Re: [asterisk-users] Iptables configuration to handle brute, force registrations?
fail2ban might be good for this. On 04/05/2011 01:00 PM, asterisk-users-requ...@lists.digium.com wrote: Date: Tue, 5 Apr 2011 08:44:41 -0700 (PDT) From: Steve Edwardsasterisk@sedwards.com Subject: Re: [asterisk-users] Iptables configuration to handle brute force registrations? On Tue, 5 Apr 2011, Gilles wrote: I'm no expert of iptables, and it seems like it can handle banning IP's that are trying to register and fail too many times. Is there a good iptables configuration that I could use as reference? Gordon Henderson posted a link to his script that handled failures above a threshold and some other cool stuff a few months back. smime.p7s Description: S/MIME Cryptographic Signature -- _ -- 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-users Digest, Vol 81, Issue 12
On 04/05/2011 03:06 PM, asterisk-users-requ...@lists.digium.com wrote: Message: 12 Date: Tue, 5 Apr 2011 13:36:21 -0500 From: Sherwood McGowansherwood.mcgo...@gmail.com Subject: Re: [asterisk-users] Iptables configuration to handle brute, force registrations? To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Cc: Bill Michaelsonb...@cosi.com Message-ID:banlktimqrbfmqpoinrphr_rjekolbwp...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 On Tue, Apr 5, 2011 at 1:31 PM, Bill Michaelsonb...@cosi.com wrote: fail2ban might be good for this. I think you missed the point, which is reducing the need for an external application that searches logs in order to determine whether or not to block an IP. Why run fail2ban and add overhead when you can just do the same thing with iptables itself? I apologize for jumping into the middle without reading the beginning of the discussion in which this central requirement to avoid an external application was stated, as I now infer from Mr. McGowan. Sorry for missing the point. I'll have to read up on fail2ban also. I thought it monitored the tails of logs. I did not know that it searched them. My intent was to suggest using an established tool that would consolidate the IP blocking and unblocking function for all ports into a single application without imposing additional maintenance overhead of new code for this purpose. Obviously, I'm not seeing the big picture. Sorry for my myopic comments and for cluttering the list. I won't make the mistake of offering worthless contributions in the future. smime.p7s Description: S/MIME Cryptographic Signature -- _ -- 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] Shorr/Call quality issues
This is why I don't do this kind of work anymore. Impossible to distinguish the phantom problems from the real ones - and I'm convinced there ARE phantom problems when you install new telephones on people's desks. Suggestion: learn to use the facility in Wireshark that can log a SIP/RTP stream and report actual latency and packet delivery stats. That will give you some solid info on at least one aspect of call quality. Message: 2 Date: Tue, 15 Dec 2009 11:03:07 -1000 From: Ben Schorr b...@rolandschorr.com Subject: Re: [asterisk-users] Can't get G.729 to work... To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: 02bd9be03009b24b9bf9e00257ad8103012d7...@hnl-pai-exh001.pacificatelier.com Content-Type: text/plain; charset=US-ASCII Yes, the routers are another issue we're dealing with. We've configured them to prioritize traffic to/from our Asterisk server but I'm not convinced that setting is really working as expected. So we're working with the vendor on that. The effective bandwidth is 6Mb/sec down x 1.4Mb/sec up (so basically 1.4x1.4 on the VPN). For 8 users, where rarely more than 2-3 of them are on the phone at any given time, that should be sufficient I think. They DO have to share the connection with their web browsing and e-mail and such but as best we've been able to tell they aren't saturating their connections - usually not more than 4-5 of the 8 are using their computers at any given time and most of them just do e-mail and local apps that shouldn't touch the Internet connection. Frankly I'm puzzled that they have these issues and the problems rarely seem to happen when I call them. I'll go to their site and make a few calls from one of their phones and it sounds perfect to me. But three days later all I hear is how frustrated they are because these new VOIP phones suck and they can never hear anybody and... sigh smime.p7s Description: S/MIME Cryptographic Signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Good phone near $125
Polycom IP 430 or 330. asterisk-users-requ...@lists.digium.com wrote: Date: Mon, 16 Mar 2009 18:24:33 -0400 From: David Ruggles da...@safedatausa.com I was looking at the aastra 9133i, however I was informed that this phone is no longer supported. What are good phones around the $100 - $125 price point? (Need POE) smime.p7s Description: S/MIME Cryptographic Signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] $20 Bounty
It's conceivable that the combined effort of these two responders
required less than ten minutes of time, yielding a theoretical pay rate
of $120/hour.
I wonder how much effort went into the other responses.
That will be $6 for my commentary, please.
Folks wrote:
Message: 1
Date: Tue, 3 Mar 2009 22:51:15 -0500
From: David Backeberg dbackeb...@gmail.com
Subject: Re: [asterisk-users] $20 Bounty
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID:
3de056a30903031951o60d6b94u3ebd87205ac64...@mail.gmail.com
Content-Type: text/plain; charset=windows-1252
On Tue, Mar 3, 2009 at 10:25 PM, David Backeberg dbackeb...@gmail.com wrote:
exten = 123,s,1 Playback(enterzipcode)
exten = 123,s,n Read(zip||5)
exten = 123,s,n System(wget http://pathtoyahooservice${zip} -o forecast.txt)
exten = 123,s,n System(wget --post-file forecast.txt -o wav.url)
exten = 123,s,n System(wget --input-file wav.url -o voice.wav)
exten = 123,s,n Playback(voice)
exten = 123,h,1 Hangup
On Tue, Mar 3, 2009 at 7:12 PM, Dean Collins d...@cognation.net wrote:
I?ll pay anyone a $20 bounty for someone to replicate the USA Asterisk
Weather App on Tropo.
All you have to do is violate the ToS on a few services:
wget the weather from yahoo, for instance:
http://weather.yahooapis.com/forecastrss?p=06513
Conditions for New Haven, CT at 9:53 pm EST
Current Conditions:
Fair, 20 F
Forecast:
Tue - Clear. High: 25 Low: 13
Wed - Mostly Sunny. High: 34 Low: 19
do a wget post of that output from the previous wget to
http://www.research.att.com/~ttsweb/tts/demo.php
do a wget on the wav file that demo generates.
It would be nicer if you record a prompt before asking for the
zipcode, but it's not strictly necessary.
You can paypal me the cash to my email. The legitimate license for
ATT Natural Voices is more than $20, and nothing built into Asterisk
for free is going to give you free-form text-to-speech.
smime.p7s
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[asterisk-users] call file concurrency
Is there a convenient way to limit the number of call files (outgoing directory) that are processed concurrently? smime.p7s Description: S/MIME Cryptographic Signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-users Digest, Vol 55, Issue 52
What economic downturn? I'm sick and tired of hearing this mantra. I wish you the best of luck in maintaining your immunity. Same here (in the UK). As long as people need to make phone calls ... Gordon The economy (and indeed humanity as a whole) needs periods of removing sludge, deadwood, and general stupidity. Profound. In a way, I agree. Having said that I would think that many of the list contributors are based on the USA, so no surprises they would feel that way. Because the US is economically isolated, I suppose. I thought our media left a bit to be desired until Youtube came along and I could see the propaganda and mindnumbing dross trotted out in the corporate controlled media there. My mind is being numbed. So people should decide for themselves whether to think positive or negative thoughts. And they should whistle a happy tune, too! I for one intend to take full advantage of the opportunities presented by this period of transition in human affairs. Good to see the entrepreneurial spirit alive in this, uh, period of transition in human affairs. John Todd would like to hear how you are doing this. smime.p7s Description: S/MIME Cryptographic Signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VPN and Asterisk
David @ULC ucoms2...@gmail.com wrote One of my user was asking, can he use VPN to access asterisk ? What does it mean ? And its possible ? How ?VPN Sometimes what is called a VPN is not a VPN by everyone's definition, so beware. By my definition, a (IP) VPN supports full layer 3 functionality (and sometimes more), as opposed to, say, some type of proxy that relays a limited set of protocols over a particular path with encryption. So you need to be more specific about your question. Example: I've run Asterisk over OpenVPN. In this case, no problem; it's just another networking layer and the only special consideration is increased overhead, and that same consideration applies to any application. There are other VPNs. smime.p7s Description: S/MIME Cryptographic Signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Quiet 24 port POE gig switch
How are you getting these 80 or 120mm fans in a 1U chassis? Remember you got barely 45mm to play with at the back and front of the switch. How are you going to mount a 80mm or 120mm fan on there? Are you assuming that the units mounted above (or below) your switch is a short 1U? You can't assume that... I guess one shouldn't assume as a rule, but... If I were concerned about noise, it would be because I am positioning this near people, and not in a densely-packed rack. Thus I would be OK eliminating the 1U constraint. I think these tend to be distinct deployment scenarios. And thinking about that, do PoE switches tend to be deployed near people? You tell me. I'd like to find a quiet 24-port PoE switch - even at 100 Mb. smime.p7s Description: S/MIME Cryptographic Signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ligion
Sometimes I do. It depends on my mood and purpose. And sometimes the author prefers to write last things first, for whatever reason. I'm kind of agnostic, too. Mike Dent wrote: H, not sure about you but I often pick up a book and flick from the back to the front, does nobody else do that? smime.p7s Description: S/MIME Cryptographic Signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-users Digest, Vol 53, Issue 5
From: Doug [EMAIL PROTECTED] Net Neutrality is great in principle. But ISP's need to somehow control those few percentage of users who suck down a huge majority of the bandwidth. It's dollars and cents. There is a rational solution for the traffic management issue. It just needs to be aligned with the pricing model. I would like to see a tiered offering. If packet carriers would respect, say, DSCP tags, and ISPs would cap traffic (by bandwidth and/or aggregate transfer per period) in different ways based on priority tags, we could have a palatable solution for all. Let the bulk users camp out at the fixed price all-you-can-eat packet buffet all they want - at the bulk priority. If the chafing dishes are occasionally empty, well, too bad. Let them select one of the higher priorities for the metered or limited traffic. Then the ISPs could advertise unlimited offerings on a best effort basis, and caveat emptor. Applications that require more reliability (predictability) will not suffer. smime.p7s Description: S/MIME Cryptographic Signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What kind of IAX2 client will install/run on EEE PC 1000 (stock Linux software)?
