15856b352c3f2956812075bb7...@10.254.105.188
' Method: INVITE
[2009-06-09 15:31:39.107] WARNING[19542]: app_dial.c:1502
dial_exec_full: Unable to create channel of type 'SIP' (cause 20 -
Unknown)
Does something need to be set for Dial to use TLS on the outbound leg
of the call to the user?
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Re: [asterisk-users] Proxying from one server to another
On May 13, 2009, at 5:54 AM, Adrian Marsh wrote: Hi All, I’m trying to find a software package to do the following sip proxy work: I’ve an A*k server A that needs to be decommissioned, from the USA, and replaced by server B, in the UK. Both servers are on public internet IPs. Whilst the client migration happens, I want to divert all the Register traffic from Server A to Server B to catch any clients still left out there. Unfortunately, the original Clients were configured with static IPs instead of DNS names for the SIP Registrar, so I have to proxy Server A until all the clients have been updated (which might be a long time). Obviously A*k itself wont do this (as far as I know). I’ve looked at siproxyd and party-sip, but with no success so far. I’ve also tried using IPtables to redirect at the IP level, but the public IP ranges seem to stop me from achieving this. It works in my local-lan testing, but not on the public servers. Any ideas? Do your SIP clients support SIP redirects? If so, you might want to consider configuring server A to issue 301 redirects pointing to server B. -- Eric Chamberlain, Founder RF.com - http://RF.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone with a working pfSense firewall configuration?
On May 11, 2009, at 2:30 PM, Tim Nelson wrote: > pfSense employs source-port randomization by default. You may want > to enable advanced outbound NAT which turns this behavior off. > > While I'm not sure this is the source of your problems, I've seen it > ruin otherwise acceptable SIP situations. > Thanks, I just tried using the static-port option. The source ports aren't randomized any more, but the INVITEs still disappear after the initial 401-INVITE response. -- Eric Chamberlain, Founder RF.com - http://RF.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Anyone with a working pfSense firewall configuration?
Other SIP clients behind the firewall (not using STUN, work). We have a SIP client using STUN and ICE behind a pfSense firewall. The firewall is behaving oddly. REGISTER packets work fine. But when the client tries to make a call, the first INVITE packet from the client pass through the firewall and makes it to the Asterisk server. The Asterisk server sends back a 401 client sends ACK, traffic passes fine. When the client then sends the INVITE with the authentication information, the INVITE packet never makes it to the Asterisk server. A packet trace on the WAN interface of the firewall shows the INVITE going out, but the packets never make it to the Asterisk server. Any ideas on how to configure pfSense to work with a SIP client using STUN and ICE, without having to install siproxyd? -- Eric Chamberlain, Founder RF.com - http://RF.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco SPA525G
On Apr 28, 2009, at 11:36 PM, Gondar Monn wrote: > Anyone have used one of the new Cisco SPA525G with Asterisk ? Will > be reading manual before starting to play with, but would really > appreciate if you could share some tips with me. Thanks > We tested one a few months ago. They work like the other SPA series phones. -- Eric Chamberlain, Founder RF.com - http://RF.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk & EC2
On Apr 25, 2009, at 10:31 PM, Aryan Ameri wrote: > > The second one, is built on a custom Fedora 8 image. The steps are not > repeatable on any other distro, not even a stock official Fedora 8 > one. Fedora > 8 itself is long EOLed and as such, not something I'd want to use on a > production server. Dahdi compilation as described on that guide > doesn't work > on CentOS or Ubuntu or Debian. Aryan, The original Feodra 8 image came from the Amazon EC2 team, they optimized it to run in EC2. I chose the Amazon fc8 image, because I'm not comfortable getting OS images from third-parties. When Amazon releases new images, I'll update the guide. You might want to consider installing fc8 then upgrading to a newer release. To build DAHDI kernel modules, all you have to do is setup your OS of choice to use a build environment that matches the Amazon kernel you are using. > > > Besides, I asked about anecdotal usage experiences running Asterisk > on EC2. > About whether latency is an issue if extensions are outside the EC2 > availability zone. About reliability of EC2 when used to host a real- > time > application server. Not just an installation guideline. Latency is pretty route specific. Amazon has good bandwidth, but I would avoid proxying media if possible. We haven't had any reliability problems with EC2 hosting our Asterisk real-time application servers. -- Eric Chamberlain, Founder RF.com - http://RF.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk & EC2
On Apr 24, 2009, at 3:30 PM, Aryan Ameri wrote: > Has anyone been able to get asterisk 1.6 running under Xen or Amazon > EC2? > > If yes, can you share your experience please? Is it usable in a > production > environment? How is the sound quality? Am I likely to suffer from > latency > issues if the extensions are not located in the US? > > Any pitfall that I should be aware of? > Aryan, Depending on your use case, Asterisk can run quite well on EC2. Voxilla is sharing a public Asterisk 1.6 AMI, more information is available at <http://forum.voxilla.com/asterisk-support-forum/asterisk-cloud-asterisk-1-6-0-5-optimized-amazon-ec2-33857.html >. -- Eric Chamberlain, Founder RF.com - http://RF.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there documentation explaining res_config_curl?
On Apr 13, 2009, at 8:52 AM, Tilghman Lesher wrote: > On Sunday 12 April 2009 02:34:10 am Julian Lyndon-Smith wrote: >> Eric Chamberlain wrote: >>> On Apr 11, 2009, at 12:53 AM, Julian Lyndon-Smith wrote: >>>> Eric Chamberlain wrote: >> >> [snip] >> >>> Thank you, that bug does have useful information. >>> >>> We are working on moving from res_config_odbc to res_config_curl, so >>> all asterisk requests go through our django backend, rather than >>> django and asterisk sharing database tables. >> >> We had a buggy odbc driver (a 3rd party closed one) - we went from >> 2-3 >> crashes per day to zero in the last year, running nearly 3M >> config_curl >> requests per month now ;) >> >> It's, like, wow man ! > > As an additional note, please see contrib/scripts/dbsep.cgi, which I > wrote > as a reference implementation for the CGI backend of > res_config_curl. It > implements several additional methods to what JMLS is using > (basically, for > all the methods in trunk), so it may be useful in that regard. dbsep.cgi looks very helpful, thanks. Mapping the require function looks to be a challenge. Is there any way to add additional information to the res_config_curl POST request? We need to authenticate each https request and we'd rather not put the username and password in the path info to keep the password out of the server logs. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there documentation explaining res_config_curl?
On Apr 11, 2009, at 12:53 AM, Julian Lyndon-Smith wrote: > Eric Chamberlain wrote: >> Is there any documentation that explains res_config_curl? >> > > We use the 1.4 backported version - it works so well I just can't sing > it's praises enough. We use it for realtime voicemail and realtime > queues / queue members. > > Have a look at bug #11747 for some documentation. Thank you, that bug does have useful information. We are working on moving from res_config_odbc to res_config_curl, so all asterisk requests go through our django backend, rather than django and asterisk sharing database tables. -- Eric Chamberlain, Founder RF.com - http://RF.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Is there documentation explaining res_config_curl?
Is there any documentation that explains res_config_curl? Specifically, the format of realtime calls made to the web server and what the return string for each call should look like? -- Eric Chamberlain ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ATA recommendation??
