Re: [asterisk-users] Interesting new hack attack
On 5/22/2014 12:41 PM, Steve Murphy wrote: So, these defenses can be employed to stop/ameliorate such hacking efforts: 1. Keep your phones behind a firewall. Travellers, beware! Never leave the default login info of the phone at default! 2. Never use the default provisioning URL for the phone, with it's default URL or password. 3. Use fail2ban, ossec, whatever to stymie any brute force mac address searches. 4. Use your firewalls to restrict IP's that can access web, ftp, etc, for provisioning to just those IP's needed to allow your phones to provision. 5. Keep your logs for a couple years. 6. Change your phone SIP acct passwords now, if you haven't implemented the above precautions yet. If I missed a previous post on this, forgive me. Just thought you-all might appreciate a heads-up. Encrypt your provisioning system if the phone supports it. I had a cable/voip service provider who HTTPS provisioned by MAC without encryption and the provisioning URL was stored, unlocked, in the ATA. Had I been slightly more nefarious, I could have walked the the provisioning tree nice and slow and easily grabbed everyone's SIP credentials in the clear. No hacking or cracking was involved. The ATA doubled as the NAT router they handed out and gave the admin password out freely. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CBAnn channel not going away in Asterisk 12
On 5/1/2014 10:38 AM, Richard Kenner wrote: Please go ahead and open an issue and attach the refs log and the full DEBUG log. That will allow us to understand what's occurring here. I need to wait until I'm sure this isn't something I caused somehow, so I need to first understand why I'm seeing this and nobody else is. I had seen it as well but just chalked it up to not grokking how the CBAnn channels worked. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Open Source Asterisk Polling Solution
On 4/23/2014 12:20 AM, Nick Cameo wrote: That's about as simple as it gets. A call file that goes to the dialplan. A dialplan that consists of Read (which would play the message) followed a GotoIf into a mailbox (either voicemail or Dial() to an external number). One hint for doing unattended dialing like this, make sure you're dialing using a SIP or other digital method rather than, say, out an analogue port that doesn't have decent answer detect. And you can't just dump a whole bunch of call files into the system at once, you'll need to meter them out based on the number of concurrent outbound calls your provider will allow. Hello James, Good to see you here, and thank you very much. Though my basic idea of how it will look using call files and dialplan is like what you and others on here have pointed out. Yes, we are using SIP for both origination and termination (just helping my friend use some of our accounts used for PBX, and prepaid). I have been using * for many years now however, never for call center/predictive dialer type processes. Once I have got this thing to call out and get calls coming in. It would be nice to write to a database all the users that press option on. I have a strong Java, PHP and SQL background. Will probably need to make a call using AGI or such? N. You can go AGI, but there are direct ODBC handles available in the dialplan if you build Asterisk properly with the ODBC resources enabled. That'd my personal preference from a performance standpoint. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Open Source Asterisk Polling Solution
On 4/22/2014 5:54 PM, Nick Cameo wrote: Hello Everyone, Thank you all for your response. The people I am doing it for run a non-profit charity, and are legally able to reach out to their customers. I will wire it up to the DNC however, for starters, I would like to get asterisk to: i) Iterate through a list of numbers ii) Play a pre-recorded message asking if they have waste they need picked up iii) If they press one, forward the call to mailbox That's about as simple as it gets. A call file that goes to the dialplan. A dialplan that consists of Read (which would play the message) followed a GotoIf into a mailbox (either voicemail or Dial() to an external number). One hint for doing unattended dialing like this, make sure you're dialing using a SIP or other digital method rather than, say, out an analogue port that doesn't have decent answer detect. And you can't just dump a whole bunch of call files into the system at once, you'll need to meter them out based on the number of concurrent outbound calls your provider will allow. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Open Source Asterisk Polling Solution
On 4/21/2014 1:47 PM, Mitul Limbani wrote: Use vicidial for achieving the same. Or call files (or AMI originate), a short bit of dialplan logic, and maybe a call to Queue(). -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Open Source Asterisk Polling Solution
On 4/21/2014 3:58 PM, Nick Cameo wrote: On Mon, Apr 21, 2014 at 2:01 PM, James Sharp ja...@fivecats.org mailto:ja...@fivecats.org wrote: On 4/21/2014 1:47 PM, Mitul Limbani wrote: Use vicidial for achieving the same. Or call files (or AMI originate), a short bit of dialplan logic, and maybe a call to Queue(). This is a nice and easy solution however, I do not know where to begin. Can you gents kindly elaborate or point us to the right directions (ie, howto tutorials) Asterisk call files: http://www.microalcarria.com/descargas/documentos/Linux/varios/Asterisk/asteriskdocs-docbook/docs-html/x1512.html (Replace Channel with dial commands to your telephony provider, change the context and extension with an appropriate). Dialplan: You'll probably just need some Playback and Read commands to play your message and get the digit response. Agents and Queues: http://www.voip-info.org/wiki/view/Asterisk+call+queues You'll probably want to use static/fixed agents unless you have a whole bunch of agents logging in and out. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Numbers hackers call
On 3/26/2014 12:20 PM, Michelle Dupuis wrote: If this is to 972 area code then the next digits should be 0X or 0XX but they are not. This differs from what I found documented for that area code - I thought someone from the region might add to the discussion. Not sure if this reflected a premium service etc. (But someone jumped in with an explanation) 0X or 0XX is only if you're in country and need to dial with the 0 national trunk code (much like dialing 1+ in the US for an in country but long distance call). Someone dialing from outside the country doesn't need to add the zero, so they just use the 972 country code + 59 prefix. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Which is more efficient for 1 to many broadcasting?
Putting a whole bunch of people into a listen-only/muted Confbridge conference or getting the broadcaster audio into a MOH class and then just having callers attach to that MOH class? Does the the muted side of a Confbridge Room still try to mix in audio from the muted channels or does it just disregard those channels and only run mixes against unmuted channels? Now, if the answer is MOH is more efficient, can someone suggest a way for a channel to be the source of a MOH class? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which is more efficient for 1 to many broadcasting?
On 3/18/2014 6:58 PM, Paul Belanger wrote: On Tue, Mar 18, 2014 at 1:02 PM, James Sharp ja...@fivecats.org wrote: Putting a whole bunch of people into a listen-only/muted Confbridge conference or getting the broadcaster audio into a MOH class and then just having callers attach to that MOH class? Does the the muted side of a Confbridge Room still try to mix in audio from the muted channels or does it just disregard those channels and only run mixes against unmuted channels? Now, if the answer is MOH is more efficient, can someone suggest a way for a channel to be the source of a MOH class? What sort of channel count are you looking for? We did some load testing recently and found less people in a bridge is better then more. Audio source location didn't really matter much. A few hundred to start with, but as with everything, I'd like to scale up as far as I can. And, of course, it makes sense that less people in a bridge is better than more but that's not quite what I'm asking. Is it more efficient to have, for example, 701 people in a confbridge room (700 muted users + 1 person yapping) or to have 700 people dialed in and just running the MusicOnHold application with said person yapping away via some audio source. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Host = Dynamic in a Register Free Setup
On 2/18/2014 2:09 PM, Eric Wieling wrote: No. Asterisk will accept calls from unregistered devices, but you have to enable guests I sip.conf and hope your dialplan is secure. No sane person does this. Asterisk cannot send calls to a device unless it knows the address from a register or from a host= entry for the peer. You may not like it, but this is the way Asteirsk has worked for the past 15 years. Isn't there also autocreatepeer? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] answering machine screening with MixMonitor
On 2/5/2014 12:09 PM, G. Paul Ziemba wrote: I'm using asterisk 1.8 as an answering machine. I'd like to hear the calls it answers aloud in case I want to pick up and interrupt the call. There are a few articles describing, for example, three-way calling a monitor phone set to auto-answer, but I couldn't find anything that described how to just send the audio to a local speaker. A local speaker connected to the Asterisk box itself? Console channel driver, chan_alsa (or chan_oss for old drivers). You'll probably end up with kind of a Rube Goldbergish approach, probably something involving ChanSpy or a conferencebridge to take the place of mixmonitor. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No compatible codecs, not accepting this offer!
On 1/15/2014 3:59 AM, Francesco Namuri wrote: Hello, I'm having this issue on my pbx, it appears that asterisk is refusing the codecs that my providers is proposing. My trunk configuration is: Pretty simple - --- username=5x5x7x9x0x3 type=friend secret=CRcxn7sqwm qualify=yes port=5060 insecure=port,invite host=sip.txtxlxoxp.it fromuser=5x5x7x9x0x3 fromdomain=sip.txtxlxoxp.it disallow=all context=from-trunk allow=alaw Here you're disallowing all codecs except alaw. --- A typical invite from my provider is: --- SIP read from UDP:xx.yy.xx.yy:5060 --- INVITE sip:5x5x7x9x0x3@192.168.1.168:5060 SIP/2.0 Via: SIP/2.0/UDP xx.yy.xx.yy:5060;branch=z9hG4bKt5sfh7nrvok3d5gqc3ritdv7b7 From: sip:3x8x6x3x3x@10.39.1.19;user=phone;tag=SDdgce901-90915 To: SIPLineUser SIPLineUsersip:5x5x7x9x...@sip.txtxlxoxp.it Call-ID: SDdgce901-9cb68ba025684f03a4094ed71e6e04f8-ao92gd1 CSeq: 59458 INVITE Content-Type: application/sdp Contact: sip:3x8x6x3...@xx.yy.xx.yy:5060;user=phone;transport=udp User-Agent: Nortel SESM 14.1.0.12 Max-Forwards: 19 Supported: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec,join,x-nortel-sipvc,gin,com.nortelnetworks.im.encryption,replaces,100rel Remote-Party-ID: sip:3x8x6x3...@sip.txtxlxoxp.it;user=phone;screen=yes;screen-ind=0;party=calling;counter=0;npi=NPI_E164;ton=TON_NATIONAL P-Asserted-Identity: sip:3x8x6x3x3x@10.39.1.19;user=phone Allow: UPDATE,REFER Content-Length: 293 v=0 o=- 0 138163748 IN IP4 xx.yy.xx.yy s=IMSS e=unkn...@invalid.net c=IN IP4 xx.yy.xx.yy t=0 0 m=audio 43718 RTP/AVP 8 18 3 101 a=fmtp:18 annexb=no a=rtpmap:18 G729/8000 a=rtpmap:3 GSM/8000/1 Butprovider will only send GSM or G729. So either you need to talk your provider into sending alaw or you need change your allow line to allow=alaw,gsm. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No compatible codecs, not accepting this offer!