On Sat, Nov 15, 2008 at 03:49:40PM -0700, Joseph wrote: What kind of IAX2 client will install/run on EEE PC 1000 (stock Linux software)? I'll eventually replace this crippled Linux with something better but I don't time to play around with it as most divers and modules are still too new and not fully available in all distros. This doesn't answer your question at all, but might be helpful nevertheless... I hosed the window manager on my Xandros eeepc 1000, and since I was frustrated by my disorientation caused by the Xandros, I but Ubuntu on it. But then I had grief with the rt2860 wireless driver. So I started again and put a simple Debian on it that supported the wireless from installation bootstrap on. Built from there, eventually installing KDE, and then twinkle. Still having problems with mic gain, but I'm not pushing hard. The net install from USB key, in case you are interested, can be obtained at: http://wiki.debian.org/DebianEeePC/HowTo/Install THe whole process took less than an hour from boot to desktop GUI and networking configuration with openvpn. smime.p7s Description: S/MIME Cryptographic Signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] network design philosophy and practice
I'm wondering how prevalent the practice of physically segregating voice and data networks is in the Real World. What are the factors that typically lead to such a decision? DIscussions of pros and cons are most welcome by me. Experiences, anybody? smime.p7s Description: S/MIME Cryptographic Signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] network design philosophy and practice
Alex Balashov wrote: Send asterisk-users mailing list submissions to asterisk-users@lists.digium.com I'm pretty sure they meant two logical networks. At least, I hope they did. Unfortunately, I was indeed referring to two physical networks. Cabling, switches, everything, all the way back to the TDM connection to the PSTN. David Gibbons wrote: Two separate networks? Did I miss something? I feel like I'm taking crazy pills! Two separate physical networks means twice the hassle, twice the maintenance, twice the cost, twice the headache. Not to mention the fact that the whole idea of VOIP is to simplify IT and focus on converging data and voice networks. This is what VLANs and QOS do best. I dare say it's what they were designed foe. I can't think of any reason that I would ever recommend two ports per desk to support telephony -- ever. It's ludicrous to think that two ports will be better than one if we're setting up our VLANs and QOS properly. A phone takes very, very little bandwidth away from the desktop and a decent one will support tagging its frames for the alternate voice VLAN. I agree, especially about QoS design intent. But I posted my question as a sanity check, and there seems to be no shortage of opinions. Now mine: I can think of two valid reasons to physically segregate the networks: 1) Insurance. I.e., to eliminate the possibility that otherwise properly configured QoS mechanisms become broken, either by accident, incompetence, or badly-designed or rogue software or hardware - or are otherwise handled carelessly as Jerry Jones suggested. But this is not a compelling argument to me in any but the most critical scenarios such as public-safety applications, etc. 2) Customer preference. If you need the business, then the customer is always right. You might not have adequate credibility with the customer or influence over the design decision, and if a customer in such a situation gets it in their heads that voice and data can't coexist on wires, then it can't. There is a variety of opinions, but no general consensus about where QoS failures typically occur, when they occur. I'm wondering if anyone has anyone has ever experienced QoS issues caused by contemporary Polycom phones like IP330s that had workstations hanging off their builtin switches? If you did, were you able to identify the cause, and was it due to any inherent failure of the phone, such as not marking packets or prioritizing dispatch correctly? smime.p7s Description: S/MIME Cryptographic Signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sonicwall potentially causing long ping times to SIP phones
Kristian Kielhofner wrote: On 10/23/08, Bruce Komito [EMAIL PROTECTED] wrote: We've had LOTS of problems with Sonicwalls doing bad things to SIP and RTP connections. I've seen the delay thing, as well as the Sonicwall throwing away entries from the ARP table because of inactivity. I've also seen sporadic, intermittent problems with transfer from one phone to another. I have no doubt that a new, properly configured Sonicwall can be made to function properly in a VoIP environment, but we are not Sonicwall experts, nor are many of the purported experts. In every case where we've had problems with VoIP behind a Sonicwall, the problems ALL disappear when we put the phones on a LAN segment that does not pass through the Sonicwall. So, now that's our going in position. If it works, great, but if it doesn't, our solution is to take the Sonicwall out of the picture. My $.02 . Bruce Komito WPTI Telecom (775) 236-5815 I wouldn't single out SonicWalls when it comes to breaking SIP traffic. Most of the anything but simple PAT devices I've seen that implement any SIP specific fixups usually end up breaking something along the line. Unless the product is from a company where SIP is their core competency (like Ingate, or /maybe/ Cisco) it's best to stay away and/or disable the SIP specific fixups wherever possible. I'm looking forward to the day when SIP-TLS is the norm and these devices have no idea what kind of traffic is flowing through them! - I sympathize, especially since a client of mine is facing the same situation. A potential update to their configuration involves exactly what you (Kristian) suggest: layering TLS in-between. I've run SIP/RTP and IAX over openVPN without issue routinely. What worries me is that the problem is not related to SIP awareness, and that some erratic performance by the Sonicwall that is benign in most circumstances manifests as a quality issue when carrying media streams. Seems unlikely, but does anybody have any clarity on this? smime.p7s Description: S/MIME Cryptographic Signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sonicwall potentially causing long ping times to SIP phones
Sorry for asking the obvious question, but are there other elements of the slow path besides the Sonicwall? I mean, what is in front of the Sonicwall? Also, might the Sonicwall be positioned as some kind of choke point in the topology, thus leading to genuine sporadic congestion? James Lamanna wrote: Date: Wed, 22 Oct 2008 11:35:12 -0700 From: James Lamanna [EMAIL PROTECTED] Subject: [asterisk-users] Sonicwall potentially causing long ping times toSIP phones Hi, I'm having an issue where some phones behind a sonicwall are auto-congesting. The status on sip show peer shows ping times anywhere from 80ms all the way up to 1100ms. PCs behind the same firewall have a ping time of about 30ms to the PBX itself. Does anyone know if the sonicwall is inserting delay into the SIP signaling path and lagging the OPTIONS messages for qualify? Thanks. -- James smime.p7s Description: S/MIME Cryptographic Signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OT: Polycom IP330 user problem
I recently sent this email to a user in response to a problem report of phone calls going to voicemail without the phone ringing. I'm wondering if I've covered all bases, or whether there is some logical explanation I haven't considered, and generally what others' opinions/experiences are that relate. This is an Asterisk system, of course. --- I looked at the server logs for the phone call missed by . They indicate that the call came in at 15:32:25, and was routed to her telephone at 15:32:32. This timed out after about 25 seconds as it should if unanswered, and was sent to voicemail at 15:32:58. I called BB and asked her to check the phone display. She told me that the phone logged an unanswered call at 15:32:32, precisely in accordance with the server log. This leaves two possible conjectures: * The telephone, for whatever reason, did not ring in response to the incoming call signal which it obviously received. * The telephone ringer was not audible or noticeable to for some other reason. For the first possibility, I can think of three circumstances that would cause this: * If the handset is slightly ajar, i.e., off-hook, the phone will make no sound, but log the call. Upon receipt of the message waiting notification, it will start blinking. Eventually, the phone reverts to on-hook status by itself even if the handset is still ajar. * If the alert code for silent ring is set, the line annunciator will flash silently to indicate the call coming in. * If the phone is malfunctioning anything can happen. There is no indication that silent ring alert was set, nor is there any current configuration setting that should cause this. That leaves three bullet points for us to consider. I can follow up with one: I will research this as thoroughly as I can to see if there are any reports of malfunctions by Polycom IP330 phones that conform to this behavior, or if there are any other possible explanations for the events that I've overlooked. If you would like to follow up in any other way, let me know what I can do to help. smime.p7s Description: S/MIME Cryptographic Signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] is there a way
Steve Totaro wrote: My only wish is that Linux had a facility like XP to bridge NICs without running all sorts of commands for brctl. Just a GUI like XP. Last time I setup a bridge in Linux, I had to change many kernel options and rebuild the entire kernel to get bridging working properly. With XP, you just select the NICS, right click and select add to bridge. For linux, I find that running firestarter, ICS/Firewall is fine, my end game is to get all of my traffic to go over an OpenVPN tunnel at my colo which is the default gateway over OpenVPN. Windows seems to have the easiest method of getting this done. I've taken to using Debian derivatives lately, so your YMMV, but maybe this is helpful to you... I haven't had to rebuild any recent kernels for bridging. I do have to apt-get bridge-utils, but that's a trivial thing I do on any box I install. I also typically apt-get other userspace stuff like vlan, nmap, tcpdump and wireshark, etc. I've been using the following type of code in /etc/network/interfaces to effect bridges. When I want to bridge a tap device with openvpn, I do something similar to establish a bridge at boot time with only one physical ether attached. Then I put the final brctl add into a script which is invoked via the up option line in the openvpn conf file. Then it's all automatic. I don't (yet) know how to do it on other distros. The following fragment is used to connect to a redundant pair of asterisks for failover: # bridge of two ethers for alternative paths to SIP clients auto eth1 iface eth1 inet static address 0.0.0.0 netmask 255.255.255.0 auto eth2 iface eth2 inet static address 0.0.0.0 netmask 255.255.255.0 auto sipbr0 iface sipbr0 inet static address 192.168.1.13 netmask 255.255.255.0 broadcast 192.168.1.255 network 192.168.1.0 bridge-ports eth1 eth2 smime.p7s Description: S/MIME Cryptographic Signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: headsets
Jay R. Ashworth wrote: In our call center (250 agents), we use Panasonic KX-TCA60 headsets, which I think we get for 7 or 8 bucks apiece, and sell to the agents at cost. Thanks for that - they look good, and I found several recommendations for them after I got yours and started looking for them. Further to this, I'm in the client office today and dealing directly with the users who are reporters and editors for a periodical and conduct many telephone interviews. They want to use their old recording devices with the new phones, but are finding unpleasant audio experiences when they switch them over from the Nortel meridians to the Polycom IP330s. So I'm looking for kit to use here as well. Recommendations most welcome. And in the case of one user, she is adamant she not be required to use a different recording device. I don't know how to approach this except to try a different telephone or mess with Polycom gain settings that the manual advises not to touch. Anybody been down this road - have any wisdom? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OT: headsets
Some users at a new Asterisk installation with Polycom IP330 phones are complaining about echo with the amplified headsets they used to use with their Nortel phones. I listened myself, and I here my own voice annoyingly loudly, and no headset/phone combination of volume control manipulation produces a very good experience. I've avoided messing with the Polycom internal settings for gain. So I tried a Plantronics M10 device with the phone, and I think it works fine. I'm going to bring this to the client tomorrow. They've been using the headsets by unplugging the handsets and putting the headset box between the handset and the phone. So I'm also going to bring an IP430 phone which has an extra RJ11 jack for a more elegant wiring setup and intelligent mode control directly from the phone (trying to stroke a fickle user). This all leads to the general question: The IP330 has a subminiature jack for headset/mic combos. Are there quality headsets anyone would recommend for in-office use for heavy users with these phones? Using any wiring path? I've tried a cell phone earphone/mic, and it sounds OK, but it's flimsy for this application. smime.p7s Description: S/MIME Cryptographic Signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk in VM.
My experience is very limited, but you asked for any perspective, so... I put an Asterisk with freePBX on a linode server (linode.com), just to play with it a few months ago. I can say that it worked to the point of being able to dial out with my Polycom phone on a FiOS connection, through the * box, and a SIP termination service like Vitelity, and to receive calls in the other direction. No problems with that, and kinda cool to be able to throw a virtual PBX out there with so little expense. I did not stress test it, nor did I examine resource usage to gain perspective on scalability. More of a proof of concept. One issue that comes up with regard to this is about timing sources for MOH, etc. Related to this and of general use to know, I believe one can associate PCI cards with particular VMs, but it's been a few months since I configured a Xen box of my own, so the details have already fled from my feeble brain... But I hope that's helpful. Alex Balashov wrote: Does anyone have any perspective on how well Asterisk performs and scales inside a Xen hypervisor environment? Obviously, the answer depends largely on what sort of hardware it's running on, whether it's in PAE mode, whether it's a newer CPU that has some paravirtualisation instruction sets available to assist it, how much memory is allocated to each VM, and other architectural considerations. Any perspective would be helpful, however. smime.p7s Description: S/MIME Cryptographic Signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [Fwd: asterisk-users Digest, Vol 51, Issue 2]
From: Joseph L. Casale [EMAIL PROTECTED] Does anyone have any perspective on how well Asterisk performs and scales inside a Xen hypervisor environment? I tried on many different pieces of hardware with various recent Xen versions and it always had some level of unpredictability and was not as reliable as running on bare hardware. I wouldn't do it for production but it was fine for testing (sort of :). All other things being equal, certainly the bare HW will win out. I'd like to also note that Xen provides a mechanism to dedicate a CPU core to a virtual machine in a system appropriately equipped. Might be useful. As to whether all critical sources of contention can be controlled adequately to achieve an equivalent or sufficiently robust environment for Asterisk -- I can't say authoritatively. It's reasonable to think it might be possible. smime.p7s Description: S/MIME Cryptographic Signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Maybe OT - routing calls in PSTN
That is my position, and I appreciate the affirmation, as well as the offer to determine the carrier. I might email you about that. But having no business relationship with the other carrier, it is at best awkward for me to initiate contact on this matter, and this should be obvious to Vitelity staff. Worse, they are now telling me to contact the user of the number to ask them what provider they use. I think this is apalling. So I'm more concerned with the practicality of relying on Vitelity for service in general and in the future. Their tech support has been absolutely cavalier to the point of insulting in refusing to deal with this basic issue of connectivity. I'm wondering if my experience is unique. From: Alex Balashov [EMAIL PROTECTED] It is their responsibility to contact the underlying origination carrier to resolve the issue. I have a Vitelity DID which generally works, but calls from a particular caller do not reach it. Vitelity has thus far disavowed any responsibility for working through this problem. smime.p7s Description: S/MIME Cryptographic Signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-users Digest, Vol 50, Issue 89
Interesting to see it done. Vitelity claims it is impossible. The number is 212-651-5632. BTW, if you provide the originating number, the underlying carrier can be determined, either by the pooling or NANPA block it is assigned to, or its LRN if ported. If you want, you can privately e-mail me the number and I'll tell you who the carrier is. smime.p7s Description: S/MIME Cryptographic Signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Maybe OT - routing calls in PSTN