On Mar 20, 2009, at 7:49 AM, Oguzhan Kayhan wrote: > Hello, > I want to ask that if thee are some ATA decives that i can use to > connect > mutliple analog phone lines to my VOIP system.. > I mean for example an ATA device with 24 ports with 24 independent SIP > accounts. > > For example for some dormitories in my area, i want to put an ATA > device > and move existing lines to VOIP accounts. > Only problem is, if i dont give seperate SIP accounts for all ports, > i can > not control who is calling where... And the billing system will also > be a > problem in that case. > The Cisco SPA8000 is an 8-port unit with a low per port cost. -- Eric Chamberlain, Founder RF.com - http://RF.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FreePBX in a Cloud With a Click - FreePBX secured and optimized for Amazon EC2
FreePBX in a Cloud With a Click (http://voxilla.com/2009/02/23/freepbx-in-a-cloud-with-a-click-1436 ) Since making freely available Voxilla’s “pre-built” Asterisk installation for Amazon’s Elastic Compute Cloud (EC2) (Asterisk on the Cloud With a Click), we’ve received many requests for a version that includes a graphical administration front end for the well-regarded open-source PBX. Answering the demand, we settled on FreePBX, itself an open-source Asterisk GUI developed under the guidance of internet and SIP phone services provider Bandwidth.com, because it is itself becoming very popular in the Asterisk world and, though tricky to install (requiring some knowledge of Linux and the installation of large external packages), it is very easy to use once it’s running correctly. To make it as easy as possible, we built an Amazon machine instance (AMI) with FreePBX, some custom modules, and Webmin (a web based server administration tool). What we came up with is an installation of FreePBX that is usable in three easy steps: • Start the AMI; • Retrieve the FreePBX interface passwords from SSH; • Securely access the FreePBX web interface. Full story available at Voxilla.com <http://voxilla.com/2009/02/23/freepbx-in-a-cloud-with-a-click-1436 > -- Eric Chamberlain, Founder RF.com - http://RF.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Managing SIP hardphones call history
On Feb 18, 2009, at 10:47 PM, Olivier wrote: > Hi, > > I've been asked sometimes to tailor call history features embeded in > SIP hardphones. > For example, a cutomer wanted internal call to be taken out. > Another wanted calls to sorted according specific criteria. > > 1. Have you identified a phone offering the possibility to display > as Call History, an XML list produced on a distant web server ? > With this feature, you would simply have to tell the hardphone which > query to send and then, you would get a customized Call History. > The Cisco SPA962 and SPA525 support RSS feeds, you could do a call history RSS feed for each phone. -- Eric Chamberlain, Founder RF.com - http://RF.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk on the Cloud With a Click - pre-built Asterisk Amazon EC2 instance
Asterisk-users, Our two-part tutorial explaining how to use VoIP and Asterisk in Amazon’s Elastic Compute Cloud (EC2) has garnered quite a bit of attention. But due to the time required to complete the many steps needed to get up and running, some of you have asked if it is possible to create a much simpler to install “pre-built” Asterisk EC2 “instance.” In short, yes it is. And we’ve done just that for you. With the power of the cloud, it’s not necessary have to wait days or hours for servers to be rebuilt. We don’t even need to start with a server that has nothing more than an operating system on it. Someone (Voxillans) can do all the grunt work: building, compiling, installing software; then share the complete server with others (you). Amazon calls this sharing Amazon Machine Images (AMIs). Now you have two choices, you can either build the Asterisk server yourself, or you can use Voxilla’s pre-built image to eliminate a lot of the heavy lifting. Learn more at http://voxilla.com/2009/02/18/asterisk-on-the-cloud-with-a-click-1405 -- Eric Chamberlain, Founder RF.com - http://RF.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on EC2 cloud computing - price assumptions - your brain needed
On Feb 13, 2009, at 9:59 AM, John Todd wrote: > > I've been involved with getting better data for running Asterisk on > the Amazon EC2 cloud computing system. Here are some calculations > I've made on costs based on current published prices on Amazon's > system. Feel free to tell me that I'm wrong with these calculations - > but be specific if you find any problems, as I suspect others may glom > onto these figures as gospel and I'd hate to have the wrong data in > there. > > http://www.loligo.com/asterisk/misc/amazon-ec2.xls > > The net of my calculations is that a small instance of 20 users in a > standard office environment would cost about $75 per month, which when > compared to running a server in-house works out to be (raw cost, not > including admin time and not discounting out-of-office bandwidth) only > $38.56 more. Very interesting. > The big advantage I like, is the ability to have identical production and development environments, without having to continuously run the development environment. When writing up how to install DAHDI on an Asterisk EC2 instance, I went through several instances, I could bring up an instance in minutes, use it for 10 or 15 minutes and then throw it away. I could do something similar with VMware ESX attached to a SAN, but it is much more capital intensive, even with leasing. For a single box, EC2 probably isn't going to be cheaper. But if you have a dynamic environment; need web, asterisk, and database servers; scaleable storage; and off-site backup, EC2 starts getting more cost effective. Storing Asterisk realtime data in Amazon's SimpleDB and voicemail in S3, would make for a very interesting and scalable solution. -- Eric Chamberlain, Founder RF.com - http://RF.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk and Amazon EC2 cloud service tutorial
Voxilla published a two-part series on using Amazon’s cloud services to meet your business telephony needs. Part 1 covers Amazon EC2 and how it is used in a VoIP setting. Part 2, covers all the steps necessary in getting the open-source Asterisk PBX to work on Amazon’s cloud. http://voxilla.com/2009/02/12/amazon-ec2-voip-1096 -- Eric Chamberlain ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] USA BRI -- any hope at all?
On Jan 27, 2009, at 9:49 AM, Michael Higgins wrote: > Folks -- > > First, apologies for not lurking for weeks or months to get the > culture of the list. I read the recent post about improvement to the > quality of posts with some amusement and full agreement. The problem > is a big and very real one. I hope I'm not deepening it. > > But my question isn't explicitly asked with this subject line or > definitively answered in the archives -- that I have found. > > What I did find left me with the impression that USA 'BRI', uh, > '2B1Q' protocol(?) is not supported by *any* hardware vendor, at > all, period, nor is it tested and proved in the software... > stack(?), in one related branch or another on the OS side. You might want to look into Cisco hardware, their WIC-1B-U cards work fine in the US, or they did 10 years ago when I last used them for VoIP. Used the WIC-1B-U is going for under $50 on eBay. An old 1600 or 1700 series router with an IOS that supports SIP wouldn't cost much either. > > -- Eric Chamberlain, Founder RF.com - http://RF.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities
Yehavi, You might want to check out some of the EDUCAUSE <http://www.educause.edu > mailing-lists to find out what other universities are doing. -- Eric Chamberlain ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] iPhone SIP or IAX client (without proxy)?
On Nov 16, 2008, at 9:07 AM, Lincoln King-Cliby wrote: I checked the app store and haven’t found anything promising, but I figured I’d ask here. Does anyone know of a SIP or IAX client for a non-jailbroken iPhone that will communicate directly with a machine running Asterisk? Lincoln, Nothing is publicly available yet. There's some incompatibilities between some open source licenses and the Apple licensing, that has delayed porting. Siphon <http://code.google.com/p/siphon/> is what you are looking for, but it is currently in beta. The beta does work with Asterisk over a Cisco VPN from the iPhone. -- Eric Chamberlain, Founder RF.com - http://RF.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What are the minimum realtime fields for sipusers?
To answer my own question after reviewing chan_sip.c. sipusers has been de-implemented in 1.6.0.1 and doesn't do anything anymore other than appear in sip show settings. On Nov 12, 2008, at 9:04 AM, Eric Chamberlain wrote: > I'm trying to get sipusers working with a realtime odbc database on > Asterisk 1.6. We have sippeers working from the database, but need > sipusers to be in a separate table for other implementation reasons. > > sip show user test load returns results from the database. > > CLI> sip show user test load > > * Name : test > Secret : > MD5Secret: > Context : inbound > Language : > AMA flags: Unknown > Transfer mode: open > MaxCallBR: 384 kbps > CallingPres : Presentation Allowed, Not Screened > Call limit : 0 > Callgroup: > Pickupgroup : > Callerid : "" <> > ACL : No > Sess-Timers : Accept > Sess-Refresh : uas > Sess-Expires : 1800 secs > Sess-Min-SE : 90 secs > Codec Order : (none) > Auto-Framing: No > > But when the user tries to register, the registration fails. > > [Nov 12 11:59:18] NOTICE[16779]: chan_sip.c:18293 > handle_request_register: Registration from '' > failed for '10.10.10.10' - No matching peer found > > What are the minimum fields required for the sipusers table? > > Currently we have: > name|secret| type | nat | host | context > | qualify | regserver | regseconds > > Also is there a way to enable debugging to show the SQL calls being > made to the database? > > -- > Eric Chamberlain > > > > > > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- Eric Chamberlain, Founder RF.com - http://RF.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] What are the minimum realtime fields for sipusers?
I'm trying to get sipusers working with a realtime odbc database on Asterisk 1.6. We have sippeers working from the database, but need sipusers to be in a separate table for other implementation reasons. sip show user test load returns results from the database. CLI> sip show user test load * Name : test Secret : MD5Secret: Context : inbound Language : AMA flags: Unknown Transfer mode: open MaxCallBR: 384 kbps CallingPres : Presentation Allowed, Not Screened Call limit : 0 Callgroup: Pickupgroup : Callerid : "" <> ACL : No Sess-Timers : Accept Sess-Refresh : uas Sess-Expires : 1800 secs Sess-Min-SE : 90 secs Codec Order : (none) Auto-Framing: No But when the user tries to register, the registration fails. [Nov 12 11:59:18] NOTICE[16779]: chan_sip.c:18293 handle_request_register: Registration from '' failed for '10.10.10.10' - No matching peer found What are the minimum fields required for the sipusers table? Currently we have: name|secret| type | nat | host | context | qualify | regserver | regseconds Also is there a way to enable debugging to show the SQL calls being made to the database? -- Eric Chamberlain ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fring: Open VPN client to be installed on the mobile, which mobile?
On Oct 27, 2008, at 3:48 AM, bilal ghayyad wrote: > Hi All; > > I do not know if anyone faced such case in dealing with open vpn (as > we need it for fring to be used from the mobile: > > Which mobile can be used to install the open vpn client on it, so we > can use it to do a vpn channel with the server that has open vpn > server? You might have better luck looking for mobile devices that natively support Cisco's VPN client, iPhone, etc. Also, your VPN may not work the way you think it will if Fring is in the middle of your call traffic flow. -- Eric Chamberlain ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How Secure Is Asterisk
On Oct 20, 2008, at 12:01 PM, Steve Anness wrote: I am sure this has been discussed prior, however, I am sitting here and being asked this very question by my superiors. They are loving what I have done with our two Asterisk servers here; however, they keep asking me if it is secure or not. Of course, as with anything, I suspect that on a secure network they can be reasonably safe. However, realistically if I am using the asterisk server to make internal calls and discussion very private matters, how possible is it for someone to listen to calls? How good is the encryption if any over an IAX trunk? Steve, This question gets asked a lot and a majority of the time the phones in question are in cubicles or other open spaces. The answer is that it really depends on the someone, the situation, and how much an organization is willing to spend. Yes, it's possible to encrypt voice traffic between SIP phones, but there is no standard that works across vendors. In most cases, it is more practical and economical to follow network security best practices. What alternative solution would they use to encrypt the voice traffic between analog or digital phones? -- Eric Chamberlain ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Does asterisk 1.6 support an authname with a domain?
We need to include the domain information in the Authentication digest username SIP header field. Using SIP/username[:password[:md5secret[:[EMAIL PROTECTED]:port] in the dialplan breaks if authname needs [EMAIL PROTECTED], is there a way to specify this value from the dialplan? -- Eric Chamberlain ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Is there a way to specify the fromdomain from the dialplan?
