Re: [Asterisk-Users] Why is the internet connection important to LAN and PSTN calls?
I've had this problem too. It would get so bad, that it wouldn't even answer incoming calls, and if I tried to dial out via pstn, I would have hung up before it got around to dialing (which it would eventually do, unfortunately). A short-short term solution was to install bind, and use it as your primary nameserver. Hopefully it'll cache dns queries long enough to survive an outage. A slightly better (in my opinion) solution would be to code a pure caching dns server, whose sole purpose is to look up specific domains and resolve them to their ip address. It'll record the result, and will check every so often (once a minute, hour, day, whatever) and update its results. If it cannot get an answer, it keeps using the last known ip address. If anyone knows of a really bare-bones, standards-breaking dns server that would say, check a flat file database each time a request is made, we could run a daemon that would check the domains we need to resolve; if no answer is received, we just skip that line. That way the daemon will be sitting there waiting for a dns answer, and not asterisk. The best solution would be to fix asterisk (I say fix, as I'm sure many will say it's not broken, that's fine). If your internet connection fails, there's no reason to have internal calls and calls in and out of your pstn lines failing too. Personally, I have a toll-free line that runs over voip, and if it can't reach my server, it'll fall back and dial a landline I have. In this case though, if my internet connection is down for an extended period of time, even those calls won't make it through. Joseph Tanner On 4/11/06, picciuX [EMAIL PROTECTED] wrote: because, a this time, the sip stack doesn't have asynchronous DNS... so ALL the sip code is stucked waiting timeouts for DNS queries (that are long timeouts). When you try to dial a LAN device, the sip code is trying to resolve your voISP service providers' addresses. We workaround this putting all external sip peers in a separate file, say sip_peers.conf, included in sip.conf with #include filename. Then, a daemon on the box try to resolve well-known addresses on well-known DNS servers on the net, every 5 minutes. If the demon fails ALL the well-known DNS queries, it assumes no internet connection is available: then it renames sip_peers.conf, and ask asterisk a sip reload. So all external sip references are out, and sip still continue working for internal phones. Needless to say, when connection come up again, the daemon do the opposite thing. hope this helps 2006/4/11, Brent Torrenga [EMAIL PROTECTED]: Out internet connection was out this morning. It seems that the SIP extensions on our LAN were affected. Behavior like: Call comes in over POTS to a TDM400P, there is a delay then before the Cisco 79[46]0's start to ring. If we were lucky enough to get a call through, then we could not transfer the call, or place the call on hold, or park the call. Outbound calls seemed to have a delay between the time they were dialed at the SIP phone and when they were connected. I know this has been brought up before, in fact there is a bit of a discussion going on now about DNS SRV (in sip.conf, set srvlookup=no, or put all the phone ip's on /etc/hosts). But what is really causing the issue here? Yes, it is DNS, or something related to DNS, but why does that have anything to do with * trying to make a phone ring on the LAN? I would think that by using qualify=yes for any outbound voip trunks we avoid an issue of trying to call out is the net is down, but why are any operations on the LAN affected? Sincerely, Brent A. Torrenga [EMAIL PROTECTED] Torrenga Engineering, Inc. 907 Ridge Road Munster, Indiana 46321-1771 +1 219 836 8918 x325 Voice +1 219 836 1138 Facsimile www.torrenga.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] OT: local calling guide
It's still up for me. They did get a new domain, which currently just redirects to the old site, but it may be a good idea to update your bookmarks anyways in case they have it redirect to a different site in the future. http://www.localcallingguide.com/ Joseph Tanner On 4/7/06, Jonathan k. Creasy [EMAIL PROTECTED] wrote: Anyone know what has happened to the local calling guide? http://members.dandy.net/~czg/search.html -Jonathan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can anybody get me setup with a hosted [EMAIL PROTECTED] box or virtual server in the next 24 hours?
I'm sure lots of people are. I could direct you to some places off-list, or you could ask in the more appropriate asterisk-biz list. Joseph Tanner On 3/16/06, Chuck Fletcher [EMAIL PROTECTED] wrote: I'm looking for a hosting company who's willing to host a [EMAIL PROTECTED] instance for under $100 a month. Experienced and professional asterisk hosting only. Thanks, Chuck ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Native Sounds - in case you missed it...
This is just an idea. I personally love the idea of lower cpu utilization, even more so than better sound quality. So take all your gsm files, and convert them to ulaw, alaw, g729, etc. Now, when someome calls in they'll always get the same quality sound files (i.e., crappy), but cpu usage will be much lower, as it doesn't have to transcode to the correct codec. Best of both worlds! Consistently crappy sound files, and lower cpu usage! BTW, it doesn't really bother me, I have a 2.something GHZ cpu and only a handful of calls are handled at any one time. I just downloaded the sound files to play with. They do sound different from the old ones, probably a combination of the better sound quality and the fact they are new recordings (you can record yourself saying the same thing a dozen times, each one will sound slightly different). Joseph Tanner On 3/15/06, Kristian Kielhofner [EMAIL PROTECTED] wrote: Douglas Garstang wrote: Aren't you bothered by the fact that the sound file quality goes up and down as different sound files are played? It's quite obvious to hear the difference between a ulaw file and a gsm file. Douglas, I know that you have had a hard time grasping this, but not EVERY person uses the sound files in the OPTIONAL asterisk-sounds package. For instance, if you were using Asterisk as a voicemail or conference solution, you would NOT need the prompts in asterisk-sounds. I'll (hopefully) get around to doing those someday... -- Kristian Kielhofner ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] System Design
The DSL speed at the remote office is 1.5 to 6.0 Down and 384 to 608 up. The DSL does have a static IP address and it's pretty rock solid in regards to stability. Curious, why the huge range in numbers? I have 1.5mb/s down and 512kb/s up, it's always been that. Or do you mean you have 6.0mb/s down and 608kb/s up, but in testing sometimes the actual speed tests lower? Anyways, just curious. If you could keep the upload at 608 that'd be great. 384 is a tad on the low side, but even handling uncompressed calls you could handle 3-4 calls. Using compression will help out a lot, especially if you're using that link for non-voip purposes. You'll definitely need some kind of qos. I'd go with another box, preferably one that doesn't require cooling and has no hard drive. I'm just picky, but I hate having multiple points of failure. Another server thrown in could have a fan fail (either locking it up or burning up the processor) or have the hard drive fail (and your whole network is brought down until you take the qos server out of the way). That's another reason I run qos on the asterisk box (I'm cheap, and it's one less possible point of failure, and one less thing to plug into my ups). I'm guessing you're going to run some kind of nat? Whatever you run the nat server on, have that handle qos too. Actually, throw on a decent firewall too. Any low-end cpu should be able to handle the load no problem, heck a 486 would do (again, personally I'd look at a newer cpu that needs no fan to keep cool, feel free to put one on it, but you'll know if it konks out your network is a-ok). Would the remote office * need a couple of POTS lines to make those local calls? It all depends. How many local calls do you plan on making at a time? If generally you need four total incoming/outgoing calls via a local line (that's incoming and outgoing combined, not separate), but will very rarely need say, 5-6 or more, it may be cheaper to just get four lines and any time a fifth call needs to go out, make it as a long-distance call. 1.1cents/minute for a few calls will be cheaper than paying for that fifth or sixth line. Even if you have enough pstn lines to handle all local calls, I'd still have it setup to automatically let them make the call as a long-distance call, never know when that important call needs to be made. You can do the same for incoming calls btw, get a feature called Call Forward Busy and do NOT get call-waiting on the line. When someone calls in, and the line's busy, it'll forward to another number you have via voip (whether it's a local number, or a toll-free number, doesn't matter). Now on those incoming calls, you may get the callerid of the original caller, or the callerid of your regular line (since in effect it's calling your other number, forwarding it on). In fact, you can get this to simulate your own PRI with just a few cheap PSTN lines. It'd be setup something like this: 555-1000: If busy, forward to 555-1001 555-1001: If busy, forward to 555-1002 555-1002: If busy, forward to 555-1003 555-1003: If busy, forward to 555-1004 555:1004: If busy, forward to 555-2000 (a voip number) 555-2000: Unlimited inbound calls Actually, you may want 555-1000 to immediately forward to 555-2000, if bandwidth isn't a concern and the number you're forwarding to is a local call. In my case, there's no local voip providers and I have to forward to a toll-free number, so I would want to keep the calls on the pstn line. Other than the possible caller-id issue (callerid may be of your own pstn line, or of the caller), this setup should work fine. Once again thanks for all of your replies! They are definitely clearing things up for me. - Jason -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joseph Tanner Sent: Wednesday, March 08, 2006 6:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] System Design Lot of questions, lots of variables, but I'll touch base on a few things. 5-10 concurrent calls is hardly anything. A plain T1 will more than handle that, even at ulaw or alaw (non)compression. Throw in a decent codec, and 10 calls won't even put a dent in your T1. Heck, it'd handle all 20 users in your main office, and the 5 users in your remote office with G729, no problem. How reliable is the remote office's DSL connection? I'd make sure you have a static ip for it (dynamic ips are just slightly problematic, especially if you have slightly flaky service, coupled with a slightly flaky modem). If it's reliable, then just keep that. What's the connection speed? Need to know the upload and download. If it's ADSL, then the upload will be a fraction of the download, and will be the limiting factor. Since I don't know your specific setup, I can't tell you specifically what to do. I'll make some guesses though. Keep DSL. No need to use VPN just for asterisk. Make sure
Re: [Asterisk-Users] HW Echo Cancellers
Just a note: This vendor is selling cards with local side echo cancellation. Most of the cards that I purchased didn't have it. The 3 that I've purchased from him did. Two questions. One, why the need for local side echo cancellation? I thought you could just reverse the connection and it would now disable echo in the opposite direction? Just curious, I don't have a T1, and this is just based on what I've read. Two, is there any way to tell what cards have this option just by looking at them? I bought a large lot (40+) and intend to resell them, probably on ebay. I would like to know what extras they have or don't have, so I can list them appropriately. Thanks! Joseph Tanner Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk @ Home 2.6 Call hangs up
My guess, is nat problems. Just for fun, try dialing your inbound number from something not connected to that asterisk box, say a cellphone. I know you're using IAX and SIP, so you'd think you wouldn't run into a double-nat problem (nat going out, nat coming in), but you never know. I have odd issues pop up sometimes when I try calling from my asterisk box right back into it, and I don't even have any nat in the way. Do outgoing calls generally work fine? How do incoming calls work when dialing from an outside line? For the heck of it, try calling out normally, and use a cellphone (or whatever) to dial into the asterisk box. Can it handle an outgoing AND incoming call at the same time, as long as it's not calling itself? If incoming calls still fail, then look into nat issues. Perhaps you can permanently forward port 5060 or 5061 (whichever you use, probably 5060) to your asterisk box, see if that helps any. May need to forward ports 1000-2000 as well. Joseph Tanner On 3/9/06, Jerry Rasmussen [EMAIL PROTECTED] wrote: I have installed asterisk @ home 2.6. I am using a Telasip VOIP account. When I make outbound or inbound calls the calls seem to connect and then get hung up. I was wondering if there was something that I am misisng. I have tried several different sip.conf configurations. Here is what they are currently. telasip-gw context=telasip-in dtmfmode=rfc2833 fromuser=jrasxxx host=gw4.telasip.com insecure=very nat=yes secret=xyz type=peer username=jrasxxx 551212 context=from-pstn dtmfmode=rfc2833 host=gw4.telasip.com insecure=very nat=yes qualify=yes secret=xyz type=peer username=jrasxxx The odd thing is it worked once or twice then stopped. If anyone could shed some light it would be greatly apperciated. Here is what the asterisk output looks like: -- AGI Script fixlocalprefix completed, returning 0 -- Executing SetVar(IAX2/100-2, OUTNUM=770555) in new stack -- Executing Cut(IAX2/100-2, custom=OUT_2|:|1) in new stack -- Executing GotoIf(IAX2/100-2, 0?16) in new stack -- Executing Dial(IAX2/100-2, SIP/telasip-gw/770555) in new stack -- Called telasip-gw/770555 -- SIP/telasip-gw-3091 is ringing -- SIP/telasip-gw-3091 answered IAX2/100-2 == Spawn extension (macro-dialout-trunk, s, 14) exited non-zero on 'IAX2/100-2' in macro 'dialout-trunk' == Spawn extension (from-internal, 770555, 1) exited non-zero on 'IAX2/100-2' -- Executing Macro(IAX2/100-2, hangupcall) in new stack -- Executing ResetCDR(IAX2/100-2, w) in new stack -- Executing NoCDR(IAX2/100-2, ) in new stack -- Executing Wait(IAX2/100-2, 5) in new stack -- Executing Hangup(IAX2/100-2, ) in new stack == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'IAX2/100-2' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'IAX2/100-2' -- Hungup 'IAX2/100-2' ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Question from a newbie on finding digium hosts
What exactly do you need? A digium card could be anything from one pstn line, to multiple t1 lines, to who knows what else. And serial number authentication...what's this for? Does a user dial in, enter in a serial, then get access to something? Like a calling card, or something completely different? If all you need is rack space, I'm sure there's some people here who could help you out. Even I have rack space available, and I'm not exactly a big host. Maybe you could ask this on the biz list? If all you need is an internet connection (don't need a voice T1 line), then just about anybody who can colocate a server will do. Might even be cheaper to lease a server (seems odd, but leasing a server can be cheaper than just renting space for a server you own). WebHostingTalk.com is a good place to look for a host, but first we need to know exactly what you need, then we can steer you in the right direction. Joseph Tanner On 3/7/06, Gene Expression [EMAIL PROTECTED] wrote: Hey all, I have a client whose previous programmer ditched. I'm his webmaster, and now he wants me to have an asterisk system set up for serial number authentication...and I have a digium card from the previous guy. Are there hosts that will set this up for me? ie, rack space somwhere? Are there guides online I can look at? Thanks Razib ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Any way to change dns timeout value? Asterisk hangs if internet unreachable
I don't have the most reliable internet connection in the world. Whenever it goes out, I can't receive any incoming calls at all, not even from pstn. When it first goes out I can still make outgoing calls through pstn, but eventually that fails too (as does voicemail, everything's out). Yes, asterisk and the local phones are all on the same network and can communicate fine. Ok, that's the symptom, and I believe I know what's causing it. Asterisk seems to be hanging on dns lookups. After a while, it gets so bad that it won't process anything at all. The reason incoming calls via pstn won't work is because I have a calleridname.agi script that runs as soon as a call comes in. Instead of trying for say, 5 seconds and then giving up, asterisk just sits there forever waiting for it to resolve. Once asterisk gives up, the caller has hung up ages ago. Obviously, I don't want pstn calls to be dependent on my internet connection, kinda defeats having a pstn line at all. Now, as soon as the internet connection craps out, I can still make outgoing calls via pstn, access voicemail, etc. If it's a long outage (like this morning, some fiber cut and the whole county is without internet, redundancy anyone?), eventually everything stops. I think it's because asterisk is re-trying to register with a host, before the dns timed out, and the built-up dns queries just bring the whole thing to a halt eventually. This morning after I noticed the internet connection was down, I tried to call the phone company (through the pstn line) and could not. When I watched the CLI, I noticed it try to call a minute or two after I hung up, quite a delayed reaction. Also could not access voicemail. When the connection came back up for a minute and crapped back out again, I was suddenly able to access voicemail and make a call. Shortly after that, I'd dial a number and it'd connect after 10 seconds or so. After that, it wouldn't try to connect until after the phone received a fast busy. A workaround was to backup my sip.conf and iax.conf files, then edit them taking out every single host reference that wasn't an ip address. If I left them in and tried to restart asterisk, it would hang on the first host trying to resolve. A minute or so later it'd give up and move on to the second. Obviously very bad news if you have several hosts that it needs to resolve (side note, why can't asterisk try to resolve multiple hosts at once; say one every 5 seconds, so it doesn't flood your network with dns requests, but also if one host hangs it can try resolving other hosts while waiting?). I've looked in dns.c and dnsmgr.c and can't see where I can set a timeout. Perhaps it's somewhere else? Maybe hiding in several files? Any ideas? I'd like to set it to five seconds, this should give most hosts that aren't down plenty of time to respond. Perhaps even better, I could cache dns results and save them to a file? Run a background application to query dns servers, if it hangs then asterisk uses the last good values (and if it's not reachable, no big deal, asterisk will just move on). I promise I searched on google before posting here. The closest thing I could find is this: http://bugs.digium.com/view.php?id=3946 Doesn't seem to have a real solution. Joseph Tanner ___ --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] System Design