I have a Vitelity DID which generally works, but calls from a particular caller do not reach it. Vitelity has thus far disavowed any responsibility for working through this problem. I recognize that some action might be required by another provider which is outside Vitelity's control, but it seems that they should at least be trying to help resolve the problem by helping me determine the responsible party and facilitating contact - because it is their DID/service that cannot be reached. In the past when I had a similar problem with a Junction DID, the folks at Junction resolved it with no hassles and zero intervention on my part. But Vitelity just keeps closing out my trouble tickets while responding in a way that indicates that they are not reading my reports carefully. How does this compare to others' experiences with Vitelity and other providers? Is there a way that I can determine whom to contact given only an originating number? Any words of wisdom? Documents I can read for educating myself? smime.p7s Description: S/MIME Cryptographic Signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] EM wink/no audio
I am preparing to connect an asterisk box with a redfone fonebridge to a T1 service provider. I am doing this by testing first with another asterisk and a Sangoma card playing the role of telco. I formerly had this test configuration operating flawlessly as a PRI connection. But I discovered that I will need to use EM, thus I've chosen the parameters as described in the subject line. So far, I am able to initiate a call from the Sangoma/telco side to the fonebridge side, and basic robbed bit/ABCD call supervision seems to work. That is, I can see the flags going up and down on but sides with zttool, and a Zap channel is allocated on each side. But it seems that DNIS is not going through, and so I had to fudge it by creating a s extension on the called side to pass the call through to a SIP telephone. When the call is answered, the caller hears silence, and the call recipient hears a soft squeal. Am I being reasonable in assuming that the absence of a valid audio stream is the likely reason that the called number is not being passed through successfully? In any case, what type of stuff should I be looking for to diagnose this? FYI, the calling side issues a log message that seems relevant, but it's precise implications elude me: chan_zap.c: Ignoring wink on channel 1 (see below). I hope someone can give clue on these matters. [Sep 22 11:27:05] VERBOSE[27487] logger.c: -- Executing [EMAIL PROTECTED]:19] Dial(SIP/366-a41a4560, ZAP/g1/1222333|300|wW) in new stack [Sep 22 11:27:05] DEBUG[27487] chan_zap.c: Dialing '1222333' [Sep 22 11:27:05] DEBUG[27487] chan_zap.c: Deferring dialing... [Sep 22 11:27:05] VERBOSE[27487] logger.c: -- Called g1/1222333 [Sep 22 11:27:06] DEBUG[27487] chan_zap.c: Ignoring wink on channel 1 [Sep 22 11:27:06] DEBUG[27487] chan_zap.c: Sent deferred digit string: T1222333w [Sep 22 11:27:19] VERBOSE[27487] logger.c: -- Hungup 'Zap/1-1' smime.p7s Description: S/MIME Cryptographic Signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI auto-configure - continued from DEV list
Tzafrir Cohen wrote: I usually configure the entire span of 24 channels (23 B + 1 D) and only the turned up channels go into service. This is good for a couple of reasons. Also note that Zaptel will anyway reserve all the 24 (for T1) or 31 (for E1) Zaptel channels for the span. So the Zaptel channel numbers will not change whether the span is fractional or full. What do you mean by reserve? Seriously, I'm trying to get a good grasp. I have always assumed that the signal presented by the Adtran TSU120e appears as a full 24 channels. But it was not clear to me how those channels are transformed on the TDM side of the fork, if at all, by the Adtran. I supposed they might be remapped within the frames. But thinking (out loud) about it some more, I realize that remapping any of the channel positions would likely invalidate some references embedded within the Q.931 data stream on the D channel, vastly complicating the process by requiring the Adtran to be aware of the content structure at a protocol layer that would otherwise be unnecessary. So I suppose that it almost certainly does not remap these channels. In fact, the nature of this animal is such that I suppose for a PRI, each entire frame could be passed to the TDM side unmodified and it would work just fine, with the PBX ignore the IP channels. And following this same line of reasoning, the zaptel code would have little need to be told through its configuration which B channels are available because such information is implicitly available via Q.931 - and thus the channels specifications in zaptel.conf serve only to restrict usage. Have I got this right? Steve, Thanks, I like this idea, and I appreciate the tip. I will try it. Meanwhile, I'm finding from others' comments that it is extremely common to find the D channel on 24, which is primarily what concerned me - and my inability to divine this precisely in my case led to my suggestion/inquiry on the dev list. I've seen enough docs that indicate that the D channel could be anywhere in the group, also implying that it's not unlikely to be at 13 or 6, IIRC. I have visions of sitting in a lonely room repeatedly editing zaptel/zapata.conf and smacking it again, and again... Please give a list of variables. At least the ones you can think of. I guess you are referring to variables in the broadest sense, as I was, so to wit... Having never attached asterisk to a T1, I have no working reference system, and I don't have a personal finite checklist of completion items. So not knowing what I don't know is the biggest variable! But I have placed configuration info in redfone.com, zaptel.conf and zapata.conf (see below). I have built the ethmf module and it loads, and I can observe a stream of data on the designated ethernet interface with tcpdump. It is a bidirectional stream of fixed length blocks that look something like what I might expect, but I have been unable to decipher any content upon superficial inspection. I am supposing it is functioning correctly, but it's validity is still a variable to me, albeit only a small source of doubt. Basic info such as alarm state is definitely getting transmitted, as zttool and the asterisk app are able to detect state changes... When I move the DSX-1 cable from the Nortel box (which works for actual phone calls, so this is not a variable) and I plug it into the redfone TDMoE box, the LED goes from yellow to green, implying that it sees the data (I guess). Similarly, zttool tells me there are no alarms and that I have the number of channels configured as specified in my configuration. It has thus far only indicated that 0 are active, which based on googling, I suppose means 0 live calls established. Now it seems that the only configuration that causes asterisk to start without complaint has been with the D channel on 24. I'll omit detail on this for the moment. Now I am at a point where I can use the pri command to get status. With the cable out, I see this: left*CLI pri show spans PRI span 1/0: Provisioned, In Alarm, Down, Active and with the cable connected, I see this: left*CLI pri show span 1 Primary D-channel: 24 Status: Provisioned, Down, Active Switchtype: National ISDN Type: CPE Window Length: 0/7 Sentrej: 0 SolicitFbit: 0 Retrans: 0 Busy: 0 Overlap Dial: 0 T200 Timer: 1000 T203 Timer: 1 T305 Timer: 3 T308 Timer: 4000 T309 Timer: -1 T313 Timer: 4000 N200 Counter: 3 Note that this cable is ordinarily attached to a Nortel PBX which is fully functioning with the T1 service. Perusing the net, I've decided that the Down status is what I must understand and correct. So the variable is the meaning of Down. Other clues seem to indicate that my box is sending stuff down the line, but hearing nothing in return. But I haven't seen any messages that elaborate. For example, the pri command provides certain trace options which yields stuff
[asterisk-users] PRI auto-configure - continued from DEV list
On Tue, Sep 9, 2008 at 7:17 AM, Bill Michaelson [EMAIL PROTECTED] wrote: I'm faced with an installation at a client site with supposed PRI service on a fractional T1. Steve Totaro wrote: I usually configure the entire span of 24 channels (23 B + 1 D) and only the turned up channels go into service. This is good for a couple of reasons. 1. No configuration changes are needed if the client decides to light up some more B channels 2. All B channels that are lit up will come up but not the B chans that are not in service, so configuring the entire span in Asterisk will not effect anything negatively. Channels that do not come up are not used by Asterisk. I have had issues with this only once, the entire span came up, not just what was provisioned, so calls going out on those channels did not work. The carrier put a Cisco box at the demarc that was configured for a full PRI going to the Asterisk box. - Steve, Thanks, I like this idea, and I appreciate the tip. I will try it. Meanwhile, I'm finding from others' comments that it is extremely common to find the D channel on 24, which is primarily what concerned me - and my inability to divine this precisely in my case led to my suggestion/inquiry on the dev list. I've seen enough docs that indicate that the D channel could be anywhere in the group, also implying that it's not unlikely to be at 13 or 6, IIRC. I have visions of sitting in a lonely room repeatedly editing zaptel/zapata.conf and smacking it again, and again... Of course, due to my inability to assure everything else in the configuration is correct, I could do all that smacking for nothing. I want to eliminate variables or otherwise devise a logical step-by-step procedure for getting this running. In my case, I've got an Adtran TSU120e doing a split between the old Nortel PBX (which I'm trying to replace) and a Cisco router for the IP side of the service. From fiddling around with the Adtran panel, I've been able to determine that there are 12 channels being sent to the DSX-1, but it tells me no more than that. If I could safely assume that D is on 24, and configuring the other 23 per your suggestion will be OK, maybe there is hope. smime.p7s Description: S/MIME Cryptographic Signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ztd-ethmf
I expected to find th module ztd-ethmf[.c...] in support of the redfone TDMoE product in my zaptel distro (I have 1.4.11). But it's not there. I am awaiting a response to a trouble ticket from redfone. Can anyone give me a jumpstart? I can't seem to google this up. smime.p7s Description: S/MIME Cryptographic Signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] in-call start monitoring
I suppose, too. So see below. I also verified that the dial command is
using Ww (which I had to fudge), but still, no monitoring.
Anything else I can check?
pig*CLI feature show
Builtin Feature Default Current
--- --- ---
Pickup *8 *8
Blind Transfer # #
Attended Transfer
One Touch Monitor *1
Disconnect Call * **
Park Call
Dynamic Feature Default Current
--- --- ---
(none)
Call parking
Parking extension : 70
Parking context : parkedcalls
Parked call extensions: 71-79
From: Paul Hales [EMAIL PROTECTED]
I suppose the bit to check is the features ('show features') and then
try to record a call (*1) and see what the terminal says...
Bill Michaelson wrote:
My client needs call recording features and would like to initiate the
process in-call (typically *1). I'm installing Asterisk 1.4.x and
FreePBX 2.4+. I'm using Polycom phones. I can't make it work. Would
somebody please give a checklist of items for me to compare my list
against - in the hope I've overlooked something?
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[asterisk-users] in-call start monitoring
My client needs call recording features and would like to initiate the process in-call (typically *1). I'm installing Asterisk 1.4.x and FreePBX 2.4+. I'm using Polycom phones. I can't make it work. Would somebody please give a checklist of items for me to compare my list against - in the hope I've overlooked something? smime.p7s Description: S/MIME Cryptographic Signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] It's telling me too much...
In case this is useful to others, a tip... I moved one of my Polycom 501's off it's subnet to another one (I've got an ether bridge glued to the back of the phone and a wireless card in the * box acting as AP). Now it is still served by the same Asterisk box, albeit through another ethernet port. It works just fine, except that on incoming calls, the full SIP address is displayed for the caller number. I was initially puzzled until I realized that the phone was simply qualifying the address of the caller because it was in a domain that is different than it's own - technically. But my users don't do technically, and come to think of it, I don't like it either. The fix: use iproute2 to mangle the packets: BEFORE: [EMAIL PROTECTED]:/home/ftp/polycom4# ip ro sh 192.168.20.0/24 dev eth0 proto kernel scope link src 192.168.20.3 192.168.99.0/24 dev ath0 proto kernel scope link src 192.168.99.1 71.245.116.0/24 dev eth2 proto kernel scope link src 71.245.116.10 169.254.0.0/16 dev eth2 scope link metric 1000 192.168.0.0/16 via 192.168.20.65 dev eth0 default via 71.245.116.1 dev eth2 metric 100 FIX: [EMAIL PROTECTED]:/home/ftp/polycom4# ip ro rep 192.168.99.0/24 dev ath0 proto kernel scope link src 192.168.20.3 AFTER: [EMAIL PROTECTED]:/home/ftp/polycom4# ip ro sh 192.168.20.0/24 dev eth0 proto kernel scope link src 192.168.20.3 192.168.99.0/24 dev ath0 proto kernel scope link src 192.168.20.3 71.245.116.0/24 dev eth2 proto kernel scope link src 71.245.116.10 169.254.0.0/16 dev eth2 scope link metric 1000 192.168.0.0/16 via 192.168.20.65 dev eth0 default via 71.245.116.1 dev eth2 metric 100 Now the phone thinks it's routing to the * box, but ignorance is bliss. Hope this helps someone. smime.p7s Description: S/MIME Cryptographic Signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ?? Vitelity dtmfmode=rfc2833 started working!
I appreciate your report (below), but it's a strange and disturbing coincidence for me. DTMF out through Vitelity was not working for me until 1-2 days ago when I changed it from rfc2833 to inband! Maybe I just missed the change date and I should change it back? Date: Tue, 22 Jul 2008 12:23:39 -0400 From: Mark G. Thomas [EMAIL PROTECTED] Subject: [asterisk-users] Vitelity dtmfmode=rfc2833 started working! To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Hi, Last week my outbound (dtmfmode=inband) DTMF via Vitelity started acting more weird than usual, and for outbound calls, incoming DTMF tones would consistenly get stuck, breaking a call screen macro I had set up. I checked sip show peer and saw that Vitelity for inbound was now reporting DTMFmode : rfc2833 (it didn't used to), so switched my ountbound dtmfmode to rfc2833 and my problems went away! Yay! It looks like Vitelity now supports rfc2833 on SIP channels. I thought others might be interested in knowing this, as at least in my case it broke things until I changed my settings, and I see this has been a prior source of frustration for many others. smime.p7s Description: S/MIME Cryptographic Signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-users Digest, Vol 48, Issue 56
Message: 1 Date: Fri, 18 Jul 2008 20:35:47 -0700 From: Dave Platt [EMAIL PROTECTED] I'm preparing for a client install of * by doing a fresh one in-house. Unlike my earlier installation that runs asterisk as superuser, my current experimental box runs without such privilege. This is causing it to moan that it can't set TOS. I absolutely don't want to install it on the client LAN without this capability. If need be, I'll set the binary to run setuid root. But I'm looking for something more elegant. While googling, I found a suggestion to use iptables mangle rules to set TOS for all packets going out of the box on ports like 5060 and 1:2. Not a bad hack, but indiscriminate and this box will be handling other traffic besides the RTP. I'd like to do better. It is possible for an iptables filter/rule to match packets in the OUTPUT chain based on the UID or GID of the process which created them, if you have the owner module loaded. You should be able to add a rule to the OUTPUT chain of the mangle table which will set the TOS properly for any and all outbound packets generated locally by the non-root user ID which you're using to run Asterisk. I've used LARTC and I'm aware of the capability, but keying on UID did not occur to me. Thank you - it's a good solution. Come to think of it, I think I need to do this myself. I'm using the ultimate Linux traffic conditioning configuration (modified very slightly) to prioritize my system's outbound traffic into multiple queues by TOS, and it's probably mis-queueing the RTP traffic because my Debian install of Asterisk is running under a non-root UID. Glad to be of assistance. I thought of using POSIX access control to enable asterisk to do TOS setting without being root (would this be CAP_NET_RAW?), which sounds perfect, but so far I'm operating with stock ubuntu hardy, and I would like to avoid a kernel build to add this capability. Any other ideas? Seems like iptables -t mangle -A OUTPUT -m owner --uid-owner $ASTERISK would be along the lines of what you want? Mark the packets with the TOS you want... and then consider using the Linux traffic-shaping system to make sure that they really do get transmitted ahead of non-urgent packets: Traffic-shaping in the box would probably be overkill for my purpose because the nature of the routing in this box will limit the contention from this source. I think I just need to have the packets treated well once they hit the local network. But this is also a worthwhile consideration, and probably useful in other circumstances. Again, thanks for the reply - it's right on target and solves my problem nicely. smime.p7s Description: S/MIME Cryptographic Signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] automon=*, Dial(, , Ww), rfc2833, canreinvite=no, but...