On 1/15/2014 5:50 AM, Gareth Blades wrote: On 15/01/14 09:39, Francesco Namuri wrote: Hello James, thanks for your answer, I supposed this too, but my provider answered me that as m=audio 43718 RTP/AVP 8 18 3 101 ^ ^ ^ GSM proposal ^ ^--- G729 proposal ^-- aLaw proposal And that a=rtpmap:18 G729/8000 proposed as media conversion a=rtpmap:3 GSM/8000/1 because the call is made by a mobile I would agree with what your service provider has said. If you look at the RFC http://tools.ietf.org/html/rfc4566#section-5.14 the '8 18 3 101' parameters are a list of media formats. The first is the one which should be used but (preferred choice) but the other may be used. Numbers in the range 96-127 are dynamic payload types and these must have a corresponding 'a=' line specifying the payload type and the codec options. Lower numbers have static payload assignments and according to that RFC dont have to have corresponding 'a=' lines. A list of types can be found at http://www.iana.org/assignments/rtp-parameters/rtp-parameters.xhtml However in all SIP traces I have seen there has always been a 'a=' line for every payload type offered. The static payload type numbers are used but there is still the 'a=' line. I missed the RTP/AVP line. I'll go back to lurking for a bit :) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk AMI - PHP or Node.js?
On 12/28/2013 7:04 AM, Shahid H wrote: Thanks Daniel, that was useful, I will check those links :) I am pretty good with PHP and jQuery. So I guess learning Node.js shouldn't be too difficult. If I decided to use Node.js - what is the best way to communicate with a browser to AMI process? Send a XML or HTTP command from a browser to AMI process .. or whatever I execute on the browser - it save the commands to the database.. a process will listen the commands from a database. Other options are http requests and do the ipc... or maybe WebSocket? My code will probably end up on Thedailywtf.com for it, but I have a PHP-based system that uses PAMI to talk to the AMI interface. There's a daemon process that listens for commands via a FIFO buffer. Several other processes web scripts just send commands to the controller process via that FIFO. Seems to work well for me. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] link to MySQL connection
On 12/3/2013 10:11 AM, Don Kelly wrote: In the php routines, I would like to use the persistent connection that is established in the dialplan, rather than creating a new connection each time they run. How can I do this? You can't, they are completely separate processes and code. Joshua Colp Thanks--that's not the answer I wanted, but it sure was quick. :) Is there anything that would enable me to use a persistent connection in the agi? --Don Yes. Use func_odbc in your PHP AGI. In Asterisk dialplan functions are treated like dialplan variables so you can get and set them just like you would other dialplan variables. If it takes 5 seconds to open a PDO DB connection inside PHP you have some OTHER problem. Eric Wieling Thanks, Eric. If I can't figure out why the connection takes so long, I'll try the func_ODBC approach. look into DNS problems. Something may be trying to do an reverse DNS lookup and taking too long to either give up or get an answer. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Amazon, Asterisk and reliability beyond a hobby system?
On 11/24/2013 2:47 AM, Todd R. wrote: Did you have the externalip setting in sip.conf set to the Elastic IP? I believe I did. But I didn't really get a chance to plow into it too much, I had a client holding me at gunpoint. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Amazon, Asterisk and reliability beyond a hobby system?
On 11/22/2013 12:52 PM, Todd R. wrote: Just checking one more time to see if anyone has an opinion on this. I am primarily interested in using a cloud type setup such as Amazon AWS for the redundancy, easy backup and recovery options. It's not about price but the idea that it will be very hard for a single piece of hardware to ruin my day. I have only one small datapoint. I ran an EC2 microinstance with Asterisk and a dozen offboard users. The only problem I had was SIP wasn't dealing well with the Elastic IP one-to-one NAT that Amazon uses. I had the usual Asterisk/NAT issues of one-way audio. I eventually moved from Amazon to Linode to get away from the NAT issues. Once I did that, everything worked fine, but again it was only a dozen users. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Interconnect Radio units using asterisk
On Aug 26, 2013, at 8:11 PM, Manolo Quijano manolo.quij...@gmail.com wrote: Hi all, This my first mail in the community. My name is Manolo. I'm new in Asterisk. My objective is to control some radio using Asterisk via web. In google I could see that the application app_rpt had this goal, but currently is not in the new Asterisk 1.8 or higher. Thinking in future, I thought the easer way to do that is to buy some RoIP unit and do the communication with asterisk via SIP or IAX2. Someone with experience in these kind of units, could give me some link or P/N of some equipmet that work fine with current version of Asterisk? Thanks in advance, Manolo Hi Manolo, While not specifically RoIP boxes, I've used the Multitech MVP-200 and 400 series VOIP gateways to interface radio systems over SIP. The MVP series boxes come in a model that supports 4-wire EM interfaces which are pretty common in radio systems. I programmed the units to auto dial a SIP address when the EM interface was seized. They didn't specifically use Asterisk, but I don't see why you couldn't stick Asterisk somewhere in there since SIP is SIP. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] autoanswer
On 07/10/2013 01:04 PM, bilal ghayyad wrote: Hello; To let the Phone answer automatically, this can be configured from asterisk (at the sip.conf for the phone)? Or it has to be from the IP Phone? Because, some phones does not support auto answer, also we do not need to do it for each Phone. Depends on the phone. Some phones you have to set auto answer on the phone itself, other phones can be told to auto answer via configuration file, and yet other phones can be told to auto answer if you send them the right SIP headers via the dialplan. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] moving calls from one E1 to another
Yes, you will be able to transfer calls between the E1 ports without a problem. As far as timing goes, the PBX will not be a clock source. You need to configuring your timing setup so that Asterisk takes timing from the Telecom PRI and then sources clock to the PBX. The PBX will take timing from Asterisk. Sent from my iPad On Jun 24, 2013, at 12:17 PM, Dimitar Dimitrov ddimit...@consult.bg wrote: Hello everyone. I want to migrate an old PBX which uses the Е1-PRI from one Telecom to VoIP by transparently moving the numbers one by one. I mean that the numbers that the PBX handle must be transparently moved from one operator to another. The old connection to the PBX is Е1-PRI and we must preserve that because no one knows how to configure this PBX. So my idea is to connect a PC with 2 ports E1 module between the PBX and the old telecom. One port of the module will be connected to the telecom's wire and the other port will be connected to the PBX. This PBX will be powered with Asterisk of course so it will be able to connect by SIP trunk to the alternative VoIP telecom. So my question is will I be able to transfer the calls from one E1 port to other? Could I be able to specify the difference clock source for the both ports? Thanks in advance. Dimitar -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PCI Passthrough of T1 cards
On Jun 16, 2013, at 4:27 PM, Nick Khamis sym...@gmail.com wrote: Anyone try this? I saw a post here: http://www.elastix.org/index.php/en/component/kunena/1-installation-issues/94041-setup-of-sangoma-a101-in-my-elastix.html But not sure if it's possible. What I am asking is if there are any T1 cards with virtual functions implemented in their drivers to allow pci-passthrough? PCI pass through is a function of the virtual machine's host system, not with the t1 card drivers. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Impromptu conferencing
On 11/7/2012 2:01 PM, martin f krafft wrote: Dear list, we would really like to be able to invite a third and fourth party to our current one-on-one call. At the moment, we have to agree to dial into MeetMe 10 minutes later, then make calls to the third parties, and hope it all works out. I have found a couple of examples on the Internet for converting channels into conferences, but I could not get any of them working. Does anyone have a working example they would be willing to share? Why not blind transfer people to a conference extension and then dial into it yourself? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Failover router recommendation
On 10/9/2012 3:52 PM, Niccolò Belli wrote: http://www.traverse.com.au/geos21-dual-adsl2-x86-router-appliance I achieved fallback in less than 10 seconds flushing routing cache and nat tables with nearly zero false positives (I can do even better but I prefer having less false disconnections). I don't use this router but a Traverse Solos PCI Adsl2+ card and a linux box. Do you have your phones set for a short register time? Otherwise the far end might have stale contact information to send incoming calls back to. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call me now outbound calls in a queue
On 9/28/2012 12:42 PM, Mitch Claborn wrote: I want to put a call me now button on the web site that will place the request into an asterisk call queue and then when an agent picks up the call in the queue, place the outbound call to the customer. The following AMI command works, but it calls the customer first, before an agent is necessarily available. Action: Originate Channel: SIP/voipms/customer_number_here Context: external Async: true Application: Queue Data: sales Callerid: Company 8005551212 How can I get an available agent before the customer call is placed? Use AMI to do a queue status. In the status reply, you'll get a list of all the agents and their status (Not in use, in use, busy, unavailable, or ringing). -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trigger Asterisk after data inserted in mysql
On 9/18/2012 3:41 PM, Ahmed Munir wrote: Hi all, I would like to know, is there a way to trigger Asterisk after data inserted into mysql DB? Like here what I'm trying to do, when the new data inserted into MySQL DB, it sends the request to Asterisk along with the new data (that is inserted in DB) for making outbound call i.e. Realtime. Currently I've set a cron job that execute my script every 30 seconds and checks for a new data in DB. If new data is inserted in 30 seconds that script will run and sends the data to Asterisk for making calls. (This is the case which I'm thinking to avoid) Please advise. You could create a trigger in mysql that calls a shell script that pokes Asterisk properly. Look here for a start: http://forums.mysql.com/read.php?99,170973,236208#msg-236208 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] kernel: dahdi: Master changed to TE2/0/2 --- Is a normal message
On 9/13/2012 2:31 PM, equis software wrote: After exchanging the cable with other equipment was running smoothly in my computer the problem persisted while the other team the cable that could be bad worked. With this test done, now suspect the problem I have it on the Digium card. I perform the test specified in: http://kb.digium.com/articles/Troubleshooting/How-do-I-run-a-pattern-loopback-test-on-a-Digium-E1-T1-card?retURL=%2Fpopup=false With this results Have you done loopback testing with your telco to make sure your line is clean? I'd point fingers there before blaming the Digium card or a cable. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] best free fax solution with asterisk
On 8/12/2012 3:57 AM, Steve Underwood wrote: On 08/12/2012 10:32 AM, James Sharp wrote: On 8/11/2012 8:05 AM, virendra bhati wrote: Hi team, I want to configure fax with asterisk. there a lot of fax link i found by google but not working perfectly. my setup as follow asterisk 10.x centos 5.8 Want to used T.38 with SpanDSP... Please suggest me the best way. and how to test FoIP ? I use Asterisk 10.3.1, SpanDSP 0.0.6, and Ubuntu 11.10 connecting to Gafachi.com. It works with probably 95% success rate talking via T.38. 95% is pretty bad. Do you know if the failures are mostly during the initial negotiation, or somewhere in the actual FAX exchange? Usually dies in the initial negotiations. 95% was just an off the top of my head number. I'll crunch some CDRs and get an actual percentage. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] best free fax solution with asterisk
On 8/11/2012 8:05 AM, virendra bhati wrote: Hi team, I want to configure fax with asterisk. there a lot of fax link i found by google but not working perfectly. my setup as follow asterisk 10.x centos 5.8 Want to used T.38 with SpanDSP... Please suggest me the best way. and how to test FoIP ? I use Asterisk 10.3.1, SpanDSP 0.0.6, and Ubuntu 11.10 connecting to Gafachi.com. It works with probably 95% success rate talking via T.38. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to Auto Answer a sip phone
From my experience with xlite, the soft phone itself must be configured for auto answer. There is no way for the dialplan to control this. On Jul 13, 2012, at 2:35, upendra uppi...@gmail.com wrote: i am using a x lite phone. regards upendra On Fri, Jul 13, 2012 at 10:29 AM, James Sharp ja...@fivecats.org wrote: Different phones use different methods. What kind of sip phones do you have? On Jul 13, 2012, at 12:17 AM, upendra uppi...@gmail.com wrote: Hi, i wanted to make dial plan in such a way that the any incoming call to the sip phone should auto answer.(auto pickup) . Help. regards Upendra -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to Auto Answer a sip phone
Different phones use different methods. What kind of sip phones do you have? On Jul 13, 2012, at 12:17 AM, upendra uppi...@gmail.com wrote: Hi, i wanted to make dial plan in such a way that the any incoming call to the sip phone should auto answer.(auto pickup) . Help. regards Upendra -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI trunk between Asterisk servers does not work.