Lot of questions, lots of variables, but I'll touch base on a few things. 5-10 concurrent calls is hardly anything. A plain T1 will more than handle that, even at ulaw or alaw (non)compression. Throw in a decent codec, and 10 calls won't even put a dent in your T1. Heck, it'd handle all 20 users in your main office, and the 5 users in your remote office with G729, no problem. How reliable is the remote office's DSL connection? I'd make sure you have a static ip for it (dynamic ips are just slightly problematic, especially if you have slightly flaky service, coupled with a slightly flaky modem). If it's reliable, then just keep that. What's the connection speed? Need to know the upload and download. If it's ADSL, then the upload will be a fraction of the download, and will be the limiting factor. Since I don't know your specific setup, I can't tell you specifically what to do. I'll make some guesses though. Keep DSL. No need to use VPN just for asterisk. Make sure each end has a static ip (dynamic ip will work, but is harder to setup and more prone to errors). Have each asterisk box register to the other. For normal incoming and outgoing calls, just have the asterisk box at that particular location handle it (no need for the remote office to connect to the main office's asterisk box, then call out via iax or sip for a long-distance phone call). You can create local extensions that when dialed, will ring a person on the other asterisk box. I.e., a user at the main office can dial 2001, and get a user at the remote office. If you deal with call queues you can group users from both offices together, no problem. A T1 or a point to point connection at the remote office would work, but is probably unecessary. If their DSL connection is flaky and unreliable, then start looking at both options. I'd probably go with whichever is cheapest, be sure to factor in equipment costs (you can generally lease equipment with a T1 line, but not with a point to point connection). As far as server specs, if all it's going to run is asterisk, then that's overkill even if it was handling all the calls. If you think you need that much server but are on a budget, then get one setup for dual processors but with just one installed, and less ram but that has room to add more. If budget's not a problem, I say go for it! That system should last you for quite a while. As for QOS, sorry I can't help you there. You could get a cheap router that has QOS built-in, or run a separate low-end server just for QOS. Personally my asterisk box also serves as my nat server, so I just run QOS directly on it. It's probably not something you want to do in an office environment, but it's better than no QOS at all. Hopefully someone else will give you some good advice on QOS equipment. Joseph Tanner On 3/7/06, Jason Adams [EMAIL PROTECTED] wrote: Hey Everyone, We are in the works of planning a new * installation for our company. We have 20 users in our main office and 5 users in a remote office a couple of states away. Our call volume for the main office will be anywhere from 5-10 concurrent calls. The remote office will have about 3 heavy users with two users making calls occasionally. Right now we have an existing PBX. We have a T-1/PRI coming into the main office and a DSL connection at the remote office. We have a Cisco 2610/PIX 501 at the main office a cheesy linksys router at the remote site. We are planning on purchasing new Cisco IP phones for everyone. My main question is this: What type of hardware/network design would be best for this situation? Would a full T-1 at the remote site work with a VPN between the offices? Or would a higher bandwidth DSL work with a VPN? Or should we move to a Point-to-Point connection? What type of hardware would be best for the end-to-end communication in regards to QoS? I know the PIX 501 doesn't support it. Would it be best to have two * servers in each office or for that call volume at the remote office does it make sense? I was thinking of a Dell Power Edge server with 4GB of ram and a dual processor.. is that enough? Sorry for all the questions! - Jason ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PAP2 won't make two g729 calls at the same time
The PAP2 can only handle one g729 call at one time. Whether that's a hardware limitation, or licensing, or both, I don't know. Joseph Tanner On 3/8/06, Warren Burstein [EMAIL PROTECTED] wrote: I have a Linksys PAP2. Identical setups for the two channels in both the unit and in Asterisk. In particular, both channels enable g729 and set it as the preferred codec, and have disallow=all and allow=g729 in sip.conf. If we make a call on one channel, it works (and uses g729), but if we make a call on the other channel when the first one is still connected, it fails. We have three g729 licenses, and no others were in use at the times this happened, but even if we didn't have enough, how would the PAP2 know that? Here's a good, and a bad INVITE message, from the log file with sip debug enabled. Has anyone seen anything like this? INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.254.44:5060;branch=z9hG4bK-3e521aa From: PAP 220 sip:[EMAIL PROTECTED];tag=6b66e68deef168b2o0 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 101 INVITE Max-Forwards: 70 Contact: PAP 220 sip:[EMAIL PROTECTED]:5060 Expires: 240 User-Agent: Linksys/PAP2-3.1.3(LS) Content-Length: 246 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Content-Type: application/sdp v=0 o=- 261305180 261305180 IN IP4 192.168.254.44 s=- c=IN IP4 192.168.254.44 t=0 0 m=audio 16392 RTP/AVP 18 100 101 a=rtpmap:18 G729a/8000 a=rtpmap:100 NSE/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.254.44:5060;branch=z9hG4bK-3fda1e15 From: PAP 220 sip:[EMAIL PROTECTED];tag=b8b86be991749af5o0 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 101 INVITE Max-Forwards: 70 Contact: PAP 220 sip:[EMAIL PROTECTED]:5060 Expires: 240 User-Agent: Linksys/PAP2-3.1.3(LS) Content-Length: 267 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Content-Type: application/sdp v=0 o=- 261589835 261589835 IN IP4 192.168.254.44 s=- c=IN IP4 192.168.254.44 t=0 0 m=audio 16400 RTP/AVP 0 8 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:100 NSE/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Memory Problems
The answer's just below the part you bolded. Use a HIGHMEM enabled kernel. Joseph Tanner On 3/8/06, Dumpolid Exeplish [EMAIL PROTECTED] wrote: Hello, This is not a question directly related to asterisk. I am currently rinning ansterisk on a Debian server and i just upgraded my memory from 1GB to 2GB. However, my linux OS does not recognise the memory upgrade. The BIOS does, but the Debian Linux refuses to use the entier memory, currently, it registered only 900MB. Can anyone tell me why thi is and a solution to this?? My Debian version is Linux asterisk 2.6.12.3 #1 Mon Aug 1 19:33:51 WAT 2005 i686 GNU/Linux The server is currently routing calls from SIP internal users through an E1 card (TE410) OUTPUT FROM dmesg command 009dc00 (usable) BIOS-e820: 0009dc00 - 000a (reserved) BIOS-e820: 000f - 0010 (reserved) BIOS-e820: 0010 - 7fee (usable) BIOS-e820: 7fee - 7fee3000 (ACPI NVS) BIOS-e820: 7fee3000 - 7fef (ACPI data) BIOS-e820: 7fef - 7ff0 (reserved) BIOS-e820: fec0 - 0001 (reserved) Warning only 896MB will be used. Use a HIGHMEM enabled kernel. 896MB LOWMEM available. found SMP MP-table at 000f5a20 On node 0 totalpages: 229376 DMA zone: 4096 pages, LIFO batch:1 Normal zone: 225280 pages, LIFO batch:31 HighMem zone: 0 pages, LIFO batch:1 -END ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Memory Problems
Something like: up2date -i kernel-hugemem Then make the appropriate changes in /etc/grub.conf, reboot, and see if it works. Of course, that's an overly simplified explanation, if this is a production system please research this first. If it's a test system, well what's the worst that could happen? Joseph Tanner On 3/9/06, Dumpolid Exeplish [EMAIL PROTECTED] wrote: So how do I enable a High mem Kernel? Do i have to recomplile the kernel to use highmem ?? On 3/9/06, Joseph Tanner [EMAIL PROTECTED] wrote: The answer's just below the part you bolded. Use a HIGHMEM enabled kernel. Joseph Tanner On 3/8/06, Dumpolid Exeplish [EMAIL PROTECTED] wrote: Hello, This is not a question directly related to asterisk. I am currently rinning ansterisk on a Debian server and i just upgraded my memory from 1GB to 2GB. However, my linux OS does not recognise the memory upgrade. The BIOS does, but the Debian Linux refuses to use the entier memory, currently, it registered only 900MB. Can anyone tell me why thi is and a solution to this?? My Debian version is Linux asterisk 2.6.12.3 #1 Mon Aug 1 19:33:51 WAT 2005 i686 GNU/Linux The server is currently routing calls from SIP internal users through an E1 card (TE410) OUTPUT FROM dmesg command 009dc00 (usable) BIOS-e820: 0009dc00 - 000a (reserved) BIOS-e820: 000f - 0010 (reserved) BIOS-e820: 0010 - 7fee (usable) BIOS-e820: 7fee - 7fee3000 (ACPI NVS) BIOS-e820: 7fee3000 - 7fef (ACPI data) BIOS-e820: 7fef - 7ff0 (reserved) BIOS-e820: fec0 - 0001 (reserved) Warning only 896MB will be used. Use a HIGHMEM enabled kernel. 896MB LOWMEM available. found SMP MP-table at 000f5a20 On node 0 totalpages: 229376 DMA zone: 4096 pages, LIFO batch:1 Normal zone: 225280 pages, LIFO batch:31 HighMem zone: 0 pages, LIFO batch:1 -END ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Two asterisks on one machine
You could run a virtual machine. I'd try xen, uml, and vmware in that order (vmware would be the easiest/quickest to setup, but is more of a resource-hog than xen or uml). Assign a separate ip to the virtual server, setup asterisk, and you're all set. BTW, just curious but why can't you run one asterisk install with both h323 and sip? It'd simplify things and use less resources than running a virtual server, assuming it works for you. Another idea, if one's solely for h323 and the other's solely for sip (neither will be running both), then you could compile asterisk twice, using different directories for each install. I don't think this would work if both needed to use the same ports. I'm guessing you want to bridge the h323 asterisk to the sip asterisk? If not, but you do want to use sip on both, perhaps you can use port 5060 on one and 5061 for the other. Couldn't bridge them, but both could talk to the outside world (that is, maybe they could, I haven't tried this and do not know what's involved). Running one in a virtual server is probably going to be the easiest way to get two asterisk processes to coexist on the same physical server. Joseph Tanner On 3/6/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hello friends, Can I run two asterisks running simultaneously on the same machine? I want one to run v1.0.2 for h323 ( which is an old and running production system ) and one for sip implementation. I wonder how it can be done since they will want access to the same ports and ip addresses. Does anyone know to do this or has done this before? Please share your experiences please. With warm regards. Vivek J. Joshi. [EMAIL PROTECTED] Trikon electronics Pvt. Ltd. --New opinions often appear first as jokes and fancies, then as blasphemies and treason, then as questions open to discussion, and finally as established truths. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk download file locations
If it's a commercial product, you should definitely mirror the files. Not only because you're benefiting financially, but because you need full control. Perhaps you'd like to incorporate a patch or two in the source? Or maybe you'd like to use a stable label, so the script downloads stable.tar.gz. Once you've tested a new version and it works with your customizations/patches/whatever, you just upload it and rename it as stable.tar.gz, and any customer who runs your script automatically gets the latest and greatest. You could simulate some of this without mirroring asterisk though. Have the script check your server for a value, say the location to download asterisk. This will let you update the URL if it changes, or have it point to a newer version of asterisk, etc. Of course, I would hardcode in some values that the script could use, in case it can't reach your server but can reach digium's. Just some thoughts. Joseph Tanner On 3/6/06, Peter Fern [EMAIL PROTECTED] wrote: Still, if you mirror them yourself, this problem all but goes away. Alistair Cunningham wrote: Colin, Because having the logic is not the correct thing to do from an engineering point of view. Consider: - What if Digium change the directory structure again? Having a published directory structure is the elegant thing to do. - Not only does it break build scripts but it breaks search engines too. - Our scripts already have more conditional logic than I'm happy with, dealing with all the inconsistencies that Linux distributions throw at us. Anything which makes the installation process less brittle is a good thing. Alistair Cunningham, Integrics Ltd, +44 20 799 39 799 sip:[EMAIL PROTECTED] http://integrics.com/ Colin Anderson wrote: Why wouldn't you build in trivial conditional logic into your script or mirror the Asterisk builds yourself? -Original Message- From: Alistair Cunningham [mailto:[EMAIL PROTECTED] Sent: Monday, March 06, 2006 8:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk download file locations This is a request to the website manager for asterisk.org. The build scripts for our ITSP product include the URLs to download the Asterisk files, such as: wget http://ftp.digium.com/pub/asterisk/asterisk-1.2.5.tar.gz; However, if a new version is released, asterisk-1.2.5.tar.gz is moved to the old directory. This breaks our scripts until we can update them and send them to our resellers. Would it be possible to have a fixed address for a particular asterisk release that will never (or at least not for a long time) change? Perhaps put all (except very old) versions in the same directory, with a 'latest' link to the latest one? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Does an entry in AstDB stay after reboot?
Yes, the AstDB is not erased on a reboot. All values are saved. I
don't know the exact location, but you can back it up and restore on
another server if you wished, useful for upgrading.
Joseph Tanner
On 3/3/06, Min Hwan Chang [EMAIL PROTECTED] wrote:
I set up a call forwarding script in extensions.conf which uses the
AstDB but I'm wondering if I reboot the server, will the entry in
AstDB still reside?
What the script does is when a call comes in, it check to see if there
is a null value or a call forward number. If null, it will call the
local office connections. If there is a number, it calls that. Now I
just need to know if I reboot the system, the current value will still
be in the AstDB, I don't want to reboot the system and then find out
the key is always getting erased. Thanks all for the help.
exten = s,1,Answer()
exten = s,2,DBget(WHEE=CFIM/temp)
exten = s,3,GotoIf($[${WHEE} = 0]?s-NoCFIM,1:s-CFIM,1)
; If call forward number in AstDB, then call that number
exten = s-CFIM,1,DBget(WHEE=CFIM/temp)
exten = s-CFIM,2,Dial(Zap/g0/${WHEE},20,tm)
exten = s-CFIM,102,Voicemail,u203
exten = s-CFIM,103,Voicemail,b203
; The 0 signifies no call forwarding.
exten = s-NoCFIM,1,Dial(${PHONE1}${PHONE2}${PHONE3},10,tr)
exten = s-NoCFIM,2,Wait(1)
exten = s-NoCFIM,3,DigitTimeout(3)
exten = s-NoCFIM,4,ResponseTimeout(7)
exten = s-NoCFIM,5,Background(/var/lib/asterisk/sounds/hello)
;exten = s-NoCFIM5,Background(/var/lib/asterisk/sounds/vacation)
exten = s-NoCFIM,6,Wait(1)
exten = s-NoCFIM,7,Voicemail,u203
exten = s-NoCFIM,8,Voicemail,b203
exten = s-NoCFIM,9,Hangup()
[call forward options]
; Creates a call forward
exten = _44X.,1,DBput(CFIM/temp=${EXTEN:2})
exten = _44X.,2,Hangup
; Null character.
exten = 4455,1,DBput(CFIM/temp=0)
exten = 4455,2,Hangup
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Re: [Asterisk-Users] How to route incoming calls to different contexts?
First, tell us if it's sip, iax, or zap. Then tell us what provider (most will use the same general config, but some like ipkall are special and a bit tricky). joseph Tanner On 3/3/06, Zach A [EMAIL PROTECTED] wrote: Hi everybody, It should be a simple thing to do but I don't know how to do it. Now I have 2 DIDs and I want one of them go to [context1] and other one to go to [context2]. How can I achieve this. Thanks, Zach A ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] really need help with outgoing calls..PSTN errors
Put a w or 2 (ww) in front of your number being dialed, it should work
then. If not, try more w's.
Joseph Tanner
On 3/3/06, sdgesa gaeharth [EMAIL PROTECTED] wrote:
I cant seem to get outgoing calls to be placed properly .. No matter what I
try I get an error from the PSTN company saying that the call can not be
completed as dialed or you need to dial a one... The asterisk debugging
seems to show the correct number being dialed out of the zap interface...
the 9 is being stripped and there is a 1 where it is supposed to be. I
am thinking it is a problem between the zap interface and the PSTN.
thanks
extensions.conf
[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=no
[globals]
ATTENDANT=1001
OUTBOUNDTRUNK=ZAP/g1
[extentions]
exten = _10XX,1,Ringing
exten = _10XX,2,Dial(SIP/${EXTEN},20)
exten = _10XX,3,Answer
exten = _10XX,4,VoiceMail([EMAIL PROTECTED])
exten = _10XX,5,Hangup
[voicemail]
exten = _910XX,1,Wait(1)
exten = _910XX,2,VoiceMailMain(${EXTEN:[EMAIL PROTECTED])
[local]
include = extentions
include = voicemail
[incoming]
exten = s,1,Answer
exten = s,n,Wait(2)
exten = s,n,Set(TIMEOUT(response)=15)
exten = s,n,Background(company-intro)
exten = s,n,WaitExten()
exten = s,n,Playback(vm-goodbye)
exten = s,n,Hangup()
exten = 0,1,Dial(SIP/${ATTENDANT},20)
exten = 1,1,Directory(voicemail,extentions,f)
exten = 2,1,Directory(voicemail,extentions)
exten = 1234,1,Playback(abandon-all-hope)
include = extentions
exten = i,1,Playback(vm-goodbye)
exten = i,2,Hangup()
exten = t,1,Playback(vm-goodbye)
exten = t,2,Hangup()
[outbound]
ignorepat = 9
exten = _9XX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
exten = _9XX,2,Congestion()
exten = _9XX,102,Congestion()
exten = _91800NXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
exten = _91800NXX,2,Congestion()
exten = _91800NXX,102,Congestion()
exten = _91888NXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
exten = _91888NXX,2,Congestion()
exten = _91888NXX,102,Congestion()
exten = _91877NXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
exten = _91877NXX,2,Congestion()
exten = _91877NXX,102,Congestion()
exten = _91866NXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
exten = _91866NXX,2,Congestion()
exten = _91866NXX,102,Congestion()
exten = _91900NXX,1,Congestion()
exten = _91976NXX,1,Congestion()
exten =
_91[1234567]XXNXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
exten = _91[1234567]XXNXX,2,Congestion()
exten = _91[1234567]XXNXX,102,Congestion()
exten = 9911,1,Dial(${OUTBOUNDTRUNK}/911)
exten = 9411,1,Dial(${OUTBOUNDTRUNK}/411)
exten = 0,1,Dial(${OUTBOUNDTRUNK}/0)
[local-access]
include = local
include = outbound
zapata.conf:
[channels]
group = 1
language=en
context=incoming
signalling=fxs_ks
switchtype=national
usecallerid=yes
hidecallerid=no
callwaiting=yes
callerid = Dulles Micro, LLC 703 450 5000
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
channel = 1
zaptel.conf:
fxsks=1,2,3,4
loadzone = us
defaultzone=us
Brings words and photos together (easily) with
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Re: [Asterisk-Users] really need help with outgoing calls..PSTN errors
Like this:
exten = _9XX,1,Dial(${OUTBOUNDTRUNK}/ww${EXTEN:1})
Joseph Tanner
On 3/4/06, sdgesa gaeharth [EMAIL PROTECTED] wrote:
You mean like this
exten = ww_9XX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
thanks
Joseph Tanner [EMAIL PROTECTED] wrote:
Put a w or 2 (ww) in front of your number being dialed, it should work
then. If not, try more w's.