After much checking and puzzling, I cannot get my Polycom 601 to toggle call recording with my Asterisk 1.4.21.1. Via FreePBX, I can set a user to always record, and the recording will show up in /var/spool/asterisk/monitor. But if I try to start recording by toggling in-call, no luck. I can see this in the feature*.conf file set: automon=*1 and I can see a 'Ww' in the logged/traced call to dial(). and I can see the RFC2833 RTP packets going through Asterisk, both with rtp debug and with wireshark. So my questions are: 1) How do I verify that asterisk actually saw the feature code spec upon restart/reload? I can't find any clues. 2) Are there any other parameters that have a bearing on this? 3) Is there anything I haven't thought of? Finally, it might be worth noting that the packet traces show three RFC2833 end events for each DTMF code pressed. This might be perfectly normal, and I even tried fudging the automon string to ***111 just to compensate as an experiment, but it had no effect. If I've done everything necessary to configure enabling the toggle function, then where should I see the failure/refusal to comply in any logs. I'm getting nothing in logs/traces. A side question: freepbx is generating include statements with a leading #, a la C includes - or a la Perl/Shell/et al comments! This is OK? I've floundering with the suspicion that I'm overlooking something really dumb... I would be grateful for some explicit diagnostic suggestions. smime.p7s Description: S/MIME Cryptographic Signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] automon followup
A followup to my own inquiry... pig*CLI feature show Builtin Feature Default Current --- --- --- Pickup*8 *8 Blind Transfer# # Attended Transfer One Touch Monitor Disconnect Call * * Park Call Dynamic Feature Default Current --- --- --- (none) Call parking Parking extension : 70 Parking context : parkedcalls Parked call extensions: 71-79 I guess this narrows it down. So presumably, my feature code specs are not finding their way into the process, but why? I'm looking, but comments are most welcome. smime.p7s Description: S/MIME Cryptographic Signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] automon follup #2
OK, I had broken the feature.conf fileset, but I just fixed it. Now I can confirm: pig*CLI feature show Builtin Feature Default Current --- --- --- Pickup*8 *8 Blind Transfer# ## Attended Transfer *2 One Touch Monitor *1 Disconnect Call * ** Park Call Dynamic Feature Default Current --- --- --- (none) Call parking Parking extension : 70 Parking context : parkedcalls Parked call extensions: 71-79 but, still no evidence of recording upon sending *1 through box. smime.p7s Description: S/MIME Cryptographic Signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TOS and security
I'm preparing for a client install of * by doing a fresh one in-house. Unlike my earlier installation that runs asterisk as superuser, my current experimental box runs without such privilege. This is causing it to moan that it can't set TOS. I absolutely don't want to install it on the client LAN without this capability. If need be, I'll set the binary to run setuid root. But I'm looking for something more elegant. While googling, I found a suggestion to use iptables mangle rules to set TOS for all packets going out of the box on ports like 5060 and 1:2. Not a bad hack, but indiscriminate and this box will be handling other traffic besides the RTP. I'd like to do better. I thought of using POSIX access control to enable asterisk to do TOS setting without being root (would this be CAP_NET_RAW?), which sounds perfect, but so far I'm operating with stock ubuntu hardy, and I would like to avoid a kernel build to add this capability. Any other ideas? smime.p7s Description: S/MIME Cryptographic Signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] D-Link DVG-3104MS
This appears to be a SIP gateway to four FXO ports for ~$250. Has anybody used it with Asterisk? Comments? http://www.ipphoneshack.com/products/D_Link_DVG_3104MS_VoiceCenter_4_Port_PSTN_Gateway-193-12.html Any good reason to pay for a Mediatrix 1204 or some other box instead? smime.p7s Description: S/MIME Cryptographic Signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] redfone fonebridge2
I'm looking for reports of recent experience with redfone fonebridge2 (with echo can) TDMoE gizmos. Anybody? Good? Bad? smime.p7s Description: S/MIME Cryptographic Signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PoE budget
I'm considering using a PoE switch like this... http://www.tigerdirect.com/applications/SearchTools/item-details.asp?EdpNo=3023334CatId=2800 ...to power as many as 24 Polycom phones of varied kinds. The sales lit indicates 190 watts available for PoE devices. But I'm concerned about a problem someone reported elsewhere... They said... Is there a reason that Polycom phones do not support PoE classes? We ran into a scenario recently where we could only power 11 Polycom 550's on a 24 port switch. This is because the Polycoms do not announce themselves as being in a specific PoE class, even though the phones only need 6W the switch assumes they need as much power as possible and allocates 14.5W to each port. We have had to resort to running unsupported firmware on the switch to get it to power 24 phones. Does anybody here have insight about this? smime.p7s Description: S/MIME Cryptographic Signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-users Digest, Vol 37, Issue 46
I've found OpenVPN to be easy to configure and very robust. It has a zillion options, but they are just that - options. I haven't used it for VoIP, but I've put it to good use doing layer 2 bridging which has eliminated many problems with certain programs traversing NAT and load-balancing routers. I can't think of any reason why it would not work well with Asterisk. On 8/10/07, *MOSBAH ABDELKADER* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hello, Is the OpenVPN the ideal solution to set a tunnel between two asterisk servers or there is a better solution. Thanks. http://www.api-digital.com-- smime.p7s Description: S/MIME Cryptographic Signature ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] exits in NJ
Hooyoo kiddin? Exit 34, I-80. And betta Inglish, myass... Bill, Exit 8, NJTP Date: Tue, 03 Jul 2007 18:13:47 -0400 From: Mark Phillips [EMAIL PROTECTED] Subject: Re: [asterisk-users] Suing Dell||Dull Computers for CID abuse Damn!!! Beat me to it ;-} As an Englishman now living in New Jersey (strangely nowhere near an exit) I have to say that the local idiom and accent leaves a significant amount to be desired. Terms like New Joisey, Shuwa ,wadder, badderies, congradulations etc make me wonder if I'm in an English speaking country at all. I've heard better English spoken in Nigeria. Mark smime.p7s Description: S/MIME Cryptographic Signature ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] got-name
Is it just me, or is the AGI interface at cnam.got-name.com failing for others? Anyone know how to contact them without sending postal mail or telegram? smime.p7s Description: S/MIME Cryptographic Signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-users Digest, Vol 35, Issue 81
Yes, of course. What happens when you dial the number, Daryl? Daryl Jones wrote: Bill Michaelson wrote: Is it just me, or is the AGI interface at cnam.got-name.com failing for others? Anyone know how to contact them without sending postal mail or telegram? I don't know how to contact them, but I am having the same problem. Is this who you mean? http://got-name.com/contact.php Got Name, Inc. 12345 Lake City Way NE Seattle, WA 98125 Phone: 1-727-254-4000 Email: [EMAIL PROTECTED] smime.p7s Description: S/MIME Cryptographic Signature ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] T1 interface
Would anyone care to recommend a T1 interface method for Asterisk that would function as an (external) alternative to a PCI card like the Digium TE120P? Like some sort of T1-SIP gateway? Also, would anyone with experience using these products care to comment on the practical value of the TE207P vs. the TE205P? smime.p7s Description: S/MIME Cryptographic Signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: asterisk-users Digest, Vol 33, Issue 102
[EMAIL PROTECTED] wrote: Date: Sun, 22 Apr 2007 19:38:04 +1000 From: Rob Hillis [EMAIL PROTECTED] Subject: Re: [asterisk-users] Softphone that supports central provisioning? To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 I put such a request for enhancement in sometime, and as is seeming to be frustratingly common for CounterPath, it was completely ignored. Were it not for the Plantronics CS-50 headsets that we bought that have support in a /very/ limited number of softphones, I'd be dumping EyeBeam /and/ X-Lite like the sack of crap that it's proving to be. Steve Davies wrote: On 4/20/07, James FitzGibbon [EMAIL PROTECTED] wrote: I went around this loop with CounterPath a couple of months back. It seems that their idea of provisioning revolves around customising the software before selling it, so that it is locking the end-user into using your (the seller's) SIP server. They had trouble understanding that the user just paid money for this software, which they want to be provisioned by a server on their own network, and they do not support this. I gave up at this stage, but perhaps if more people apply pressure, it will become possible to extend their current (quite useable) provisioning interface, but have a user-configurable setting to determine where the configuration is fetched from. At present the configuration server setting is fixed at compile-time by CounterPath. smime.p7s Description: S/MIME Cryptographic Signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Mediatrix 1204 trix
Thank you, C F and Florian. Now I must expose my ignorance about SIP and Mediatrix... I've adapted my sip.conf to essentially conform with what you've posted. So when I restart the Asterisk server, ethereal indicates that a NOTIFY goes to the Mediatrix (at 192.168.20.188), which responds with a 481, resulting in this message: -- Got SIP response 481 Subscription does not exist back from 192.168.20.188 My guess is that I'm missing a piece of the puzzle on the Mediatrix side of the configuration. Similarly, I've configured the Mediatrix via snmpset commands such that: telephonyAttributesAutomaticCallEnable[*] = 1 and telephonyAttributesAutomaticCallTargetAddress[*] = my desired extension(s) When I call the Mediatrix from POTS, it sends INVITE to Asterisk with the appropriate extension, but Asterisk responds with 404. I think I'm missing something involving REGISTER, but I'm foggy... would somebody clear the haze, please? In my floundering, I tried putting this into sip.conf: register = [EMAIL PROTECTED]/441 But the Mediatrix was unimpressed, rebuffing my entreaty with a: 405 Method Not Allowed I don't take rejection well, and so I'm loathe to speak with the Mediatrix again. I really need someone wiser to advise me... Message: 15 Date: Sat, 16 Sep 2006 21:59:34 -0400 From: C F [EMAIL PROTECTED] Subject: Re: [asterisk-users] Mediatrix 1204 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed I have the same setup as Florian, however I have dtmfmode set to rfc instead of inband On 9/16/06, Florian Overkamp [EMAIL PROTECTED] wrote: Bill Michaelson wrote: Would anyone be kind enough to post a sip.conf fragment as a sample for use with a Mediatrix 1204? Ours works with: [mtrix1] type=peer host=172.28.4.46 mask=255.255.255.255 context=in-mtrix1 qualify=no canreinvite=no dtmfmode=inband disallow=all allow=ulaw Best regards, Florian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Mediatrix 1204
Would anyone be kind enough to post a sip.conf fragment as a sample for use with a Mediatrix 1204? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Imposter binary
I found a bogus binary in my (obviously) hacked system in /usr/sbin. I am still investigating. FWIW, it was 608828 bytes big. It appears to have arrived recently, but I haven't determined how. Here is some more info... sum /usr/sbin/asterisk.suspect 15139 595 I'm just posting this in case it's helpful to anyone. The only reason I noticed this is that asterisk stopped working properly. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom TOS
My 501 admin manual refers to a precenence field, and in another place it refers to a seven bit value. So I would guess it uses diffserv format. Does that help? Date: Mon, 10 Apr 2006 15:32:29 -0400 From: Jonathan k. Creasy [EMAIL PROTECTED] Subject: [Asterisk-Users] Polycom TOS To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Does anyone know the format for the TOS element in the Polycom config? -Jonathan Jonathan Creasy Network Engineer BluegrassNet Development ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom 501's for sale
From: Martin Joseph [EMAIL PROTECTED] It sounds to me like you are suggesting that a QoS infrastructure can be utilized over the internet at large? Is this only true for big guys that have an SLA in place? I would love to discover some QoS mechanism that is respected in general, but that doesn't seem to be the case? Even Speakeasy, when I called them to see if there was an existing QoS modality in place on there network, refused to provide any info. This seems like the single biggest problem for a broader adoption of Voip ? - With Speakeasy, you get a point to point connection like DSL or T-1 to their facility. Presumably from there, they can give QoS through to the TDM network on their own facilities. But I'm assuming... Also, I've received offers of MPLS service from vendors that will tie multiple locations together. I guess one could leverage such an offering to effectively share a TDM gateway among geographically dispersed service sites for VoIP. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: G729 and Meetme
I suppose that in order to accomplish conferencing, Asterisk must produce a broadcast audio stream (waveform) which is a numerically combined derivative of all of the input audio streams. In order to do so, it almost cetainly will work with uncompressed data. Therefore, encoding such as G.729 is unsuitable for this purpose. It must be decoded first. - I have noticed that when I try to connect multiple G729 VoIP devices into a MeetMe conference that I can only add up to the number of G729 licenses I have. Now I would think that because all the devices are G729, this wouldn't be the case and the only license that would ever be used would be if a non G729 device or Zap channel was a part of the Meetme conference. This is apparently note the case. Can anyone explain to me exactly why this is. I don't really mind buying more licenses if I need to but I can't seem to wrap my head around where the Codec translation that is requiring the license is taking place. Regards, Raymond McKay ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GPS-enabled cell phone/PDA
Date: Fri, 24 Feb 2006 14:56:54 + From: Steve Kennedy [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] GPS-enabled cell phone/PDA On Fri, Feb 24, 2006 at 07:17:52AM -0600, Rich Adamson wrote: Its my understanding the cell phone coordinates are sent to the cell phone provider and their equipment reads (and holds) that data. Its not part of any data available to you in any form unless you talk to the cell provider and convience them you have a valid need. Highly unlikely in the US anyway. Even if you could convience them to provide it, they would likely demaand some sort of out-of-band data transmission facility. GSM networks have the Cell ID available to the phone, however that's not much use without the location of the cellsite. There are now location based services, whereby you can query the network and they'll give out an approximate location (most cells are sectored [6 sectors per cell) which gives a direction, the cell also knows what power the phone is transmitting with, and the power it's received so can make a good approximation of where the phone is (within 60 degrees angle). However it's likely a phone will be picked up by several cells, so the network can triangulate and make a better aproximation. Making the information available to end-users is problematic due to privacy issues, unless the user explicitly agrees to give the info away. With GPS units, the info is stored in the phone and can send it out using SMS or other means. - It was my impression that only a handful of cellphones have full GPS units in them. Benefon and some Motorola units made for the former Nextel come to mind. The Benefon units do send SMS reports, and in fact, I have written code to control and track these units via SMS using a Nokia 31 GSM terminal. Unfortunately, aside from their unique GPS/SMS capability, the Benefons are not very attractive products, in my opinion. And they are expensive. The Motorola units contain Java machines and a well defined API for accessing the location data. I have not worked with them. There have undoubtedly been changes in the marketplace since I did this work about 2 years ago. As I understand it (but don't have thorough knowledge and could be mistaken), other units generally only receive GPS satellite signals and relay the data to cellular provider networks where the actual location calculation is done. This can be done with assistance of data obtained based on tower proximity, which jumpstarts the iterative process of approximation. I think it is called assisted GPS or some such... ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] I must be missing something zimple...