On 6/28/2012 3:53 PM, Ernie Dunbar wrote: We have three Asterisk servers, Voip1, Voip2, and Voip3. Voip1 has a Wildcard TE405P (3rd Gen), Voip2 has a Wildcard TE410P/TE405P (1st Gen), and Voip3 has a Wildcard TE110P. Voip1 is the server that handles our PRI to the PSTN and we hope will allow us to failover to other Asterisk servers (ie, Voip2 and Voip3). Voip2 is our current production server, and Voip3 is being turned into our next production server. We're trying to build a PRI trunk between Voip1 and Voip3. Curiously enough, we've already done this between Voip1 and Voip2, so one would think that the same configuration would work between Voip1 and Voip3 as well. However, it hasn't gone so smoothly. If you're wondering why we don't just use SIP trunking between these servers, it's because faxes are not reliable over SIP trunks. I am open to suggestions however. At any rate, the PRI trunk between Voip1 and Voip3 isn't working, and that's my current problem. - I have built a T1 crossover cable, and it's plugged in between Span 3 on Voip1, and Span 1 on Voip3. - I have a green light on both PRI cards for the appropriate spans. - Both servers detect their cards on boot. - DAHDI is installed on both servers, and all diagnostics are good, ie. dahdi_test returns good results, dahdi_tool shows that the alarms are OK, and executing 'dahdi show status' on the Asterisk console shows the same. The chan_dahdi.conf configuration for spans 3 and 4 on Voip1 looks like this: ; Span 3: TE4/0/4 T4XXP (PCI) Card 0 Span 4 group=3 context=default switchtype = national signalling = pri_net channel = 49-71 group = 63 ; Span 4: TE4/0/4 T4XXP (PCI) Card 0 Span 4 group=4 context=default switchtype = national signalling = pri_net channel = 73-95 context = default group = 63 Span 4 goes to Voip2, which has a working PRI trunk. The chan_dahdi configuration for Voip3 looks like this: group=1 signalling=pri_cpe switchtype=national context=local channel=1-23 dchannel=24 ;channel=25-47,49-71,73-95 rxgain=0 txgain=0 busydetect=yes busycount=5 resetinterval=1800 I have a test DID, the dialplan for which on Voip1 looks like this: exten = 604484,1,Dial(DAHDI/g3/604482) But when I call 604484 from my cell phone, I get no output on the Asterisk console on Voip3, and this output on Voip1: -- Executing [604484@local:1] Dial(DAHDI/5-1, DAHDI/g3/604482) in new stack [Jun 28 12:43:32] WARNING[4219]: app_dial.c:1286 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion) == Everyone is busy/congested at this time (1:0/1/0) == Auto fallthrough, channel 'DAHDI/5-1' status is 'CONGESTION' -- Accepting call from '778839' to '604484' on channel 0/5, span 1 I've also tried connecting span 3 to one of the other ports on Voip2 with the same configuration, and I get the same results. I've run loopback tests on the TE110P and tested the cable thoroughly. Any input on this problem is greatly appreciated. You've got the spans configured as group = 63 but you're trying to dial out on group 3 (DAHDI/g3 rather than DAHDI/g63). -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Getting channel 'SIP/192.168.1.69-00000000' refused to negotiate T.38
Does your VoIP provider support t.38? Sent from my iPad On Jun 22, 2012, at 11:05 AM, Ahmed Munir ahmedmunir...@gmail.com wrote: Hi, I recently configured T.38 on Asterisk 10.4.2. When I send the fax to Asterisk, it gives the errors as listed below; WARNING[25986]: app_fax.c:442 transmit_audio: channel 'SIP/192.168.1.69-' refused to negotiate T.38 WARNING[25986]: app_fax.c:174 span_message: WARNING T.30 ECM carrier not found As in sip.conf the configuration is listed below; t38pt_udptl = yes,fec,maxdatagram=400 faxdetect = t38 And the rest are the standard configuration. Please advise to resolve this issue. -- Regards, Ahmed Munir Chohan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8.13.0 / problem with cdr logging (mysql, odbc)
On 6/18/2012 11:52 AM, Thorsten Göllner wrote: Hi, I am trying now for over 4 hours setting up cdr-logging via odbc into a mysql database. But with no success. Do you have any hint for me? *SNIP* But after a call hangup I get the following error: cdr_odbc.c: Unable to retrieve database handle. CDR failed. What is going wrong here, please? The DSN that you specify in cdr_odbc.con should be the DSN you configured in res_odbc.conf, in this case mysql versus MySQL-asterisk. I beat head against the desk for hours because of this same issue. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - Incoming fax cuts ADSL line
On 5/16/2012 12:07 PM, Tim Nelson wrote: - Original Message - Hi, I'm facing a strange situation. Though it's not directly related to Asterisk, I do think it is interesting to this mailing list. The setup is a single line which is split between an ADSL modem/routeur and a fax machine (Asterisk was removed from the equation). Any time the fax machine rings (incoming fax), the ADSL service is troubled to the VPN users are disconnected. It can be reproduced at will. I've changed the ADSL filter twice (a different unit, then a different model) without any visible change. What could explain this ? I've experienced this quite a few times, and after working with a local telco, it has become policy to not place ADSL on lines where fax is going to be used. I'm unsure of the exact technical reasons behind this other than 'the fax signals/frequencies interfere with the ADSL signalling/frequencies used on the circuit'. It sounds like you might want to separate your fax/ADSL lines. --Tim You might also be able to limit the Fax machines maximum transmission rate so the modem's transmission spectrum doesn't inch up into where the ADSL service is. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax .pdf from Asterisk
On 5/4/12 1:57 AM, Bruce B wrote: James, That is amazing details. I can use all of this. Thank you for sharing. I am assuming you installed res_fax from repository? *yum install asterisk18-res_fax_digium.i386* No. I built Asterisk 10 from source. Once I installed spandsp, the menuconfig had the option of installing app_fax. And how did you install SpanDSP? Is there a guide you used? Downloaded and built from source from http://soft-switch.org/downloads/spandsp/ It builds easy. I am aiming for multi-channels fax so the digium one won't do for me as it's one channel limit like you mentioned. I probably don't need T.38 but hey it won't hurt to have it. Well, the Digium one isn't always limited to 1 channel, just 1 channel for the free version. And honestly, if you can get T.38 support it will make your life a lot easier. Faxing over VOIP is hit miss. Doing T.38 Fax over IP makes faxing a lot more reliable. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax .pdf from Asterisk
On 5/3/12 9:16 PM, Bruce B wrote:
Lee,
Much appreciated for the input.
I am running all VoIP. SIP and IAX2 to our ITSPs. Please elaborate on
HylaFax and IAXmodems. Is there a guide posted on to get it running, or
is it part of the repository? Once installed how would one send .pdf as fax?
You can either use the Free Fax for Asterisk single channel at a time
fax system or app_fax that uses the SpanDSP library. I'm working on a
similar system myself that uses the SpanDSP system because I could never
get FFFA to talk T.38 right to my provider (Gafachi).
I use spoolfiles to create a call that lands in my dialplan and from
there, I can pick up the fax results in the dialplan.
Here's my callfile
Channel: SIP/1771655@gafachi1a
CallerID: 18005551212
MaxRetries: 0
RetryTime: 60
WaitTime: 30
Context: faxout
Extension: 5
Priority: 1
And my extensions file
[faxout]
exten = 5,1,SendFAX(/tmp/test.tiff,a)
exten = 5,n,Noop(${LOCALSTATIONID});
exten = 5,n,Noop(${LOCALHEADERINFO});
exten = 5,n,Noop(${FAXSTATUS});
exten = 5,n,Noop(${FAXERROR});
exten = 5,n,Noop(${REMOTESTATIONID});
exten = 5,n,Noop(${FAXPAGES});
exten = 5,n,Noop(${FAXBITRATE});
exten = 5,n,Noop(${FAXRESOLUTION});
Note that I use the a option in SendFAX. That makes it change
behaviors on negotiating things like T.38. It was needed for my
provider, but it may not be needed for yours. You may or may not
need/even have access to T.38 faxing.
You will have to convert the PDF to a TIFF file before you can send it
with either Fax subsystem.
gs -q -dNOPAUSE -dBATCH -sDEVICE=tiffg4 -sPAPERSIZE=letter
-sOutputFile=/tmp/test.tiff test.pdf
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Re: [asterisk-users] Getting Mac Address on connected IP phones
On 3/13/12 5:53 PM, Danny Nicholas wrote: Ping the phones, then run arp. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of resea...@businesstz.com Sent: Tuesday, March 13, 2012 4:52 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Getting Mac Address on connected IP phones I am struggling to get the mac-addresses of IP phones that are connected to asterisk as the phone are in different VLAN with * and they were manually configured. I want to centralize their configuration using res_phoneprov or tftp I have tried nmap and arp in vain. Any idea? ping + arp isn't going to work if they're on a different VLAN. I believe this will work: 1) Set up your TFTP server, but do not put any configuration files in the /tftpboot directory (or whatever the directory is). 2) Set the DHCP server on the phones' network to hand out the TFTP server address. 3) Reboot the phones 4) Watch the TFTP server logs and you should see each phone request a file based on its MAC. With no downloaded config file, the phone should revert to what it already has in nvram. 5) Collect MAC addresses out of the server logs 6) Profit? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - T.38 unreliable on a LAN : truth or obscurantism ?