Joseph Tanner
On 3/3/06, sdgesa gaeharth wrote:
I cant seem to get outgoing calls to be placed properly .. No matter what
I
try I get an error from the PSTN company saying that the call can not be
completed as dialed or you need to dial a one... The asterisk debugging
seems to show the correct number being dialed out of the zap interface...
the 9 is being stripped and there is a 1 where it is supposed to be. I
am thinking it is a problem between the zap interface and the PSTN.
thanks
extensions.conf
[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=no
[globals]
ATTENDANT=1001
OUTBOUNDTRUNK=ZAP/g1
[extentions]
exten = _10XX,1,Ringing
exten = _10XX,2,Dial(SIP/${EXTEN},20)
exten = _10XX,3,Answer
exten = _10XX,4,VoiceMail([EMAIL PROTECTED])
exten = _10XX,5,Hangup
[voicemail]
exten = _910XX,1,Wait(1)
exten = _910XX,2,VoiceMailMain(${EXTEN:[EMAIL PROTECTED])
[local]
include = extentions
include = voicemail
[incoming]
exten = s,1,Answer
exten = s,n,Wait(2)
exten = s,n,Set(TIMEOUT(response)=15)
exten = s,n,Background(company-intro)
exten = s,n,WaitExten()
exten = s,n,Playback(vm-goodbye)
exten = s,n,Hangup()
exten = 0,1,Dial(SIP/${ATTENDANT},20)
exten = 1,1,Directory(voicemail,extentions,f)
exten = 2,1,Directory(voicemail,extentions)
exten = 1234,1,Playback(abandon-all-hope)
include = extentions
exten = i,1,Playback(vm-goodbye)
exten = i,2,Hangup()
exten = t,1,Playback(vm-goodbye)
exten = t,2,Hangup()
[outbound]
ignorepat = 9
exten = _9XX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
exten = _9XX,2,Congestion()
exten = _9XX,102,Congestion()
exten =
_91800NXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
exten = _91800NXX,2,Congestion()
exten = _91800NXX,102,Congestion()
exten =
_91888NXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
exten = _91888NXX,2,Congestion()
exten = _91888NXX,102,Congestion()
exten =
_91877NXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
exten = _91877NXX,2,Congestion()
exten = _91877NXX,102,Congestion()
exten =
_91866NXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
exten = _91866NXX,2,Congestion()
exten = _91866NXX,102,Congestion()
exten = _91900NXX,1,Congestion()
exten = _91976NXX,1,Congestion()
exten =
_91[1234567]XXNXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
exten = _91[1234567]XXNXX,2,Congestion()
exten = _91[1234567]XXNXX,102,Congestion()
exten = 9911,1,Dial(${OUTBOUNDTRUNK}/911)
exten = 9411,1,Dial(${OUTBOUNDTRUNK}/411)
exten = 0,1,Dial(${OUTBOUNDTRUNK}/0)
[local-access]
include = local
include = outbound
zapata.conf:
[channels]
group = 1
language=en
context=incoming
signalling=fxs_ks
switchtype=national
usecallerid=yes
hidecallerid=no
callwaiting=yes
callerid = Dulles Micro, LLC 703 450 5000
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
channel = 1
zaptel.conf:
fxsks=1,2,3,4
loadzone = us
defaultzone=us
Brings words and photos together (easily) with
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Yahoo! Mail
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Re: [Asterisk-Users] Re: Asterisk at large
Well, this may be a workaround, but you could run that retrieve_extensions_from_mysql.pl script from cron (once an hour, day, whatever) and reload the extensions right afterwards (also doable using cron). So any changes you make in your central MySQL database will be reflected the next time cron runs. If you use it to write to a second file which is included in the main extensions.conf file, you have the added benefit of being able to customize each installation, if need be (and if you don't want to, well you don't have to, just leave extensions.conf alone except for the include statement). Joseph Tanner On 3/2/06, Douglas Garstang [EMAIL PROTECTED] wrote: Yikes. Managability! It's a lot easier to manage multiple Asterisk systems configuration from a single MySQL database then it is to manage .conf files on several redundant Asterisk boxes. I can't believe you asked that question. I'll apologise in advance because I must be missing part of this thread. -Original Message- From: C F [mailto:[EMAIL PROTECTED] Sent: Thursday, March 02, 2006 10:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: Asterisk at large Can you explain why you would want asterisk only thru realtime? and not thru the /etc/asterisk/ ? The wiki is located at: http://www.voip-info.org/ the archives for this list is located at: http://lists.digium.com/ The asterisk irc channel is at: irc://irc.freenode.net/#asterisk Google is located at: http://www.google.com/ The asterisk docs project is located at: http://www.asteriskdocs.org/ On 3/2/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi Group, Please read my previous message below, I want to configure Asterisk with Mysql and make Asterisk dynamic so that Asterisk will read everything from Mysql and we can make changes to mysql data directly. Please tell how can we do this and point me to related documentation. Thanks for your help and time, Manoj. Quoting [EMAIL PROTECTED]: Hi Group, I was able to install Asterisk and its addons successfully. Now I want to eliminate sip.conf and extensions.conf and use everything from Mysql DB, Is this possible? I have seen this page http://www.voip-info.org/wiki/index.php?page=Asterisk%20extensions%20from%20mysql and learnt that we still get the data from Mysql DB and write it as sub file to actual sip or extensions.conf before starting Asterisk. Can we eliminate config files completely? If it is possible then please point me to the links explaing how can we do this? I also found very less information on using Asterisk with Mysql, if there are any articles discussing this please send me those links. Thanks for your help all the time, Manoj. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk at large
Well, I still think my idea would work, until you could find a better implementation. Just tell users that sign up or make changes to their config that changes take up to an hour (or whatever schedule you run your cron job on) to take effect. Or, you could customize the web page you use to let users make changes to automatically run a script that somehow signals your asterisk servers that they need to regenerate and reload the extensions.conf file immediately. I'd still have it run automatically every hour or so, just in case. Also, I haven't used this feature, so I'm not sure what would happen if the mysql server went down and the remote asterisk box(es) tried to access it. Would the script just error out, leaving the existing extensions.conf file intact? Or would it wipe it out, and then error out, leaving you with a blank extensions.conf file? I suppose if it wasn't well-behaved, you could edit it yourself to check for the database link first, before wiping out the extensions.conf file. Hope that helps some. Joseph Tanner On 3/2/06, Douglas Garstang [EMAIL PROTECTED] wrote: Ok then, a single Asterisk server. It's still going to be easier to manage from a database. This is especially true when you factor in self provisioning, ie providing users with a way to make changes to their own configuration. -Original Message- From: C F [mailto:[EMAIL PROTECTED] Sent: Thursday, March 02, 2006 12:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: Asterisk at large Douglas, a lot easier? If it's like you say with multiple servers. But the OP did not indicate this in his/her question, in fact s/he sounded clueless. Also, what is the purpose of NOT having *any* configs from /etc/asterisk/ On 3/2/06, Douglas Garstang [EMAIL PROTECTED] wrote: Yikes. Managability! It's a lot easier to manage multiple Asterisk systems configuration from a single MySQL database then it is to manage .conf files on several redundant Asterisk boxes. I can't believe you asked that question. I'll apologise in advance because I must be missing part of this thread. -Original Message- From: C F [mailto:[EMAIL PROTECTED] Sent: Thursday, March 02, 2006 10:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: Asterisk at large Can you explain why you would want asterisk only thru realtime? and not thru the /etc/asterisk/ ? The wiki is located at: http://www.voip-info.org/ the archives for this list is located at: http://lists.digium.com/ The asterisk irc channel is at: irc://irc.freenode.net/#asterisk Google is located at: http://www.google.com/ The asterisk docs project is located at: http://www.asteriskdocs.org/ On 3/2/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi Group, Please read my previous message below, I want to configure Asterisk with Mysql and make Asterisk dynamic so that Asterisk will read everything from Mysql and we can make changes to mysql data directly. Please tell how can we do this and point me to related documentation. Thanks for your help and time, Manoj. Quoting [EMAIL PROTECTED]: Hi Group, I was able to install Asterisk and its addons successfully. Now I want to eliminate sip.conf and extensions.conf and use everything from Mysql DB, Is this possible? I have seen this page http://www.voip-info.org/wiki/index.php?page=Asterisk%20extensions%20from%20mysql and learnt that we still get the data from Mysql DB and write it as sub file to actual sip or extensions.conf before starting Asterisk. Can we eliminate config files completely? If it is possible then please point me to the links explaing how can we do this? I also found very less information on using Asterisk with Mysql, if there are any articles discussing this please send me those links. Thanks for your help all the time, Manoj. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE
Re: [Asterisk-Users] Polling Asterisk for Life
He's probably having a similar problem I had, where asterisk stops responding to any commands at all (whether it's bluetooth show peers, sip show registry, or even stop now; all that works is exit). Well, I guess I can't say any commands at all, I haven't tried every single one. I upgraded to 1.2.4 to try to resolve this problem (was running 1.2.1). It seems to be running ok now, but has not been long enough to tell for sure. If it messes up again, I will try to remember to issue the command show license. If it doesn't respond to that, then that should be a good command to use to test if asterisk is responding. Just grep for part of the output it should give you, and if it's missing just killall -9 asterisk, and restart, and hopefully it works then. You could give your script some smarts if you wanted. If it passes the test, then the script is done. If it fails, then kill asterisk, restart, wait an appropriate amount of time to let it come back online fully (if it usually takes 30 seconds, give it 2 minutes just in case, if it takes 2 minutes give it 5, etc.). Re-issue the command, if asterisk still does not respond (possibly killall didn't work, or there's another issue keeping asterisk from working properly) then issue a reboot (or alternatively you could shut down more services such as zaptel, attempt to bring it all back online again, and if it still doesn't work then reboot; personally I'd rather just reboot and send some kind of alert to me). Joseph Tanner On 3/2/06, Cosmin Prund [EMAIL PROTECTED] wrote: AFAIK there are problems with repeatedly connecting and disconnecting the manager interface. Also you're probably using a proxy (all manager interfaces I've seen are using proxies), it might not be a good idea to pool something out of the manager that often. Did you consider running a cron job on the server, using asterisk -rx to run a command and then decide rather asterisk is down or not based on the result? This way you'd be doing on the server, working around the problems with the manager interface and saving some bandwidth :). You might also be able to call /sbin/reboot directly from the cron script! If on the other hand the whole server is going down you may simply use ping! -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Matt Sent: Thursday, March 02, 2006 7:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Polling Asterisk for Life Hi, Occassionally Asterisk will go down and I have to restart it.. not often.. but sometimes. When it does the manager interface stops working, as does the CLI. My thoughts was to poll the manager interface once every 5 minutes for a value. If I don't get the value back then alert me that the server is possibly down. Does anyone know what a good value to poll for might be? I was thinking I could poll my SIP account for the CallWaiting value, but would like something that was not linked to my account. Just something that returns a single line is fine. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Changing caller id on transfer
Hrm, well it depends exactly how you're transferring calls as to how
you'd write it in extensions.conf. Is it being transferred to an
internal line or to an external line? If external, then of course you
need to be able to set the outgoing callerid (you'd basically be
spoofing it, but that shouldn't be an issue).
I have done something similar, but not exactly like what you're
wanting. I'm not sure what the best way to do it would be. Perhaps
you could set the callerid early in asterisk in a variable (name it
something like, ${OUTGOINGCALLERID}). Before making an outgoing call,
check asterisk's built-in callerid variable, if it's empty then set it
to your special variable. If it's not empty, then use it (so a normal
outgoing call wouldn't already have callerid set, and would use your
value, but if an incoming call came in then the callerid variable
would be set, and we'd use that instead).
The way I did it would require that a user start off in a different
context based on whether they're receiving a call, or making an
outgoing call. Perhaps you can check for a flash, or make them dial a
special extension to make an outgoing, transferred call? I dunno, my
setup's unique and I'm not sure how you can adapt it to your needs.
Anyways, if you can get them in a different context, then it's simple.
In your normal outgoing context, the very first line should be what
sets the callerid. In the special incoming then outgoing context, do
something like this:
exten = _1NXXNXX,8,Goto(cell-out,${EXTEN},2)
In this case, _1NXXNXX is the extension matched when I dial a
normal long-distance number (such as 1-931-555-1212). It jumps to the
[cell-out] context (can name this anything you want, this is just my
setup with calling out via bluetooth), it keeps the extension the same
(so in [cell-out] we would need an extension of _1NXXNXX), and
goes to priority 2. This bypasses the first priority, which is where
you set callerid for regular outgoing calls, so now you'll use the
existing value for the outgoing callerid, instead of changing it.
You could just as easily recreate your dialplan for outgoing calls
that are transferred, but I prefer to jump to an existing context,
that way I only have to change one part of extensions.conf. I know
that if I can make a long-distance call from a local extension, then
it'll work when someone calls in and gets bridged, because the code is
exactly the same except for setting callerid.
Hope that helps more than it confuses.
Joseph Tanner
On 3/2/06, Cosmin Prund [EMAIL PROTECTED] wrote:
As usual, this is most likely a easy question, but here it goes any way:
How can I change the caller id on a transferred call so the called party
knows the call has been transferred from a colleague and it's not coming
directly from our outside lines?
The story goes like this:
1) Client calls. All phones ring.
2) Someone picks up the phone.
3) The phone gets transferred to someone.
4) The person that gets the transferred call sees the original caller id and
doesn't know the call has been transferred. I'd like the person that gets
the transfer to see the caller id with a digit prefix. Ex: Original
caller-id: 0269123456; Caller id if the call has been transferred:
1*0269123456
I know I can use SetCallerId(1*${CALLERIDNUM}) but how do I know I'm doing a
transfer and not calling someone?
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Re: [Asterisk-Users] Polling Asterisk for Life
The problem isn't that asterisk isn't running, it's that asterisk is not responding. When asterisk is in this funky state, I can still run asterisk -r from the command line and get access to the CLI. While in the CLI, the only command that asterisk will respond to is exit which drops me back to the shell. If I try to issue a stop now, asterisk just immediately returns to the CLI prompt. It does this for every single command, except for exit. So, simply respawning asterisk, or checking to see if it's running isn't good enough, because asterisk is indeed running. We need to access asterisk and issue a command, and see if asterisk responds appropriately. If not, we can assume it has died, and we can kill it off (killall -9 asterisk) and then start it back up again (or reboot the whole server if necessary). Yes, it's an odd problem, but I've noticed it so I can confirm it is a state asterisk can get into, and can confirm its symptoms. Hopefully all that is over with now after I upgraded (also fyi, I also moved my x101p around so it'd get its own irq, so it's possible that was the problem, though I doubt it). If it turns out I still have the problem, I'll probably whip up a script to check asterisk's condition and restart if needed. Joseph Tanner On 3/2/06, David Cook [EMAIL PROTECTED] wrote: Obviously if Asterisk keeps going down there is another problem to be found. However, why not start it from /etc/inittab with respawn??? Else, poll from cron or a script with ps ax | grep asterisk | grep -v grep | wc -l to find out if it is running. dbc. Date: Thu, 2 Mar 2006 22:01:01 +0200 From: Cosmin Prund [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Polling Asterisk for Life To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii AFAIK there are problems with repeatedly connecting and disconnecting the manager interface. Also you're probably using a proxy (all manager interfaces I've seen are using proxies), it might not be a good idea to pool something out of the manager that often. Did you consider running a cron job on the server, using asterisk -rx to run a command and then decide rather asterisk is down or not based on the result? This way you'd be doing on the server, working around the problems with the manager interface and saving some bandwidth :) . You might also be able to call /sbin/reboot directly from the cron script! If on the other hand the whole server is going down you may simply use ping! -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Matt Sent: Thursday, March 02, 2006 7:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Polling Asterisk for Life Hi, Occassionally Asterisk will go down and I have to restart it.. not often.. but sometimes. When it does the manager interface stops working, as does the CLI. My thoughts was to poll the manager interface once every 5 minutes for a value. If I don't get the value back then alert me that the server is possibly down. Does anyone know what a good value to poll for might be? I was thinking I could poll my SIP account for the CallWaiting value, but would like something that was not linked to my account. Just something that returns a single line is fine. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polling Asterisk for Life
Just to add to this train of thought, I have noticed some strange errors in /var/log/asterisk/messages that I think occur when asterisk gets in this strange state. Errors such as SetGroup and NoOp aren't valid commands. Initially I thought perhaps 1.2.4 got rid of the SetGroup command, but noticed these errors were before I upgraded (I know the command's deprecated). Also NoOp hasn't been deprecated so that was odd. I believe these errors happened around the same time asterisk stopped responding (but was still running, and apparently trying to handle some calls, but of course failing since it couldn't parse my dialplan). Hope that helps more than it confuses. Joseph Tanner On 3/2/06, Joseph Tanner [EMAIL PROTECTED] wrote: The problem isn't that asterisk isn't running, it's that asterisk is not responding. When asterisk is in this funky state, I can still run asterisk -r from the command line and get access to the CLI. While in the CLI, the only command that asterisk will respond to is exit which drops me back to the shell. If I try to issue a stop now, asterisk just immediately returns to the CLI prompt. It does this for every single command, except for exit. So, simply respawning asterisk, or checking to see if it's running isn't good enough, because asterisk is indeed running. We need to access asterisk and issue a command, and see if asterisk responds appropriately. If not, we can assume it has died, and we can kill it off (killall -9 asterisk) and then start it back up again (or reboot the whole server if necessary). Yes, it's an odd problem, but I've noticed it so I can confirm it is a state asterisk can get into, and can confirm its symptoms. Hopefully all that is over with now after I upgraded (also fyi, I also moved my x101p around so it'd get its own irq, so it's possible that was the problem, though I doubt it). If it turns out I still have the problem, I'll probably whip up a script to check asterisk's condition and restart if needed. Joseph Tanner On 3/2/06, David Cook [EMAIL PROTECTED] wrote: Obviously if Asterisk keeps going down there is another problem to be found. However, why not start it from /etc/inittab with respawn??? Else, poll from cron or a script with ps ax | grep asterisk | grep -v grep | wc -l to find out if it is running. dbc. Date: Thu, 2 Mar 2006 22:01:01 +0200 From: Cosmin Prund [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Polling Asterisk for Life To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii AFAIK there are problems with repeatedly connecting and disconnecting the manager interface. Also you're probably using a proxy (all manager interfaces I've seen are using proxies), it might not be a good idea to pool something out of the manager that often. Did you consider running a cron job on the server, using asterisk -rx to run a command and then decide rather asterisk is down or not based on the result? This way you'd be doing on the server, working around the problems with the manager interface and saving some bandwidth :) . You might also be able to call /sbin/reboot directly from the cron script! If on the other hand the whole server is going down you may simply use ping! -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Matt Sent: Thursday, March 02, 2006 7:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Polling Asterisk for Life Hi, Occassionally Asterisk will go down and I have to restart it.. not often.. but sometimes. When it does the manager interface stops working, as does the CLI. My thoughts was to poll the manager interface once every 5 minutes for a value. If I don't get the value back then alert me that the server is possibly down. Does anyone know what a good value to poll for might be? I was thinking I could poll my SIP account for the CallWaiting value, but would like something that was not linked to my account. Just something that returns a single line is fine. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polling Asterisk for Life