I'm configuring a box with a TDM400P with 2 FXS and 2 FXO. I configured the FXO's first to try them, and they worked (I could talk to myself thru the PSTN. But when I add the FXS's to zapata.conf and restart *, I have a problem... [chan_zap.so] = (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found -- Registered channel 4, FXS Kewlstart signalling -- Registered channel 3, FXS Kewlstart signalling Feb 17 16:38:09 WARNING[2613]: chan_zap.c:923 zt_open: Unable to specify channel 1: No such device Feb 17 16:38:09 ERROR[2613]: chan_zap.c:6879 mkintf: Unable to open channel 1: No such device here = 0, tmp-channel = 1, channel = 1 Feb 17 16:38:09 ERROR[2613]: chan_zap.c:10311 setup_zap: Unable to register channel '1' Feb 17 16:38:09 WARNING[2613]: loader.c:414 __load_resource: chan_zap.so: load_module failed, returning -1 -- Unregistered channel 1 -- Unregistered channel 2 Feb 17 16:38:09 WARNING[2613]: loader.c:554 load_modules: Loading module chan_zap.so failed! Now, ztcfg happily shows... televox:~# ztcfg -vvv Zaptel Configuration == Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) Channel 02: FXO Kewlstart (Default) (Slaves: 02) Channel 03: FXS Kewlstart (Default) (Slaves: 03) Channel 04: FXS Kewlstart (Default) (Slaves: 04) 4 channels configured. and my zaptel.conf has this in it: ;### signalling=fxo_ks context=demo channel = 1 ;### signalling=fxo_ks context=demo channel = 2 So, what am I missing? TIA ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] re: Polycom IP501 with Asterisk - distinctive ring
The answer is yes, I think, but I don't recall precisely how off the top of my head, and I'm walking out the door in a moment. The phone will hold more than a dozen distinct ring tones which you can create for yourself, and you can have asterisk direct it to use a ring tone independently of line appearance. The most direct way would be with the SIP Alert-Info field, but the phone itself can associate ring tones with specific callers from its contact directory too. Hopefully, someone will chime in with more precise (and helpful) detail before I return to the office, but I hope my reassurance is helpful anyway... Date: Thu, 9 Feb 2006 10:49:08 -0500 From: Andrew Kohlsmith [EMAIL PROTECTED] Subject: [Asterisk-Users] Polycom IP501 with Asterisk - distinctive ring? To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii The Polycom SoundPoint IP 501 phones have been fantastic so far. I still have a lot to learn when it comes to them, but the manual seems pretty extensive and so far Asterisk has been playing well with them. I have a need to be able to identify incoming calls based on some factor (could be time of day, caller ID, dialed number, it doesn't matter.) -- Assuming Asterisk can differentiate between the calls I want, how do I inform the IP501? There are only three line appearances -- I can't simply just ring a different appearance since there aren't enough of them. Is there a way to get Asterisk to tell the IP501 to use a different ring, put something up on the display, *something* on a dynamic basis? The wiki doesn't seem to have a lot of information about this kind of thing. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] * point to point t1 solution? / alternatives
This has been an interesting discussion for me (except for the sniping). The last post led me, out of curiosity, to this wiki entry: http://www.voip-info.org/wiki-Asterisk+TDMoE I was unaware of this feature, and it looks pretty good. I've been pondering replacing some T1's by leveraging IP capacity but of course have run up against the QoS issue. My idea was different... I don't have production experience with precisely this type of application, but I ask for validation from this list. Pardon me for stating what is undoubtedly obvious to many... The key to assuring adequate performance in replacing a TDM link with IP is to assure that adequate idle time is reserved for voice on the IP segment(s) involved in the route. In this way, latency can be stabilized, and if maintained below a certain (arbitrary) threshold, performance can be deemed acceptable. The first step, of course, is to assure that the virtual TDM allocation does not exceed the available IP bandwidth (so leave a margin, which is huge in the example given). The next step is to use routers which respect the TOS field (however it is used; diffserv/whatever), and finally, to assure that no non-VoIP traffic can be injected into the path with higher routing priority. On a point-to-point link, a pair of typical Linux boxes can do all this. Given the original problem, I would place Asterisk boxes at either end of the link, and have them blend the ordinary traffic with the VoIP traffic (which would probably use IAX to relay calls between the T1s), while assuring (enforcing) that VoIP packets are marked as highest priority. There are varied ways of accomplishing this, and a good reference which I've used in the past can be found at: http://www.lartc.org/lartc.html Additionally, I think one could use the tunneling techniques described in that guide to encapsulate the non-VoIP traffic such that its packets' originally marked TOS values are preserved for transit outside the segment used for TDM emulation. In this way, that part of the segment bandwidth not required for VoIP would function as a dedicated link, allowing other prioritization of traffic such as interactive vs. bulk (or even other voice!), with the added advantage that it could use the reserved VoIP bandwidth when it is otherwise not required (albeit in the case of a T-1 over 10Mb, that's insignificant). Is this easier or harder than TDMoE as described? Does the TDMoE shared idle bandwidth? What about stability (I'm thinking of SW releases)? What other drawbacks or advantages are there? Date: Wed, 25 Jan 2006 23:53:59 -0700 From: Damon Estep [EMAIL PROTECTED] Subject: [Asterisk-Users] * point to point t1 solution? To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Can anyone point me to a reference or sample config for bypassing a nailed up (point to point) t1 between two PBXs with asterisk and a pair of t1 cards? Right now I have 2 Nortel norstars connected to each other via a leased line t1. I also have a solid 10mbps low latency microwave link between the 2 sites. My goal is to run an asterisk box at each end with a t1 card and Ethernet card to act as a TDMSIP gateway to bypass the nailed T1 in a relatively dumb configuration, with the goal of migrating off of the norstars eventually. In past situations I would have done this with a pair of Cisco routers with T1 interfaces in them but in this case I want to get asterisk into the picture as an eventual replacement for the norstars. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] * point to point t1 solution?
You've clarified your requirements for me. Please indulge me - I really want to understand - what are the application implications of this? In other words, what system behavioral changes will your users experience in the various scenarios (pure circuit emulation vs. relay via IAX or similar)? Date: Thu, 26 Jan 2006 07:00:02 -0700 From: Damon Estep [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] * point to point t1 solution? To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 Lets put the TDMoE aside for a minute... The same trunking could be achieved with SIP or IAX, could it not (with higher latency)? The rest of the question remains - is there a way to get asterisk to output, bit for bit, on a t1 interface, the same data that is input on a remote asterisk box t1 interface - using any trunking protocol. This is what would be required to truly emulate a signaling un-aware point to point t1 like one that you would get from a telco if you ordered a point to point esf/b8zs t1 from A location to Z location. Pure circuit emulation - not ISDN/CAS/EM signaled voice. Does that clarify the question at all? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] * point to point t1 solution? / alternatives
Right - so I will assume this makes it slightly more efficient in that respect. And of course, any solution that uses multiple hops brings in a raft of considerations for limiting interference by other data streams - the essential QoS question. Date: Thu, 26 Jan 2006 15:16:25 - From: Steve Langstaff [EMAIL PROTECTED] Remember, however that TDMoE is TDMoE, not TDMoIP - it's not routable (unless you encapsulate it somehow, I guess). ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: * point to point t1 solution? / alternatives
I can appreciate the desire to avoid reconfiguring existing hardware, but that is part and parcel of what we are discussing: reconfiguring hardware. Without further specification, it has no bearing on how to preserve application behavior, which is what we are trying to accomplish with this discussion. I don't wish to second-guess your analysis of the business requirements - you are the authority - but your initial post expressed a desire to move toward an Asterisk configuration as one of your goals. Toward that end, development of an appropriate dialing plan ultimately must happen, and I would think if done properly, would not change dialing patterns or extension numbering unless this is what you desired. I must agree that fax and modem performance is problematic, but here again, this would be an issue anyway when you transition completely to Asterisk, as you implied about your long-term plan. So perhaps now is the time to address this matter. Are you sure you really want to do this at all? Date: Thu, 26 Jan 2006 08:52:04 -0700 From: Damon Estep [EMAIL PROTECTED] what are the application implications of this? In other words, what system behavioral changes will your users experience in the various scenarios (pure circuit emulation vs. relay via IAX or similar)? circuit emulation will; 1. eliminate the need to reconfigure the exisitng hardware. 2. improve the chances that fax and analog modem devices will still work. 3. NOT change any dialing patterns or extensons numbering. there are other, but they are less significant My goal is to run an asterisk box at each end ... with the goal of migrating off of the norstars eventually. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: OT: Legacy systems / fax
Around 1978, when I was consulting to a multinational company in the business of agriculture, I witnessed this configuration in their communications center in NYC: A paper tape punch attached to a teletype machine was busily punching out a tape that was being spewed into a wastebasket. Somehow, running behind it by several feet of tape, was a paper tape reader on another teletype drawing the tape out of the basket, sending the data to who-knows-where. Amazingly, it wasn't getting tangled. To me, this was emblematic of how tradition dies hard... T.30 will be with us for a while to come. Wise managers will limit this to the outer boundaries of their enterprises wherever practical, ASAP. From: Jean-Michel Hiver [EMAIL PROTECTED] Doing a analog (piece of paper) - digital (scanning process) - analog (modulation over TDM) - digital (conversion to TDMoIP) - analog (demodulating on the other fax) - digital (reconstructing the image in fax memory) - analog (printing) conversion doesn't make any kind of sense... It might be great for legacy systems, but it's so not the right way of doing it. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk-Users Digest, Vol 18, Issue 144
I use dbget to set callerid, but it's based on account code, and set statically with the station, not the agent. Users can set callerid by dialing a function coded in the dialplan for that purpose. Overhead is not a problem. In your case, perhaps you can set the desired callerid into a channel variable using the inheritance syntax (double underscore) prior to the queue() operation. Then you can pick it up on the outbound side. - Original Message - From: Franklin Webb To: asterisk-users@lists.digium.com Sent: Tuesday, January 24, 2006 7:34 AM Subject: [Asterisk-Users] Fw: setting outgoing caller ID by the queue anextension is logged into Greetings fellow list members, I am trying to add some tricky functionality to Asterisk dialplan and I was curious if anyone else has come up with a solution to something like this. Basically I have phone representatives that log into one of several queues (not using chan Agent, we log in by the extension), and frequently these agents have to make attended transfer calls to outside numbers. This transfer basically amounts to a new outgoing call. I have been asked to set the caller ID for these outgoing calls based on the queue the phone representative is currently logged in to. Unfortunetly I cannot think of a way to do this. The incomming and outgoing calls are two different calls. I have considered using DBPut and DBGet to store this information in a database. This might work, but I am also concerned about the overhead involved. I cannot think of a way to do this using global variables since I need to store a seperate value for each extension. Has anyone run into an issue like this and come up with a solution? Any thoughts are much appreciated. Thank you, Franklin Webb Assistant IT Project Leader Inter Media Marketing Solutions ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cannot change distinctive ring polycom phones
In sip.cfg, add something like this: alertInfo voIpProt.SIP.alertInfo.1.value=ring3 voIpProt.SIP.alertInfo.1.class=4/ ...to correspond to something like this... SPECIAL_RING se.rt.4.name=ring3 se.rt.4.type=ring-answer se.rt.4.timeout=2000 se.rt.4.ringer=2 se.rt.4.callWait=6 se.rt.4.mod=1/ Please note that I just hacked this example out of my own cfg, modified it for you, and possibly introduced an error, because it is untested now! But it was lifted from working code, so it should get you on the right track, and I hope it helps. Date: Tue, 24 Jan 2006 14:54:31 +0100 From: Giorgio Incantalupo [EMAIL PROTECTED] Subject: [Asterisk-Users] cannot change distinctive ring polycom phones To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed Hi, I'm using asterisk 1.2.1 on a debian sarge distro. I've followed notes in http://www.voip-info.org/wiki/view/Polycom+auto-answer+config and http://www.voip-info.org/wiki/index.php?page=OptiPoint+600+SIP+-+Distictive+ring+using+ALERT_INFO but I still cannot change ring style via asterisk using exten = 666,1,SipAddHeader(ALERT_INFO=ring3) in extensions.conf . Is it possible to do it on polycom ip phones? If yes...how? Is there something missing?? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hardware recommendations