On 02/15/2012 03:03 PM, Olivier wrote: Hi, When someone says T.38 is not reliable on a (normally loaded and managed) LAN, would you rather agree or disagree ? In this case, fax calls are coming in through an analog gateway, passing trough Asterisk and then going out to ISDN through a digital gateway. Comments ? While I can't speak for Asterisk's T.38 performance (it was barely past the point of okay, it compiles at the time of this datapoint), T.38 in general can handle nasty network conditions without a problem as long you enable some sort of error correction (either FEC or packet redundancy). Case in point, I ran several hundred SIP-based T.38 calls a month over VSAT links. The links ran anywhere from 550 to 750ms latency and would average around 1-2% packet loss (averaged over a 5 minute period). Those were with a Quintum ASG400 at the far end and a Quintum CMS960 going into PRIs at the VSAT hub. So if T.38 can handle that, it can handle just about anything. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Virtual Server
I run my Asterisk system on a quad core Opteron system running VMWare ESXI 5. On Feb 10, 2012, at 21:18, Carlos Rojas crt.ro...@gmail.com wrote: Hello everybody someone in this list, has installed asterisk, in a virtual server like proxmox? I'm thinking install some asterisk servers in a machine dell xeon 64 processor, but I'm not sure, about virtual Server software. I heard, about proxmox, but I don't know if works fine. Regards Carlos -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Router that support Asterisk
On 02/01/2012 02:17 PM, bilal ghayyad wrote: Hi All; I heard from some friends that there are a dsl router that has Linux OS and it has asterisk on it, so the ip phone can register on this router, also if the router has FXS or FXO ports then it can be used to place calls through them. Is it really? Where I can these routers? Did anyone try it to tell us if it is stable and working fine? Regards Bilal The Cisco DDR2200 that I just got from Centurylink for DSL appears to be just that. I haven't tested the FXS ports on it yet, though. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] create table in mysql using asterisk
On 01/09/2012 02:44 AM, Eyal wrote:
Hi,
I try to create a new table using MYSQL command in asterisk.
This is what i write:
*Query resultid ${connid} CREATE TABLE IF NOT EXISTS conference_600
(id int(11) NOT NULL auto_increment, channel_id varchar(40),
number_in_line int(2), PRIMARY KEY(id))*
and this is the warning that i get in the cli:
*app_addon_sql_mysql.c:383 aMYSQL_query: aMYSQL_query: mysql_query
failed. Error: You have an error in your SQL syntax; check the manual
that corresponds to your MySQL server version for the right syntax to
use near 'conference_600 (id int(11) NOT NULL auto_increment,
channel_id varchar(40)' at line 1
*
What is the problem do you think?
Do I in the direction or have a completely different way to do this?
That's a MySQL syntax error, not an Asterisk error. However, the
solution is to not put quotes around your table and field names. That
will make MySQL happy.
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Re: [asterisk-users] Ringing agents cell as an alert?
On 01/03/2012 01:06 PM, Todd Routhier wrote: Happy New Year to all! Asterisk 1.8.x I have a queue to which I add agent channels like SIP/300 dynamically using the manager interface. Once logged in, there SIP/300 of course rings when a call is distributed to them. How can I also get the agents cell phone to ring without actually adding it to the queue? I mean id I add something goofy like SIP/MyProvider/1555444 to the queue, I don't know what will happen at this point, haven't tested it. Even if it works (asterisk channel state etc) it will mess with the queue and treat the cell phone like a separate agent, messing up call distribution etc. I am trying to be as clear as possible, sorry if my questions are cloudy. Basically, I have the queue doing what I want right now, I just want to add the ability to have an agent's cell phone ring as a means of alerting them if they are away from their desk. If they can answer the call and the queue will handle it just as they answered it from their SIP device, that would be a bonus. I know this can all be done, just not sure how to tackle it at the moment. Any guidance would be appreciated, thanks in advance. Perhaps blend the agent's SIP phone + cell phone together as a local channel and then add that local channel to the queue instead of SIP phone + cell phone. Asterisk will see the local channel as one agent rather than two. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ringing agents cell as an alert?
On 01/03/2012 01:22 PM, Todd Routhier wrote: Sounds perfect, I will need to look into how to blend them together like that. Put them in extensions conf like so: [agentblends] exten = bob,1,Dial(SIP/300SIP/12102263232@myprovider) Then put Local/bob@agentblends into your queue. I wonder though, will channel state still work using that method? I think it's needed by something in the queue but I can't remember at the moment. Channel state for ringing/answer? The state will be ringing until Bob either answers his cell phone (or his cell voicemail answers, that may be a quirk. Adjust the timeout parameter on the dial command to work that out) or his SIP phone. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mark queue agent as away
On 01/03/2012 09:42 PM, Raj Mathur (राज माथुर) wrote: Hi, I have a queue with a number of (static) agents. Is there an easy way for an agent to indicate that she is away from her seat, so that her phone is not rung when a call comes in? And the converse, of course: be able to notify Asterisk when she is back and ready to accept calls? There's the PauseQueueMember function. You could write an extension that your agent can dial to, look up her Agent info via callerid or something, then pause her via that function. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to listen on different sip port for a device?
On 12/26/2011 04:15 PM, sean darcy wrote: Thanks for the response. Home asterisk : 10.0.0 - Office: 1.8.8.0 So I thought I'd leave all the sip providers on udp, and move the home-office to tcp. And registration just work Just Worked over the default tcp registry port - which I was surprised to learn was 5060, and not 5061. So it appears cablevision only blocks udp 5060. sean How about using multiple IP tables entries to redirect several ports down to 5060. Dunno if it would work, just a wild-hair guess. I should test it later. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best PBX for Call Centers?
On 12/15/2011 01:33 PM, Tarek Sawah wrote: Hello List, I have customer with a 40 Agents call center. and is looking to install a PBX switch that can serve those agents. As per my experience i suggested Asterisk as i have tested it with Call Centers, however he has been advised not to use it although his provider is using Asterisk to send him calls. He has been advised to use Sippy which they claim is more stable than Asterisk. i'm not an expert with Sippy so i'm looking for a piece of an advise here.. if i'm doing an Asterisk Vs Sippy comparison. can anyone help? Regards I think the answer you're going to get on an Asterisk mailing list is Asterisk is the best. I'd find out why your customer is being advised against Asterisk. Bet you'll find political reasons rather than technical. And if they are technical reasons, make sure they're applicable to a recent version of Asterisk rather than hearing It doesn't do Feature X and then finding out that Feature X was added in 1.4 (and the Asterisk world is on the equivalent of 1.10) and the person just has experience with 1.2. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best PBX for Call Centers?
On 12/15/2011 01:43 PM, James Sharp wrote: On 12/15/2011 01:33 PM, Tarek Sawah wrote: Hello List, I have customer with a 40 Agents call center. and is looking to install a PBX switch that can serve those agents. As per my experience i suggested Asterisk as i have tested it with Call Centers, however he has been advised not to use it although his provider is using Asterisk to send him calls. He has been advised to use Sippy which they claim is more stable than Asterisk. i'm not an expert with Sippy so i'm looking for a piece of an advise here.. if i'm doing an Asterisk Vs Sippy comparison. can anyone help? Regards I think the answer you're going to get on an Asterisk mailing list is Asterisk is the best. I'd find out why your customer is being advised against Asterisk. Bet you'll find political reasons rather than technical. And if they are technical reasons, make sure they're applicable to a recent version of Asterisk rather than hearing It doesn't do Feature X and then finding out that Feature X was added in 1.4 (and the Asterisk world is on the equivalent of 1.10) and the person just has experience with 1.2. I should have probably read the whole message. Especially the part about the more stable claim. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to query Microsoft SQL server for caller-id source
Build Asterisk with ODBC support and then use the ODBC functions to do the database dips. On Dec 12, 2011, at 13:44, Douglas Mortensen d...@impalanetworks.com wrote: Any suggestions from people who have done this before? Thanks, - Doug Mortensen Network Consultant Impala Networks Inc CCNA, MCSA, Security+, A+ Linux+, Network+, Server+ A.A.S. Information Technology . www.impalanetworks.com P: (505) 327-7300 F: (505) 327-7545 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple route failover zaps registration
On 12/11/2011 07:22 PM, Mike Diehl wrote: Hi all, I've got a customer who is bringing up a second Internet connection for fail- over. I've configured a WRT54 with 2 LAN ports and arranged for it to fail over when one of the routes is no longer available. That works just fine at the IP level. However, when the router fails over, the phones lose their registration, presumably because their IP address has changed from Asterisk's point of view. The phones happen to be Polycom 335's, and I'm running Asterisk 1.6.2.9. What is the best way to manage this situation so that the phones don't become unavailable during failover? I'm considering using the Tinc VPN solution to prevent the IP address from chaing, but I'm hoping for a more simple solution. Any ideas? Shorten the registration expiration time on the phones? It'll make them a bit chattier, but it'll make sure Asterisk has the most recent IP address. Shorten it down to as long as you feel the phones can be without service, but also make sure you don't make them too chatty so they're reregistering every second. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple route failover zaps registration