Is there a way to override the fromdomain specified in the sip.conf and instead set the value from the dialplan? If we use: Set(CALLERID(num)[EMAIL PROTECTED] The SIP From header turns into: [EMAIL PROTECTED]@10.10.10.10 We want [EMAIL PROTECTED], and we can't have an entry in sip.conf for every provider. -- Eric Chamberlain ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Text messaging and Asterisk
On Oct 13, 2008, at 11:28 AM, C. Savinovich wrote: > > I mean is if someone know of an sms server or service that allows > me to > send outgoing text messaging. > Are you sending SMS to known users or to any mobile phone user? If you are sending to a fixed user base, track down the email to SMS gateways for their carriers. Then sending an SMS is no different than sending an e-mail. -- Eric Chamberlain, Founder RF.com - http://RF.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ifbyphone/google analytics
On Oct 13, 2008, at 11:21 AM, Dean Collins wrote: Any thoughts? http://www.websitemagazine.com/content/blogs/posts/archive/2008/10/13/google-analytics-track-phone-calls.aspx It wouldn't be to hard to duplicate this with Asterisk. One could export the entire call path through an IVR or call center, time on call, etc. to Analytics. -- Eric Chamberlain, Founder RF.com - http://RF.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there a way to test SIP credentials withoutmaking a call?
On Oct 11, 2008, at 8:28 PM, Rob Hillis wrote: > Eric Chamberlain wrote: >>> Is there a particular reason you /can't/ register? It would seem >>> that >>> registration would provide the functionality you require, even if >>> you're >>> only making outbound calls. >>> >> In the case of a server like Asterisk, wouldn't sending a register >> disrupt the flow of inbound calls until the UA that normally handles >> inbound calls re-registers? > > Are you using the same credentials as existing extensions to make > calls > from different extensions? That would seem to be a particularly bad > idea. You should be configuring /one/ sip extension per SIP phone. > Those extensions that handle outgoing calls only could be put in a > different number range, or have a letter prefixed or suffixed to the > extension, but you should /not/ be using one configured extension for > two different purposes. We're developing the client and don't have control over the server, which may or may not be Asterisk. Adding extra extensions isn't possible. Can OPTION packets be used to verify authentication? -- Eric Chamberlain ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there a way to test SIP credentials withoutmaking a call?
On Oct 11, 2008, at 1:41 PM, Rob Hillis wrote: > Eric Chamberlain wrote: >> I should have clarified, we're only making outbound calls, not >> inbound, so there is no registration. >> > > Is there a particular reason you /can't/ register? It would seem that > registration would provide the functionality you require, even if > you're > only making outbound calls. In the case of a server like Asterisk, wouldn't sending a register disrupt the flow of inbound calls until the UA that normally handles inbound calls re-registers? -- Eric Chamberlain ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there a way to test SIP credentials withoutmaking a call?
I should have clarified, we're only making outbound calls, not inbound, so there is no registration. On Oct 11, 2008, at 9:27 AM, Meftah Tayeb wrote: > hi, > (i am no sur): > the user credential is tested during SIP Registration Step > thanks and tel me if this is a error > - Original Message - > From: "Eric Chamberlain" <[EMAIL PROTECTED]> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Sent: Saturday, October 11, 2008 5:20 PM > Subject: [asterisk-users] Is there a way to test SIP credentials > withoutmaking a call? > > >> Is there a SIP packet that a SIP client can send to Asterisk to >> confirm that the credentials entered by the user are correct, without >> placing a call? >> >> We'd like to test the credentials when the user enters them, rather >> than wait until they try to make their first call. >> >> -- >> Eric Chamberlain >> >> >> >> >> ___ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- Eric Chamberlain ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Is there a way to test SIP credentials without making a call?
Is there a SIP packet that a SIP client can send to Asterisk to confirm that the credentials entered by the user are correct, without placing a call? We'd like to test the credentials when the user enters them, rather than wait until they try to make their first call. -- Eric Chamberlain ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.6.0 CDR billsec and duration not working from h extension
Can someone tell me what I am doing wrong? Why doesn't CDR(duration)
or CDR(billsec) return the correct values?
cdr.conf
endbeforehexten=yes
extensions.conf
[macro-Dial]
; ${ARG1} - Dial String
exten => s,1,Dial(${ARG1},,M(post-dial))
exten => h,1,NoOp(Call was hung up - ${CDR(duration)} seconds long,
billed for ${CDR(billsec)} seconds)
The log shows:
-- Executing [EMAIL PROTECTED]:1] NoOp("SIP/10.10.10.170-b7d94f78",
"Call was hung up - 0 seconds long, billed for 0 seconds") in new stack
But cdr-csv/Master.csv has logged time values for duration and billsec:
"","510555","+410001","pop-inbound","""1510555""
<510555>","SIP/10.10.10.170-b7d94f78","SIP/
voipprovider.com-089ae8a0","Dial","SIP/1510555:password::[EMAIL PROTECTED]
,,M(post-dial)","2008-10-09 20:59:00","2008-10-09
20:59:03","2008-10-09 20:59:08",
8,5,"ANSWERED","DOCUMENTATION","1223585940.35"
--
Eric Chamberlain
___
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asterisk-users mailing list
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http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FW: Google Alert - "dean collins"
Dean, You retain copyright to your e-mail. Send a DMCA take down notice to blogspot. The real problem is the ad networks, no ads, no incentive to profit from stealing content for ad revenue. It's funny how some advertisers with really good search engines can't seem to use the same search engine to figure when they are profiting from stolen content. On Sep 27, 2008, at 12:11 PM, Dean Collins wrote: Who is Chris Langford in Huntsville Alabama and is he seeking Digium’s permission in order to report the asterisk mailing lists out onto the internet http://asteriskbizrss.blogspot.com/ http://www.blogger.com/profile/04174728129647374395 What can be done to stop people doing this and making money out of selling ads on these crappy blogsites? Cheers, Dean From: Google Alerts [mailto:[EMAIL PROTECTED] Sent: Saturday, 27 September 2008 2:38 PM To: Dean Collins Subject: Google Alert - "dean collins" Google Blogs Alert for: "dean collins" [asterisk-biz] Philippines By Chris Langford(Chris Langford) I have a friend who is setting up a domestic outbound call center with about 20 agents initially looking for a simple low cost implementation. Email me with reference information and I’ll send you contact details. Regards,. Dean Collins ... Asterisk Biz - http://asteriskbizrss.blogspot.com/ This as-it-happens Google Alert is brought to you by Google. Remove this alert. Create another alert. Manage your alerts. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Eric Chamberlain, Founder RF.com - http://RF.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Write Asterisk CDR MySQL records to multiple servers
On Sep 10, 2008, at 7:11 PM, Tilghman Lesher wrote: > On Wednesday 10 September 2008 19:55:15 Eric Chamberlain wrote: >> On Sep 10, 2008, at 2:01 PM, Tilghman Lesher wrote: >>> On Wednesday 10 September 2008 13:22:51 Ricardo Melendez wrote: >>>> Hi to all, I actually have an asterisk server configured to write >>>> CDR mysql >>>> records in the same machine (localhost), but I want to write this >>>> records >>>> to another machine also in mysql at the same time, It is possible? >>>> It >>>> means that I want save the records in both machines. >>> >>> You can either use MySQL replication or you can use 2 different CDR >>> drivers at >>> the same time, such as ODBC, with the Mysql-ODBC-Connector and the >>> MySQL CDR >>> driver. Also, in 1.6, cdr_adaptive_odbc allows you to specify >>> multiple CDR >>> backends within the same configuration file. >> >> Are there any sample config's explaining how to setup >> cdr_adaptive_odbc? Is cdr_adaptive_odbc used with cdr.conf? >> cdr.conf >> makes no mention of how to use cdr_adaptive_odbc, yet >> cdr_adaptive_odbc.com doesn't explain how to set things like >> usegmtime >> or loguniqueid. > > I thought that the sample cdr_adaptive_odbc.conf was rather clear, but > apparently not. The point of this module is to allow you log > whatever you > like in terms of the CDR variables. Do you want to log uniqueid? > Then simply > ensure that your table has that column. If you don't want the > column, ensure > that it does not exist in the table structure. If you'd like to > call uniqueid > something else in your table, simply provide an alias in the > configuration > file that maps the standard CDR field name (uniqueid) to whatever > column > name you like. Perhaps you'd like some extra CDR values logged that > aren't > in the standard repertoire of CDR variables (some that come to mind > are > certain values used for LCR: route, per_minute_cost, and > per_minute_price). > Simply set those CDR variables in your dialplan, i.e. > Set(CDR(route)=27), > ensure that a corresponding column exists in your table, and > cdr_adaptive_odbc > will do the rest. > > I do agree that I have overlooked gmtime as a possible setting for > datetime > fields in cdr_adaptive_odbc, and that's probably something that I > need to add. > However, I think that the method by which specifying which columns > you'd like > to have is certainly much more intuitive than the old "These are > your columns. > You must have them, or all CDRs will fail" approach. Hopefully, you > will also > see what I think is a rather innovative approach to CDRs and wonder > how you > ever got along without it. In fact, the adaptive approach has been > now ported > to most of the other CDR drivers, including mysql, postgres, and > sqlite, and > another developer (who is more familiar with that API) is working on > TDS > support. > Thanks for the explanation, it is clear now. The confusing part for a first time ODBC user is weeding all the old unneeded cruft. Does cdr_adaptive_odbc have any support for spooling records while a database connection may be down? -- Eric Chamberlain ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Write Asterisk CDR MySQL records to multiple servers
On Sep 10, 2008, at 2:01 PM, Tilghman Lesher wrote: > On Wednesday 10 September 2008 13:22:51 Ricardo Melendez wrote: >> Hi to all, I actually have an asterisk server configured to write >> CDR mysql >> records in the same machine (localhost), but I want to write this >> records >> to another machine also in mysql at the same time, It is possible? >> It >> means that I want save the records in both machines. > > You can either use MySQL replication or you can use 2 different CDR > drivers at > the same time, such as ODBC, with the Mysql-ODBC-Connector and the > MySQL CDR > driver. Also, in 1.6, cdr_adaptive_odbc allows you to specify > multiple CDR > backends within the same configuration file. > > Are there any sample config's explaining how to setup cdr_adaptive_odbc? Is cdr_adaptive_odbc used with cdr.conf? cdr.conf makes no mention of how to use cdr_adaptive_odbc, yet cdr_adaptive_odbc.com doesn't explain how to set things like usegmtime or loguniqueid. -- Eric Chamberlain ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Is there a way to get the Call-ID into the CDR?