Are you guys perchance using Local/[EMAIL PROTECTED] in your installations? I am, not in my extensions.conf but in a .call file. I started using the .call files around the same time I originally installed 1.2.1, so I can't say which one caused the problems. I'll keep running 1.2.4 right now, and if it acts up again I'll remove the Local/[EMAIL PROTECTED] in my .call file. I really need to use it though, I don't know how to use the functionality of SetGroup and CheckGroup in a .call file, which is absolutely necessary in my situation (my cellphone will happily let me make a second outgoing call, which will screw up the first call and possibly connect the two outgoing calls if the second person hangs up, not good!). Joseph Tanner -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polling Asterisk for Life
No TDM400 cards on my end (don't know what the original poster has). Just a simple X101P card. It's a clone, but an exact clone (made sure the layout was 100% the same, I have other modems with the same chipset, but different layout, so tossed those aside). BTW, came home tonight, and it's messed up again. Unfortunately, it DID respond to show license, so that won't work. However, it did NOT respond to help, so perhaps that could be used to see if it's responding or not. I'll try not using the Local command in my .call file to see if there's any difference. May be a few days until I know though. Joseph Tanner On 3/2/06, Mike Clark [EMAIL PROTECTED] wrote: Joseph Tanner wrote: The problem isn't that asterisk isn't running, it's that asterisk is not responding. When asterisk is in this funky state, I can still run asterisk -r from the command line and get access to the CLI. While in the CLI, the only command that asterisk will respond to is exit which drops me back to the shell. If I try to issue a stop now, asterisk just immediately returns to the CLI prompt. It does this for every single command, except for exit. So, simply respawning asterisk, or checking to see if it's running isn't good enough, because asterisk is indeed running. We need to access asterisk and issue a command, and see if asterisk responds appropriately. If not, we can assume it has died, and we can kill it off (killall -9 asterisk) and then start it back up again (or reboot the whole server if necessary). Yes, it's an odd problem, but I've noticed it so I can confirm it is a state asterisk can get into, and can confirm its symptoms. Hopefully all that is over with now after I upgraded (also fyi, I also moved my x101p around so it'd get its own irq, so it's possible that was the problem, though I doubt it). If it turns out I still have the problem, I'll probably whip up a script to check asterisk's condition and restart if needed. Joseph Tanner Do you happen to have TDM400 cards in your system? I have learned to set up any machine with TDM400 cards to do a nightly auto-reboot. If we don't, they will eventually exhibit behaviour identical to what you describe above. Mike Clark ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polling Asterisk for Life
I just really can't live without the Local channel, so I did some research. It appears that maybe putting /n on the end fixes my problem. So instead of this: Local/[EMAIL PROTECTED] I should use this: Local/[EMAIL PROTECTED]/n Curiously enough, without the /n on the end, I get this interesting error message: Mar 3 00:48:39 WARNING[5548]: channel.c:3054 ast_do_masquerade: Channel type 'BLT' does not have a fixup routine (for BLT/Motorola)! Bad things may happen. And, just as it says, bad things do indeed happen. Sometimes I get no audio on the call, in fact as soon as the zap channel picks up (I called back into my server), it'd often stop ringing on my end. Also the call file sometimes does not get removed. Found this out the hard way when my aunt called me back wondering what I needed (when I got back home, I had to restart asterisk, some time afterwards the call file was triggered, called my cellphone, got voicemail which it interpreted as being answered, then promptly called her). Note to self, check for call files before restarting asterisk, especially past 11PM. With the /n on the end, I don't get that error message, and after more than a dozen calls I haven't had a single issue. Yes, I did revert back to no /n, and again had problems right off the bat. Put it back, and again things work. I understand what /n does, I'm just not sure why it'd make my setup less buggy. Well, I'll continue to test, but for now this seems to help if you're using the Local channel. Joseph Tanner On 3/2/06, Joseph Tanner [EMAIL PROTECTED] wrote: No TDM400 cards on my end (don't know what the original poster has). Just a simple X101P card. It's a clone, but an exact clone (made sure the layout was 100% the same, I have other modems with the same chipset, but different layout, so tossed those aside). BTW, came home tonight, and it's messed up again. Unfortunately, it DID respond to show license, so that won't work. However, it did NOT respond to help, so perhaps that could be used to see if it's responding or not. I'll try not using the Local command in my .call file to see if there's any difference. May be a few days until I know though. Joseph Tanner On 3/2/06, Mike Clark [EMAIL PROTECTED] wrote: Joseph Tanner wrote: The problem isn't that asterisk isn't running, it's that asterisk is not responding. When asterisk is in this funky state, I can still run asterisk -r from the command line and get access to the CLI. While in the CLI, the only command that asterisk will respond to is exit which drops me back to the shell. If I try to issue a stop now, asterisk just immediately returns to the CLI prompt. It does this for every single command, except for exit. So, simply respawning asterisk, or checking to see if it's running isn't good enough, because asterisk is indeed running. We need to access asterisk and issue a command, and see if asterisk responds appropriately. If not, we can assume it has died, and we can kill it off (killall -9 asterisk) and then start it back up again (or reboot the whole server if necessary). Yes, it's an odd problem, but I've noticed it so I can confirm it is a state asterisk can get into, and can confirm its symptoms. Hopefully all that is over with now after I upgraded (also fyi, I also moved my x101p around so it'd get its own irq, so it's possible that was the problem, though I doubt it). If it turns out I still have the problem, I'll probably whip up a script to check asterisk's condition and restart if needed. Joseph Tanner Do you happen to have TDM400 cards in your system? I have learned to set up any machine with TDM400 cards to do a nightly auto-reboot. If we don't, they will eventually exhibit behaviour identical to what you describe above. Mike Clark ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: SV: [Asterisk-Users] Problems with voicemail
This probably has nothing to do with your problem, but I had a problem with similar symptoms, except asterisk was actually crashing whenever I tried to access voicemail. It would sometimes say some digits, but never got far (never got as far as the actual message). Problem turned out, crazily enough, to be having zaptel compiled with CONFIG_ZAPTEL_MMX. Commented that out, recompiled, worked fine. Uncommented again, recompiled, and it would crash every time I accessed voicemail. I'm running CentOS 4, with a 2.6 kernel, and did use the make linux26 command. Oh, and I did read the warning about compiling mmx with an AMD processor, but this server has an Intel Celeron in it, so it should have been ok. Oh well. Joseph Tanner On 2/24/06, Roger Lewau [EMAIL PROTECTED] wrote: I checked the permitions and updated the ones with the wrong permissions. No it is reading the number of messages correct, but as soon as I press 1 to listen it stops again. So again, I checked the permissions on the messagefolder but it seemed ok. I see now that another person on this lista has the exact same problem. Kind regards Roger -Original Message- From: Dinesh Nair [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Thu, 23 Feb 2006 20:00:30 +0800 Subject: Re: SV: [Asterisk-Users] Problems with voicemail On 02/22/06 23:11 Roger Lewau said the following: Connected to Asterisk 1.2.4 currently running on ns2 (pid = 47562) Verbosity is at least 9 -- Remote UNIX connection -- Executing VoiceMailMain(SIP/asterisk-0946, @sip) in new stack -- Playing 'vm-login' (language 'se') -- Playing 'vm-password' (language 'se') -- Playing 'vm-youhave' (language 'se') == Spawn extension (sip, 990, 1) exited non-zero on 'SIP/asterisk-0946' it's borking when attempting to read numbers. is sounds/digits populated with adequate perms ? -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo=== ===+ | for a in past present future; do | | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +== ===+ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on SPA-3000 rings extensions only afterthird ring
There really is no way to completely eliminate the lag, even if you disable callerid. Another workaround would be to connect a loud phone directly to your pstn line. When you hear it ring, jump up and grab your regular phone. It'll start ringing by the third ring. You'll have callerid pop up between the third and fourth ring (the 3-4 ring that the caller hears, it'll be between the 1-2 ring on your regular line). I discovered this workaround when I left a fax machine (which had a very loud ring) directly connected to my pstn line. I was surprised at how many calls came in and never got past the voice greeting (all were unfamiliar numbers, probably wrong numbers or what-not). I'm afraid for your situation, there's no way to do what you want without some kind of workaround, especially since you need callerid information. I suppose you could disable callerid detection in asterisk, and get callerid delivered to you another way (on your TV if you have a satellite receiver that supports callerid, a device that reads off the callerid to you, one of those nifty globes I've seen at radio shack, etc.). You'll still have a little bit of delay until asterisk rings your extensions, but it'll be more like 1-2 rings instead of 3. I honestly think a voice recording being played just before it rings your extensions isn't as bad an idea as you think. I use one for my residential line in addition to my business line. Haven't heard a single complaint yet. In fact I've gotten a few nice comments from it (I can customize the recording used based on callerid, leaving nice cute messages for family/friends, and the default recording for everyone else). Hope you find a solution that suits your needs. Joseph Tanner On 2/22/06, Zach A [EMAIL PROTECTED] wrote: If not in spa3k, then how about digium hardware, will that be faster in picking up caller IDs or is it possible to make it work faster. I need only one FXS/FXO. Is X101P single FXS/FXO? Zach A. -Original Message- From: Rich Adamson [mailto:[EMAIL PROTECTED] Sent: Wednesday, February 22, 2006 9:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk on SPA-3000 rings extensions only afterthird ring Thanks for your replies and sharing your experiences. Is there any way in SPA3000 to send the rings to sip phones on asterisk while still waiting for the caller ID? This will affect the dial plan sequence but maybe user will have the option to pickup right away or wait until the caller ID displays. Or maybe there is a way for SPA3000 to find the caller ID a littler faster, as all the other phones do which are directly connected to the Bell line. No, there is no way to do that in the spa3k. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on SPA-3000 rings extensions only after third ring
Same problem here with X101P. In my case (and I'm guessing your situation is similar, but not sure since the hardware is different) asterisk needs to see the first ring come through before doing anything. Sometime between the first and second ring it gets callerid information, then sometime after the second ring it can start processing the call (i.e., execute what you have set in extensions.conf). My workaround, is to have asterisk actually answer the call, and a voice (which my lovely wife recorded) tells the caller to please hold while their call is connected. Once the caller hears ringing tones again, your internal lines are ringing at the same time. No more having people hang up after 4-5 rings, when all you've heard is 1-2. This setup has the nice side effect of letting me force unknown callers to press 1 before being connected. Anyone I know (and have entered their phone number in extensions.conf to recognize) won't have to dial 1. All others will, and this has so far eliminated all telemarketing calls and even all wrong numbers (they know right away they got the wrong number, and hang up without pressing 1). It also lets me gain access to various functions no matter where I'm calling from. I can enter a password while the recording is playing, and get dialtone. From there I can call out (like a calling card), or check voicemail, etc. It's just like I'm dialing from an internal extension, which can come in handy (say I need to reach my wife in the middle of the day, when she's usually asleep and GotoIfTime directs the calls to voicemail, and I'm calling from an unknown number; I just enter my password, get dialtone, dial 6 which I have setup to ring all internal extensions regardless of time, and voila! I have a grumpy wife). If asterisk didn't automatically answer the call, none of this would be possible (well, I suppose I could press 0 during voicemail, and have the o extension setup, but this way works better). Maybe not the solution you were looking for, but personally I think this workaround opens up a lot of possibilities you may not have thought of previously. Joseph Tanner On 2/21/06, Zach A [EMAIL PROTECTED] wrote: Hi, My telephone extensions on asterisk which itself is connected to the Bell line using SPA-3000, ring only after third ring from the caller. Why is this happening and what is the solution? Thanks Zach A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] Good VoIP providers that support Asterisk PBX's
I have voicepulse connect too. I had occassional problems with incoming calls, but not many and not recently. Have had more problems with outgoing calls which is fine for me, as I have more than one backup (I use voxee as my primary due to lowest price, then voicepulse, and failing that I can use my cellphone or my landline). I am a bit disappointed with the price, it was decent before they upped it to $11. Seems a bit high to me, for just an incoming line with no outgoing minutes. Many other places charge about that and give you a bunch of minutes, or an unlimited local calling plan (in-state, in-area code, etc.). But, it's been very reliable, no complaints about uptime. Joseph Tanner On 2/19/06, David Blomquist [EMAIL PROTECTED] wrote: I've been using voicepulce connect for several months with very few problems. Occasionally I get all circuits are busy messages when trying to dial out but no too often. d From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael J. Liberatore Sent: Sunday, February 19, 2006 4:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Good VoIP providers that support Asterisk PBX's I had voicepulse connect but had to transfer IAX2 had non stop drop outs in audio all the time. Tried everything to fix it, even with 14ms ping times it just didnt want to work right. I never figured out why, just canceled. Although i didnt like the no-name on incoming caller id either though, From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of andrew matthews Sent: Tuesday, February 14, 2006 8:52 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Good VoIP providers that support Asterisk PBX's http://connect.voicepulse.net They support astrisk, with iax2 :) On 2/14/06, Jim Robinson [EMAIL PROTECTED] wrote: Hi Folks, Can anyone give me some good recommendations for VoIP providrs that support Asterisk PBX's? We're based in Georgia and I having a hard time finding anyone Regards, Jim PS - If you could CC me in on the reply I would greatly appreciate it! jim(-A T-)linux-sp.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Software E.C. Along with Tellabs
Shouldn't hurt, I'd give it a try. But first you may want to fiddle with the Tellabs configuration some more. This has some good information: http://www.voip-info.org/wiki/view/Tellabs+Hardware+Echo+Cancellers Joseph Tanner On 2/15/06, Doug Lytle [EMAIL PROTECTED] wrote: Since putting my Tellabs EC into place around 2 weeks ago, the echo problem has almost been eliminated. Reports of some very faint echo, but everybody is happy. My question is, if I were to also turn on the Asterisk Software EC, would this remove any residual echo that may make it past the Tellabs Hardware EC. Thanks, Doug ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ChanIsAvail
Perhaps I'm missing something here, but why not just have asterisk
dial all the phones regardless? No need to check what's available or
not, just dial all of them. If you don't want users on the phone to
hear a call-waiting beep, just make sure call-waiting is disabled.
Any phones that are able to ring will do so, the ones that are busy
obviously will not.
If I am missing something, let me know, but this seems to be the
easiest solution and will do what you said you need. Dial all phones,
and all that are available will ring, the rest will just return a busy
message which asterisk should ignore, as long as one phone somewhere
is not busy. I haven't run into this, but I would assume if all
phones were busy that asterisk would then go to priority +101, so you
could send them straight to voicemail.
Joseph Tanner
On 2/14/06, Jayson Navitsky [EMAIL PROTECTED] wrote:
Hi,
So I've done my research on Chanisavail, read the wiki, checked the
archive but can't seem to find anything to suit my scenario. I've
played around with it a lot, but I'm still scratching my head on what
I need to do.
What I need is to be able to accept a call by SIP and ring all
telephones that are not in use (which just so happen to be on Zap
interfaces, but might be SIP in the future).
What I have now is this (I know it's really bad):
exten = 1646555,1,Answer()
exten = 1646555,2,Dial(Local/[EMAIL PROTECTED]Local/[EMAIL
PROTECTED]Local/[EMAIL PROTECTED]Local/[EMAIL PROTECTED],30)
exten = 1646555,3,ChanisAvail(Local/[EMAIL PROTECTED]Local/[EMAIL
PROTECTED])
exten = 1646555,4,Cut(DESK3=AVAILCHAN||1)
exten = 1646555,5,ChanisAvail(Local/[EMAIL PROTECTED]Local/[EMAIL
PROTECTED])
exten = 1646555,6,Cut(DESK4=AVAILCHAN||1)
exten = 1646555,7,Dial(${DESK3}${DESK4},30,tr)
exten = 1646555,8,Busy
(Each local is 1 zap interface)
Which is sort of my temporary work around to the problem for now,
first if there are no phones in use all phones will ring, if not it
will return busy and then it is checked to see if there is anything
available to ring between those 2 groups there. If only one phone
is in use only 2 channels will ring right now (obviously).
What I need is for any available channel to ring.
Any thoughts?
Thanks,
Jay
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Re: [Asterisk-Users] ChanIsAvail
Eventually I will learn to read the message twice before responding.
I see you are dialing all the phones first. So if just one is busy
asterisk won't dial any? That's odd. Now, I don't have any phones on
a zaptel card (just an X101P for incoming), but when dialing multiple
sip phones, it'll ring the ones that are available no problem. I can
call from one to the other, obviously the one I'm calling from can't
be rung and returns a busy message, but asterisk happily dials the
rest. In fact, I have asterisk dial several extensions that aren't
even online (test extensions that are sometimes online, sometimes not)
and I've never had a problem.
Sorry I couldn't be of more help :(
Joseph Tanner
On 2/14/06, Joseph Tanner [EMAIL PROTECTED] wrote:
Perhaps I'm missing something here, but why not just have asterisk
dial all the phones regardless? No need to check what's available or
not, just dial all of them. If you don't want users on the phone to
hear a call-waiting beep, just make sure call-waiting is disabled.
Any phones that are able to ring will do so, the ones that are busy
obviously will not.
If I am missing something, let me know, but this seems to be the
easiest solution and will do what you said you need. Dial all phones,
and all that are available will ring, the rest will just return a busy
message which asterisk should ignore, as long as one phone somewhere
is not busy. I haven't run into this, but I would assume if all
phones were busy that asterisk would then go to priority +101, so you
could send them straight to voicemail.
Joseph Tanner
On 2/14/06, Jayson Navitsky [EMAIL PROTECTED] wrote:
Hi,
So I've done my research on Chanisavail, read the wiki, checked the
archive but can't seem to find anything to suit my scenario. I've
played around with it a lot, but I'm still scratching my head on what
I need to do.
What I need is to be able to accept a call by SIP and ring all
telephones that are not in use (which just so happen to be on Zap
interfaces, but might be SIP in the future).
What I have now is this (I know it's really bad):
exten = 1646555,1,Answer()
exten = 1646555,2,Dial(Local/[EMAIL PROTECTED]Local/[EMAIL
PROTECTED]Local/[EMAIL PROTECTED]Local/[EMAIL PROTECTED],30)
exten = 1646555,3,ChanisAvail(Local/[EMAIL PROTECTED]Local/[EMAIL
PROTECTED])
exten = 1646555,4,Cut(DESK3=AVAILCHAN||1)
exten = 1646555,5,ChanisAvail(Local/[EMAIL PROTECTED]Local/[EMAIL
PROTECTED])
exten = 1646555,6,Cut(DESK4=AVAILCHAN||1)
exten = 1646555,7,Dial(${DESK3}${DESK4},30,tr)
exten = 1646555,8,Busy
(Each local is 1 zap interface)
Which is sort of my temporary work around to the problem for now,
first if there are no phones in use all phones will ring, if not it
will return busy and then it is checked to see if there is anything
available to ring between those 2 groups there. If only one phone
is in use only 2 channels will ring right now (obviously).
What I need is for any available channel to ring.
Any thoughts?