On 12/12/2011 12:35 AM, Mike Diehl wrote: Actually, I've configured the phones to use DNS SRV records to find the Asterisk server, and this works very well. The problem is that when the router fails over, the phones IP address changes and this causes them to be unavailable from Asterisk's point of view. And that's why I was suggesting the short registration timer. I'm assuming that the LAN connections/WRT54 is for client phones, not the server. The phone registers, Asterisk sees it as coming from 75.0.0.1 port 5060 (the external IP of the WRT54). And the phone re-registers every 30 seconds, and Asterisk just refreshes the registration. Now, the primary internet connection falls over and the WRT switches to the other connection. If you set the registration timeout to a short interval, the phone sends out a register message and Asterisk sees it as coming from 85.0.0.1, port 5060. Asterisk then updates the registration IP for that peer and it can deliver and receive calls from those phones. The only thing that will happen is that active calls *will* get dropped. There is no way to hide the IP change from that without the VPN tunnel. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Confrence call is not make
I check in CLI [Dec 6 17:46:58] WARNING[16264]: pbx.c:4088 pbx_extension_helper: No application 'MeetMe' for extension (employees, 777, 1) == Spawn extension (employees, 777, 1) exited non-zero on 'SIP/phone1-' Plz tell me , where i am wrong in configuration. Chances are you didn't install DAHDI before building Asterisk, so you never built the MeetMe application. Go download install DAHDI, then rerun ./configure menuselect for Asterisk and make sure the MeetMe app is available. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Becoming a CLEC
On 11/16/2011 10:30 AM, eherr wrote: But what is the correct physical setup of a CLEC. Do you get rack space at a carrier hotel and equipment in there? Do you get rack space at the local ILEC CO?; which is Verizon here. What are the types of voice platforms used by CLECs? Just as a point of reference, the CLEC I have experience with had 2-3 racks in a colo facility and a handful of circuits coming in from the access tandem and SS7 provider. The core switch was a SUMMA4, but I forget who made the SS7 hardware (it may have been SUMMA as well). -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] trouble with sip connection and registration
You're not going to get a telnet connection on port 5060, since that's tcp and sip uses UDP. Use tcpdump/wireshark on your office pbx to see if the packets are getting to you. If not, then there's something wrong inbetween. A firewall misconfig, perhaps. Or the unthinkable: your home ISP has started filtering 5060. On Nov 14, 2011, at 18:51, sean darcy seandar...@gmail.com wrote: I have a home asterisk box which connects to the office asterisk, so I can just dial extensions. This used to work just fine. I'm using 10.0-rc1 on the home box, 1.8.7.0 on the office. But it doesn't work now: [Nov 14 18:38:19] NOTICE[21563]: chan_sip.c:13161 sip_reg_timeout:-- Registration for 'sip@officePBX' timed out, trying again (Attempt #86) I first thought it was some fall out of the new upgrade to 10.0-rc1, but now I'm not sure. Even with verbose at 6 I don't see anything on the office console about the attempted registration. And on the office: lsof -i:5060 COMMAND PID USER FD TYPE DEVICE SIZE/OFF NODE NAME asterisk 2045 asterisk 15u IPv4 23030 0t0 UDP *:sip but: telnet localhost 5060 Trying 127.0.0.1... telnet: connect to address 127.0.0.1: Connection refused iptables is set to allow 5060 udp and tcp. And I've flushed iptables, but still no luck. I can ssh into the office box from the home box. The office box is directly connected, the home nat'ed. Any help really appreciated. sean -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] trouble with sip connection and registration
On 11/14/2011 09:57 PM, sean darcy wrote: Unthinkable!! Used wireshark: I can see the REGISTER packets going out from the home router, but nothing from home:5060 shows up at the office. Bummer. Now I get to think about how to set up special ports between home and office. A great evening activity. I've always used a VPN to get around BS like that. A home router running dd-wrt connecting via PPTP to your server, problem solved. I'd also call your ISP and hold them accountable and ask why your traffic is being filtered. And which ISP are you using? That way we all know which ones like to play games with SIP packets. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DID from Direct from Telco
On 11/03/2011 07:20 PM, Nick Khamis wrote: Hello Everyone, Unlike going through DIDx, DIDLogic etc.., we have an option of getting DIDs directly from local telco Bell Canada. Currently our SIP Trunk provider assigned a DID to us, and as you know, they just redirect requests it to our PBX. However, when dealing directly with a telco, what equipment will we need? Basically giving us the same capability as a DID provider. If someone can paint a picture on how the DID suppliers function it would be greatly appreciated. If I were to guess it would be: Telco Lines - Gateway E1/T1 - SIP Proxy - Media Servers? With this scenario, do we now have control over the number of channels? Thanks in Advance, Simplest (with 3-4 T1s): Telco Lines - Asterisk box with T1 card (and possibly a codec processor card) - Customer More complex (with a bunch of circuits) : Telco Lines - Gateway T1 - SIP Proxy - Media Servers - Customer And if your question of number of channels is Can I control the number of channels a customer can use simultaneously?, then the answer is With Asterisk, Yes -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DID from Direct from Telco
On 11/03/2011 09:16 PM, Nick Khamis wrote: Hello James, Thank you so much for your response. We just purchased an AudioCodes MP124 for this job. And setting up OpenSIPS as the proxy. As I mentioned earlier, Bell Canada is the Telco here in Toronto. As for other Telcos around the world, for example Bell South in the states, is it possible to have them route a block of Florida phone numbers to our FXS port here in Canada, or do we have to have a T1 gateway + SIP Proxy in Florida, routing the calls to our setup in Toronto and vice versa? Routing Florida numbers up to Canada would get you charged LD per minute fees. You can go with a provider like Level 3 or Global Crossing and they can hand you a T1 circuit that has DIDs from many different areas in the US. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS ports on TDM410P card...
On 10/15/2011 05:31 AM, Michael C. Robinson wrote:
[Oct 15 01:48:02] NOTICE[3747] channel.c: Dropping incompatible voice
frame on SIP/2006- of format ulaw since our native format has
changed to 0x8 (alaw)
[Oct 15 01:48:49] WARNING[3750] pbx.c: Channel 'DAHDI/1-1' sent into
invalid extension 's' in context 'default', but no invalid handler
--
I have the code to set up an extension for toggling Telco pass through
working I think. What isn't working is the pass through. I get the
above error messages when I try to call the POTS line connected to
DAHDI/1 from my Comcast line.
I'm noticing other warning messages cropping up about this file or that
file not existing and modules not loading, but mostly the system seems
to be working so I'm wondering if these warnings are relevant. I'm
using Asterisk 1.8.
I think that [from-pstn] isn't working...
You're not landing in [from-pstn]. Incoming calls are landing in
[default]. That's not a problem in extensions.conf, that's a problem in
dahdi.conf for those channels. They're not in the right context.
For those who don't know what I'm after, I'm trying when a phone company
call comes in to ring SIP phones and local FXS lines on my TDM410P. The
purpose of the toggle is to be able to disable this feature. Sometimes,
I really want to use this system as a private intercom system where at
other times, ringing remote SIP phones for an incoming telephone company
call might be needed. Say you are at extension 2000 or 2002, SIP phones
in other buildings, and you want or need to be able to receive calls
from the PSTN.
I'm in the U.S., under the [external] section am I blocking long
distance outgoing phone calls? In the U.S., you dial 1 and then
the number for long distance. Essentially, what I need to do is
block dialing 1 and then a number with the exception of 1-800 or
1-866.
[external]
exten = _91800NXX,1,Dial(${CENTURYLINK}/${EXTEN:1})
exten = _91888NXX,1,Dial(${CENTURYLINK}/${EXTEN:1})
exten = _91877NXX,1,Dial(${CENTURYLINK}/${EXTEN:1})
exten = _91866NXX,1,Dial(${CENTURYLINK}/${EXTEN:1})
exten = _91NXXNXX,1,Playback(im-sorry-Dave-you-cant-call-that)
exten = _91NXXNXX,n,Congestion
exten = _81800NXX,1,Dial(${COMCAST}/${EXTEN:1})
exten = _81888NXX,1,Dial(${COMCAST}/${EXTEN:1})
exten = _81877NXX,1,Dial(${COMCAST}/${EXTEN:1})
exten = _81866NXX,1,Dial(${COMCAST}/${EXTEN:1})
exten = _81NXXNXX,1,Playback(im-sorry-Dave-you-cant-call-that)
exten = _81NXXNXX,n,Congestion
That'll let you dial US toll free numbers out the channel specified by
dialing 9 or 8. It will playback a message and then generate a
congestion tone if some other number is dialed.
--
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Re: [asterisk-users] Maybe slightly OT but..
Then there's also the point where it makes more sense to drop a GSM card into your Asterisk box and get a cheap unlimited mobile to mobile plan for a SIM and use that to transit your calls to VoIP. Although that won't help the original asker, though, since he mentioned Verizon. On Oct 11, 2011, at 10:22, Daniel Tryba dan...@tryba.nl wrote: On Tue, Oct 11, 2011 at 08:26:22AM -0400, john Millican wrote: Thanks to all for the responses. Boss calls overseas a lot and has an unlimited data plan, so this coupled with the rates that we get for our VoIP calls it is much cheaper than what Verizon charges. My own experience with my 3G provider and VOIP provider is that creating a callback or originator setup is still cheaper then using the GSM carrier (esp. long distance), has better quality (lower latency) and still works if data is not available. -- Daniel Tryba -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Maybe slightly OT but..
On 10/10/2011 05:35 PM, john Millican wrote: Hello all, Does anyone know of a good free/inexpensive 3G SIP client for the iPhone? If anyone is using one that works good for them could you please let me know. I use VaxVoip on my 3GS and iPad. It works great over both 802.11 and 3G. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Maybe slightly OT but..