Here's the use case: call comes in, extension match is made on caller ID and dialed number, dial plan dials a number and connects the two call legs. Is there a way to get the Call-ID from the SIP header of the outbound call leg and store it in the CDR? -- Eric Chamberlain ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Is including a linefeed in the JabberSend message possible?
Is there a way to include a linefeed in the message sent by JabberSend? -- Eric Chamberlain ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Anything to convert from JSON into Asterisk dialplan variables?
Is there anything already out there that can efficiently convert a JSON string into Asterisk dialplan variables? Our current backend speaks JSON and we need to parse the response to construct the dialstring. -- Eric Chamberlain ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there a way to encrypt passwords stored in the realtime database?
On Aug 20, 2008, at 12:34 PM, Igor Hernandez wrote: > Hey SIP, > > I understand what you're saying but keeping the key in memory > permanently doesn't protect you for very long, it just makes the > attacker waste a bit more time scanning the memory to get at the key. > > In other words, if the key is available to asterisk it will be > available > to anyone else in the system with sufficient privileges. > Assume I'm using a FIPS 140-2 Level 4 HSM, now, how can I protect my passwords when they are in the database? -- Eric Chamberlain Founder RF.com http://RF.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there a way to encrypt passwords stored in the realtime database?
On Aug 20, 2008, at 10:19 AM, Tzafrir Cohen wrote: > On Wed, Aug 20, 2008 at 10:00:55AM -0700, Eric Chamberlain wrote: >> We are exploring using Asterisk for a project and we are looking >> for a >> way to encrypt/decrypt the peer passwords stored in the realtime >> database (postrges). >> >> Ideally, we want to use a public key to encrypt the passwords before >> they go into the database and have Asterisk use a private key to >> decrypt the password as part of the call out process. >> >> Has anyone developed something like this? > > What is the point in that? What threats does it help you to mitigate? > Passwords are added/changed on a web front-end and stored in a database. We want to limit exposure to the Asterisk boxes, we don't want compromises of the web front-end or database to result in revealing passwords. These passwords are used to authenticate with other SIP systems, so storing a MD5 hash wouldn't work, hence the need to encrypt and decrypt. -- Eric Chamberlain Founder RF.com http://RF.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Is there a way to encrypt passwords stored in the realtime database?
We are exploring using Asterisk for a project and we are looking for a way to encrypt/decrypt the peer passwords stored in the realtime database (postrges). Ideally, we want to use a public key to encrypt the passwords before they go into the database and have Asterisk use a private key to decrypt the password as part of the call out process. Has anyone developed something like this? -- Eric Chamberlain Founder RF.com http://RF.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ASTERISK/ENSWITCH ON EC2
On Jul 11, 2008, at 12:28 PM, Robert McNaught wrote: > Has anyone deployed a hosted environment like enswitch using EC2? I > was wondering if anyone had any thoughts on concerns on the > feasibility in doing this using cloud computing? > > For setting up a VoIP service provider and not having the headache of > dealing with the hassle and expenses of hardware, racks, cages etc, it > looks pretty attractive. > > Any thoughts? > If you are setting up a VoIP service provider, I would be concerned about the service uptime using the EC2 cluster, it is after all still in beta and had a number of outages over the past year. Have you considered other hosting solutions? There are a number of high quality hosting offerings, that offer 24/7 phone support, without requiring any long term contracts. -- Eric Chamberlain Founder RF.com http://RF.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sippyskype
On Jul 7, 2008, at 7:46 AM, Emmanuel Favre-Nicolin wrote: > Hi, > > I'd like to know if someone already succesfully installed sippyskype. > > Here on gentoo, I'm starting to build necessary stuff. > I've got sippyskype running on CentOS 5.1. > > You shouldn't have to build anything, the application comes with the > necessary JAR files. > > As per the sippyskype instructions, you'll want the authors patched > version of mjsip. The original mjsip implementation is not code > complete and not RFC compliant in several areas. > > The author is very responsive to fix submissions. He only has a > SPA3102 and a windows machine to code for, so offering space on an > asterisk box could go a long way. > > > You might want to post your question on the sippyskype blog <http://blog.mhspot.com/ >. I've got sippyskype running on CentOS 5.1. You shouldn't have to build anything, the application comes with the necessary JAR files. As per the sippyskype instructions, you'll want the authors patched version of mjsip (source included in the download). The original mjsip implementation is not code complete and not RFC compliant in several areas. The author is very responsive to fix submissions. He only has a SPA3102 and a windows machine to code for, so offering space on an asterisk box could go a long way. > -- Eric Chamberlain Founder RF.com http://RF.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk as a component in Jabber network
Thanks, I was hoping that there was a way for Asterisk to send a subscription request as part of the call setup. On Jul 2, 2008, at 1:37 AM, Philippe Sultan wrote: > Hi Eric, > > no that's not possible, Asterisk being connected either as a regular > client or component. > > In the configuration you describe, Asterisk needs to be allowed by the > XMPP user (gmail.com) to send messages to his/her address. This is > achieved under the mutual subscription mechanism. > > However, the opposite call configuration works ok if you use the > JabberReceive application, which gives a flexible way to access > telephony resources from Gtalk/Jingle through Asterisk. That is, back > to your example, a GoogleTalk user can pass any SIP URI to an Asterisk > gateway. JabberReceive is still at a testing stage, if you want to > give a try, please check : http://bugs.digium.com/view.php?id=12569 > > Cheers, > > -- > Philippe Sultan > > -- Eric Chamberlain Founder RF.com http://RF.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk as a component in Jabber network
Philippe, Is it possible to configure Asterisk in a component configuration such that any sip user could call any Jabber user? A sip user could dial , for example, and the call would get routed to that person, even if the two parties have never spoke before and are not buddies. On Jun 30, 2008, at 2:13 AM, Philippe Sultan wrote: > Hi Antonio, > > here is the corresponding section of my jabber.conf file, that allows > our Asterisk server to connect to our local XMPP server (jabberd2), as > a component. > > [asterisk-component] > type=component > serverhost=jabber.inria.fr > username=asterisk > secret=*** > port=5347 > > Depending on your XMPP server, the port number may be different. Can > you tell us more about what you want to achieve? > > Cheers, > > Philippe > > On Fri, Jun 27, 2008 at 4:54 PM, Antonio Anderson M. de Souza > <[EMAIL PROTECTED]> wrote: >> Hi Everybody, >> >> Does anybody have some tutorial how to configure Asterisk in the >> component >> mode in a Jabber service, i already configured, and tested it in >> the client >> mode, and it worked fine, but i think the component is the best >> solution for >> what i need to implement. >> >> Thank you very much, >> >> -- >> Antonio Anderson M. Souza >> Project Manager >> Voice Technology >> Rua: Libero Badaró, 293 >> Cj 30D - 30o. andar >> CEP: 01009-907 >> [EMAIL PROTECTED] >> phone: +55 11 3588-0188 >> mobile: +55 11 8863-0693 >> ___ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> AstriCon 2008 - September 22 - 25 Phoenix, Arizona >> Register Now: http://www.astricon.net >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > > > -- > Philippe Sultan > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- Eric Chamberlain Founder RF.com http://RF.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip proxy failover
You can use DNS SRV records to specify more than one proxy and their order of usage. -- Eric Chamberlain, CISSP Chief Technical Officer Voxilla - http://voxilla.com/ _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert McNaught Sent: Wednesday, November 21, 2007 11:25 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] sip proxy failover Hi, Is it possible to failover from Outbound SIP proxyA to Outbound SIP ProxyB on the event that Outbound Sip Proxy A became unavailable - using the qualify option for sip peers - it should be possible to monitor the ping/back time, which would give us a good indication of whether a host is up and running. I have had a look in the mailing list archives, but cant see this having been asked before? How would someone do this in Asterisk - would this have to be done with Dialplan programming, before placing the call, it would check the most recent qualify ping time and route based on that? As far as I am aware it is only possible to put host=xxx.xxx.com once in sip.conf Has anyone got this to work, to have a failover outbound proxy in asterisk, which automatically fails over? Thanks :-) Robert McNaught ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Softphone to be installed on the Mobile
Divitas Networks <http://divitas.com/> has an asterisk based solution that allows seamless roaming between the Wi-Fi and GSM network. An appliance connects to or is the PBX on the office LAN and a client runs on the smartphone. The appliance and client then coordinate which network to use based on signal strength and availability. -- Eric Chamberlain, CISSP Chief Technical Officer Voxilla - http://voxilla.com/ > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of bilal ghayyad > Sent: Wednesday, November 21, 2007 1:29 AM > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] Softphone to be installed on the Mobile > > Hi All; > > Is there a softphone that can be installed on a mobile > (new mobile models), so it can work with Asterisk as > following: > > 1) As SIP or H323 client, with the ability to add > button functionalities (call pickup, call transfer, > ...) so if there is a wireless network, then it can > use it to connect to Asterisk and work as client, but > from the Mobile. > > 2) If there is no wireless network, then it can > receive calls via the GSM (doing a special settings on > Asterisk to forward the call to the mobile number), so > he can receive the call and do the PBX functionalies > (transfer, pickup, forward)? > > I saw this in AVAYA, AL Catel, Cisco, ... > > Any help? > Regards > Bilal > > > > __ > __ > Get easy, one-click access to your favorites. > Make Yahoo! your homepage. > http://www.yahoo.com/r/hs > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] California based PSTN connections
We use VoicePulse Connect. They now have a POP in San Francisco. -- Eric Chamberlain, CISSP Chief Technical Officer Voxilla - http://voxilla.com/ _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adrian Marsh Sent: Saturday, November 17, 2007 5:33 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] California based PSTN connections Hi, Can anyone recommend any company that can provide PSTN termination for SIP calls, at least USA based, preferably California area. One of my A*k servers is US based and I don't want my traffic flowing back and forth via my current UK PSTN provider for US<>US calls. Thanks, Adrian ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wanted: tutorial on troubleshooting SIP issues