Actually, call groups are yet another layer of complexity. Let me try another explanation. With VoIP, the distinction between a call appearance capability and a line is artificial to an extent. Think of a line as the analog for a pair of copper wires. Think of a call appearance as call waiting capability on a line. Well, not exactly, but it works for me. In practice, "lines" are identities as understood by the phone and the SIP server (Asterisk). So when a call arrives, the actual line it arrives on is indicated in some fashion, depending on the phone. It probably has separate line buttons and/or LEDs to indicate which line is ringing or to press to answer. And because there are different lines, you can specify different behaviors to associate with the lines for whatever purpose, such as call forwarding, anonymous call reject, or whatever. Similarly, you may select from the various lines in order to place an outgoing call which affects, among other things, the call record and caller ID. Whether this is useful to you depends on your organizational requirements. This leaves aside the question of how you direct calls to the phone based on your dialplan, which provides another layer of identity in some sense - a topic for a separate discussion, perhaps. On top of this, each identity (line) can have mutliple call appearances. This simply means that you can have multiple calls in progress (originated or answered) simultaneously. The mechanism for managing this varies by phone and configuration. Beyond this, Asterisk can be programmed to ring multiple lines for an incoming call. Call groups/pickup groups, are a way of defining associations so that a user of a line that is not ringing can answer a call directed to another line (or other lines) which is (are) ringing. Does this make sense? Date: Tue, 24 Jan 2006 10:49:38 -0800 From: Gary Richardson [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Hardware recommendations To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1 From my understanding this is more like a 'Key' system than a 'PBX'. You can make all you phones ring when a certain number is dialed. The first one to pick up gets the call. I can't think of exactly what this functionality is called, but I believe there are menus for it in [EMAIL PROTECTED] Perhaps it's call groups? You need to think of asterisk as a multiplexor -- you have x number of lines coming in from the PSTN and y number of phones. Not all phones are active at one time and it is completely indescriminate when it comes to the next available line. It doesn't matter which line gets picked up when you dial 9, just that you get an outside line. You should be able to get your telco to assign the same phone number on mutliple lines and it will ring through to the next available line (similar to how a T1 works). On 1/24/06, Dane Reugger [EMAIL PROTECTED] wrote: Maybe I am getting this wrong - every phone I look at says it handles a given number of lines. I don't want to spend the extra for 4 appearances when all I need is 2. Where I must be missing something is: Imagine w/ have 2 appearances phones - no operator - the phones just ring. Lets say a call comes in and its for Joe, Joe picks up another call comes in, this time for Fred - he picks up now a call comes for me - wouldn't their above calls occupy all of our appearances? If not I would think we would need some type of operator forwarding the call to the phones instead of just having them ring. Sorry, I'm not getting it - maybe I'm just too old fashioned. I'm trying to do this as simply and economically as possibly w/o sacrificing quality. Your help is GREATLY appreciated. -Dane Kerry Garrison wrote: You need to separate lines from call appearances. Asterisk has lines (actual phone lines) and phones have call appearances (number of simultaneous calls the phone can handle). You could have 1000 lines going into your Asterisk box but the typical user doesn't need more than 2 - 4 simultaneous calls. On the flip side, you could have 4 "lines" coming into your asterisk server and have 100 phones with 4 call appearances each. By using Asterisk to manage the lines, you don't need 400 phone lines to support 100 phones w/4 call appearances each. Kerry Garrison Publisher - http://GeekGazette.com - http://VOIPSpeak.net (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Dane Reugger Sent: Tuesday, January 24, 2006 9:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Hardware recommendations If you have 16 call appearances or lines - how do you get to line 16 - type in some code? Adam Goryachev wrote:
RE: [Asterisk-Users] Polycom FW
Thank you to all who responded to my inquiry below. As explained by a few people, Polycom has a policy of withholding current firmware releases from users, thus forcing them to contact "authorized" resellers for support should they need this code. Similar to another reported experience, I attempted to contact the reseller from whom I purchased this telephone, but was unable to penetrate the screening process because I had not purchased a support contract at additional cost. So I was sitting with a one day old phone that had become a paperweight, and the other sources that *-users list correspondents had kindly referred me to were just not working for me. Perhaps I was doing something wrong. Having painlessly downloaded and used FW from other manufacturers' web sites in the past, I found this predicament exasperating. I contacted another reseller and agreed to purchase another Polycom model if they agreed to provide me with current firmware. After placing an order, I was given ftp access to a site for obtaining the material. Minutes later, I had restored the phone to working order. The sales rep at the second reseller explained that Polycom has this policy in order to "strengthen the reseller channel" which I thought was rather ironic. Well, he did make a sale, I suppose, but it felt like blackmail to this customer. I'm not terribly impressed by the second reseller, and I'm soured on the first. While I have no objection to paying for any kind of support, it does seem that the deliberate creation of impediments such as this do not enhance the image of Polycom, up and down the distribution channel. But of course, that is their privilege. As I stated, I like the phone, and it compares very favorably to others I'm evaluating. However, if all other things were equal (and they could become so with just some firmware improvements), this nonsense will drive me into the arms of a competing manufacturer in a heartbeat. It's needless grief to put your customers through, and that's just stupid, in my opinion. -Original Message- From: Bill Michaelson [mailto:[EMAIL PROTECTED]] Sent: Thursday, January 19, 2006 4:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Polycom FW Anyone know how to obtain firmware and starter .cfg files for Polycom phones? Despite registering at the Polycom web site, I can't locate this stuff. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom FW
Anyone know how to obtain firmware and starter .cfg files for Polycom phones? Despite registering at the Polycom web site, I can't locate this stuff. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom 301 DTMF
Just got a Polycom 301 and I'm configuring. Examples given in wiki recommend using dtmfmode=inband, so that's what I set in sip.conf for this phone, as I have for various other IP phones on my network. But the telephone does not seem to send DTMF tones up thru the network (although I hear them in the handset when I bang the buttons). Also, I can't seem to find a corresponding parameter in the web-based config pages of the phone. Can anyone give me some hints about how best to configure this? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Possible bug with GotoIfTime
Running a fairly recent subversion release of Asterisk, I'm running into a problem using labels (as opposed to priorities) with this application. Here is the dialplan segment: ; isolate gotoiftime bug with labels ;exten = 806,1,GotoIfTime(8:00-20:00,*,*,*?2:4) exten = 806,1,GotoIfTime(8:00-20:00,*,*,*?light:dark) exten = 806,n(light),noop(light) exten = 806,n,hangup exten = 806,n(dark),noop(dark) exten = 806,n,hangup As coded, this is what happens when it executes: -- Executing GotoIfTime(IAX2/hack-2, 8:00-20:00|*|*|*?light:dark) in new stack Jan 7 18:38:09 NOTICE[28137]: pbx.c:1705 pbx_extension_helper: No such label 'light:dark' in extension '806' in context 'default' Jan 7 18:38:09 WARNING[28137]: pbx.c:6312 ast_parseable_goto: Priority 'light:dark' must be a number 0, or valid label == Spawn extension (default, 806, 1) exited non-zero on 'IAX2/hack-2' -- Hungup 'IAX2/hack-2' But if I disable the second exten line instead of the first, it works properly. Beware. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] transfer application
I am having trouble understanding how to use this. I want to transfer
certain incoming calls from an IAX ITSP based on caller ID. From what I
can make of the docs, I thought I need to do something like this...
exten = _NXXNXX,n(nocid),transfer(1000)
exten = _NXXNXX,n,noop(boo,${TRANSFERSTATUS})
exten = _NXXNXX,n,hangup
exten = 1000,1,Dial(IAX2/jnctn_out/16665551234,45,t)
exten = 1000,n,hangup
When the call comes in, the console shows that TRANSFERSTATUS contains
SUCCESS, but there is no evidence that the lines at extension 1000 ever
execute. The caller hears silence; he is not disconnected.
I'm wondering what is happening or what I'm doing wrong, or if it can be
done.
___
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[Asterisk-Users] New Manager Client Program
Here is a work-in-progress that provides pop-up note-taking windows based on caller-ID, outgoing call dialing from directory lookup selection, and other stuff. I hope it's useful to folks. http://asteroid.from.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk-Users Digest, Vol 17, Issue 176
I'm probably mistaken and unaware of a feature, but I thought the concept of dialing an agent does not exist. An agent is not a channel, but rather, someone who associates themself with a station from which they service a queue. You dial the queue with queue() Message: 8 Date: Fri, 30 Dec 2005 20:04:38 +0530 From: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Can we dial agents from extensions.conf To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Thanks a lot Mr. Alexander Lopez for your prompt attension. I tried the same thing but it wouldnot happen. I use it as:- exten = 12,1,Dial(Agent/12) exten = 12,2,Hangup where agent 12 is configured as :- agent = 12,12, vivek ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sixtel
Just curious... Is there anyone out there who has given this outfit money and actually received any service from them? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] manager interface behavior
I'm working on a manager client that I designed to hold open TCP connection to asterisk while it is running for varoius purposes. After being puzzled by unexpected behavior, I realized that the server closes the connection after it completes an originate action - or at least it does in the case of my test transactions. I solicit opinions: is this a feature or a bug? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: manager interface behavior
snacktime wrote: On 11/23/05, Bill Michaelson [EMAIL PROTECTED] wrote: I'm working on a manager client that I designed to hold open TCP connection to asterisk while it is running for varoius purposes. After being puzzled by unexpected behavior, I realized that the server closes the connection after it completes an "originate" action - or at least it does in the case of my test transactions. I solicit opinions: is this a feature or a bug? I've never seen that behavior and I've written several clients for the manager api. I guess it's possible that a particular combination of variables in the request could trigger an error that makes asterisk do that. I would try issuing the same originate by telneting in manually and see what happens. That way you can positively rule out your client being the one that's disconnecting. to which I reply: That's the first thing I did, and it confirmed the behavior (see below). To be precise, the disconnect occurs after the Newchannel report. So I infer that you think it is inappropriate. I've recoded the client so that it immediately reconnects. Anybody actually tried this? I can imagine that the developer might have assumed that such a request would likely come from a transient client, and that it would be helpful to terminate the connection. But if so, I don't think it's the right decision. Maybe it's just an oversight. Any other opinions? I'm too lazy to read the server side code. [EMAIL PROTECTED]:~ telnet hack.cosi.com 5038 Trying 192.168.10.26... Connected to hack.cosi.com. Escape character is '^]'. Asterisk Call Manager/1.0 action: login username: bill secret: dontell Response: Success Message: Authentication accepted action: originate callerid: 00 context: default priority: 1 exten: 212 channel: Local/762 Response: Success Message: Originate successfully queued Event: Newchannel Privilege: call,all Channel: Local/[EMAIL PROTECTED],2 State: Ring CallerID: unknown CallerIDName: unknown Uniqueid: 1132773921.72 Connection closed by foreign host. [EMAIL PROTECTED]:~ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] setting caller ID with Voicepulse
Due to some change I've been unable to identify, my Asterisk box is no longer successfully passing caller ID to the called party with calls placed through Voicepulse. This worked just fine until recently. Also, identical code functions correctly (caller ID arrives) when the call is sent via Junction Networks. I could post a fragment of extensions.conf, but before I do, I wonder if any other users of Voicepulse might want to check for problems. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM400 FXO Screech
A nasty screech. That's what callers here sometimes when they dial into my FXO port from the PSTN. But usually, it works OK. Is this common? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ATA-488 FXO
Is anyone using a Grandstream ATA-488 FXO port to connect a PSTN trunk to their Asterisk box (via SIP, of course)? Is it possible to have such a beast operate reasonably? If so, is it also possible to use the FXS port concurrently and independently? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chanisavail - queuing
Is there anyway to code for queuing for an available trunk. I thought of this while reading about Erlang C. Basically, the idea is that when a caller at an internal extension tries to place a call via PSTN, but all available trunks are busy, the call is placed in a FIFO queue for the first available trunk while the caller hears an appropriate announcement. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] gotta be a dumb question...