On 10/10/2011 10:31 PM, linux guy wrote: On Mon, Oct 10, 2011 at 8:08 PM, Andresand...@telesip.net wrote: I would recommend Acrobits. Not free but only a few bucks. It works fine with ATT 3G. This begs the question... which is more expensive (and where)... making a regular cell call or making a SIP call over 3G ? Depends on where you're calling and your cellular LD plan. For example, an ATT World Connect (paying an addition $3/mo) call to Nigeria on my iPhone would cost me about $0.60USD/min (A non-World Connect call would be $3.50/min, so lets just go with World Connect). Patching that same call into my Teliax account via SIP over 3G would be (worst case) $0.16USD/min, assuming that I didn't go over the 2GB of dataplan that I pay for whether I use or not. Now if I don't have a data plan and have to pay the the equivalent of $25/2GB, the numbers go quite the opposite way. Using Google's calculator = (25 U.S. dollars) per (2 gigabytes) = 0.000149 USD/kilobit (note kilobit here). A G.711 call is about 80kilobits/sec each way. That comes out to 0.011920USD/sec * 2 (bidirectional). So, round to $0.024USD/sec, times 60 seconds in a minute, comes out to $1.44USD/min data + $0.16USD/min in termination. I think. My math could be off by an order of magnitude somewhere. And your mileage may vary depending on codec used. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium FFA + Gafachi T38 outgoing issues
On 10/08/2011 02:38 PM, Ryan Wagoner wrote: I signed up with Gafachi a few weeks ago to use them for T38 as well. I haven't had any luck getting it to work. I have been mainly trying to use Asterisk in T38 pass through mode and have tested with a Linksys SPA2102 and Zoiper. Gafachi basically told me they have many customers utilizing their T38 implementation and that it works. When asked for a list of compatible devices they said there were too many combinations and it was up to me to find a working solution. I wonder how many of these customers are just getting fallback to G711 when the T38 stack falls over. Heck, I thought I was getting T38 until I realized that I had SendFAX running with the audio fallback option. Turned that off, and fax fails 100% of the time. I am still looking a PAYG service provider that has a working T38 implementation. It seems like these are impossible to find. I found t38faxing.com. I was going to try them until I saw that their opening credit is $10. More than I want to spend to try for just home faxing. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium FFA + Gafachi T38 outgoing issues
On 10/07/2011 12:27 AM, Nasir Iqbal wrote: Check firewall and NAT settings! Monitoring sip and media flow from asterisk cli can help, use sip set debug on, rtp set debug on and udptl set debug on No NAT involved and I shut off IPTables. Still no luck. Debug shows the SIP invite, RTP frames going in out, the SIP reinvite, and then UDPTL frames coming in until timeout. See the entire transaction at http://pastebin.ca/2087758 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium FFA + Gafachi T38 outgoing issues
On 10/07/2011 04:04 PM, Kevin P. Fleming wrote: First, we can see that Gafachi's T.38 implementation still has some breakage in it (although the problems are ones that Asterisk has been taught to deal with). In its 200 OK to the T.38 re-INVITE, it has a=T38FaxRateManagement:transferredTCFlocalTCF; this is not valid (the valid values for this are 'transferredTCF' and 'localTCF'). In addition, even though Asterisk sent a=T38FaxUdpEC:t38UDPRedundancy, Gafachi replied with a=T38FaxUdpEC:t38UDPFEC. For T.38 version 0 (which is in use here), the only valid response was either what Asterisk sent, or no T38FaxUdpEC value at all. However, it is unlikely those are causing the call failure here. It's hard to say for sure without seeing the contents of the UDPTL packets, but based on their sizes, they are very likely t38-nosignal packets, and if that's all the FAX stack in Asterisk ever received, it would not trigger a FAX transaction to begin. Another possible problem is the repeated 'seq 0' in all the UDPTL packets, but this could be a problem with the UDPTL stack debugging messages themselves (this was just fixed in the Subversion branches for Asterisk 1.8 and later a couple of days ago). Theres a few t30-nosignal packets at the beginning, but then they transition to other t30 packets, including CNG, CED, preambles, training and data. Wireshark says the sequence number is always 0, so it appears that Asterisk is not mis-displaying http://pastebin.ca/2087784 I can provide the raw tcpdump file if needed. If you would, please retry this with the HEAD of the Asterisk 10 branch instead of 10.0.0-beta1, and also capture the UDPTL packets themselves so we can see what they contained. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium FFA + Gafachi T38 outgoing issues
On 10/07/2011 04:42 PM, Kevin P. Fleming wrote: You shouldn't be *receiving* CNG, as you are the calling endpoint. You're right. Hadn't even thought about that. If you are seeing UDPTL packets containing T.38 CED, V.21 preamble, DIS, etc. then something is badly wrong. ... and, that thing is probably the sequence number. Once Asterisk sees a packet with sequence number 0, any subsequent packets received with the same sequence number will be dropped (because according to the T.38 recommendation, they must be retransmissions... new packets would have higher sequence numbers). So these UDPTL packets are never making their way up to the FAX stack, and the FAX transaction never gets started. I guess it must be common for UDPTL stacks out there to just not care about repeated sequence numbers, although the one in Asterisk sure does (and it's based on the same code as the one in CallWeaver, FreeSwitch and maybe other packages too). If you'd like to experiment, you can comment out lines 495 and 511 of main/udptl.c, which will make Asterisk's UDPTL stack just not care at all about the incoming sequence number. HEAD out of SVN + the sequence number change still gets no fax transmit. I do get a few addition fax status messages on the console, though. I'm getting -- FAX handle 0: [ 000.062327 ], STAT_EVT_RX_IMG_STRT st: WT_DIS rt: UNEXPECT 4-9 times (with changing timestamps, of course), then nothing until disconnect. It sounds like that Gafachi's T38 implementation is horribly, horribly broken. I'm not tied to them at all, so if their stuff is broken, I'll go somewhere else. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Digium FFA + Gafachi T38 outgoing issues
Hi, folks. I'm having a heck of a time trying to get outgoing T38 faxing (I don't need inbound right now) working with FFA and Gafachi. G711 faxing works (as well as can be expected over the internet), but I want the higher reliability of T38. I'm running Asterisk 10-beta1. When I drop my callfile in to make the call, I get this: -- Attempting call on SIP/18884732963@gafachi1a for application SendFAX(/srv/httpd/htdocs/upload/scantest2.tiff,dz) (Retry 1) == Using UDPTL CoS mark 5 == Using SIP RTP CoS mark 5 Channel SIP/gafachi1a-000a was answered. Launching SendFAX(/srv/httpd/htdocs/upload/scantest2.tiff,dz) on SIP/gafachi1a-000a -- Channel 'SIP/gafachi1a-000a' sending FAX: --/srv/httpd/htdocs/upload/scantest2.tiff -- Channel 'SIP/gafachi1a-000a' FAX session '6' started -- FAX handle 0: [ 000.000594 ], STAT_EVT_STRT_TX st: IDLE rt: IDLENSTX -- FAX handle 0: [ 000.001139 ], STAT_EVT_TX_HW_RDY st: WT_TX_HW_RDY rt: TRDYNHTY -- FAX handle 0: [ 000.001724 ], P30EVN_SEND_STARTED [Oct 6 04:21:36] ERROR[11616]: res_fax.c:1421 generic_fax_exec: channel 'SIP/gafachi1a-000a' FAX session '6' failure, reason: 'fax session timed-out' (TIMEOUT) [Oct 6 04:21:36] NOTICE[11616]: pbx_spool.c:373 attempt_thread: Call completed to SIP/18884732963@gafachi1a THIS PART HAPPENS ABOUT 15 SECONDS LATER -- FAX handle 0: [ 040.000211 ], STAT_EVT_T1_EXPst: WT_DIS rt: WDISNT1X -- FAX handle 0: [ 042.499953 ], STAT_EVT_HW_CLOSE st: WT_HW_CLSrt: WCLSNCLS -- FAX handle 0: [ 042.500083 ], STAT_SES_COMPLETE -- FAX handle 0: [ 042.500110 ], P30EVN_COMPLETE -- Channel 'SIP/gafachi1a-000a' FAX session '6' is complete, result: 'FAILED' (FAX_NO_FAX), error: 'T1_TIMEOUT', pages: 0, resolution: 'unknown', transfer rate: '2400', remoteSID: '' A tcpdump trace shows the initial invite, ringing, answering, some G711 frames back and forth, the send-T38-invite-after-10-seconds reinvite (as specified by the Z option), then the far end sends a bunch of T38 traffic until Asterisk times out and drops the call. What also confuses me is this (and this may just be semantics or a true bug): asterisk*CLI fax show stats FAX Statistics: --- Current Sessions : 0 Reserved Sessions: 0 Transmit Attempts: 8 Receive Attempts : 0 Completed FAXes : 7 Failed FAXes : 7 How can I have 8 attempted transmits, 7 completed faxes, and 7 failed faxes? I know 1 transmit didn't go through because I tried to place one call while another was in progess and I only have one licensed channel. Thanks, James Sharp ja...@fivecats.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ANI II/Payphone indication
Quickie: Does anyone out there have experience with PRI delivery of ANI II information? Specifically, I want to know if it's possible from within Asterisk to know if the inbound call (which may or may not be to an 800 number) came from a payphone or not. I know with some 800 providers it's possible to block inbound calls from payphones (due to the FCC surcharge etc) but was wondering how accessible that information is once the call hits my box. I'm not sure about PRIs, but when I did it with Feature Group D trunks, the information came in as ANI II info digits prepended to the ANI. I had to modify * a bit, though, because it was stripping off the info digits and throwing them away. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] B-channels resetting every 60 minutes?
We have one other error (twice today) we get Out of trunk data space on call number , dropping How do I determine what is causing this error? we have a point-to-point T1 between 2 * boxes, with 3 phone in the remote office. I have no idea how the trunk could be out of space. The end goal is trying to figure why we are dropping calls! Are you running IAX between the boxes? Are you running IAX trunking between the boxes? If so.. Do you have trunking configured identically on both ends (trunk=yes in iax.conf)? Do you have a zaptel device (or ztdummy) in both ends? If either of those questions is no, then you'll get the out of trunk data space error and drop calls. Make sure both ends are configured the same for trunking and you have a zaptel device or ztdummy. Or just don't use trunking (trunk=no). ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PCI front mount chassis?
I am looking for a good case to house my Digium PCI cards, I was hoping to mount them in the front for cleaner access then in the back. Unfortunately I haven't found much, does anyone have a good recommendation for chassis to use up to six digium cards? Probably not cost effective, but i had looked into a few 6U CompactPCI chassis...stick a CPCI to PCI adapter card in there. *rummagerummage* Ah-hah. http://www.pcisystems.com/PCI-ADAPTER.html Something like that. Except that particular one wouldn't work for some cards because it is PCI 2.1, not 2.2. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] wct1xxp module and the T100P
Steven, Perhaps I should have posted my question differently to the list: After installing the CVS version of Asterisk, I type, modprobe xct1xxp. The machine accepts the command but the LED on the T100P does not flash. How do I know that the T100P module has loaded correctly? Do you see any errors showing up in the output from dmesg? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Surveys
Is it possible to have the system outdial and take surveys. either by receiving DTMF or voice? Yup. Just have the system use the outgoing queue (see sample.call) and have it call an AGI script upon answering. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie Question. Is asterisk right for my scenario?
Hi, Please excuse me if my question seems too simplistic. I have been reading the mailing list for some time and I am still a bit confused. Here is the scenario that I would need to achieve and am wondering if asterisk is the correct software to use. (h323) (h323/SIP) (h323) pstn---cisco--Asterisk??-cisco---pstn | | | -sip phone I have an existing h323 structure doing h323 pstn termination and would like add sip to part of the structure, also at the same time would like asterik to act as a softswitch to store dial plans and make routing decisions. Asterisk at the same time will do h323/SIP translation. My question, can Asterisk do all these? Or am I totally off? Quite simply, yes. Asterisk is a softswitch more than anything else. And it can take an incoming call on any of its available protocols (PSTN, IAX, SIP, H323, plus many more) and route it back out any of them. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] The Smallest Asterisk Server Ever?
Now, here's the real question: can you install it on a toaster? It builds and runs on NetBSD, minus the hardware part (for the moment)...so yeah. Asterisk on NetBSD/Vax. Hrm. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] The Smallest Asterisk Server Ever?