The NOTIFY packets are most likely keepalive packets from the SPA942 to the Asterisk server. In a NATed WAN environment they are used to keep the firewall ports open. In your LAN situation, you can turn them off. The NAT keepalive is configured on a per extension basis. To turn them off, log into the administrative interface on the SPA942 and go to each Ext tab. On each Ext tab, there will be a NAT Settings section, change the NAT Keep Alive Enable to No. -- Eric Chamberlain, CISSP Chief Technical Officer Voxilla - http://voxilla.com/ > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Philip Prindeville > Sent: Thursday, November 08, 2007 4:11 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [asterisk-users] Wanted: tutorial on troubleshooting SIP issues > > For someone that's network-aware, but hasn't sat down and plowed through > umpteen SIP-related RFC's and memorized the standards, is there a good > primer on troubleshooting SIP issues? > > I'm seeing a lot of NOTIFY/603 messages on my network between Asterisk > and my Sipura 942's, for instance... > > Not sure what these are... perhaps the qualify keepalives? In which > case, I guess the 603 is moot... but since the messages are originating > from the Sipuras to Asterisk and not vice-versa, it wouldn't seem to be > the "qualify"... Next guess would be that they're NAT keepalives, but > Asterisk and the phones are on the same private subnet (which in turn > *is* NATted)... > > Anyway, pointers for someone wanting to learn to quickly diagnose SIP > conversations would be great. > > Thanks, > > -Philip > > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk as a SIP to XMPP Jingle voice gateway
Philippe, Thanks for the info. -- Eric Chamberlain, CISSP Chief Technical Officer Voxilla - http://voxilla.com/ > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Philippe Sultan > Sent: Friday, November 09, 2007 2:39 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Asterisk as a SIP to XMPP Jingle voice > gateway > > Hi Eric, > > > I'm looking for a SIP to XMPP Jingle voice gateway. > > > > > > > > I see that Asterisk has Jabber and Jingle support, but it looks like > > Asterisk acts as a Jabber client. > > Asterisk can connect as a client or component to a XMPP server. XMPP > components are typically used as gateways between XMPP and other IM > services such as MSN or Yahoo. > > You can connect Asterisk to GoogleTalk's XMPP network as a client > only, which will therefore be accessible through a presence > subscription mechanism just like a usual client. > > On the other hand, you can connect Asterisk as a component to your > locally administered XMPP server, for example. A 'service discovery' > request to the server will show the Asterisk server as being > available. > > > Are there any Jabber server solutions, where Jabber users can call SIP > users > > by using the SIP URI and vice versa? > > Asterisk can be used to call Gtalk users from SIP phones, and vice > versa. Configuration examples are given here : > http://www.voip-info.org/wiki/view/Asterisk+Google+Talk > The call configuration is handled in the Dialplan in that case. > > If you need to place a call from a XMPP client to a SIP URI, you'll > also have to find a client that's able to to so. I know that > GoogleTalk and Jabbin both speak XMPP + Gtalk. However, the GoogleTalk > client's user interface does not allow you to place a call to anything > but another XMPP client from your buddy list, without offering the > ability to enter either a SIP URI or phone number. A possible > workaround was available here : > http://bugs.digium.com/view.php?id=8659 > > As for Jingle, Asterisk tries to follow the latest set of > specifications (code only available from SVN trunk), which are not > completed yet. > > Philippe > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk as a SIP to XMPP Jingle voice gateway
Hello, I'm looking for a SIP to XMPP Jingle voice gateway. I see that Asterisk has Jabber and Jingle support, but it looks like Asterisk acts as a Jabber client. Are there any Jabber server solutions, where Jabber users can call SIP users by using the SIP URI and vice versa? -- Eric Chamberlain, CISSP Chief Technical Officer Voxilla - http://voxilla.com/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
What is the use case? Linksys, Polycom, Snom, and Aastra all have their strengths and weaknesses. -- Eric Chamberlain, CISSP Chief Technical Officer Voxilla - http://voxilla.com/ > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] > Sent: Monday, October 29, 2007 10:42 AM > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] (no subject) > > Hi all, > > We have a client that needs to setup about 80 desk phones (about 50 > in one location and about another 30 in 5 different locations). Which > brand/model would you recommend. We were personally thinking in > recommending either Cisco, Aastra, Polycom, or Snom, for we've heard > great things about them. However, having no real experience with them > makes it hard in recommending one to our customer. The only > experience we've had is a very frustrating one trying to load the IP > software on a Cisco 7970G and so we assume that if we have to go > through that for all 80 phones, we'll probably commit suicide :) > > Thanks > > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail playback on iPhone
I tested this again, and wav files do play as attachments with firmware 1.1.1. -- Eric Chamberlain, CISSP Chief Technical Officer Voxilla - http://voxilla.com/ > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Eric Chamberlain > Sent: Wednesday, October 24, 2007 3:29 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Voicemail playback on iPhone > > Convert the voicemail to a mp3 file. > > As of firmware version 1.1.1, the iPhone mail application will recognize, > but not play wav attachments. But the mail application does, recognize > and play mp3 file attachments. > > -- > Eric Chamberlain, CISSP > Chief Technical Officer > Voxilla - http://voxilla.com/ > > > -Original Message- > > From: [EMAIL PROTECTED] [mailto:asterisk-users- > > [EMAIL PROTECTED] On Behalf Of Jason Lixfeld > > Sent: Wednesday, October 24, 2007 7:46 AM > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: Re: [asterisk-users] Voicemail playback on iPhone > > > > Sorry, it's clear my question was far too vague. > > > > To clarify, is there a recipe to make * record voicemail in a format > > that allows playback on iPhone's media/music player playback for > > voicemails that are received say, in an email message. > > > > It seems the * voicemail defaults don't work. This link seems to > > describe codecs that do work, however I haven't seen any indications > > as to whether * voicemail can be tweaked to record in any of the > > supported formats: http://www.kehlet.cx/ > > > > Any success out there? > > > > On 22-Oct-07, at 7:38 PM, Ron Stephan wrote: > > > > > > > > Trick question I assume? > > > > > > It was mind numbingly simple on my iPhone...(though none of the > > > voice mail worked when London a few weeks ago). > > > > > > - tap voice mail - > > > - tap speaker (upper right) until it turns blue (is activate) > > > - tap the message you want to playback > > > - use assorted controls to delete - replay etc. > > > > > > > > > Now...if the question is ... how do you get asterisk voice mail to > > > show up on an iPhone...I am all ears. Groovy concept - if > > > anybody has a hack - I'd love to see it. > > > > > > > > > > > > Elvis > > > > > > > > > > > > > > > > > > > > > > > > -Original Message- > > > From: [EMAIL PROTECTED] [mailto:asterisk-users- > > [EMAIL PROTECTED] > > > ] On Behalf Of Jason Lixfeld > > > Sent: Monday, October 22, 2007 4:16 PM > > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > > Subject: [asterisk-users] Voicemail playback on iPhone > > > > > > Anyone managed to get this to work? What's the recipe? > > > > > > ___ > > > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > > > > > asterisk-users mailing list > > > To UNSUBSCRIBE or update options visit: > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > > __ NOD32 2607 (20071022) Information __ > > > > > > This message was checked by NOD32 antivirus system. > > > http://www.eset.com > > > > > > > > > > > > ___ > > > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > > > > > asterisk-users mailing list > > > To UNSUBSCRIBE or update options visit: > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > ___ > > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail playback on iPhone
Convert the voicemail to a mp3 file. As of firmware version 1.1.1, the iPhone mail application will recognize, but not play wav attachments. But the mail application does, recognize and play mp3 file attachments. -- Eric Chamberlain, CISSP Chief Technical Officer Voxilla - http://voxilla.com/ > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Jason Lixfeld > Sent: Wednesday, October 24, 2007 7:46 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Voicemail playback on iPhone > > Sorry, it's clear my question was far too vague. > > To clarify, is there a recipe to make * record voicemail in a format > that allows playback on iPhone's media/music player playback for > voicemails that are received say, in an email message. > > It seems the * voicemail defaults don't work. This link seems to > describe codecs that do work, however I haven't seen any indications > as to whether * voicemail can be tweaked to record in any of the > supported formats: http://www.kehlet.cx/ > > Any success out there? > > On 22-Oct-07, at 7:38 PM, Ron Stephan wrote: > > > > > Trick question I assume? > > > > It was mind numbingly simple on my iPhone...(though none of the > > voice mail worked when London a few weeks ago). > > > > - tap voice mail - > > - tap speaker (upper right) until it turns blue (is activate) > > - tap the message you want to playback > > - use assorted controls to delete - replay etc. > > > > > > Now...if the question is ... how do you get asterisk voice mail to > > show up on an iPhone...I am all ears. Groovy concept - if > > anybody has a hack - I'd love to see it. > > > > > > > > Elvis > > > > > > > > > > > > > > > > -Original Message- > > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] > > ] On Behalf Of Jason Lixfeld > > Sent: Monday, October 22, 2007 4:16 PM > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: [asterisk-users] Voicemail playback on iPhone > > > > Anyone managed to get this to work? What's the recipe? > > > > ___ > > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > __ NOD32 2607 (20071022) Information __ > > > > This message was checked by NOD32 antivirus system. > > http://www.eset.com > > > > > > > > ___ > > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which LDAP OID for iphones
The idea of adding every network device to directories was popular five years ago, but didn't really move beyond iPlanet, NDS, and AD marketing and a few enterprise network management solutions. Unless you have a specific application this sounds like more trouble than it's worth. Typically you want to associate your devices with other records (logs, etc.), making a relational database a much easier to manage solution with fewer moving parts. -- Eric Chamberlain, CISSP Chief Technical Officer Voxilla - http://voxilla.com/ _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olivier Sent: Friday, October 05, 2007 2:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Which LDAP OID for iphones Hello, I'm new to LDAP. I've read Device class exists (oid 2.5.6.14) in rfc2256. I've heard a Pluggable Device sub-class (a device with a MAC address) also exists though I can't find its OID at the moment. 1. Does any standard class specifically defines IP Phones or SIP hardphones or ATAs or Trunk lines ? What's their OID ? 2. How does one can find by himself if such classes exist ? I discovered this http://www.oid-info.com , there might be other sources. 3. Beside that, would you even try to use LDAP to store Resources data ? Many use it for User data but what about Resources (Trunk lines, ...) ? Is it worth the effort ? Cheers ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What's the deal with ATAcomm?