Thanks,
Jay
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Re: [Asterisk-Users] Re: Why is asterisk ignoring my context?
If you send it to a different context, you still have to have the appropriate extension, i.e.: [test] Exten = 555-555-,1,NoOp(test) I've also noticed that with providers that I don't register with, who just blindly send the call to the same address (i.e., IPKall), context seems to be ignored. If the default context is [default], and you want it to be sent to the [test] context, just use a goto line, i.e.: [default] Exten = 555-555-,1,Goto(test,s,1) And then it'll be sent to: [test] Exten = s,1,NoOp(test) You could send it to any context/extension. I use this trick to send calls from multiple providers coming in different ways (iax, sip, zap) to the same context/extension, so I only have one context to edit instead of many. Hope that helps some. Joseph Tanner On 2/13/06, Bromont Quebec [EMAIL PROTECTED] wrote: Do you also have a SIP phone you are dialing from? This is what I would have setup: sip.conf: [sipphone] Bla Bla Bla context=local-phones [someprovider] Bla bla bla context=someprovider-in extensions.conf [local-phones] exten = 55,1,Noop(test) [someprovider-in] exten = s,1,Dial(SIP/sipphone) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: RE : [Asterisk-Users] lists problem, Gmail????????
May be some truth to it though :( Personally I use gmail, but use a different email address that is forwarded to my gmail account. With this setup, I haven't had any issues. I use gmail because it's easily accessible from any PC, and I like how it groups conversations (probably why you see a lot of gmail addresses signed up on mailing lists). Joseph Tanner On 2/13/06, Olivier.taylor [EMAIL PROTECTED] wrote: Pfff, What for an answer :( I use gmail and have no problems. Olivier -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Martin Joseph Envoyé : lundi 13 février 2006 20:36 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [Asterisk-Users] lists problem, Gmail On Feb 13, 2006, at 2:45 AM, Simone Cittadini wrote: C F ha scritto: Am I the only one having trouble with this list? Since the begining of the week I have not been receiving mail from the list like I used to, is this a gmail problem? or is it subscription problem? or is something wrong with the list? anybody else using gmail having any problems? Yes, I'm also getting some lag sometimes, one or two days without receiving mails get a real mail server and it works great! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk with USB
I believe it's just being recognized as a modem. Feel free to try it out, but I haven't seen anything describing how to accomplish what you want with just a data cable (and I have searched). Please prove me wrong, I'd love to ditch the bluetooth dongle (already have too many 2.4GHz devices as it is, I think they're starting to cause interference). This is just going on gut instinct here, but if you're really persistent, maybe you can use the data cable to send the dial commands, and have some kind of adapter cable going from the 2.5 plug you have, to a 3.5, put that into the line-in of a sound card, and then configure asterisk to send the dial commands (to dial numbers, hangup, anything that needs a key pressed on the phone) through usb (should be able to access the tty device and issue commands there), and use the soundcard for audio. If I'm not mistaken, that's basically what the dock-n-talk and cellsocket devices do. You may run into a few problems, but I think it'd work. Joseph Tanner On 2/8/06, Facundo Ameal [EMAIL PROTECTED] wrote: But my cell phone is recognised as a ttyACM device... Is it the same? 2006/2/7, Joseph Tanner [EMAIL PROTECTED]: Far as I know, you cannot use a usb cable to connect a cellphone directly to asterisk. You need something called a cellsocket or a dock-n-talk. You use these to connect directly to a regular telephone, so to connect to asterisk you'll need an FXO port. I'd love to find something that would directly connect a cellphone to asterisk that didn't cost a fortune. A usb cable to the cellphone would be perfect, just a plain gsm-sip gateway would be nice too but are $. Joseph Tanner On 2/7/06, Joe Tahan [EMAIL PROTECTED] wrote: I've read something on connecting a cellphone to asterisk with bluetooth, I'm not really sure about connecting to a usb phone. I think Joseph Tanner can help us out, as he did it with bluetooth. Truely/ Joe From: Facundo Ameal [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk with USB Date: Tue, 7 Feb 2006 11:55:07 -0300 Hello everybody! I've seen that you can connect your cellphone via bluetooth, but I've a Motorola V300 and it doesn't have that feature, so I wish to connect it via USB cable, is it pissible con use my cellphone with asterisk like that? I 've not been able to find information on how to do this, I'l appreciate any help. Thanks in advance! -- Facundo Ameal. famealatgmaildotcom Linux User #395088 FWD: 741664 MSN: asadoatlamorcilladotcomdotar ICQ: 74005793 Open your mind, use open source. ___ --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 Don't just Search. Find! Try MSN Search: Fast. Clear. Easy. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Facundo Ameal. famealatgmaildotcom Linux User #395088 FWD: 741664 MSN: asadoatlamorcilladotcomdotar ICQ: 74005793 Open your mind, use open source. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk with USB
You can do it with a bluetooth connection because someone wrote a driver, chan_bluetooth, to interface with the cellphone (makes asterisk look like a normal headset to a bluetooth-enabled phone). Maybe looking at the source will give you some ideas on how to proceed, but chan_bluetooth will only work with a bluetooth dongle (and only certain ones at that) and a bluetooth phone. You may need to write your own custom driver, though you may be able to get away with using agi. Joseph Tanner On 2/8/06, Facundo Ameal [EMAIL PROTECTED] wrote: If you can do it with a bluetooth conectionm, why not with the USB? Do you know which is the differece? Does it detect the cellphone as another device? I don't have a phone with bluetooth capability so I cannot test it to see how te OS recognizes it. 2006/2/8, Joseph Tanner [EMAIL PROTECTED]: I believe it's just being recognized as a modem. Feel free to try it out, but I haven't seen anything describing how to accomplish what you want with just a data cable (and I have searched). Please prove me wrong, I'd love to ditch the bluetooth dongle (already have too many 2.4GHz devices as it is, I think they're starting to cause interference). This is just going on gut instinct here, but if you're really persistent, maybe you can use the data cable to send the dial commands, and have some kind of adapter cable going from the 2.5 plug you have, to a 3.5, put that into the line-in of a sound card, and then configure asterisk to send the dial commands (to dial numbers, hangup, anything that needs a key pressed on the phone) through usb (should be able to access the tty device and issue commands there), and use the soundcard for audio. If I'm not mistaken, that's basically what the dock-n-talk and cellsocket devices do. You may run into a few problems, but I think it'd work. Joseph Tanner On 2/8/06, Facundo Ameal [EMAIL PROTECTED] wrote: But my cell phone is recognised as a ttyACM device... Is it the same? 2006/2/7, Joseph Tanner [EMAIL PROTECTED]: Far as I know, you cannot use a usb cable to connect a cellphone directly to asterisk. You need something called a cellsocket or a dock-n-talk. You use these to connect directly to a regular telephone, so to connect to asterisk you'll need an FXO port. I'd love to find something that would directly connect a cellphone to asterisk that didn't cost a fortune. A usb cable to the cellphone would be perfect, just a plain gsm-sip gateway would be nice too but are $. Joseph Tanner On 2/7/06, Joe Tahan [EMAIL PROTECTED] wrote: I've read something on connecting a cellphone to asterisk with bluetooth, I'm not really sure about connecting to a usb phone. I think Joseph Tanner can help us out, as he did it with bluetooth. Truely/ Joe From: Facundo Ameal [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk with USB Date: Tue, 7 Feb 2006 11:55:07 -0300 Hello everybody! I've seen that you can connect your cellphone via bluetooth, but I've a Motorola V300 and it doesn't have that feature, so I wish to connect it via USB cable, is it pissible con use my cellphone with asterisk like that? I 've not been able to find information on how to do this, I'l appreciate any help. Thanks in advance! -- Facundo Ameal. famealatgmaildotcom Linux User #395088 FWD: 741664 MSN: asadoatlamorcilladotcomdotar ICQ: 74005793 Open your mind, use open source. ___ --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 Don't just Search. Find! Try MSN Search: Fast. Clear. Easy. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Facundo Ameal. famealatgmaildotcom Linux User #395088 FWD: 741664 MSN: asadoatlamorcilladotcomdotar ICQ: 74005793 Open your mind, use open source
Re: [Asterisk-Users] Asterisk with USB
Actually, there is such a thing as a usb headset for a cellphone. My Razr V3 only has a mini-usb connection, used for data, charging, and apparently voice if you want a wired headset. It may be possible to rig up a driver for THIS phone, but I doubt anyone will bother, since this phone has bluetooth also. Then again, after more testing I think that there's some interference caused if a second phone is powered on and you're using the bluetooth connection to interface to asterisk. Made some calls yesterday while on the road, but were full of static (just like when I place test calls from home). Got home, and noticed a second cellphone had been left on. Seems that the cellphone works fine with the bluetooth dongle, as long as there is not a second phone turned on (the second phone is also a razr, but bluetooth was off). Two cellphones calling each other directly have no static. A direct usb connection to asterisk might be beneficial in THIS case, but as there aren't many phones that have a usb connection for a headset, I doubt it'd be practical (and still may not be possible, I'm not sure how the usb headset works, maybe it sends the signal digitally using a usb standard, or just uses some of the pins in the usb jack in a non-standard way). Joseph Tanner On 2/8/06, Cosmin Prund [EMAIL PROTECTED] wrote: Bluetooth enabled phones to talk to Bluetooth headsets; I guess there's a protocol for the phone to talk to any Bluetooth headset, no matter who made it. This protocol would have to include something to allow voice to pass from the phone to the headset and vice versa. It might also include something for dialing out. I suppose chan_bluetooth is emulating a headset, so there's support in the phone for passing voice around. There's also documentation for how to do it. I've never seen a USB headset so I doubt there's any support in the phone for passing voice over that connection. Maybe this is why there's no such thing as chan_usb Then again, I'm no expert on this matter, so maybe I'm plain wrong. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Facundo Ameal Sent: Wednesday, February 08, 2006 3:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk with USB If you can do it with a bluetooth conectionm, why not with the USB? Do you know which is the differece? Does it detect the cellphone as another device? I don't have a phone with bluetooth capability so I cannot test it to see how te OS recognizes it. 2006/2/8, Joseph Tanner [EMAIL PROTECTED]: I believe it's just being recognized as a modem. Feel free to try it out, but I haven't seen anything describing how to accomplish what you want with just a data cable (and I have searched). Please prove me wrong, I'd love to ditch the bluetooth dongle (already have too many 2.4GHz devices as it is, I think they're starting to cause interference). This is just going on gut instinct here, but if you're really persistent, maybe you can use the data cable to send the dial commands, and have some kind of adapter cable going from the 2.5 plug you have, to a 3.5, put that into the line-in of a sound card, and then configure asterisk to send the dial commands (to dial numbers, hangup, anything that needs a key pressed on the phone) through usb (should be able to access the tty device and issue commands there), and use the soundcard for audio. If I'm not mistaken, that's basically what the dock-n-talk and cellsocket devices do. You may run into a few problems, but I think it'd work. Joseph Tanner On 2/8/06, Facundo Ameal [EMAIL PROTECTED] wrote: But my cell phone is recognised as a ttyACM device... Is it the same? 2006/2/7, Joseph Tanner [EMAIL PROTECTED]: Far as I know, you cannot use a usb cable to connect a cellphone directly to asterisk. You need something called a cellsocket or a dock-n-talk. You use these to connect directly to a regular telephone, so to connect to asterisk you'll need an FXO port. I'd love to find something that would directly connect a cellphone to asterisk that didn't cost a fortune. A usb cable to the cellphone would be perfect, just a plain gsm-sip gateway would be nice too but are $. Joseph Tanner On 2/7/06, Joe Tahan [EMAIL PROTECTED] wrote: I've read something on connecting a cellphone to asterisk with bluetooth, I'm not really sure about connecting to a usb phone. I think Joseph Tanner can help us out, as he did it with bluetooth. Truely/ Joe From: Facundo Ameal [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: Asterisk Users Mailing
Re: [Asterisk-Users] Fedora Core 3 or Fedora Core 4? yum update ornot?
On 2/8/06, Paul [EMAIL PROTECTED] wrote: Maybe some people think that a PBX should come with a few games just like so many cell phones these days :) Unfortunately, mine has to sit on the front line, it can't hide behind a firewall. I only have one IP, and it's either assign it to asterisk (and thus force it to serve as a nat server, occassional ftp server, etc.) or have to deal with having asterisk behind nat. Configuring sip without nat is soo easy. Yes, I took the easy way out! Of course, in my situation I make sure to keep it fairly up to date. Joseph Tanner Technical Support wrote: I think that some people try to make their asterisk box a do-everything super server. Can you image a traditional PBX with direct access via the internet, serving web pages via apache, running sendmail, etc. Our approach has been keep it simple. We lock each Asterisk PBX down has hard as possible. This includes no direct internet connection (it should sit behind a real firewall), minimal services running, etc. With this philosophy, one can treat the PBX as an appliance: don't touch it if it's working. If you must run host web pages, run mail servers, offer SQLnet connections, make visible to the internet, etc. then other users are correct - you better continually patch/update ASAP. MD -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex Barnes Sent: Wednesday, February 08, 2006 4:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Fedora Core 3 or Fedora Core 4? yum update ornot? -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: 08 February 2006 08:41 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Fedora Core 3 or Fedora Core 4? yum update ornot? However, if you expose the box to the internet, you might want to upgrade those components that are known to have vulnerabilities. If you don't, count on the box being compromised sooner or later. This is sound advice worth taking. If you get a system stable in production, LEAVE IT ALONE!! We have just switched from SUSE to Fedora4 for our new installs and are very happy with it. Personally I much prefer it and bonus is it's free. Something that might be of interest is before I deployed the box live I did a full yum update I guess it must have updated the kernel or something as after I rebooted the box zap stopped working with some weird errors. Quick recompile of zaptel had everything working a charm but its something worth keeping in mind. I think the once it's working, leave it alone advice is very sound indeed :) HTH Alex Information contained in this e-mail and any attachments are intended for the use of the addressee only, and may contain confidential information of Ubiquity Software Corporation. All unauthorized use, disclosure or distribution is strictly prohibited. If you are not the addressee, please notify the sender immediately and destroy all copies of this email. Unless otherwise expressly agreed in writing signed by an officer of Ubiquity Software Corporation, nothing in this communication shall be deemed to be legally binding. Thank you. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Chan_BT question WAS: Asterisk with USB
Yes, you can dial out just fine. Joseph Tanner On 2/8/06, Aldo Bergamini [EMAIL PROTECTED] wrote: [EMAIL PROTECTED] is believed to have said: Bluetooth enabled phones to talk to Bluetooth headsets; I guess there's a protocol for the phone to talk to any Bluetooth headset, no matter who made it. This protocol would have to include something to allow voice to pass from the phone to the headset and vice versa. It might also include something for dialing out. I suppose chan_bluetooth is emulating a headset, so there's support in the phone for passing voice around. There's also documentation for how to do it. I've never seen a USB headset so I doubt there's any support in the phone for passing voice over that connection. Maybe this is why there's no such thing as chan_usb Then again, I'm no expert on this matter, so maybe I'm plain wrong. Hello, let me ask something related, but on chan_bluetooth. If this driver is faking Asterisk as a headset to the BT cellphone, does this mean that it can only answer incoming calls, but NOT ask the cellphone to dial a number? I am asking because I would like to see if I can use this driver to bypass any hardware gsm-gateway to obtain a 'gsm_trunk'. Would something like Dial(bluetooth/33512345594, 20) work? Of course using the correct channel indication for the chan_bluetooth module... TIA Aldo ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Welltech USA? and Wellgate Products?
Dug through my old doorstops, found what appears to be an unbranded Welltech 3502. When I first got it, I had a heck of a time trying to get callerid to work. Tried every setting there was, eventually found out that while it could be configured, it would not work for the sip version! Had to load the H323 version, then callerid would work. Didn't work quite like the sip version, so I tried to find newer firmware anywhere I could. Completely killed one of them (in desperation, I tried the 3502a firmware). Today I flashed the newest firmware on the one that worked (had to go through telnet, as the web-based config went through the steps, but never actually did the upgrade). I now have callerid, but only if there's no name. If it's number-only, then it shows up perfectly. Name, and nadda (if the name ends in a number, then that part'll show up). Also, there's no dialplan settings to alter. You just punch in the number, and when you're done it'll pass it to asterisk. You can set it to have an end key, you can choose *, #, or none, but even with it set to # it'll still time out after several seconds and send the digits already entered (might be a problem if you're a slow dialer). So, if you're looking at the 3502, it seems to work well enough, but they still have some callerid issues. Can't comment on other models. Joseph Tanner On 2/8/06, Martin Joseph [EMAIL PROTECTED] wrote: On Feb 8, 2006, at 9:22 AM, Ariel Batista wrote: I normally don't like talking bad about products. But I would like to say that the Welltech/Wellgate are not products that are support to work with asterisk. I have invested many hours of work in getting there device to work with Asterisk. They do not. And also as of Last Nov. They told me that they did not plan on supporting Asterisk. Good luck if you are able to get them to work since they go and sell there product with other names please post the settings you get for them to work. I have 2 of them as paper holders. And since there really bad I will not even sell them on ebay. FYI, They released newer firmware as of 12/2005 that is supposed to make most of there devices Asterisk compatible. If you try it, please let us know... If you have the 3701a unit (1FXS 1FXO) or really any FXO unit that you want to get rid of, please contact me off list, and I will take my chances and experiment with it a bit. Thanks for the feedback, Marty ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk and week-ends
Yes. Google GotoIfTime. I use this to not ring our phones during the day (we're night people), you can just as easily set it up to play a message during times that you're closed and send directly to voicemail (you can specify certain times of the day on certain days, or whole days such as saturday and sunday, and a lot more). Joseph Tanner On 2/7/06, demigor [EMAIL PROTECTED] wrote: Hello, I would like to know if it's possible to configure asterisk to play something nice to a person calling me during week-ends when there is noone available at the phone and switch back to normal calls receiving on Monday morning. Please help. Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_bluetooth - concurrent calls?
I thought more about this, perhaps you were instead asking if you could have multiple phones setup, and have multiple calls processed? It should work, I haven't attempted it, so I'm not sure if chan_bluetooth will handle multiple phones at the same time, but otherwise it should be fine. Just make sure to setup each individual phone with its own call group, set the limit to one, so when the first is busy asterisk will try the second, and so on down the line. Joseph Tanner On 2/7/06, Peter Molnar [EMAIL PROTECTED] wrote: Hi, i was reading about connecting a cellular phone over chan_bluetooth. I was wondering, if one is able then to make/receive concurrent calls or if you can make just one at time? Peter ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_bluetooth - concurrent calls?