On Tue, 3 Feb 2004, Chris Albertson wrote: Smallest Asterisk server? No. That old Gateway box must be about 2 cubic feet. 1.5 ft^3 at a minimum. I've got one that is about 0.2 ft^3 a factor of maybe 10 smaller. Hehehe.. As far as Form Factor goes, I'm sure there are smaller boxes out there. How about Most resource challenged Asterisk server ever? :) How about one of the 1Ghz Soekris boards with a 802.11 board in it? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Internal Lines Dialing Out
exten = _.,1,Dial(Zap/1/$EXTEN)
exten = _.,1,Dial(Zap/1/${EXTEN})
Gotta put the name of the variable in brackets for it to work.
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Re: [Asterisk-Users] Adtran 750 DID question.
Hello All, I've mostly solved my DID problem from a few days ago. Apparenly the lines weren't configured properly. Now heres the next question. 12 EM wink lines from telco. I have them all plugging into an Adtran 750 with FXS cards. The Adtran ports are configured DPO. How do I signal this from Zaptel. I have them setup EM in zaptel.conf and EM_W in zapata.conf. They work, however, no DNIS info is being passed. Do I need to signal these something different like loopstart or kewlstart, so the DNIS info gets passed? I watch the Tx/Rx bits from zttool, and everything looks okay coming from the Adtran. It looks like asterisk isn't winking properly. I had a similar problem. I ended up setting the trunks to either just plain em or featd (I don't remember). I chased through the chan_zap source code and decided (maybe incorrectly) that asterisk doesn't look for DNIS digits in EM Wink mode. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Auto dial in Off Hook situation.
Thanks John, I think it is not that simple. I am not using a phone but a Cisco ATA. The scenario: - User--(Multitech VOIP MVP200 (FXS))--Internet--(Multitect VOIP MVP100 (FXO))--Cisco ATA--Asterisk--Any extension Any reason you can't use the H.323 load for the MVP200? I've not tried it in a year or so, but it mostly worked last time I tried it. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Incoming Voice/Fax Discrimination?
I'm evaluating * to replace the crap set of peered smart phones we have now in our small office, but I haven't been able to find out about this anywhere yet: I need to know if * can discriminate _incoming_ FAX calls on a voice line and route them to a specific extension? Yes, it can. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Example of TDM20B
; FXS Port 1 context=local signalling=fxs_ls usecallerid=yes echocancel=yes echocancelwhenbridged=yes ; ;FXS Port 2 context=local signalling=fxs_ls usecallerid=yes echocancel=yes echocancelwhenbridged=yes Change the signalling here to fxo_ls. Its gotta match what's in zaptel.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream transfer solution + DTMF translation possible?
The solution to the problems with the Grandstream 1.0.4.39 firmware is to use inband (in-audio) DTMF. Neither the RFC2833 nor INFO seem to work. Don't the Grandstreams send a DTMF 'F' INFO message on a hookflash? Shouldn't be that hard to change chan_sip to register an 'F' as an AST_FLASH control message. I'd do it, but I couldn't test it...since it seems that Grandstream hasn't found it in the goodness of their heart to add hookflash support to the HandyTone ATA. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] re: hardware requirement -asterisk
# ifconfig xl0 xl0: flags=8843UP,BROADCAST,RUNNING,SIMPLEX,MULTICAST mtu 1500 address: 00:01:02:78:11:e8 media: Ethernet autoselect (10baseT) status: active inet 203.219.167.126 netmask 0xfffc broadcast 203.219.167.127 inet6 fe80::201:2ff:fe78:11e8%xl0 prefixlen 64 scopeid 0x2 But ifconfig seems to suggest that it is running in simplex mode. If the DSL modem is running in full duplex mode and the card in your machine has auto-negotiated to half-duplex, things go to hell quickly. Auto-negotiation sucks anyway. It works about as well as Plug Play. ifconfig xl0 media 10BaseT mediaopt full-duplex Or hell, try ifconfig xl0 media 100BaseTX mediaopt full-duplex Run netstat -I xl0 -i 1 and watch for collisions. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to Order Disconnect Supervision from SBC using Adit 600?
Can anyone help me with the term that SBC uses to refer to disconnect supervision? I have an Adit 600 channel bank which has helped improve the disconnect detection time down to about 8 seconds. This is still causing some issues in particular with call progress enabled in * we are having a few disconnects while calls are in session (about 2 reported in first 5 days of use). I have talked both to a local phone contractor and SBC directly and no one seems to know what I am talking about. The phone contractor knew about the issue with other phone systems in the area but didn't know there was a way to fix it and SBC reps seem to never have heard of disconnect sup or calling party disconnect. I've never seen a line from SBC that DIDN'T come with disconnect supervision (some SBC line monkeys I know call it battery drop disconnect). The * Handbook refers to loop start with call sup as kewlstart are there other names for this protocol? One of the local contractors thought that SBC automatically drops line voltage on remote hangup, in which case I need to know what signalling to program into the ADIT 600's fxo channels. I also have the option of going to groundstart signalling if this would fix the problem, but it would cause some line downtime so it is not my preferred method. Kewlstart is also an alias for battery drop disconnect. The Adit 600 manual lists the following options for mapping FXO ports to the T1 DSO. DPT = Dial Pulse Termination EMDW = EM Delayed Wink start EMI = EM Immediate start EMICPD = EM Immediate Start with Calling Party Disconnect EMW = EM Wink start GS = Ground Start GSRB = Ground Start with Reverse Battery LS = Loop Start LSCPD = Loop Start Calling Party Disconnect LSRB = Loop Start with Reverse Battery VoIP = Voice over IP (CMG only) I believe I currently have the lines set to LSCPD which improved the hangup situation, but hasn't completely fixed it. That should be right. If you're really interested in looking, take a cheap voltmeter and put it across the line. If everyone is on hook, you'll see 48V. If someone goes off hook, you'll see it drop to about 6V. If you see a quick drop to 0V when the far end hangs up, you've got battery drop disconnect. I don't know if this has any relevance but I am also originating the clock source from the * side with Wildcard T1 card. That's really the only way it'll work. The channel bank can't generate clocking. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to Order Disconnect Supervision from SBC using Adit 600?
I have a little more info on this. Following the suggestion of another post on this topic I tracked down an analog phone with lighted buttons powered by the phone connection. I directly connected the phone to one of my inbound lines and called it with my cell phone. Picked up the analog phone, verified call completion and then hung up my cell. I watched and waited for the lights to go out. Sure enough they did, but it took 8 seconds from the time of the hangup. After the flash more phone started emitting a dialtone sound. Is this correct? I was under the impression the voltage drop would happen almost immediately. Do you have another analog line that's on the same Central office as the line in question? The delay could be lag time betwee the time you hang up your cell phone, the cell provider MTSO processes the hang up, passes it on to their termination provider, who then passes it on to your termination provider, who then passes it on to you. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] far end disconnect supervision
If some channel banks don't support this, how on earth do they know when the telco side of the call has hung up ? They don't. They rely on either a timeout or the called party hanging up to disconnect the call. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 911
FYI there is a way to do 911 its called E-911 enhanced 911 the user has to set it up with the local emergency services to it and you setup your pbx to xmit the data. There's PS/ALI (Private Switch Automatic Location Information) that's quickly becoming state mandated for all PBX systems. The problem with it is if your customers are spread out across multiple PSAPs or RBOCs. Then you've got to interface with the PSAP or RBOC in every area where you've got a customer. Then you've also got to assign a DID to every customer that can be transmitted back to the PSAP over the PRI or CAMA trunks (which are necessary to use E-911). Its fine if you're limited to one or two PSAPs in your service area, but a company like Vonage or NuFone has an almost unlimited number of PSAPs in their coverage area. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hardware to build an Enterprise AsteriskUniversal Gateway
Andrew Kohlsmith wrote: I would set the Enterprise Class bar at five 9's reliability (about 5.25 minutes per year of down time) the same as a Class 4/5 phone switch. This would require redundant design considerations in both hardware and software. To turn around, let's discuss what we need to focus on to get Asterisk there: Here's a few bullet points, there's certainly a lot more * Linux platform stability - how? Even more than Linux itself is the x86 platform... I've thought about this a bit when considering * boxes for big customers. When one actually comes along, I'll have to actually make a decision :-). From where I stand, the best thing to do for smaller customers is give them a box with RAID and redundant power supplies, if they can afford it. You can overcome most of those problems by buying good quality hardware. If you buy your * server from your local Taiwanese clone shop, you're asking for trouble. A big, beefy machine from Dell would be better. But if I were to have a big customer with deep pockets, I'd really like * on a big Sun beast with redundant-everything (i.e. you can hot swap any component and there's usually n+1 of everything). The problem is that I don't think there's any Solaris support for Digium cards, since it's kind of a chicken-and-egg problem. Nope. No Solaris support, but you might be able to get away with Linux/Solaris...but then you lose a lot of the hot-swapability. In my experience, though, the only things I've ever been able to hotswap were power supplies and hard drives...and thats not software/os dependant. One of these days, I may convince myself to buy a modern Sun box (maybe the ~$1000 Blade 100s) and see what can be done. The only problem I could conceive would be endian-ness, but I read about Digium cards in a PowerPC box, so that won't be a problem, right? Nick Endian-ness is really only a driver issue. Its when programmers who believe that the world revolves around Linux/i386 that you have problems. Personally, I'd stick my Digium cards into an Alpha of some sort. A DS-10L for 1U mounting with 1 card or a DS-20 for multiple cards where you need lots of processor zoobs. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hardware to build an Enterprise AsteriskUniversal Gateway