You should probably post that question on the Asterisk business forum. -- Eric Chamberlain, CISSP Chief Technical Officer Voxilla - http://voxilla.com/ > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Andrew Joakimsen > Sent: Saturday, September 29, 2007 3:44 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] What's the deal with ATAcomm? > > That's horrible. I don't buy too many IP phones these days, but can > anyone suggest a place better than the scumbags at VoIP supply? > > ___ > > Sign up now for AstriCon 2007! September 25-28th. > http://www.astricon.net/ > > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Point-to-Point SIP link without registration
Correct you want to set those settings to yes. Search the Voxilla Linksys forums for hotline or ringdown and you will find several examples. The examples are mostly for the spa3000, but the configuration is mostly the same. You are basically setting up ip or sip uri speeddials that are automatically dialed when the line goes off-hook. -- Eric Chamberlain On Sep 27, 2007, at 11:14 AM, "Mojo with Horan & Company, LLC" <[EMAIL PROTECTED] > wrote: > err... you'd set them to 'yes', right? Sorry if I'm missing the > obvious. > > Eric Chamberlain wrote: >> You can do this with any of the Linksys SPA series ATA's or phones, >> just set "Make Call Without Reg" and "Ans Call Without Reg" to no. >> > > > ___ > > Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ > > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Point-to-Point SIP link without registration
You can do this with any of the Linksys SPA series ATA's or phones, just set "Make Call Without Reg" and "Ans Call Without Reg" to no. -- Eric Chamberlain, CISSP Chief Technical Officer Voxilla - http://voxilla.com/ > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Chris Bagnall > Sent: Tuesday, September 25, 2007 11:34 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [asterisk-users] Point-to-Point SIP link without registration > > Greetings list, > > I need to set up a point to point SIP connection between two devices > without either of them registering with a registrar/proxy/etc. at all. The > devices I've tested so far all seem to insist on having a registration > before they'll make or take calls. > > One of the devices needs to be an ATA with an FXO port (e.g. > Sipura/Linksys SPA-3000/3102), the other device can be either an ATA or a > SIP Phone. > > Does anyone have any hardware recommendations that'll work in this > scenario? > > Regards, > > Chris > -- > C.M. Bagnall, Director, Minotaur I.T. Limited > For full contact details visit http://www.minotaur.it > This email is made from 100% recycled electrons > > > > > > ___ > > Sign up now for AstriCon 2007! September 25-28th. > http://www.astricon.net/ > > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Samsung Sprint CDMAoIP
> -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of C F > Sent: Friday, September 21, 2007 7:34 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] OT: Samsung Sprint CDMAoIP > > On 9/21/07, Alexander Lopez <[EMAIL PROTECTED]> wrote: > > > > > > Snip headers > > > On 9/20/07, Jason Parker <[EMAIL PROTECTED]> wrote: > > > > C F wrote: > > > > > AFAIK, the calls are free when you use it thru that device. Sprint > > > > > however charges $15 a month per phone or $30 for family plan. > > While I > > > > > agree that sprint should pay me for this, it's not as bad. > > T-mobile on > > > > > the other hand, does the same thing with wifi enabled phones, it > > > > > doesn't cost extra, and is completely free. > > > > > > > > > > > > > If you're referring to T-Mobile's "[EMAIL PROTECTED]" service, it's > > actually > > > $20 > > > > per month, per line on the account (unless it's changed very > > recently). > > > > > > > > > > I don't know about that, could be you are right. > > > > > > > As far as how it works on T-Mobile, I recently had some questions > > > answered by > > > > them about that.. They use UMA over wifi, and it does automatic > > > switching > > > > between the wifi and the gsm towers (ie; your call stays up). > > > > > > The same goes for Sprint. > > > > > > > > > > > Quote from the tech I talked to: > > > > "[EMAIL PROTECTED] does not use a VoIP protocol, as the voice data is > > > > transferred from the Internet directly to our UMA Gateway and then > > > > through our regular Mobile Switching Centers." > > > > > > I know it's a quote from the tech, but isn't it voice packets that > > > travels over the Internet (a packet switched network) instead of over > > > GSM (TDM switched network) which makes that statement incorrect? It > > > doesn't matter what the higher level protocol is, it's still VoIP. > > > > > Your right it is "STILL VoIP" by definition but its not... > > Yes it is, not only in definition but in practice as well. See below. > > > > > From: http://www.newstep.com/our%20market/technologies.asp > > > > Gateway-based Solutions > > By placing special gateways at the edge of a GSM network, Unlicensed > > Mobile Access (UMA) allows users with dual-mode handsets to access > > mobile phone services via both cellular and Wi-Fi links. In cellular > > mode, voice traffic travels over standard GSM radio waves. In Wi-Fi > > mode, an IP tunnel carries GSM traffic across the enterprise network > > and/or the Internet to a UMA gateway. The gateway looks like a base > > station controller (BSC) to the cellular network, so when a handset > > moves between cellular and Wi-Fi coverage, the network handles it as an > > ordinary BSC-to-BSC handoff. MSC emulation-also known as IP VLR-is > > similar to UMA, except that the gateway mimics a mobile switching center > > (MSC) and a visitor location register (VLR) instead of a BSC. > > > > Intimately tied to cellular technology and dual-mode handsets, > > gateway-based solutions provide access only to mobile network services > > and can be deployed only by facilities-based mobile network operators. > > Moreover, gateway-based solutions cannot leverage the full capabilities > > of IP and VoIP because all voice traffic remains in TDM format. Service > > When the above line is taken out of context (which is what I > understood from your response) then it could be said that it suggests > it's not VoIP. But really all the paper is saying is that to the > network (GSM) it doesn't look like VoIP but like TDM, since it's in > that format, and can therefore not take advantage of most VoIP > features. But it's still VoIP. > It's GSM tunneled over IP, some of the data traffic may or may not be voice traffic in the GSM traffic in the IP tunnel. Calling it VoIP would be a stretch. -- Eric Chamberlain, CISSP Chief Technical Officer Voxilla - http://voxilla.com/ ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Samsung Sprint CDMAoIP
> -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Jason Parker > Sent: Thursday, September 20, 2007 11:49 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] OT: Samsung Sprint CDMAoIP > > > If you're referring to T-Mobile's "[EMAIL PROTECTED]" service, it's actually > $20 > per month, per line on the account (unless it's changed very recently). > > As far as how it works on T-Mobile, I recently had some questions answered > by > them about that.. They use UMA over wifi, and it does automatic switching > between the wifi and the gsm towers (ie; your call stays up). > > Quote from the tech I talked to: > "[EMAIL PROTECTED] does not use a VoIP protocol, as the voice data is > transferred from the Internet directly to our UMA Gateway and then > through our regular Mobile Switching Centers." > > Pretty interesting stuff. > Interesting from a marketing and sales perspective that one can get people to buy a box, pay for the bandwidth used by the box, and then pay an extra $20/month per phone, all for coverage problems the carrier should address. But then again these carriers have managed to convince people to pay close to a thousand dollars per megabyte for SMS messages. -- Eric Chamberlain, CISSP Chief Technical Officer Voxilla - http://voxilla.com/ ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Samsung Sprint CDMAoIP
The device is a femtocell device; I would bet that they keep it in a format that works with their existing equipment, rather than use SIP. The device is also licensed for a specific frequency that is owned by the carrier, you wouldn't be able to use this device for any other purpose without their permission. The cell carriers are trying to get people excited about femotcell technology, so they can shift the cellular infrastructure costs off onto their customers. The traditional model is that the cell providers pay property and tower owners rent when they site an antenna. The cell providers don't even offer you a cheaper rate when you use the infrastructure you paid for. Notice how the device supports up to three simultaneous calls? You're even paying them to provide a better signal for your neighbors. -- Eric Chamberlain, CISSP Chief Technical Officer Voxilla - http://voxilla.com/ > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of C F > Sent: Wednesday, September 19, 2007 6:57 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [asterisk-users] OT: Samsung Sprint CDMAoIP > > http://gizmodo.com/gadgets/cellphones/sprintsamsung-instant-cell+to+wi+fi- > box-is-official-named-airave-300451.php > > The above is quite interesting, it would be interesting to see if it > uses sip, which I have no reason to believe otherwise, and if it does, > can it be hacked to talk to Asteirsk? In which case one could have a > very good extension to asterisk using any Sprint Cell phone, or maybe > even any CDMA (Verizon) cell phone as well. > > ___ > > Sign up now for AstriCon 2007! September 25-28th. > http://www.astricon.net/ > > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Another State Of The Punctuation Mark question - Vonage