...but I'm gonna ask it anyway, because I can't figger it out... Every call that is bridged in my * system begins with a console message like this one... -- Attempting native bridge of SIP/215-b09e and SIP/259412-5967 Now, I've got canreinvite=no in every sip definition, but it happens anyway. Furthermore, every keypress of '*' during a call causes the message to be emitted again. It wouldn't bother me so much except that this is apparently causing the keystroke to be swallowed, thus disabling other features (*1, *2...). Clues, please? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: feature usage/digit detection
Thanks for the answer. Doesn't solve my problem, but that's only because I didn't state my goal. You have corrected a misconseption on my part, which ought to get me closer. I'll explain... Indeed, I do have the "tT" options in the dial command. This is because I thought this would enable the use of the '#' for transfers, and it works satisfactorily. I also have various '*N' definitions in features.conf, but these don't work. I suppose I do have to rethink my strategy as you've suggested, but I don't know how to have my cake and eat it.. (?) By the way, I am using various SIP phones, with various DTMF detection techniques (e.g. ZyXEL wifi:inband, Grandstream BT101 and ATA-488:INFO) with apparent success because many features do work (such as transfer with #). Message: 22 Date: Sun, 30 Oct 2005 10:57:57 -0400 From: Andrew Kohlsmith [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] gotta be a dumb question... To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset="iso-8859-1" On Sunday 30 October 2005 09:44, Bill Michaelson wrote: -- Attempting native bridge of SIP/215-b09e and SIP/259412-5967 Now, I've got canreinvite=no in every sip definition, but it happens anyway. That has nothing to do with reinvites. In Asterisk terms, a native bridge between two channels is the lowest-latency connection between those channels without dropping out of the loop entirely. Essentially a native bridge just reads voice frames from one and transmits them to the other. There is no codec translation or any other goodness going on. When you hit a DTMF digit (you must be using inband DTMF here I think), the native bridge must be dropped because Asterisk needs to prepare to do something with the DTMF (transfer, etc.) -- when Asterisk has determined that it doesn't need to do anything special, it sets up the native bridge again to minimize the latency once again. The fact that your * is getting "swallowed" tells me that you are using * in features.conf to denote special keypresses to Asterisk. In Dial() you likely have the 't' or 'T' flags set, which causes Asterisk to "think" that those DTMF digits are for it, not for the other side. Either edit features.conf, remove the 't' or 'T' flags from the Dial() command or rethink your strategy. I hope this is an acceptable answer, and I certainly hope it's accurate. It's my understanding of the system anyway.If you prefer not to have these types of messages, you need to turn DOWN the verbosity level. -A. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: FCT-11M
Thank you. After some reboots and repeated testing, I've refined my observations. The no-audio problem is gone (no explanation). Through further experimentation I've been able to observe a few consistent things about behavior in its current condition... The main problem seems to be related to disconnect signalling. Simply put, the channel didn't hang up after the GSM connection ends. Instead, I would hear a dial tone thru the bridged side of the call (it's not a North American dialtone). I considered changing the zone in indications.conf, but that is system-wide, and probably inappropriate because I have a Verizon POTS line too. Then I tried putting hanguponpolarityswitch=yes in zapata.conf and it worked - the call would be torn down. But... ...that introduced a new problem on outbound calling, because the channel would hangup immediately upon remote answer. So I added answeronpolarityswitch=yes too, but it had no effect. I also messed around with polarityonanswerdelay= but I was operating in the dark and it didn't help. So I tried to load modules with debug options (zaptel, wctdm, wcfxo), hoping to see more info about device behavior in realtime, but I see nothing new in the message log. But I'm kind of bumbling and stumbling on that. If anyone can offer more precise guidance, I'd be grateful. - Date: Mon, 24 Oct 2005 22:24:21 -0700 From: OTR Comm [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] GSM gateway for Asterisk To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 I forwarded your note below to [EMAIL PROTECTED] I found some docas on the FCT-11M at their site, but it was in Chinese, so I sent them your problem. Hope they will respond to this list and maybe to you directly. Murrah Boswel - Original Message - From: Bill Michaelson [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, October 24, 2005 9:42 PM Subject: [Asterisk-Users] GSM gateway for Asterisk I recently obtained a FCT-11M GSM-analog converter box. It arrived with no documentation. So I popped in a SIM chip, and connected the the RJ11 port to an FXO port on my Asterisk box. It worked smoothly right away for inbound and outbound calls in all respects. For about an hour. Then either spontaneously or due to some action I've been unable to identify, call supervision and other functions became flaky. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] GSM gateway for Asterisk
I recently obtained a FCT-11M GSM-analog converter box. It arrived with no documentation. So I popped in a SIM chip, and connected the the RJ11 port to an FXO port on my Asterisk box. It worked smoothly right away for inbound and outbound calls in all respects. For about an hour. Then either spontaneously or due to some action I've been unable to identify, call supervision and other functions became flaky. First, I noticed inbound calls started malfunctioning. The Asterisk box answers, but no audio is heard on either end (the dialplan bridges to a SIP phone). Also, the call never ends. I can only knock it down by using CLI soft hangup or restarting Asterisk. Then outbound calls got wierd. It will dial out thru the GSM network to my cellphone, and audio is OK in both directions, but call termination fails if initiated from the remote GSM side. In that case, the box emits three short beeps, followed by a steady beep which is audible on the SIP phone to which the call is bridged. The channels don't hangup until the SIP phone causes it to. I was initially concerned that I had fried the FXO port by using an incompatible device, but I've ascertained that the port still works OK with a POTS line. I now suspect that the FCT-11M has been reconfigured somehow, since I obtained it and it was working. But I have no clue about how to examine it's configuration if possible at all. It has a USB (master) port but I don't know what it is for. Does anyone know if English documentation is available, or otherwise have any ideas on how to debug this? Much appreciate any insights. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP-CPE Gateway
Has anyone used the GSM-SIP gateway product produced by a company at sipcpe.com? Any comments? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Budgetone 101 Hold/Xfer/Conf/Flash
Is it possible to use the Hold/Transfer/Conference/Flash keys of the Budgetone-101 (FW 1.0.5.22) with Asterisk? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk-Users digest, Vol 1 #3092 - 11 msgs
I know that the 1 denotes the Zap channel number. That's why I would
not expect it to dial a 1. But it apparently does dial a one. Hence my
original question.
If it did not dial a 1, it would not work because a 1 is required for
the called number, as coded, to work properly with the local phone service.
Furthermore, I discovered this because I originally coded it this way:
exten = _NXX,1,Dial(Zap/1/${EXTEN}|55)
...which simply timed out on the line and failed. Experimentally, I
determined that the telco was expecting 3 more digits, in spite of the
fact that 7 digit dialing is normal for the line.
From: Asterisk Learner [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Dial via X100P
It does not dial a 1. The '1' denotes the Zap channel number which in
this case is probably your X100P. Zap channels are assigned to Zap ports
depending on the order in which you do a modprobe on them.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bill
Michaelson
Sent: Saturday, March 13, 2004 2:18 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Dial via X100P
Just connected my X100P to Verizon. I stumbled across a config that
works, for the moment, with this Dial command:
;this works, because it prefixes a 1 on the dialing. But why does
it?...
exten = _NXX,1,Dial(Zap/1/609${EXTEN}|55)
The comment says it all. The card/SW seems to dial a 1 before it dials
the 609${EXTEN}
Unless I'm misinterpreting what is happening?
This obviously limits my possibilities. Can somebody explain to me why
it dials 1, or appears to?
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[Asterisk-Users] Dial via X100P
Just connected my X100P to Verizon. I stumbled across a config that
works, for the moment, with this Dial command:
;this works, because it prefixes a 1 on the dialing. But why does it?...
exten = _NXX,1,Dial(Zap/1/609${EXTEN}|55)
The comment says it all. The card/SW seems to dial a 1 before it dials
the 609${EXTEN}
Unless I'm misinterpreting what is happening?
This obviously limits my possibilities. Can somebody explain to me why
it dials 1, or appears to?
--
Bill Michaelson - COS, Incorporated - Software Development - [EMAIL PROTECTED]
Thanks for putting up with my spam filter!
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[Asterisk-Users] IAX Native bridge
I coded a dialplan that conditionally forwards a call to my cell phone if no answer on site. During a test, I received a call (via Voicepulse IAX) which correctly Dial'ed out to my cell phone (also via Voicepulse) as expected. Fine - it worked - except that the voice delay was so extreme ( 1sec). But the interesting part came next... -- Call accepted by 66.234.228.132 (format ULAW) -- Format for call is ULAW -- IAX2[voicepulse]/6 stopped sounds -- IAX2[voicepulse]/6 stopped sounds -- IAX2[voicepulse]/6 is ringing -- IAX2[voicepulse]/6 stopped sounds -- IAX2[voicepulse]/6 stopped sounds -- IAX2[voicepulse]/6 answered [EMAIL PROTECTED]/1 -- Attempting native bridge of [EMAIL PROTECTED]/1 and IAX2[voicepulse]/6 -- Channel '[EMAIL PROTECTED]/1' ready to transfer -- Channel 'IAX2[voicepulse]/6' ready to transfer -- Releasing IAX2[voicepulse]/6 and [EMAIL PROTECTED]/1 -- Hungup 'IAX2[voicepulse]/6' Native bridge? Cool! I says to myself. I figure the call will be released from * and handled entirely by Voicepulse, who I assume will bill me appropriately for the remainder of the "outgoing" call. And I'll get better quality for the remaining duration. But the call instead is dropped at this point instead - both sides disconnected from the cloud. Anybody know why and how this is controlled and what my options are? -- Bill Michaelson - COS, Incorporated - Software Development - [EMAIL PROTECTED] Thanks for putting up with my spam filter!
[Asterisk-Users] outgoing spool parallelism
Thanks for the suggestions on the hotel wake-up! Actually, I don't have a hotel, but my earlier request was unanswered because I suppose it was uninspiring. So I used a hard example that was readily identifiable. Your helpful responses led me to the facility I had not managed to find by myself in the docs. Now that I've tried it, and it works, I've got some more specific questions about it's operation... How does * manage concurrency when processing files in the outgoing directory? Does it have some kind of intelligence or controlling mechanism which serializes requests based on the capacity of resource combinations required to satisfy the requests? Or is it just a single thread/processing queue for all requests found in the spool dir? Also, is there any way to control the sequencing (priority) of the enqueued requests? Or is it a random free-for-all? -- Bill Michaelson - COS, Incorporated - Software Development - [EMAIL PROTECTED] Thanks for putting up with my spam filter! ___ 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
[Asterisk-Users] Re: Outgoing parallelism
Thanks, Scott. I'm in a general exploration mode, but I do have a small broadcast application in mind. My limited experimentation leads me to suspect that there is no queue management at all. I was testing with only a single call file just minutes ago, and the system tried to redial the destination as a retry (60 second interval had been spec'ed), even though the first call was still in progress! I suppose I will have to manage throttling with some kind of completely external process, which is likely to be cumbersome. For the immediate application, and given my current facilities, single threading will be adequate (and necessary), but from what I've seen, even this could be challenging. If I put together anything generally useful, I'll share it. From: Scott Stingel [EMAIL PROTECTED] Hi Bill- I've built some load testers for asterisk, using the outgoing call facility. It's been a little while, so you may want to test this yourself, but I recall finding a couple of problems: (a) I don't think it manages queuing very well if there are a limited amount of outbound resources. For example (again, from memory), if you define a group (g9 for example) of two lines for use in outbound calling, it works fine if the number of outbound calls to be made at any moment never exceeds 2. A third call file in this example, will be grabbed by asterisk, but will fail immediately. So I had to create a mechanism in my Perl script to ensure that the outbound calls actually completed - no easy feat since I couldn't tell when that occurs from the perl script too easily. (b) There was a problem dumping more than about 12-15 outbound calls at once in the outgoing directory, even if there were plenty of channels available to make the calls. Asterisk would grab them but would not process some of them. This is a load-testing scenario, and not too common I realise, but something to be aware of. It didn't seem to matter if I switched to a more powerful processor. These problems occurred using a December release of asterisk - maybe they are fixed now?? Please let me know if you are doing any load testing, and I'll send you some simple scripts if you like. The outgoing facility works fine at lower call volumes, if you stagger the creation of the files in the outgoing directory. ___ 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
[Asterisk-Users] Hotel wake-up
Anybody know how to implement a hotel wake-up call feature with *? -- Bill Michaelson - COS, Incorporated - Software Development - [EMAIL PROTECTED] Thanks for putting up with my spam filter! ___ 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
[Asterisk-Users] outdial broadcast
Can someone refer me to an example of an automated broadcasting operation that sends a canned voice message to a list of phone #'s? -- Bill Michaelson - COS, Incorporated - Software Development - [EMAIL PROTECTED] Thanks for putting up with my spam filter! ___ 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
[Asterisk-Users] System called seems forked up
Attempting to correct the problem about which I earlier posted - wherein
a system() call which apparently succeeds is perceived to have failed by
the * process, I changed code in app_system.c so that it would be more discerning...
res = system((char
*)data);
/*
if (res 0) {
ast_log(LOG_WARNING, "Unable to execute '%s'\n", (char *)data);
res = -1;
} else if (res == 127) {
ast_log(LOG_WARNING, "Unable to execute '%s'\n", (char *)data);
res = -1;
*/
if (res == -1) {
ast_log(LOG_WARNING, "Fork failed for '%s'\n", (char *)data);
res = -1;
} else if (WEXITSTATUS(res) != 0) {
ast_log(LOG_WARNING, "Error completion for '%s'\n", (char
*)data);
res = -1;
It is now indeed more discerning, but it has reported Fork failed. But the
fork most certainly has not failed! The shell command invoked has run, and
what's more, completed successfully, producing the expected files.
Referring to the system(2) man page (Red Hat 9, stock)...
RETURN VALUE
The value returned is -1 on error (e.g. fork failed), and the return
status of the command otherwise. This latter return status is in
the
format specified in wait(2). Thus, the exit code of the command
will
be WEXITSTATUS(status). In case /bin/sh could not be executed,
the
exit status will be that of a command that does exit(127).
...and noting that "fork failed" is only an example of an error, I'm wondering
what *other* condition might cause the -1 return value.
Does anyone have any ideas?
[Asterisk-Users] Re: System called seems forked up
It is now indeed more discerning, but it has reported Fork failed. But the fork most certainly has not failed! The shell command invoked has run, and what's more, completed successfully, producing the expected files. Does anyone have any ideas? [EMAIL PROTECTED] suggested: Can you check the errno? strerror(errno); should give you a string of why it failed. (Just be careful not to use other stuff which touches errno after the fork() Of course - very good suggestion (embarrassed I didn't think of it)... anyway... it returns 10, which perror tells me is "No child processes". Sooo, I suppose the spawned process is somehow disassociated from the process group prior to execution of the wait() embedded within the system()? Duuh... I'm still stumped, but I guess we are on to something? On the other hand, if a fork does really fail, one might expect errno to be 10 in that case too. I've half a mind to break it out into a fork/exec/wait for myself, but, uh, ugh. I guess I'm lazy. Please, briliant insights, anybody?