You can use the three-way calling feature on the cellphone, so one user could talk to two different people at once. If you have more than one cellphone, this might be tricky (you want only one actual call going out per cellphone, but go ahead and let a second call be placed through one sometimes for three-way calling, and ensure that the three-way call goes out the same cellphone, and not to another now-free cellphone that's earlier in the dial priority). If you plan on just having one cellphone connected, I think it wouldn't be too much trouble. Just have a regular extension that will only allow one call in the callgroup, then you can use a special extension that will let you dial a second time with the callgroup set to 2. Just remember you need to connect the two calls to have a three-way conversation, perhaps a blank atd command? I don't know, haven't tried it. It should be possible though. Joseph Tanner On 2/7/06, Peter Molnar [EMAIL PROTECTED] wrote: And (as GSM Restriction) one can do only one call per phone (conferences and onHold are managed by the GSM-AP). This was what i was actualy interested in. My idea was, when conferecnces work, it should be possible to make 2 calls over 1 GSM phone at a time. But apparently this wont work. Peter ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk with USB
Far as I know, you cannot use a usb cable to connect a cellphone directly to asterisk. You need something called a cellsocket or a dock-n-talk. You use these to connect directly to a regular telephone, so to connect to asterisk you'll need an FXO port. I'd love to find something that would directly connect a cellphone to asterisk that didn't cost a fortune. A usb cable to the cellphone would be perfect, just a plain gsm-sip gateway would be nice too but are $. Joseph Tanner On 2/7/06, Joe Tahan [EMAIL PROTECTED] wrote: I've read something on connecting a cellphone to asterisk with bluetooth, I'm not really sure about connecting to a usb phone. I think Joseph Tanner can help us out, as he did it with bluetooth. Truely/ Joe From: Facundo Ameal [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk with USB Date: Tue, 7 Feb 2006 11:55:07 -0300 Hello everybody! I've seen that you can connect your cellphone via bluetooth, but I've a Motorola V300 and it doesn't have that feature, so I wish to connect it via USB cable, is it pissible con use my cellphone with asterisk like that? I 've not been able to find information on how to do this, I'l appreciate any help. Thanks in advance! -- Facundo Ameal. famealatgmaildotcom Linux User #395088 FWD: 741664 MSN: asadoatlamorcilladotcomdotar ICQ: 74005793 Open your mind, use open source. ___ --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 Don't just Search. Find! Try MSN Search: Fast. Clear. Easy. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] touch tones too fast ?
I think you can add w (without the quotes) to your dialplan to wait. Perhaps putting a few in front of the number, or even one in between each number? Not sure, haven't had to use this feature, sorry. Perhaps your provider doesn't like the duration of the dtmf tones themselves. For that I think you'd have to go into the zaptel source. Joseph Tanner On 2/7/06, Eldon Neustaeter [EMAIL PROTECTED] wrote: Config: AAH 2.2 Digium TDM card connecting to 3 x Telus POTS lines Polycom 501 phones pretty basic setup, working mostly just fine... When I dial a number such as: 96045551212 Telus automation will sometimes come online and tell me that the number I have dialled cannot be completed as dialled. If I hang up the Polycom 501 and redial the EXACT same number, it will work the second time. I think that AAH or Asterisk is passing touch tones to the POTS line too fast possibly. The dialplan simply has 9|. to strip out the 9's. Any suggestions? -- Eldon Neustaeter ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] callback script?
On 2/6/06, Arne Morten Johansen [EMAIL PROTECTED] wrote: Thanks. I'm able to getting the asterisk calling back to my cellphone. But when I get to the authentication I get this message when I start to dial in my password: NOTICE[5178]: rtp.c:509 ast_rtp_read: Unknown RTP codec 96 received Is this a DTMF failure of some sort? Thanks again. -Opprinnelig melding- Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne av Joseph Tanner Sendt: 4. februar 2006 11:51 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [Asterisk-Users] callback script? This is what I use, more or less: http://mundy.org/blog/index.php?p=73 , go down to Incoming Call Context (about 1/3 down). I had to modify it a bit, as I actually need Asterisk to pick up and listen to some DTMF digits before hanging up and calling me back, but it works great for me, and requires no external agi scripts. Joseph Tanner On 2/3/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: On Thursday 02 February 2006 11:40, Arne Morten Johansen wrote: How do I setup a Callback script? This script does what I want to do. But how do I set it up? http://www.junghanns.net/en/callback.html I see it uses PHP for scriptlanguage. So where do I place it (the .agi)? /var/lib/asterisk/agi-bin and should be 755 benchev ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] callback script?
Sorry for the blank email, here's what I meant to send: I haven't seen that error before, sorry. A quick search using google turned this up though: http://www.mail-archive.com/asterisk-users@lists.digium.com/msg08901.html Not sure if it's relevant in your case. What is asterisk using to dial your remote cellphone? Is it a sip, iax, or zap channel? Or are you calling out using a cellphone connected to your asterisk server (whether by a dock connected to a zap card, or bluetooth)? I've noticed that dtmf is not processed between two cellular phones on a cingular account (tried two different motorola phones plus a sony ericsson phone). It may be an issue with other carriers too, to test just call from one cellphone to another, press some keys, and see if the other side hears any dtmf tones. If not, then you'll have to find another way to do what you're trying to accomplish. Joseph Tanner On 2/6/06, Arne Morten Johansen [EMAIL PROTECTED] wrote: Thanks. I'm able to getting the asterisk calling back to my cellphone. But when I get to the authentication I get this message when I start to dial in my password: NOTICE[5178]: rtp.c:509 ast_rtp_read: Unknown RTP codec 96 received Is this a DTMF failure of some sort? Thanks again. -Opprinnelig melding- Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne av Joseph Tanner Sendt: 4. februar 2006 11:51 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [Asterisk-Users] callback script? This is what I use, more or less: http://mundy.org/blog/index.php?p=73 , go down to Incoming Call Context (about 1/3 down). I had to modify it a bit, as I actually need Asterisk to pick up and listen to some DTMF digits before hanging up and calling me back, but it works great for me, and requires no external agi scripts. Joseph Tanner On 2/3/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: On Thursday 02 February 2006 11:40, Arne Morten Johansen wrote: How do I setup a Callback script? This script does what I want to do. But how do I set it up? http://www.junghanns.net/en/callback.html I see it uses PHP for scriptlanguage. So where do I place it (the .agi)? /var/lib/asterisk/agi-bin and should be 755 benchev ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Some feedback and issues on using chan_bluetooth
I have a Motorola Razr successfully connected to asterisk using a bluetooth dongle and chan_bluetooth. Here's some issues I've run across: - You have to ignore the instructions in bluetooth.conf, saying to run sdptool search --bdaddr xx:xx:xx:xx:xx:xx 0x111F to determine the correct channel to use for your phone. My phone reported Channel 7, but will not work with anything other than Channel 3. I would recommend trying channel 3 first, then trying what sdptool suggests, then starting at 1 and working your way up until it works. - CallerID is not passed to asterisk. The CLI shows that chan_bluetooth is indeed getting the cid information, but the cid remains blank. Here's a sample of what I get during an incoming call on the cellphone (number has been changed, obviously): Feb 6 05:48:01 NOTICE[29681]: /usr/src/asterisk-test/bluetooth/chan_bluetooth/chan_bluetooth.c:1723 ag_unsol_clip: Parsed '+CLIP: 1611212,129' number='1611212' type='129' name='' [AG] Motorola +CLIP: 1611212,129 Is this working for anyone? Maybe something changed in recent asterisk versions (I'm running 1.2.1) such as a variable name for callerid? This info is not getting passed on to asterisk, so if anyone calls I won't know who they are. Not a big issue, I won't be giving this number out to anyone, but it may be an issue for some. If DTMF worked between cellphones, then this would be semi-urgent for me, as I have to get the caller's callerid information to make sure they're allowed DISA access. Since DTMF is NOT working between cellphones for me, then I have users call a regular line first, which can detect callerid just fine, will record the number they want to call, then call them back. - There's no way (that I can tell) to limit the number of calls using chan_bluetooth. This is a pretty big issue for me. Right now if someone is connected to the cellphone next to Asterisk, and someone else also tries to connect, Asterisk will happily send the dial command, causing the cellphone to call that number (and abrubtly putting the other call on hold). I really need to limit the calls (incoming and outgoing) to a total of one. That way if Asterisk tries to call out on the phone that's already on a call, it'll just jump +101 (or the call will fail, either scenario is preferable to the current situation). - Also, don't forget to enable auto-answer on your cellphone when using a headset. Otherwise when you get an incoming call, Asterisk will think it picked it up and happily proceed, but in reality the phone is just continuously ringing. The cellphone needs to automatically answer the call in order for asterisk to actually handle it. - May just be a quirk with my setup, but any calls using the bluetooth-connected cellphone that either originate or are terminated on the local asterisk system, have bad static on one side. If I call from a cellphone here, have it connect to the cellphone connected to asterisk, and call a number that rings to my local asterisk system, the cellphone gets bad static (but the person on the regular phone hears perfect audio). If I call from a regular phone connected to my asterisk system, and that call gets routed out the cellphone connected via bluetooth, then I hear everything fine but the other end hears bad static. It's an annoyance more than anything, just means I have to pay 1cent a minute for long distance during nights and weekends instead of getting them free. Note that this setup works perfect, with no static (but perhaps a slight bit of lag) when we use a cellphone on the road, connected to the cellphone next to asterisk, calling an external number. This is what my main use is, so it's not a big deal, but might be worth looking into. - Last note, anyone who is setting up a system similar to mine, and having to work around the no-dtmf tones, note that when using a callback script it will call the second party as soon as it gets an answer. Whether you pick up your cellphone or voicemail picks up, it'll consider your end answered and then call the other party. Just a note in case you use your setup to call someone, then decide you'd rather not (after it's too late) and just ignore the incoming call; as soon as your voicemail kicks in, asterisk calls the person you wanted to dial, and they'll likely just hear dead-air or the tail-end of your voicemail message, and may be upset that your voicemail is calling them for no apparent reason. Joseph Tanner ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: delaying answer for a number of ringsor an amount
well I've heard that there are open source IP phones given away for free in WALMART, I'm seriously thinking to get couple of 'em!! What phone would this be? I didn't notice any, but there's 5-6 Wal-Marts within an hour's drive, I'd love to try to find some. Never can have too many. Are they regular IP phones that connect via ethernet, or do they plug in via usb? I wouldn't want any usb ones for myself, but my dad could use one. Joseph Tanner Truely/ Joe Tahan From: Brian J. Murrell [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Re: delaying answer for a number of ringsor an amount Date: Sun, 05 Feb 2006 09:49:08 -0500 On Sun, 2006-02-05 at 05:28 -0600, Joseph Tanner wrote: Again, give everyone in your home/office a phone connected to asterisk (whether it's a sip/iax phone, or a regular phone connected to an ATA, or what have you). Sure. Wanna send me some ATAs or even IP phones? It's all about budget dude. Not everyone has the $$ to outfit the whole house with IP and IP phones right away. b. -- My other computer is your Microsoft Windows server. Brian J. Murrell signature.asc ___ --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 Share a single photo or an entire slide show right inside your e-mail with MSN Premium. Join now and get the first two months FREE* ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: delaying answer for a number of ringsor an amount
Funny funny. In this day of free (after rebate) PAP2s, a free (again, I assumed after rebate) IP phone seemed plausible. BTW, check walmart.com, they do indeed sell ip phones. I guess I'll just have to use one of my free DTA310s or my free PAP2 instead. Joseph Tanner On 2/6/06, Gonzalo Servat [EMAIL PROTECTED] wrote: On 2/6/06, Joseph Tanner [EMAIL PROTECTED] wrote: well I've heard that there are open source IP phones given away for free in WALMART, I'm seriously thinking to get couple of 'em!! What phone would this be? I didn't notice any, but there's 5-6 Wal-Marts within an hour's drive, I'd love to try to find some. Never can have too many. Are they regular IP phones that connect via ethernet, or do they plug in via usb? I wouldn't want any usb ones for myself, but my dad could use one. Oh they're giving away both types! and if you hurry, you get a free Asterisk box to go with it! go go go! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: delaying answer for a number of ringsor anamount
Jose, Close, check the bottom of my messages, and the name sent along with my email address; it's Joseph not Jose. There are No open source IP phones, I was only joking, I assumed you should know what an open source is. There are no open source routers, no open source PBXs, no open source (insert name of open source product here). So is it impossible for an IP phone to run linux? I had to do some searching, but there are IP phones available that run linux. Just because the general public doesn't know it runs linux (or another open source OS), doesn't mean that it's not open source. Now, my definition of an open source product is probably different from yours. If it runs linux (or another open source OS), then that's good enough for me. Even if parts are closed source, I'm more concerned about the OS itself. Perhaps you meant open source as in everything is completely open, the OS, all supporting programs, etc. In that case, I am not sure an open source IP phone exists, but I would not think someone stupid for thinking that one COULD exist. Joseph Tanner Truely/ Joe From: Joseph Tanner [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Re: delaying answer for a number of ringsor anamount Date: Mon, 6 Feb 2006 06:24:43 -0600 well I've heard that there are open source IP phones given away for free in WALMART, I'm seriously thinking to get couple of 'em!! What phone would this be? I didn't notice any, but there's 5-6 Wal-Marts within an hour's drive, I'd love to try to find some. Never can have too many. Are they regular IP phones that connect via ethernet, or do they plug in via usb? I wouldn't want any usb ones for myself, but my dad could use one. Joseph Tanner Truely/ Joe Tahan From: Brian J. Murrell [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Re: delaying answer for a number of ringsor an amount Date: Sun, 05 Feb 2006 09:49:08 -0500 On Sun, 2006-02-05 at 05:28 -0600, Joseph Tanner wrote: Again, give everyone in your home/office a phone connected to asterisk (whether it's a sip/iax phone, or a regular phone connected to an ATA, or what have you). Sure. Wanna send me some ATAs or even IP phones? It's all about budget dude. Not everyone has the $$ to outfit the whole house with IP and IP phones right away. b. -- My other computer is your Microsoft Windows server. Brian J. Murrell signature.asc ___ --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 Share a single photo or an entire slide show right inside your e-mail with MSN Premium. Join now and get the first two months FREE* ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Take the effort out of being organized with MSN(r) Calendar. MSN Premium: Join now and get the first two months FREE* ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: delaying answer for a number of ringsor an amount
There's idiots that tell people about free, and cheaper-than-free deals all the time. Here's just one such idiot: http://www.fatwallet.com/forums/messageview.php?start=0catid=24threadid=524641 BTW, the idea is to get all that you want for yourself first, THEN tell everyone about the deal. Again, I'm thinkin free = FAR (free after rebate). But, I have seen at least one time, a completely, 100% free item, no rebates, no gimmicks, advertised. I believe it was OfficeMax, and was some kind of check printing software. It was advertised as being free without rebates, was not a misprint, and people were having no problems walking out the door with it without paying a penny. Now, had you said these phones were 100% free, no rebates or anything, completely unlocked, no service plans needed, etcthen I would have seriously doubted you, enough that I wouldn't even bother calling anyone up to verify the deal (though if I was already in a Wal-Mart, and in the right section, I might just glance to make sure). But, in the interest of getting this thread back on track: Boy oh boy am I stupid! Man oh man, you pulled one over on me. I must be the most stupidest idiotest dumb there is! Hahaha! Good one! Joseph Tanner On 2/6/06, Gonzalo Servat [EMAIL PROTECTED] wrote: On 2/6/06, Joseph Tanner [EMAIL PROTECTED] wrote: Funny funny. In this day of free (after rebate) PAP2s, a free (again, I assumed after rebate) IP phone seemed plausible. BTW, check walmart.com, they do indeed sell ip phones. I guess I'll just have to use one of my free DTA310s or my free PAP2 instead. ... and even if they *did* indeed give away free phones, which is unimaginable as they're in the business of MAKING money by SELLING, do you really think people are going to come here and tell the world about it? Bit gullible, aren't ya... ;-) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: delaying answer for a number of rings or an amount
Here's a step-by-step of what happens below: 1 - a call comes in and Asterisk rings SIP/Brian and SIP/joe for 30 seconds. So you don't want Asterisk to wait and see if the POTS line is picked up before ringing the SIP phones? Interesting. If it's anything like my setup, Asterisk handles ALL calls, whether from sip, iax, or zap. So when the zap line rings, asterisk will ring your internal sip phone(s), and if the call isn't picked up after so many seconds, it'll stop ringing the internal lines and go straight to voicemail. No phones are connected directly to the POTS line, just asterisk. The only downside to this approach, is the caller will hear about two rings before you beging to hear anything (takes asterisk that long to see the call, check for callerid, then start ringing your internal lines). My solution is to have a quick greeting played to the caller, then they hear ringing again when the internal lines ring. Also gives me a chance to force callers to press 1 if I don't recognize their callerid, stops telemarketers dead in their tracks (those auto-dialing machines that ring you and either hang up after you pick up, or tell you to stay on the line for an important message, will not know to dial 1 first and will be hung up on). 2 - After 30 seconds if the line is still ringing (nobody picked up POTS phone or SIP phones) * answers the line and sends to Voicemail. Asterisk never picks up the call until the 30 seconds are up. What seems to be happening here is that even if somebody picks up the POTS line within a few seconds, after the 30 seconds (Wait() in my case, but I'd imagine the same will happen after ringing the SIP lines for 30s) is up Asterisk is also on the POTS line (with the callee who picked up the POTS phone) doing the voicemail intro and recording the conversation. Again, give everyone in your home/office a phone connected to asterisk (whether it's a sip/iax phone, or a regular phone connected to an ATA, or what have you). Any call that comes in will go through asterisk. Then you won't have to worry about having it detect if a POTS line was picked up directly, if you have it pass the call to an internal phone, it'll know if that phone picked up or not, and will know whether to pass it to voicemail or not. Joseph Tanner [from-pots] exten = s,1,Dial(SIP/brianSIP/joe,30) exten = s,2,Voicemail(u2001) exten = s,3,Hangup I will try this exactly and see if it works any better. b. -- My other computer is your Microsoft Windows server. Brian J. Murrell -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (GNU/Linux) iD8DBQBD45ffl3EQlGLyuXARAobbAJoCaGeIV/gzNTyfw1h6xt+EYCdHPwCeIwfZ J3CaPbHa1j3wxqJw/aK9+NY= =ttIm -END PGP 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 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to handle provider UNREACHABLE in the dialplan?