Vonage has a "business" offering, but they aren't really structured to provide business quality support. I wouldn't use them for a business. For several years now, we've used VoicePulse Connect <http://connect.voicepulse.com/> for our Asterisk IAX and SIP trunks. Ravi and KP are both technical guys and know Asterisk extremely well. -- Eric Chamberlain, CISSP Chief Technical Officer Voxilla - http://voxilla.com/ > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Jay R. Ashworth > Sent: Tuesday, September 11, 2007 5:57 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [asterisk-users] Another State Of The Punctuation Mark question - > Vonage > > There was a flurry of "Vonage is going to unlock SIP" activity last > year; did anything productive ever come of it? > > Are *you* using your Vonage lines directly into Asterisk? > > In lieu of that, for a 4 line small business that doesn't need to pay > Vonage $150 a month, who? Broadvoice? Someone else? > > I'm a touch unimpressed with the fact that BV's website *won't quote > you BYOD pricing* until you actually place the damn order -- or so it > appears to my eyes. > > 727. > > Cheers, > -- jra > -- > Jay R. Ashworth Baylink > [EMAIL PROTECTED] > Designer The Things I Think RFC > 2100 > Ashworth & Associates http://baylink.pitas.com '87 > e24 > St Petersburg FL USA http://photo.imageinc.us +1 727 647 > 1274 > > ___ > > Sign up now for AstriCon 2007! September 25-28th. > http://www.astricon.net/ > > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T1 to SIP conversion, standalone device
There are a number of T1/PRI gateway manufacturers. Audiocodes and Mediatrix are both manufacturers that our customers have had success with. Audiocodes tends to be a much more expensive modular solution. If you only need one T1 look at the Mediatrix 3531 (street price is around $2,100). <http://www.google.com/products?q=med3531dg&rls=com.microsoft:en-us:IE-SearchBox&ie=UTF-8&oe=UTF-8&sourceid=ie7&rlz=1I7HPIC&um=1&sa=N&tab=wf> -- Eric Chamberlain, CISSP Chief Technical Officer Voxilla - http://voxilla.com/ _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michelle Dupuis Sent: Friday, September 07, 2007 11:31 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] T1 to SIP conversion, standalone device Over a year ago I saw a discussion about a standalone device which converted a T1 in/out to SIP in/out (over 10/100 LAN). Anyone recall what this device is? (I'm looking for a standalone device - not a PCI card). Thanks ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wireless VOIP Keysets? Recommendations?
I haven't come across any wireless devices that support IAX2, but we have successfully used the Linksys WIP300, Linksys WIP330, Nokia N80, Nokia E61i, and Nokia N95 with asterisk. If you just need wireless and not mobility, the Linksys WBP54G also works well to interconnect Ethernet based VoIP phones. -- Eric Chamberlain, CISSP Chief Technical Officer Voxilla - http://voxilla.com/ > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of William Stillwell (Ki4swy) > Sent: Sunday, September 02, 2007 5:45 PM > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] Wireless VOIP Keysets? Recommendations? > > Any Recommendations on a "Good" Wireless Voip Keyset that works well with > Asterisk? > > I would prefer one that is IAX2 as it works better behind a Nat'd > Firewall.. > > But I am reaching out to you guys as you all would know what would work > the best :-) > > > > > > > Sent via the WebMail system at kotbh.net > > > > > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Nokia cell connected to Asterisk
Using the phone itself as a GSM-SIP gateway is not possible with the native VoIP application, but it looks like it should be possible with a custom application for the phone. -- Eric Chamberlain, CISSP Chief Technical Officer Voxilla - http://voxilla.com/ > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Remco Barendse > Sent: Monday, August 20, 2007 11:22 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Nokia cell connected to Asterisk > > Has anyone ever tried using a Nokia phone with SIP client as channel for > Asterisk? I mean i would like to receive calls to the mobile on > asterisk and use the Nokia phone to place calls to cell destinations. > > I have enough Nokia E60's to do that and it would circumvent the need for > chan_bluetooth or something similar!! :) > > > On Mon, 20 Aug 2007, Steve Totaro wrote: > > > Well chan_bluetooth is really amazing (especially if your phone does not > > support SIP). > > > > You connect your phone via bluetooth to your asterisk box and it becomes > > a channel type. You can use it as an extension(FXS) or a phone line > > (FXO). I believe you can send and receive SMS through the > > phone/Asterisk as well. > > > > Chan_bluetooth README is in the asterisk-addons trunk and gives you > > basic instruction on setting it up. > > > > You get several added pieces of functionality with this setup. SMS send > > and receive through your phone using Asterisk?, FXO failover or LCR, FXS > > where your cell phone becomes an extension. > > > > Thanks, > > Steve > > > > Jonathan GF wrote: > >> Thanks Steve and Mitcheloc, > >> > >> in fact i was think in something more "obsolet" like connect via > >> serial/usb cable the cell to the asterisk box. Never thought in the > >> SIP stack of new Nokia's but i will start looking for info about this. > >> If you [Steve] know of a good written material of interest please let > >> me know. > >> > >> Probably Mitcheloc is right too, there are a lot of manners to achieve > >> this and the problem is mine that i don't know how to search what i > >> want. Anyway, thank you for your inputs. Any others will be welcomed, > >> for sure. > >> > >> Regards, > >> > >> Jonathan GF > >> > >> > >> > >> On 8/20/07, *mitcheloc* <[EMAIL PROTECTED] > >> <mailto:[EMAIL PROTECTED]>> wrote: > >> > >> Jonathon, > >> > >> Are you talking about using the built in SIP client on some Nokia > >> phones? I'm using an E90 with Asterisk and it works very well. I > used > >> Google for help and it returned plenty of results. > >> > >> Cheers, > >> Mitchel > >> > >> On 8/19/07, Steve Totaro <[EMAIL PROTECTED] > >> <mailto:[EMAIL PROTECTED]>> wrote: > >>> If it is bluetooth and you don't mind running Asterisk 1.4 > >> trunk, you should look at chan_mobile. > >>> > >>> Thanks, > >>> Steve Totaro > >>> > >>> > >>> > >>> From: [EMAIL PROTECTED] > >> <mailto:[EMAIL PROTECTED]> on behalf of > >> Jonathan GF > >>> Sent: Sun 8/19/2007 6:26 PM > >>> To: asterisk-users@lists.digium.com > >> <mailto:asterisk-users@lists.digium.com> > >>> Subject: [asterisk-users] Nokia cell connected to Asterisk > >>> > >>> > >>> Hi folks, > >>> > >>> i've been looking for in many sources but i cannot see clear if > >> the options i'm chasing is feasible with Asterisk. I understand > >> that should be. > >>> > >>> I would like to connect a nokia cell to Asterisk but i don't > >> know how exactly. > >>> > >>> Any ideas, inputs, docs or refs will be welcomed. > >>> > >>> Thanks in advance. > >>> > >>> Jonathan GF > >>> > >>> > >>> ___ > >>> --Bandwidth and Colocation Provided by http://www.api-digital.com- > - > >>> > >>> asterisk-users mailing list > >>
Re: [asterisk-users] Nokia cell connected to Asterisk
On the SIP side of things, we have a how-to guide for the Nokia E series and Asterisk. <http://voxilla.com/voxilla-stories/voxilla-how-to-guides/using-the-nokia-e-series-phones-with-asterisk-865.html> -- Eric Chamberlain, CISSP Chief Technical Officer Voxilla - http://voxilla.com/ _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jonathan GF Sent: Monday, August 20, 2007 4:05 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Nokia cell connected to Asterisk Thanks Steve and Mitcheloc, in fact i was think in something more "obsolet" like connect via serial/usb cable the cell to the asterisk box. Never thought in the SIP stack of new Nokia's but i will start looking for info about this. If you [Steve] know of a good written material of interest please let me know. Probably Mitcheloc is right too, there are a lot of manners to achieve this and the problem is mine that i don't know how to search what i want. Anyway, thank you for your inputs. Any others will be welcomed, for sure. Regards, Jonathan GF On 8/20/07, mitcheloc <[EMAIL PROTECTED]> wrote: Jonathon, Are you talking about using the built in SIP client on some Nokia phones? I'm using an E90 with Asterisk and it works very well. I used Google for help and it returned plenty of results. Cheers, Mitchel On 8/19/07, Steve Totaro <[EMAIL PROTECTED]> wrote: > If it is bluetooth and you don't mind running Asterisk 1.4 trunk, you should > look at chan_mobile. > > Thanks, > Steve Totaro > > > > From: [EMAIL PROTECTED] on behalf of Jonathan GF > Sent: Sun 8/19/2007 6:26 PM > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] Nokia cell connected to Asterisk > > > Hi folks, > > i've been looking for in many sources but i cannot see clear if the options > i'm chasing is feasible with Asterisk. I understand that should be. > > I would like to connect a nokia cell to Asterisk but i don't know how exactly. > > Any ideas, inputs, docs or refs will be welcomed. > > Thanks in advance. > > Jonathan GF > > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > <http://lists.digium.com/mailman/listinfo/asterisk-users> > http://lists.digium.com/mailman/listinfo/asterisk-users > > -- Mitchel Constantin Snap - A desktop user interface for Asterisk www.snapanumber.com <http://www.snapanumber.com> ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk, PAP2T and 2Wire DSL router
Carlos, Do you have the NAT keepalive options enabled on the PAP2T? It sounds like the router is timing out the connection and dropping the port mapping. -- Eric Chamberlain, CISSP Chief Technical Officer Voxilla - http://voxilla.com/ > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Carlos Chavez > Sent: Thursday, August 16, 2007 9:09 AM > To: Asterisk > Subject: [asterisk-users] Asterisk, PAP2T and 2Wire DSL router > > Here is Mexico the phone company uses a DSL router from 2Wire which > in > my opinion is quite bad. I am having problems getting PAP2T adapters > connected to Asterisk using these routers. They connect fine but after > about 5 minutes I get a message on the Asterisk console that the ATA is > unreachable. So far the only way I have found for the ATA to stay > connected more than five minutes is to put it in the DMZ but this is not > always possible either because the DMZ is already in use by some other > computer, they do not use DHCP and this router cannot assign a DMZ if > you do not use DHCP or simply because you have more than one ATA on the > network. > > Does anyone know of a tweak to this router that will allow the ATA > to > remain connected to Asterisk? This only seems to affect Linksys and > Sipura ATAs because when you have phones on the same network > (Grandstream) those do not lose the connection to the server. > > -- > Telecomunicaciones Abiertas de México S.A. de C.V. > Carlos Chávez Prats > Director de Tecnología > +52-55-91169161 ext 2001 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Patent issues, what features we can't use?