1. Moving a physical interface (whether a T1, ethernet or 2-wire pstn) is mostly trevial, however what signal is needed to detect a system failure and move the physical connection to a second machine/interface? (If there are three systems in a cluster, what signal is needed? If a three-way switch is reqquired, does someone want to design, build, and sell it to users? Any need to discuss a four-way switch? Should there be a single switch that flip-flops all three at the same time (T1, Ethernet, pstn)?) Simple idea: Have a process on each machine pulse a lead-state (something a s simple as DTR out a serial port or a single data line on a parallel port) out to an external box. This box is strictly discrete hardware and built with timeout that is retriggered by the pulse. When the pulse fails to arrive, the box switches the T1 over to the backup system. Since protecting calls in progress (under all circumstances and configurations) is likely the most expensive and most difficult to achive, we can probably all agree that handling this should be left to some future long-range plan. Is that acceptable to everyone? Its going to be almost impossible to preserve calls in progress. If you switch a T1 from one machine to the other, there's going to either going to be a lack of sync (ISDN D-channels need to come up, RBS channels need to wink) that's going to result in the loss of the call. 2. In a hot-spare arrangement (single primary, single running secondary), what static and/or dynamic information needs to be shared across the two systems to maintain the best chance of switching to the secondary system in the shortest period of time, and while minimizing the loss of business data? (Should this same data be shared across all systems in a cluster if the cluster consists of two or more machines?) 3. If a clustered environment, is clustering based on IP address or MAC address? a. If based on an IP address, is a layer-3 box required between * and sip phones? (If so, how many?) Yes. You'll need something like Linux Virtual Server or an F5 load balancing box to make this happen. You can play silly games with round robin DNS, but it doesn't handle failure well. b. If based on MAC address, what process moves an active * MAC address to a another * machine (to maintain connectivity to sip phones)? Something like Ultra Monkey (http://www.ultramonkey.org) c. Should sessions that rely on a failed machine in a cluster simply be dropped? d. Are there any realistic ways to recover RTP sessions in a clustered environment when a single machine within the cluster fails, and RTP sessions were flowing through it (canreinvite=no)? e. Should a sip phone's arp cache timeout be configurable? Shouldn't need to worry about that unless the phone is on the same physical network segment. f. Which system(s) control the physical switch in #1 above? A voting system...all systems control it. It is up to the switch to decide who isn't working right. g. Is sharing static/dynamic operational data across some sort of high-availability hsrp channel acceptable, or, should two or more database servers be deployed? DB Server clustering is a fairly solid technology these days. Deploy a DB cluster if you want. 4. If a firewall/nat box is involved, what are the requirements to detect and handle a failed * machine? a. Are the requirements different for hot-spare vs clustering? b. What if the firewall is an inexpensive device (eg, Linksys) with minimal configuration options? c. Are the nat requirements within * different for clustering? 5. Should sip phones be configurable with a primary and secondary proxy? a. If the primary proxy fails, what determines when a sip phone fails over to the secondary proxy? Usually a simple timeout works for this..but if your clustering/hot-spare switch works right...the client should never need to change. b. After fail over to the secondary, what determines when the sip phone should switch back to the primary proxy? (Is the primary ready to handle production calls, or is it back ready for a system admin to diagnose the original problem in a non-production manner?) Auto switch-back is never a good thing. Once a system is taken out of service by an automated monitoring system, it should be up to human intervention to say that it is ready to go back into service. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Sun Servers with UltraSparc Processors
I had documented the Makefile modification in an email to the list. If you search for Sparc in the mailing list, you should be able to find it. If not, drop me a line and I'll see if I still have it. I've got an Ultra 30 sitting here doing nothing. I'll see what I can come up with for Linux/Sparc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Web Dialer
I am putting together a solution that will employ the Digium TE410P with one T1 going out the PSTN and the other front-ending a PBX. The idea is that based on a URL, Asterisk will dial an employee behind the PBX. When the employee picks up, Asterisk will dial the customer (detailed in the URL). I am assuming Asterisk can work with Apache (through AGI maybe) to dial the employee and then connect to the customer via info in the URL (or related through some sort of DB lookup). Another requirement will be to record the phone call as well. You could do it through either the Asterisk manager interface or have a CGI scrip t in your web front end create an auto call file that dials the employee and runs a second Dial command upon answer. I have worked a bit with Asterisk and am very happy with what it can do -- and would prefer to stay with Asterisk. The question is, can Asterisk handle what my requirements are or would this better be served by Bayonne? Asterisk is better. Hands down. No questions. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help with x101P
Occasionally I do NPA-NXX lookups for my local exchanges to see what other carriers have prefixes in my area. I used to use telcodata.us, but they seem to have gone offline. Usually, after you find the carrier's name, you can see info on the location and type of switch being used. I can't say with any assurity that the info is accurate, but it is there if you dig http://www.dslreports.com/coinfo Has the same kinda info. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] frame buffering
Hi all. Could it be possible that video frame buffering be causing problems even if the computer is not running X ? Yes. There are known problems with systems running with either a frame buffer console or a serial console. For best results, run a plain VGA console. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Land line vs. VoIP provider.
What about having your VoIP gateway system placing a 911 call to the 911 answering center in the appropriate region and when the 911 operator answers, have a message say This is a 911 call from 123 Main Street, Nowhere Nebraska then connect the caller to the 911 operator. Legal? Maybe. Dunno. Just a random thought that I came up with on the way to the aforementioned middle of nowhere, Nebraska. Not all VoIP providers will have Vonage's 911 issues. It's perfectly possible for a VoIP provider to provide outbound caller information to the PSAPs if they spend the time and money to do so. Stephen Summary: if you're the only caller, calling only to the US, then you might be crazy to not use a land line, especially given the deals currently available and the 911 issue (but see http://www.vonage.com/features_911.php). Even then, if you already have broadband in house (or at home), VoIP amy be an attractive alternative, if only for the control it gives you over your phone service. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Headless Linux system for Asterisk
Run using a serial console (http://www.tldp.org/HOWTO/Remote-Serial-Console-HOWTO/). No monitor, VGA adapter, keyboard etc needed. Use SSH to log into the asterisk box for any maintenance, etc. If the box gets hosed, connect the serial port to a working PC and fire up minicom and your all set. You'll find this type of setup quite often in data center environments. Except there is a known problem of dropping/missing interrupts with running serial consoles with certain Digium boards. You also have the same problem if you use a framebuffer console. If you truly want it headless with a serial console without that problem, stick a PC Weasel board in it (http://www.realweasel.com). ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FAX, IAX and *....Maybe I'm dreaming...:-)
It's just my lowly opinion but I too must agree when it comes to the consumer/soho (1 to 3 line) markets. CAUTION!!, DANGER!! Marketing Hat On!! Vonage, the most visible marketer of a voip consumer product must also agree. Vontage offers an ip fax line. using cisco's ata. Vontage must see some good reason for doing so. (I assume it's h.323). AT, MCI and Time Warner will be competing directly against Vonage when they introduce their consumer voip products. I'd bet they too will be offering an ip fax line. Odds are you will be competing against them too. If I was vonage I'd be telling the world how important a ip fax line was :-) Personally, I dont think that the world in general really cares about an ip fax line. All they want is a system that works all the time/every time and doesn't require elaborate and convoluted setup. They're not ooohing and ahhhing about oooh, this uses VoIP. They just know that they can stop spending $30/mo on an analog phone line and they get their long distance either flat rate or for an absurdly low per minute rate. Don't sell it as VoIP. Sell it as a total replacement for the analog line. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: * with RADIUS
Can someone give me an idea exactly what things are intended to be tested via RADIUS, or some other AAA system? Are we talking about building SIP/IAX/H323 entries from RADIUS? This is where the PAM system I developed for * comes into play. I've got most of it working at the moment, but I'm having trouble figuring out how to actually pass the authentication information back to the requesting channel driver. It seems that the structs linked lists that the drivers use to authenticate are only built on a restart/reload. I'm not sure how to handle this part of it dynamically. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium Wildcard TE410P
It can be either. Does this card only work as PRI or can it be used like a standard T-1 wired to a PSTN Switch? TIA -Seth ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] was FXO cards
Case 1 and 2 are ties in my eyes, except the channel bank would provably be cheaper to upgrade to 8 lines. I am just afraid of the channel bank. I just don't know anything about them. If I buy the wrong crap, it gets really expensive fast, plus adds another layer of complexity. You could also talk with your local phone company and other CLECs to find out pricing on fractional voice T1s/partial PRI. Depending on the locality, the breakeven point is usually 6-8 lines. You might even be able to get a deal on a hybrid data/voice circuit. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PRI problems
I've got a couple of PRIs coming in from a SUMA 4 switch with some 800 numbers routed through it. When the calls come in, I get the following message on the console and the call never makes it through: (800 number is fake) Extension '8005551212' in context 'nonauthenticated' from '232102749585' does not exist. Rejecting the call on span 4, channel 1. I do have the following line in extensions.conf in [nonauthenticated] exten = 8005551212,1,AGI,ivr-main.pl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PRI problems
*CLI show dialplan nonauthenticated [ Context 'nonauthenticated' created by 'pbx_config' ] '8005095639' = 1. AGI(ivr-main.pl) [pbx_config] '8005095640' = 1. AGI(ivr-main.pl) [pbx_config] '8005095641' = 1. AGI(ivr-main.pl) [pbx_config] check 'show dialplan nonauthenticated' regards Martin On Fri, 21 Nov 2003, James Sharp wrote: I've got a couple of PRIs coming in from a SUMA 4 switch with some 800 numbers routed through it. When the calls come in, I get the following message on the console and the call never makes it through: (800 number is fake) Extension '8005551212' in context 'nonauthenticated' from '232102749585' does not exist. Rejecting the call on span 4, channel 1. I do have the following line in extensions.conf in [nonauthenticated] exten = 8005551212,1,AGI,ivr-main.pl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PRI problems
It seems that there's a non-printable character at the beginning of the DNIS stream I'm getting from the SUMA 4 switch. Once I chopped that off, everything works right. Hi James, Try to do exten = _8005095639,1,Agi(ivr-main.pl) Quoting James Sharp [EMAIL PROTECTED]: *CLI show dialplan nonauthenticated [ Context 'nonauthenticated' created by 'pbx_config' ] '8005095639' = 1. AGI(ivr-main.pl) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] We are thinking of Asterisk
Hi all, We are thinking of changing our Nortel Meridian PBX to Asterisk. Before we jump into this we would like to know if we can support some important for us functionalities on Asterisk. We would like to know if we can 1. Have menu based voice mail with Asterisk? (like press 1 for sales, press 2 for tech support ...) Yes 2. If we can login, logout make ourselves unavailable with Asterisk ACD? If Asterisk will route alls FIFO bases to the loggedin and available personal. Yes 3. Can we have regular PBX functionalities Call Forwarding Call Transfer Conference Call Music on Hold Yes 4. Can we point DID directly to the extension? No need to dial a main number and than an extension. Yes 5. Is it possible to monitor a call in Asterisk for training or coaching purposes. If a new person is doing a tech support can we monitor (listen to conversation)? If yes, would it be possible to record conversation - like a voicemail? Monitor, yes. Recording...maybe 6. Can we get our voicemail in our e-mail mailbox? (Can we delete, make new, forward voicemail from e-mail client? Yes 7 Can we send and receive faxes from PC? That's a work in progress. Sort of works, sort of doesn't. Thank you in advance for taking time and answering our questions. Aram ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