[Asterisk-Users] Re: System call forked - more stuff
It gets better (worse)...
I had been testing with console (-c) mode. When I allow * to run background,
it crashes after the system() call (which succeeds, by the way). The -vvv
option yields these final messages before *poof*...
== Spawn extension
(intern-post, 112, 1) exited non-zero on 'SIP/248379-bcdc'
-- Executing Macro("SIP/248379-bcdc", "record-cleanup") in new stack
-- Executing SetVar("SIP/248379-bcdc", "MONITORDIR=/var/spool/asterisk/monitor")
in new stack
Feb 21 09:56:57 WARNING[1209214528]: ast_expr.y:346 ast_yyerror: ast_yyerror():
syntax error: parse error
-- Executing GotoIf("SIP/248379-bcdc", "0?6:3") in new stack
-- Goto (macro-record-cleanup,s,3)
-- Executing System("SIP/248379-bcdc", "/usr/local/bin/wmix /var/spool/asterisk/monitor/20040221-095652-111-112-in.wav
/var/spool/asterisk/monitor/20040221-095652-111-112-out.wav /var/spool/asterisk/monitor/20040221-095652-111-112")
in new stack
I don't know what the yyerror is about either.
[Asterisk-Users] System cmd usage
Using John Todd's example for recording, from his cleanup/conversion macro...
; Turn the two in/out
.wav files into a single .wav file with both channels
exten = s,3,System(/usr/local/bin/wmix ${MONITORDIR}/${CALLFILENAME}-in.wav
${\
MONITORDIR}/${CALLFILENAME}-out.wav ${MONITORDIR}/${CALLFILENAME})
;
; Remove the old .wav files - we don't need them anymore.
exten = s,4,System(/bin/rm ${MONITORDIR}/${CALLFILENAME}-in.wav ${MONITORDIR}/\
${CALLFILENAME}-out.wav)
;
; This part of the routine compresses the .wav files into a .gsm file for
; better storage (about 1/5 the size of a .wav file). Use "untoast" to
restor\
e
; to normal wav file format. (toast and untoast are fairly standard on Linux
s\
ystems)
;
exten = s,5,System(/usr/bin/toast -F ${MONITORDIR}/${CALLFILENAME})
The wmix runs successfully (it produces the mixed file), and running "by
hand" from the shell indicates that it returns 0 to the shell. But the *
console log seems to think it failed...
-- Executing System("SIP/248379-fe6e",
"/usr/local/bin/wmix /var/spool/asterisk/monitor/20040220-121235-111-916095326873-in.wav
/var/spool/asterisk/monitor/20040220-121235-111-916095326873-out.wav
/var/spool/asterisk/monitor/20040220-121235-111-916095326873") in new stack
Feb 20 12:12:56 WARNING[1209214528]: app_system.c:57 system_exec: Unable
to execute '/usr/local/bin/wmix /var/spool/asterisk/monitor/20040220-121235-111-916095326873-in.wav
/var/spool/asterisk/monitor/20040220-121235-111-916095326873-out.wav
/var/spool/asterisk/monitor/20040220-121235-111-916095326873'
== Spawn extension (macro-record-cleanup, s, 3) exited non-zero on 'SIP/248379-fe6e'
in macro 'record-cleanup'
== Spawn extension (intern-post, s, 1) exited non-zero on 'SIP/248379-fe6e'
Any ideas why?
[Asterisk-Users] Re: System call succeed, asterisk sees failure
Then I infer that the asterisk process is improperly retrieving or interpreting the System process completion code. That would be a serious bug that could break a lot of applications. I wonder if it is specific to some installations or more widespread.
The validity of the code in app_system.c is unclear to me at first glance...
res = system((char *)data);
if (res 0) {
ast_log(LOG_WARNING, Unable to execute '%s'\n, (char *)data);
res = -1;
} else if (res == 127) {
ast_log(LOG_WARNING, Unable to execute '%s'\n, (char *)data);
res = -1;
} else {
if (res ast_exists_extension(chan, chan-context, chan-exte\
n, chan-priority + 101, chan-callerid))
chan-priority+=100;
res = 0;
}
My reading of man pages indicates that the status return by system(2) (refer to wait()) is more than just the value set by an exit() call or returned by a main() function, which seems to be restricted to the low-order byte. I haven't studied it through, but I'm wondering if the hi-order bit can be set, thus causing (res 0) in spite of successful process completion (returning 0).
Could this be the problem?
From: Eric Stanley [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] System cmd usage
Date: Fri, 20 Feb 2004 12:22:12 -0600
Reply-To: [EMAIL PROTECTED]
I saw the same thing. I think I determined that it always failed at the same
point in the macro, no matter what command was being executed. I just put
the whole cleanup process in a shell script and I execute the shell script
from the macro.
Eric
Message: 2
Date: Fri, 20 Feb 2004 12:48:36 -0500
From: Bill Michaelson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] System cmd usage
Reply-To: [EMAIL PROTECTED]
--03060507040102040002
Content-Type: text/plain; charset=us-ascii; format=flowed
Content-Transfer-Encoding: 7bit
Using John Todd's example for recording, from his cleanup/conversion
macro...
; Turn the two in/out .wav files into a single .wav file with both channels
exten = s,3,System(/usr/local/bin/wmix
${MONITORDIR}/${CALLFILENAME}-in.wav ${\
MONITORDIR}/${CALLFILENAME}-out.wav ${MONITORDIR}/${CALLFILENAME})
;
; Remove the old .wav files - we don't need them anymore.
exten = s,4,System(/bin/rm ${MONITORDIR}/${CALLFILENAME}-in.wav
${MONITORDIR}/\
${CALLFILENAME}-out.wav)
;
; This part of the routine compresses the .wav files into a .gsm file for
; better storage (about 1/5 the size of a .wav file). Use untoast to
restor\
e
; to normal wav file format. (toast and untoast are fairly standard on
Linux s\
ystems)
;
exten = s,5,System(/usr/bin/toast -F ${MONITORDIR}/${CALLFILENAME})
The wmix runs successfully (it produces the mixed file), and running by
hand from the shell indicates that it returns 0 to the shell. But the
* console log seems to think it failed...
-- Executing System(SIP/248379-fe6e, /usr/local/bin/wmix
/var/spool/asterisk/monitor/20040220-121235-111-916095326873-in.wav
/var/spool/asterisk/monitor/20040220-121235-111-916095326873-out.wav
/var/spool/asterisk/monitor/20040220-121235-111-916095326873) in new stack
Feb 20 12:12:56 WARNING[1209214528]: app_system.c:57 system_exec: Unable
to execute '/usr/local/bin/wmix
/var/spool/asterisk/monitor/20040220-121235-111-916095326873-in.wav
/var/spool/asterisk/monitor/20040220-121235-111-916095326873-out.wav
/var/spool/asterisk/monitor/20040220-121235-111-916095326873'
== Spawn extension (macro-record-cleanup, s, 3) exited non-zero on
'SIP/248379-fe6e' in macro 'record-cleanup'
== Spawn extension (intern-post, s, 1) exited non-zero on
'SIP/248379-fe6e'
Any ideas why?
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[Asterisk-Users] Inbound IAX to SIP
I've a SIP phone (GS 100) which dials out fine through a Voicepulse Connect account via *. And I've got a phone number which does DID in via IAX from Voicepulse. I want it to ring the GS phone for now. I have this in extensions.conf: [voicepulse-incoming] ; This context tells Asterisk what to do with ; incoming calls from VoicePulse (if you have signed ; up for DIDs ; ; We should now hear a congratulations recording ; on incoming calls to our VoicePulse phone number. ; Once we know that's working, we'll change this to a ; Dial statement (or something else depending on our ; needs). ;exten = _NXXNXX,1,Playback(demo-congrats) exten = _NXXNXX,1,Dial(SIP/248379) exten = h,1,Hangup exten = i,1,Hangup exten = t,1,Hangup ; busy condition N+101... exten = _NXXNXX,102,Playback(demo-congrats) And sip.conf: [248379] type=friend host=dynamic canreinvite=no mailbox=1234 context=demo disallow=gsm dtmfmode=inband But the phone won't ring... it acts busy and I don't understand why. Here is some console info... -- Accepting AUTHENTICATED call from 66.234.228.132, requested format = 4, actual format = 4 -- Executing Dial([EMAIL PROTECTED]/2, Sip/248379) in new stack Feb 17 18:17:56 NOTICE[1209214528]: app_dial.c:506 dial_exec: Unable to create channel of type 'Sip' == Everyone is busy at this time -- Executing Playback([EMAIL PROTECTED]/2, demo-congrats) in new stack -- Playing 'demo-congrats' (language 'en') == Spawn extension (voicepulse-incoming, 6094556707, 102) exited non-zero on '[EMAIL PROTECTED]/2' -- Executing Hangup([EMAIL PROTECTED]/2, ) in new stack == Spawn extension (voicepulse-incoming, h, 1) exited non-zero on '[EMAIL PROTECTED]/2' -- Hungup '[EMAIL PROTECTED]/2' There is also: *CLI sip show peers Name/usernameHost Mask Port Status 248379 (Unspecified) (D) 255.255.255.255 0Unmonitored Clues gratefully accepted. ___ 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] Digium connectivity issue?
I observed a packet routing endless loop at: 16 host-63-108-128-153.apid.com (63.108.128.153) This happened with traceroute from two distinct origination points. Seems to have been resolved. Message: 3 Date: Fri, 13 Feb 2004 20:11:44 -0500 From: John Fraizer [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Digium connectivity issue? Reply-To: [EMAIL PROTECTED] Rich Adamson wrote: Are others seeing hugh delays and/or lack of connectivity to Digium? Rich ___ 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 I saw an over 12hr outage to Digium and IAXtel and I know it wasn't OUR network so, they must have had some sort of problem. John [EMAIL PROTECTED]
[Asterisk-Users] Re: Asterisk-GS and codec selection
Regarding codec selection, I see a minor difference between the FWD and the local * box test cases, but I know nothing about the negotiation protocol... With FWD, the OK message lists 3 Media Formats: Bingo...GS chokes with GSM...just disallow it in your sip.conf: disallow=all allow=alaw allow=ulaw Thank you, very much. That got it working. Actually, I used disallow=gsm as suggested by someone else. Please forgive my ignorance, but this leaves open questions which are nagging me... I expected that the SIP dialog would be a negotiation such that the devices agree on a mutually acceptable encoding. And I think it's obvious (correct me if I'm missing any key points) that such a negotiation would involve selecting one of the encoding formats which appears in both lists presented by each side. It doesn't seem reasonable that the GS should just "flake out" as it seems to do, simply because it is offered an option it can't accept amongst ones that it can. Is this indeed what I am seeing, or am I mischaracterizing it? Also, as I noted earlier, shouldn't * wait for the ACK before spewing the audio stream? It appears to be missing the ACK because it retransmits the OK shortly after it begins sending the RTP data. These loose ends make me very uncomfortable.
[Asterisk-Users] Re: asterisk-grandstream call
I am trying to muddle my way tthrough getting something - actually anything to work - with Asterisk. I've acquired a Grandstream phone and I've got * on a Red Hat 9 box. I've gotten to a point where I can see (via ethereal) that the phone REGISTER's successfully with asterisk, and then I try to dial into voicemail. This is what I observe in the packet trace... GS: INVITE - * *: Status 100 (Trying) - GS *: Status 200 (OK with session description) - GS Does the GS then send an ACK? It should. If it doesn't then this probably means that it hasn't received the 200 response. (firewall issue?) If it is sending the ACK, then it is probably a codec issue, as has been already mentioned. GS doesn't always seem to do very well in codec selection. Doug - Thanks for that hint. I see what you mean. When configured for FWD, the GS does indeed send an ACK at an equivalent point in the protocol. But no, the GS does not send an ACK when configured for my * box. I suppose the * box is expecting it, because about one second later, the * box resends the 200 message - this in spite of the fact that has started spewing RTP furiously. Both devices are on the same LAN, with no intervening firewall, and the OK ought to be visible to the GS (the packet even contains the expected destination MAC ID, derived earlier via ARP). That makes two mysteries: 1) why doesn't the GS seem to see the OK? and 2) why does * send the RTP stream in spite of the fact that it has not received the ACK from the GS? Shouldn't it wait? Regarding codec selection, I see a minor difference between the FWD and the local * box test cases, but I know nothing about the negotiation protocol... With FWD, the OK message lists 3 Media Formats: Media Description, name and address (m): audio 10496 RTP/AVP 0 8 101 Media Type: audio Media Port: 10496 Media Proto: RTP/AVP Media Format: 0 Media Format: 8 Media Format: 101 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-16 But with the local box, it lists one other - note the addition of GSM... Media Description, name and address (m): audio 16708 RTP/AVP 3 0 8 101 Media Type: audio Media Port: 16708 Media Proto: RTP/AVP Media Format: 3 Media Format: 0 Media Format: 8 Media Format: 101 Media Attribute (a): rtpmap:3 GSM/8000 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-16 Don't see much else different in the packets. It might also be relevant that the FWD connection, which works successfully, is through a firewall with NAT. Still fishing... thanks for your attention - much appreciate not being alone here!