Shouldn't you ask your attorney these questions? Any answers you receive here will not legally protect you. -- Eric Chamberlain, CISSP Chief Technical Officer Voxilla - http://voxilla.com/ _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Zeeshan Zakaria Sent: Monday, August 13, 2007 7:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Patent issues, what features we can't use? Hi everybody, As the Asterisk community is getting larger and larger, I was wondering that the features which are provided in Asterisk and are programmed by the open source community under GPL, or GUIs like FreePBX which also come loaded with wonderful features and uses same Asterisk, are they anywhere violating any patent laws? Most of the features work the same way as Nortel, Avaya and other PBX systems. Is there anyone who owns these features and will come one day to claim his royalties? When I deploy an asterisk soultion for a customer, is there any violation of any patent or copyright laws anywhere? Of if I use my own Asterisk server to provide services to some customers, am I violating any patent laws by not paying the royalties to some patent owners? I heard people saying that IVR technology is patented and google search for patents also say so. But we all are using IVR for ourselves and our customers without paying royalties to anyone. But when it comes to using g729, all of a sudden royalty issue comes in. So what is right to use and what is not? -- Zeeshan A Zakaria ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT Provisioning http-server-enabled devices (Was: Siemens Gigaset DECT base provisioning)
What you describe is doable; we have a number of device configuration wizards. But it is generally easier to use the device's bulk provisioning methods, like https an XML configuration file to the device. The provisioning settings a pretty standard and don't change very often. The problem with using the user web interface is that the manufacturers quite often change the interface with new firmware releases, so you are constantly updating the scripts. -- Eric Chamberlain, CISSP Chief Technical Officer Voxilla - http://voxilla.com/ _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olivier Sent: Friday, August 10, 2007 3:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] OT Provisioning http-server-enabled devices (Was: Siemens Gigaset DECT base provisioning) hello, I would to define and unattended process to configure devices which are http-server-enabled, use DHCP but do not use TFTP-DCHP to configure themselves during boot. Has anyone worked on such subject ? I was thinking of something like : populating configuration file from device web pages (rendering this as generic and flexible as possible) writing a script which reads this file and set each parameter using http writing a script which monitors network environment to trigger previous when certain events occur. All this is not very clear for me, yet. Regards ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Login info from Active directory
Active Directory relies on Kerberos for authentication. Kerberos uses tickets and does not centrally track presence. You would either have to parse all the Domain Controller logs or use some other method to determine when a user is logged in, such as setting a flag with login/logout scripts. -- Eric Chamberlain, CISSP Chief Technical Officer Voxilla - http://voxilla.com/ _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olivier Sent: Monday, August 06, 2007 10:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Login info from Active directory Hello, Is it easy to retrieve user presence from Asterisk dialplan according Active directory data ? I mean how do you know a user is logged reading data from Active directory ? best regards ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware that can ring my phone?
A phone system for under $100 is asking a lot.
It can be done, but what is your time worth.
You might want to consider some other phone system if all you need is
IVR and analog support or look at hosted solutions.
--
Eric Chamberlain, CISSP
Chief Technical Officer
Voxilla - http://voxilla.com/
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Linux Lover
> Sent: Wednesday, August 01, 2007 8:30 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Hardware that can ring my phone?
>
> Yes, you understood correctly. Thank you - and all
> others who replied so quickly - for your precise and
> guiding answers.
>
> The Digium TDM11B looks looks like the perfect match
> for me:
>
> http://www.telephonyware.com/telephonyware/tw00068.html
>
> But one thing that I forgot to mention is that my
> business is only in its beginning stage and I need to
> be as thrifty as possible. While $216 is a reasonable
> price, I was wondering whether my (currently very
> modest) goal can be achieved by spending much less
> (under $100). For example, what if I buy one of those
> el-cheapo PBX boxes and connect it to an Asterisk
> server?
>
> http://www.soho-pbx.com/sp-104.htm
>
> Do you think this could work for me or did I expose a
> gross misconception on my part?
>
> Thanks,
> Lynn
>
> --- john beaman <[EMAIL PROTECTED]> wrote:
>
> > Lynn,
> >
> > If I understand you question correctly, you would
> > need:
> >
> > A computer (preferably a server) to run Asterisk
> > An analog interface card such as the Digium TDM400P
> > An analog phone line (POTS)
> > An analog (real) phone
> >
> > Calls would come in on the POTS line, get answered
> > by Asterisk. Callers would hear your voice menu,
> > and input their choice. If they opted for a live
> > person, asterisk would then send the call to your
> > analog (real) phone.
> >
> >
> >
> > John Beaman
> > Telecom Specialist
> > Voice Telecommunications Services Department.
> > Good Samaritan National Campus
> > 605-362-3331
> >
> > >>> [EMAIL PROTECTED] 8/1/2007 8:48:47 AM
> > >>>
> > Hello,
> >
> > I am a small business owner in need for a solution
> > that automatically answers an incoming call, prompts
> > the caller via touch-tone menu ("press 1 to leave a
> > message, press 0 to speak to a representative") and
> > will ring my (real) phone ONLY if requested by
> > caller.
> >
> > I know that Asterisk is capable of all the logic
> > behind what I described above. However, I couldn't
> > find a hardware product that will allow me to
> > accomplish the above (preferrable using Asterisk
> > software). Does such thing exists?
> >
> > Thanks,
> > Lynn
> >
> >
> >
> >
>
__
> __
> > Sick sense of humor? Visit Yahoo! TV's
> > Comedy with an Edge to see what's on, when.
> > http://tv.yahoo.com/collections/222
> >
> > ___
> > --Bandwidth and Colocation Provided by
> > http://www.api-digital.com--
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >
> >
> http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> > > -
> >
> > This email transmission and any documents, files or
> > previous
> >
> > email messages attached to it may contain
> > information that is
> >
> > confidential or legally privileged. If you are not
> > the intended
> >
> > recipient, you are hereby notified that any
> > disclosure, copying,
> >
> > printing, distributing or use of this transmission
> > is strictly
> >
> > prohibited. If you have received this transmission
> > in error,
> >
> > please immediately notify the sender by telephone or
> > return
> >
> > email and delete the original transmission and its
> > attachments
> >
> > without reading or saving in any manner.
> >
> >
> >
> > The Evangelical Lutheran Good Samaritan Society.
> >
> >
> -
> > > ___
> > --Bandwidth and Colocation Provided by
> > http://www.api-digital.com--
> >
> > ast
Re: [asterisk-users] Asterisk Vm functionality question
Andrew, Could you elaborate on how you configure the MWI of the mobile device to use asterisk voicemail? -- Eric Chamberlain, CISSP Chief Technical Officer Voxilla - http://voxilla.com/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Vm functionality question
Yes it's possible. It's also possible to have Asterisk try and find the person in the field and either connect the call or deliver the message. -- Eric Chamberlain, CISSP Chief Technical Officer Voxilla - http://voxilla.com/ _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James R. Stevens Sent: Wednesday, July 25, 2007 10:21 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk Vm functionality question Going over the needs of any PBX that replaces our current system (working toward Asterisk) and have VM functionality question. Currently when someone leaves a voice mail for a sales person (Who is in the field) the system takes the VM and then in turn dials over a POTS line and pages the sales person notifying them of a VM (Does not deliver the message-just notifies) Is this possible with Asterisk? 14 Channel PRI straight into a Sangoma T1/E1 card -- This message has been scanned for viruses and dangerous content by <http://www.athensdistributing.com/> Athens Hyperion Scanner, and is believed to be clean. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