This is probably a stupid question, but how do you specify multiple
fallovers? I.e., if provider1 is not reachable/busy, try provider2.
If provider2 is down, try provider3. If provider3 is down...etc. I
understand how to do it the old way, just keep adding 101 to the
extension. What would you add to a NOANSWER extension though? I
guess you could send it to a different context, then you could use
another NOANSWER, but I like keeping things short and easy.
Joseph Tanner
On 2/3/06, Florian Overkamp [EMAIL PROTECTED] wrote:
Hi Ronald,
Ronald Wiplinger wrote:
You could read out all the entries in the DNS zone and create your own
list of entries in /etc/hosts, and then create multiple asterisk
peers: voipbuster1, voipbuster2, etc... Then you can use regular
dialplan logic to cycle through all of them.
that is exactly the point what I am looking for. How can I use the next
peer in the dial logic? I was trying DIALSTATUS, ... but I could not
make it.
Should be easy; we use:
[macro-safedial]
;exten = s,1,Dial(${ARG1},${ARG2},g,${ARG4})
exten = s,1,Dial(${ARG1},${ARG2},${ARG3},${ARG4})
exten = s,2,Goto(s-${DIALSTATUS},1)
exten = s-CANCEL,1,Hangup
exten = s-NOANSWER,1,GotoIf($[${DIALEDTIME} = 0]?3)
exten = s-NOANSWER,2,Hangup
exten = s-NOANSWER,3,Verbose(1,Need failover for ${ARG1})
exten = s-BUSY,1,Busy
exten = s-CHANUNAVAIL,1,Verbose(1,Need failover for ${ARG1})
exten = s-CONGESTION,1,Congestion
exten = _s-.,1,Congestion
exten = s-,1,Congestion
Florian
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Re: [Asterisk-Users] callback script?
This is what I use, more or less: http://mundy.org/blog/index.php?p=73 , go down to Incoming Call Context (about 1/3 down). I had to modify it a bit, as I actually need Asterisk to pick up and listen to some DTMF digits before hanging up and calling me back, but it works great for me, and requires no external agi scripts. Joseph Tanner On 2/3/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: On Thursday 02 February 2006 11:40, Arne Morten Johansen wrote: How do I setup a Callback script? This script does what I want to do. But how do I set it up? http://www.junghanns.net/en/callback.html I see it uses PHP for scriptlanguage. So where do I place it (the .agi)? /var/lib/asterisk/agi-bin and should be 755 benchev ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_bluetooth: successful compile and outbound cell calls: Still tweaking inbound setup. WAS: Cannot compile chan_bluetooth on Asterisk 1.2.1
Update: I finally have multiple cingular phones to test this with. Incoming DTMF does NOT work (at least not with a Motorola V551, RAZR V3, or a Sony Ericsson T237, both Motorolas were used with bluetooth, the Ericsson does not have bluetooth so could not be tested connected to asterisk). I actually called the phones directly, and there's no dtmf being sent between the mobile phones. When one calls a landline (or voip, anything that's not a mobile phone) then dtmf works. Perhaps they've caught on to us? Temporary solution right now, is to call a regular line connected to the asterisk box, authenticate the user, ask for the number they want to dial, hangup and then call them back (and call the number they wanted to dial). It'll use a minute off your plan, but it's better than 10, 20, 60, or more minutes. If anyone knows a workaround, please let us know! It'd be nice if there was a firmware edit that'd do something like forcing inband dtmf, or what-not. Joseph Tanner On 1/27/06, Nilesh Londhe [EMAIL PROTECTED] wrote: Editing subject line to reflect current status. On 1/26/06, Nilesh Londhe [EMAIL PROTECTED] wrote: Since T616 is not answering (and incoming calls are going to Cingular voicemail after 30 sec,) I suspect the problem focus area is... -- Executing Answer(BLT/T616, ) in new stack Is http://www.thetechguide.com/howto/asterisk/bluetoothfiles.tar.gz tar xzf bluetoothfiles.tar.gz the latest source (r40?) On 1/26/06, Nilesh Londhe [EMAIL PROTECTED] wrote: Here are my findings with my experiment using Sony Erisson T616 with Cingular Service and connected to [EMAIL PROTECTED] 2.2 on a freshly installed system and following the instructions http://www.thetechguide.com/howto/asterisk/chanbluetooth.html Outbound calls (Asterisk to T616 via bluetooth): Works OK via Dial(BLT/T616/8005551212) Inbound calling (T616 to asterisk via bluetooth): My configuration for inbound calls: [bluetooth] exten = s,1,Wait(1) exten = s,2,Answer exten = s,3,Dial(SIP/1007,15,rtT) exten = s,4,VoiceMail([EMAIL PROTECTED]) exten = s,5,Hangup My observation: When I call my cell T616 from my landline, SIP/1007 rings for 2 seconds and the call is answered by Cingular voicemail not by asterisk voicemail. My cingular voicemail is set to answer in 30 seconds after first ring. Output on the asterisk CLI: [EMAIL PROTECTED] ~]# asterisk -r Asterisk 1.2.1, Copyright (C) 1999 - 2005 Digium. Written by Mark Spencer [EMAIL PROTECTED] = Connected to Asterisk 1.2.1 currently running on asterisk1 (pid = 3025) Verbosity is at least 3 [AG] T616 +CIEV: 2,4 [AG] T616 +CIEV: 2,3 [AG] T616 RING [AG] T616 +CLIP: 421212,161,,,Landline -- Executing Wait(BLT/T616, 1) in new stack -- Executing Answer(BLT/T616, ) in new stack [AG] T616 +CIEV: 2,1 [AG] T616 +CIEV: 3,0 -- Executing Dial(BLT/T616, SIP/1007|15|rtT) in new stack -- Called 1007 -- SIP/1007-d97e is ringing == Spawn extension (bluetooth, s, 3) exited non-zero on 'BLT/T616' [AG] T616 ATH [AG] T616 AT+CHUP [AG] T616 ERROR [AG] T616 OK [AG] T616 AT+BRSF=23 [AG] T616 ERROR [AG] T616 AT+CIND=? [AG] T616 +CIND: (battchg,(0-5)),(signal,(0-5)),(batterywarning,(0-1)),(chargerconnected,(0-1)),(service,(0-1)),(sounder,(0-1)),(message,(0-1)),(call,(0-1)),(roam,(0-1)),(smsfull,(0-1)) [AG] T616 OK [AG] T616 AT+CIND? [AG] T616 +CIND: 5,3,0,1,1,0,0,0,0,0 [AG] T616 OK [AG] T616 AT+CMER=3,0,0,1 [AG] T616 OK [AG] T616 AT+CLIP=1 [AG] T616 OK [AG] T616 AT+CGMI [AG] T616 SONY ERICSSON [AG] T616 OK [AG] T616 AT+CGMI [AG] T616 SONY ERICSSON [AG] T616 OK [AG] T616 +CIEV: 2,4 [AG] T616 +CIEV: 2,3 asterisk1*CLI On 1/26/06, Nilesh Londhe [EMAIL PROTECTED] wrote: BTW, I did get clear bidirectional audio when I succeded in dialing out...(with the channel = 3 in /etc/asterisk/bluetooth.conf) I have Sony Ericsson T616 connected to a cheap commodity bluetooth USB dongle that I bought ages ago from meritline. On 1/26/06, Nilesh Londhe [EMAIL PROTECTED] wrote: Thanks a billion. Outbound bluetooth dialling on the lines of Dial(BLT/DevName/8005551212) worked for me. Still trying out the inbound route. Before I created the [bluetooth] context, it tried to reach the [default] context but then I began by creating a new context [bluetooth] in extensions.conf and got my internal SIP phone to ring when I received a call on my SE T616 cell phone. However, I could not get the inbound line
Re: [Asterisk-Users] Nagios and Asterisk
I have used both, just not together. I have a possible idea though. If they're running on separate servers, you can have nagios send an email that the asterisk server receives. Have different email aliases for different alerts, or have a script parse the email to see what kind of alert it is. Have this script generate a .call file in /var/spool/asterisk/outgoing based on the type of alert. If they're running on the same server you might be able to skip having to send an email (but if not, then just have it send an email to a local user, it'll work the same). Personally, I just had Nagios send an email whenever there was a problem. If the tech is in front of their workstation, they'll get a notice immediately. If not, you could have a text message sent instead. Worked great for me. On 1/27/06, Darrell Long [EMAIL PROTECTED] wrote: Is anyone using Asterisk (and Festival) to make calls to appropriate persons (techs, etc. ) when Nagios generates a particular type of alert? If so, I would love to hear how people are doing it. Thanks, -- Darrell S. Long BestWeb Corporation ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help with Congestion error
I think it's a problem on Voicepulse's end, I'm having the same problems. If there is a problem on your end, let me know, maybe I have the same problem. Personally, I'm using voxee for all my outbound calls with voicepulse as a backup. I'd probably pick someone else as a backup (their rates are a bit high), but I like their auto-fill feature. If I forget to refill a prepaid account, I could be in big trouble. But in my case the call'd just fail, then try voicepulse next, which will always have a positive balance. Quick question, anyone recommend any other decent providers with a credit card auto-fill option? Rates aren't as big a deal as reliability, I'll use the fly-by-night, untested, prepaid-only providers as the first provider, and a reliable one as the backup. On 1/27/06, Naren Koka [EMAIL PROTECTED] wrote: I am using Asterisk with Connect.VoicePulse. Of late, we are getting too many congestion errors. Chris Icide has helped me before in setting up the server. He has done a wonderful job. It has worked very well until about 2 months ago. Now I need some help to fix this issue. I appreciate the help. Sincerely, Naren Koka (480) 829-0479 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Lockups since upgrade 1.2.3 - anyone else? Any ideas?
Quick and dirty solution: mv /usr/lib/asterisk/modules /usr/lib/asterisk/modules.bak Then go into the asterisk source directory (in my case, /usr/src/asterisk) and do a make install. Might as well re-install the asterisk-addons too, if you need anything there. Try running asterisk now and put it through its paces. If you're missing any functionality, try to put it back in (probably a module included in asterisk-addons). If you can't get it working and time is critical, just stop asterisk, do a mv /usr/lib/asterisk/modules /usr/lib/asterisk/modules.new and then a cp -r /usr/lib/asterisk/modules.bak /usr/lib/modules and restart asterisk and try to figure out what went wrong. The modules.new directory has all the new modules, modules.bak still has the old ones. Joseph Tanner On 1/27/06, Dan Littlejohn [EMAIL PROTECTED] wrote: On 1/27/06, Noah Miller [EMAIL PROTECTED] wrote: Hi Brent - Boy oh boy. This blows. I upgraded to 1.2.2 from 1.0.9, and of course had the timebomb bug. Immediately after upgrading to 1.2.3 we were ok, for 24 hours or so. Since upgrading to 1.2.3, though, the whole system has locked up twice. Once on Thursday, and then about a half hour ago. The server would reply to a ping, but no ssh login, no local console login - just locked up. This ain't good for business. We've been doing fine with 1.2.3 so far. No problems reported, though I only have it deployed in a small office. Definitely no lock-ups. On the asterisk side, just a basic question - did you make sure to remove the old modules so the new 1.2.3 versions got installed? As far as the lockups, maybe it is coincidental? I've never had asterisk (even the crazy CVS versions) lock a whole OS like that. I have had machines running asterisk lock up, but it always turned out to be caused by something else like bad hardware, or unrelated network problems. - Noah ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I was confused about the modules. Got this warning when upgrading to 1.2.3 even when using the most current asterisk-addons and even svn asterisk-addons. WARNING WARNING WARNING Your Asterisk modules directory, located at /usr/lib/asterisk/modules contains modules that were not installed by this version of Asterisk. Please ensure that these modules are compatible with this version before attempting to run Asterisk. app_addon_sql_mysql.so app_rxfax.so app_saycountpl.so app_striplsd.so app_substring.so app_txfax.so cdr_addon_mysql.so chan_modem_aopen.so chan_modem_bestdata.so chan_modem_i4l.so chan_modem.so format_mp3.so res_config_mysql.so WARNING WARNING WARNING Do not understand how to fix this? Do not know if that would also be related to the ops crashing. Dan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Lockups since upgrade 1.2.3 - anyone else? Any ideas?
Got my commands mixed up. The last should be cp -r /usr/lib/asterisk/modules.bak /usr/lib/asterisk/modules, it shouldn't be /usr/lib/modules. Sorry bout that, wasn't thinking clearly. Joseph Tanner On 1/27/06, Joseph Tanner [EMAIL PROTECTED] wrote: Quick and dirty solution: mv /usr/lib/asterisk/modules /usr/lib/asterisk/modules.bak Then go into the asterisk source directory (in my case, /usr/src/asterisk) and do a make install. Might as well re-install the asterisk-addons too, if you need anything there. Try running asterisk now and put it through its paces. If you're missing any functionality, try to put it back in (probably a module included in asterisk-addons). If you can't get it working and time is critical, just stop asterisk, do a mv /usr/lib/asterisk/modules /usr/lib/asterisk/modules.new and then a cp -r /usr/lib/asterisk/modules.bak /usr/lib/modules and restart asterisk and try to figure out what went wrong. The modules.new directory has all the new modules, modules.bak still has the old ones. Joseph Tanner On 1/27/06, Dan Littlejohn [EMAIL PROTECTED] wrote: On 1/27/06, Noah Miller [EMAIL PROTECTED] wrote: Hi Brent - Boy oh boy. This blows. I upgraded to 1.2.2 from 1.0.9, and of course had the timebomb bug. Immediately after upgrading to 1.2.3 we were ok, for 24 hours or so. Since upgrading to 1.2.3, though, the whole system has locked up twice. Once on Thursday, and then about a half hour ago. The server would reply to a ping, but no ssh login, no local console login - just locked up. This ain't good for business. We've been doing fine with 1.2.3 so far. No problems reported, though I only have it deployed in a small office. Definitely no lock-ups. On the asterisk side, just a basic question - did you make sure to remove the old modules so the new 1.2.3 versions got installed? As far as the lockups, maybe it is coincidental? I've never had asterisk (even the crazy CVS versions) lock a whole OS like that. I have had machines running asterisk lock up, but it always turned out to be caused by something else like bad hardware, or unrelated network problems. - Noah ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I was confused about the modules. Got this warning when upgrading to 1.2.3 even when using the most current asterisk-addons and even svn asterisk-addons. WARNING WARNING WARNING Your Asterisk modules directory, located at /usr/lib/asterisk/modules contains modules that were not installed by this version of Asterisk. Please ensure that these modules are compatible with this version before attempting to run Asterisk. app_addon_sql_mysql.so app_rxfax.so app_saycountpl.so app_striplsd.so app_substring.so app_txfax.so cdr_addon_mysql.so chan_modem_aopen.so chan_modem_bestdata.so chan_modem_i4l.so chan_modem.so format_mp3.so res_config_mysql.so WARNING WARNING WARNING Do not understand how to fix this? Do not know if that would also be related to the ops crashing. Dan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VOXEE Caller ID..
I'm running Asterisk 1.2.1. You're supposed to have to set callerid this way: Set(CALLERID(num)=9315551212) In fact, doing this with voicepulse works fine. However it doesn't with voxee (at least for me). I have to set callerid the old fashioned way: SetCallerID(9315551212) I even tried setting it using both methods, the correct method followed by the old method, and it still wouldn't work (at least for me). The old way still works for voicepulse too, so I just left it set that way. Joseph Tanner On 1/27/06, Ben Higley [EMAIL PROTECTED] wrote: I cannot find any means of passing my own Callerid using Voxee. It always comes across as NO ID, or nothing, or unknown. I could not find anything on their website about setting your own caller id in the system either. (their web account pages). Is anyone here using their own Callerid information through Voxee? thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No audio? Update your Asterisk
For what it's worth, I've been messing around with my install all night and haven't had a single issue. [EMAIL PROTECTED] 2.2, Asterisk version 1.2.1. Even set the date ahead, still no problems. Could be a fluke, I'm interested if anyone else is using 1.2.1 and has these issues, but for now I'm sticking with what I have. BTW, all my testing tonight involved SIP (Sipura SPA-2000) and either PSTN using an X100P card, IAX account with voxee, or a SIP account with vbuzzer. Had audio both ways all the time. Joseph Tanner On 1/25/06, BJ Weschke [EMAIL PROTECTED] wrote: On 1/25/06, Darren Ellis [EMAIL PROTECTED] wrote: Guys, I'm not familiar enough with mantis to tell what version of asterisk are affected by this bug? I have 1.09, 1.10, 1.2.1 and 1.2.2 (as a test) deployed. Can someone tell me what the real impact is going to be? Thanks 1.2.1 and 1.2.2 are likely affected. I didn't see this code in a pre-1.2 system I reviewed earlier this morning for a client. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best FXO hardware for home use
Personally, I've had great success with an X101P (it's a clone, but it's the exact same chipset and layout of the original). Now, with Asterisk 1.2 beta2 (I believe it was beta2, I could be wrong though) and a P3 933MHz PC I did get annoying echo that I couldn't get rid of, and only on outgoing calls. If someone called me, even though all the same equipment is being used, there was no echo. Anyways, I upgraded to [EMAIL PROTECTED] 2.2 with Asterisk 1.2.1 and at the same time upgraded to a Celeron 2.93GHz PC, and there's virtually no echo. Only if there's complete silence on the other end and you yell very loud, can you barely make any hint of an echo out. No idea if it was the Asterisk upgrade, the new PC, or both that fixed my problem. Also, somewhere around the pre-1.0 days, I had two of these clones (one was the exact same layout as the actual X101P, the other had a different layout but the same chipset) and the one I used with my Packet8 line had no echo, but my landline did. Didn't matter if I switched the lines, the one connected to the Packet8 device had zero echo, the one connected to my landline had a noticeable echo (again, only on outgoing calls, incoming was fine). Played with rxgain/txgain, all the echo settings, etc. But now all is fine. Guess what I'm trying to say, is a lot depends on the line itself, and your exact setup. If you can pick up an X101P clone for cheap, I'd try that first. Most you're out is a few bucks (I say a few bucks, cause even if you pay $20 and decide it won't work for you, you can sell it for about what you paid). If you build or repair PCs a lot for others, then you'll need a good cheap modem someday anyways, the clone cards work fine for that. Works fine for me, only issue I have now is callerid isn't 100% reliable, but works the majority of the time. Until I troubleshoot it further (i.e., connect a regular phone directly to my landline to at least verify it's getting callerid when asterisk isn't), I can't blame the card for that. As long as the card will work with your setup (odds are it will), I think it's the best solution for home or small business use. Joseph Tanner On 1/25/06, Rich Adamson [EMAIL PROTECTED] wrote: echo cancellation is pretty limited on these cheap devices. the spa3000 manual for example states the AEC is limited to 8ms. good AECs will handle 64ms or more. in my experience the spa3000 echo canceller is cranky. it works most but not all of the time. I have been using one for 6 months without any problems. Make sure you have the most current firmware on it and it should work just fine. Kerry, There is an issue with the spa3k (as well as the TDM04b) in terms of handling echo properly on long pstn loops. You are obviously on a relatively short loop if you've not been exposed to the variable echo cancellation issues. In other words, long pstn loops basically fall outside the limits of the echo cancellation software as someone else already noted. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cannot compile chan_bluetooth on Asterisk 1.2.1
Please note this is a work in progress: http://www.thetechguide.com/howto/asterisk/chanbluetooth.html Basically the bluetoothfiles.tar.gz has the cvs code with the Makefile that worked for me, plus the edited Makefile in /usr/src/asterisk/channels, and the bluez edits I needed (hcid.conf with the correct profile, the files needed for the pin which is set to 1234, etc.). The guide is supposed to walk a person through the entire process of getting an Asterisk box setup and bluetooth working, but it's grossly incomplete. Maybe it'll help you out. Joseph Tanner On 1/25/06, Nilesh Londhe [EMAIL PROTECTED] wrote: Hi Joseph: I still couldn't compile the newest cvs version of chan_bluetooth, so I again tried my trick of using the Makefile from an older version (which used .tmp to compile) and it worked! Can you please point to the cvs you used, the location and content of pin files you created and paste a copy of the make file that worked for you? Appreciate you sharing this information. Thanks. On 1/20/06, Joseph Tanner [EMAIL PROTECTED] wrote: Ok, I did get this going (somewhat), and in case someone else has the same issues I'll detail what I had to do. First, I was using the instructions at http://mundy.org/blog/index.php?p=79. They stated that [EMAIL PROTECTED] 2.2 already had all the rpms necessary for bluetooth and that I could skip the yum install step. I did, however, run the command anyways after a few failed attempts. There's an error in the rpm name, they tell you to install bluez-libs, the correct name is bluez-libs-devel (at least, that's what I needed to install). I still couldn't compile the newest cvs version of chan_bluetooth, so I again tried my trick of using the Makefile from an older version (which used .tmp to compile) and it worked! Once compiled, I installed and started up asterisk. I then received a message on my phone asking if I wanted to allow asterisk to connect, and then asked for a pin. This took a bit of figuring out, but I got passed that. In /etc/bluetooth/hcid.conf, there's a line that says pin_helper /usr/bin/bluepin; (it may have a different path, the important thing is the pin_helper part). Now backup the script in question, i.e. in my case mv /usr/bin/bluepin /usr/bin/bluepin.bak. Use your editor of choice to create a new file with the same name, and in it enter: #!/bin/sh -e echo PIN:1234 Replace the 1234 with whatever you want your pin to be. I don't know if this is necessary, but I also edited /etc/bluetooth/pin to read: 1234 Again, 1234 should be whatever you want your pin to be. I then stopped asterisk, stopped the bluetooth service, started the bluetooth service back up, started asterisk, then when my phone asked for a pin I put in 1234, and it worked! You may also need to make another edit to hcid.conf, under Local Device Class change it to read class 0x200404; or possibly class 0x700408;. This makes your bluetooth dongle look like a headset, and not a data device (I experienced some flakiness until I changed this). Now, I edited /etc/asterisk/bluetooth.conf appropriately (changed the channel for the phone to 7, it's a Motorola V551), started it all up, made some test calls and...no audio! The cellphone works great otherwise. It'll connect, stay connected as long as I want it to, and when I hang up the asterisk extension the cellphone will disconnect too. Too bad I didn't realize 611 was a free call until after I made a lot of test calls (it's a prepaid phone). I did call our home number directly to see if maybe I just had a one-way audio problem, but nobody could hear a thing on either end. I will continue to troubleshoot this before I ask another question about it, but it's not looking good. BTW, the usb dongle I'm using is a Linksys USBBT100. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Want to automatically park call and have caller hear ring tones
Here's the short of it. I have an Asterisk 1.2.1 system setup to
handle both personal and business calls. Now, the business callers
will hear music while on hold, so the default MOH needs to play
regular music. Personal callers should hear rings, not music. I have
this working except for one specific case. If someone calls during
the day (we're night people), asterisk will not ring the phones.
Often we'll be up before asterisk thinks we are, so we will miss quite
a few calls.
My solution was to have asterisk park the call for 15 seconds, send
the callerid information to a YAC listener on my laptop and our TiVo,
and I can pick up any phone and dial 4 to pick up the parked call.
Works perfect, except parked callers hear music, not ringing. To make
it a bit less confusing, I play a quick wav file saying to please hold
while your call is connected, which sounds rather impersonal (at least
for a personal call, it's fine for biz calls and is what I use there
too).
Is there a way to have a call parked, and have the caller hear the
default ringing tones, and not have to mess around with MOH?
Currently I'm using ParkAndAnnounce, and just announcing it to
/dev/null (which it complains about, but it works). Here's the
specific section of my extensions.conf file if anyone's curious:
[asleep]
exten = s/_931555,1,NoOp
exten = s/7205879978,1,NoOp
exten = s/4025179978,1,NoOp
exten = s,1,System(/bin/echo -n -e '${CALLERIDNAME} ${CALLERIDNUM}'
| nc -w 1 192.168.1.16 10629)
exten = s/_931555,2,NoOp
exten = s/7205879978,2,NoOp
exten = s/4025179978,2,NoOp
exten = s,2,System(/bin/echo -n -e '${CALLERIDNAME} ${CALLERIDNUM}'
| nc -w 1 192.168.1.19 10629)
exten = s,3,NoOp
exten = s,4,NoOp
exten = s/_731584,5,NoOp
exten = s/7205879978,5,NoOp
exten = s/4025179978,5,NoOp
exten = s,5,Playback(custom/pleasehold)
exten = s,6,ParkAndAnnounce(pbx-transfer:Parked|15|/dev/null|asleep,s,8)
exten = s,7,NoOp
exten = s,8,Playback(custom/voicemail)
exten = s,9,Voicemail(s1)
exten = s,10,Hangup
exten = s,107,NoOp
exten = s,108,Playback(custom/voicemail)
exten = s,109,Voicemail(s1)
exten = s,110,Hangup
Note that the call doesn't start here, rather it starts in another
context which checks the hours, then if it's in the middle of the day
it'll pass it off to the asleep context. Here's what each line does.
The first four lines are for s,1, basically if the callerid matches
one of the first three numbers, it does a NoOp. Otherwise, it
performs the normal s,1 line which uses echo and nc (netcat) to send
the callerid information to my TiVo. The s,2 lines are the same
thing, except it sends the info to my laptop (I changed the first
number to something generic, that's not what's actually in my config;
the other two are from calling cards that an annoying member of our
family uses, basically I'm ignoring all calls from them). s,3 and s,4
are both NoOp in for future expansion. s,5 gives the Please Hold
message, otherwise suddenly hearing music would be confusing. s,6
parks the call for 15 seconds, after which time it returns to the
asleep context, line s,8. s,7 is for future use. s,8 plays back a
custom voicemail greeting, s,9 is for the caller to leave a voicemail
(vm doesn't give its own greeting, just starts with a beep). s,10
hangs up. s,107 is in case the parkandannounce doesn't work for
whatever reason, it's a NoOp which passes it to s,108 which as before
plays the vm greeting, s,109 is the actual voicemail, s,110 hangs up.
BTW, I just noticed I need to add those bad callerids between s,5 and
s,6, else they will be parked but not hear the please hold message (I
want those numbers to go straight to voicemail, I'll clean it up later
so I'm not repeating myself all the time).
Is there an easy way to do what I want, parking these calls
automatically and the caller just hears the normal ringing tone?
Thanks!
___
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Re: [Asterisk-Users] Cannot compile chan_bluetooth on Asterisk 1.2.1
Again, my documentation is still sparse. I should have noted that the phone will recognize asterisk and connect even if the channel in bluetooth.conf is configured wrong. You'll just get no audio, or disconnects, or what-not until it's set correctly. So realize that later on when you're testing. Also the usb dongle must have a CSR chipset, else it won't work (well, at least probably won't work, I'll provide instructions on how to tell if it should work or not later). Here's the relevant instructions on http://www.crazygreek.co.uk/content/chan_bluetooth for how to dial out: Send a call out by using Dial(BLT/DevName/0123456). As far as dialing in, there's a special context (I think [bluetooth] maybe? I'll have to get back to you on that). I know that it should work fine, because I tried dialing the phone, asterisk picked it up then immediately disconnected because there was no context for it to go to (I think it tried to fall back on [default], which I didn't have configured to accept an incoming call). Good luck! Joseph Tanner On 1/26/06, Nilesh Londhe [EMAIL PROTECTED] wrote: Thanks a lot. I succeeded in pairing my Sony Ericson T616 using your instructions at http://www.thetechguide.com/howto/asterisk/chanbluetooth.html without any problems. I rebooted and the phone prompted me to connect to asterisk. I provided the pin 1234 and voila it connected... Couple of observations: I started off with clean slate and booted off from [EMAIL PROTECTED] 2.2 CD. skipped the initial yum -u update part to save some time. When I ran the sdptool search --bdaddr MACADDRESS 0x111F command, below is what I got: Class 0x111F Searching on MACADDRESS Service Name: HF Voice Gateway Service RecHandle: 0x10007 Service Class ID List: (0x111f) Generic Audio (0x1203) Protocol Descriptor List: L2CAP (0x0100) RFCOMM (0x0003) Channel: 6 Profile Descriptor List 0x111e Version 0x0100 Note that in /etc/asterisk/bluetooth.conf, I kept Channel = 3 (did not change it to 6) and it paired my tooth in the first attempt after I rebooted asterisk box. Now, I want to get rid of my Doc-N-Talk that I currently connect my T616 to and the other end of Doc-N-Talk goes to x100p. Although I have worked with linux a bit, I am basically an ASTERISK NEWBIE so please pardon my ignorane but I don't know what to do next...that is.. how to define this bluetooth channel to make and receive calls using this setup... Appreciate your help. On 1/25/06, Joseph Tanner [EMAIL PROTECTED] wrote: Please note this is a work in progress: http://www.thetechguide.com/howto/asterisk/chanbluetooth.html Basically the bluetoothfiles.tar.gz has the cvs code with the Makefile that worked for me, plus the edited Makefile in /usr/src/asterisk/channels, and the bluez edits I needed (hcid.conf with the correct profile, the files needed for the pin which is set to 1234, etc.). The guide is supposed to walk a person through the entire process of getting an Asterisk box setup and bluetooth working, but it's grossly incomplete. Maybe it'll help you out. Joseph Tanner On 1/25/06, Nilesh Londhe [EMAIL PROTECTED] wrote: Hi Joseph: I still couldn't compile the newest cvs version of chan_bluetooth, so I again tried my trick of using the Makefile from an older version (which used .tmp to compile) and it worked! Can you please point to the cvs you used, the location and content of pin files you created and paste a copy of the make file that worked for you? Appreciate you sharing this information. Thanks. On 1/20/06, Joseph Tanner [EMAIL PROTECTED] wrote: Ok, I did get this going (somewhat), and in case someone else has the same issues I'll detail what I had to do. First, I was using the instructions at http://mundy.org/blog/index.php?p=79. They stated that [EMAIL PROTECTED] 2.2 already had all the rpms necessary for bluetooth and that I could skip the yum install step. I did, however, run the command anyways after a few failed attempts. There's an error in the rpm name, they tell you to install bluez-libs, the correct name is bluez-libs-devel (at least, that's what I needed to install). I still couldn't compile the newest cvs version of chan_bluetooth, so I again tried my trick of using the Makefile from an older version (which used .tmp to compile) and it worked! Once compiled, I installed and started up asterisk. I then received a message on my phone asking if I wanted to allow asterisk to connect, and then asked for a pin. This took a bit of figuring out, but I got passed that. In /etc/bluetooth/hcid.conf, there's a line that says pin_helper /usr/bin/bluepin; (it may have a different path, the important thing is the pin_helper part). Now backup the script in question, i.e. in my case mv /usr/bin/bluepin /usr/bin/bluepin.bak. Use your editor of choice to create a new
Re: [Asterisk-Users] Want to automatically park call and have callerhear ring tones
Having a dummy extension ring sounds like a great idea, but I'm not sure how to implement it. Can it ring the console if there's no soundcard on the server? Already ran into an issue with not having a soundcard with ztmonitor, but was able to work around that. Is there another way to create a dummy extension? If not, I have a couple old sip boxes not in use (they don't support callerid with sip firmware, so they're just doorstops now). Guess I could set one of those up. Thanks for the tip! On 1/25/06, David S. Madole [EMAIL PROTECTED] wrote: From: Joseph Tanner [EMAIL PROTECTED] Here's the short of it. I don't think so! I have an Asterisk 1.2.1 system setup to handle both personal and business calls. Now, the business callers will hear music while on hold, so the default MOH needs to play ... My solution was to have asterisk park the call for 15 seconds, send the callerid information to a YAC listener on my laptop and our TiVo, and I can pick up any phone and dial 4 to pick up the parked call. Works perfect, except parked callers hear music, not ringing. ... Is there a way to have a call parked, and have the caller hear the default ringing tones, and not have to mess around with MOH? How about you just don't answer the call in the first place? Ring it through to an extension that doesn't actually ring (maybe the console?) and then use pickup to answer from another phone in the same pickup group. This could be used for your music on hold case also by using the m modifier on the Dial command to the non-ringing extension. David ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Want to automatically park call and have callerhear ring tones
Sorry to reply twice, but thought it might make a difference. Regarding this part of your message: How about you just don't answer the call in the first place? Before it ever reaches the [asleep] context, Asterisk has already answered the call. Biz callers get the standard Press 1 for Sales, Press 2 for Support, etc. Personal calls get a message like You have reached the Tanner residence. Press 1 and your call will be connected.. Serves two purposes, first anyone misdialing will hang up after realizing they didn't want to call anyone named Tanner, and second most telemarketers we get use a machine to mass-dial, then tell you to hold for an important message, then sit there (that is if they don't hang up on your first). Haven't seen one smart enough to press 1 yet. Yes, I'm on the Do Not Call list, but I still get calls from DirecTV and others that we have accounts with. I think your suggestion will still work fine though, I just have to try to visualize it, then when I have some free time to test it (drove my wife crazy last night with all the yelling to see if the TiVo was showing the callerid info, I'll wait till the weekend when she's at work) I'll figure something out. Thanks again! On 1/25/06, David S. Madole [EMAIL PROTECTED] wrote: From: Joseph Tanner [EMAIL PROTECTED] Here's the short of it. I don't think so! I have an Asterisk 1.2.1 system setup to handle both personal and business calls. Now, the business callers will hear music while on hold, so the default MOH needs to play ... My solution was to have asterisk park the call for 15 seconds, send the callerid information to a YAC listener on my laptop and our TiVo, and I can pick up any phone and dial 4 to pick up the parked call. Works perfect, except parked callers hear music, not ringing. ... Is there a way to have a call parked, and have the caller hear the default ringing tones, and not have to mess around with MOH? How about you just don't answer the call in the first place? Ring it through to an extension that doesn't actually ring (maybe the console?) and then use pickup to answer from another phone in the same pickup group. This could be used for your music on hold case also by using the m modifier on the Dial command to the non-ringing extension. David ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cannot compile chan_bluetooth on Asterisk 1.2.1
Ok, I did get this going (somewhat), and in case someone else has the same issues I'll detail what I had to do. First, I was using the instructions at http://mundy.org/blog/index.php?p=79. They stated that [EMAIL PROTECTED] 2.2 already had all the rpms necessary for bluetooth and that I could skip the yum install step. I did, however, run the command anyways after a few failed attempts. There's an error in the rpm name, they tell you to install bluez-libs, the correct name is bluez-libs-devel (at least, that's what I needed to install). I still couldn't compile the newest cvs version of chan_bluetooth, so I again tried my trick of using the Makefile from an older version (which used .tmp to compile) and it worked! Once compiled, I installed and started up asterisk. I then received a message on my phone asking if I wanted to allow asterisk to connect, and then asked for a pin. This took a bit of figuring out, but I got passed that. In /etc/bluetooth/hcid.conf, there's a line that says pin_helper /usr/bin/bluepin; (it may have a different path, the important thing is the pin_helper part). Now backup the script in question, i.e. in my case mv /usr/bin/bluepin /usr/bin/bluepin.bak. Use your editor of choice to create a new file with the same name, and in it enter: #!/bin/sh -e echo PIN:1234 Replace the 1234 with whatever you want your pin to be. I don't know if this is necessary, but I also edited /etc/bluetooth/pin to read: 1234 Again, 1234 should be whatever you want your pin to be. I then stopped asterisk, stopped the bluetooth service, started the bluetooth service back up, started asterisk, then when my phone asked for a pin I put in 1234, and it worked! You may also need to make another edit to hcid.conf, under Local Device Class change it to read class 0x200404; or possibly class 0x700408;. This makes your bluetooth dongle look like a headset, and not a data device (I experienced some flakiness until I changed this). Now, I edited /etc/asterisk/bluetooth.conf appropriately (changed the channel for the phone to 7, it's a Motorola V551), started it all up, made some test calls and...no audio! The cellphone works great otherwise. It'll connect, stay connected as long as I want it to, and when I hang up the asterisk extension the cellphone will disconnect too. Too bad I didn't realize 611 was a free call until after I made a lot of test calls (it's a prepaid phone). I did call our home number directly to see if maybe I just had a one-way audio problem, but nobody could hear a thing on either end. I will continue to troubleshoot this before I ask another question about it, but it's not looking good. BTW, the usb dongle I'm using is a Linksys USBBT100. ___ --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] Cannot compile chan_bluetooth on Asterisk 1.2.1
The short of it: I am unable to compile chan_bluetooth on Asterisk 1.2.1 on CentOS 4.2. I installed using the [EMAIL PROTECTED] 2.2 iso. Server is a plain Celeron 2.93GHz box. Asterisk source is in /usr/src/asterisk, newest chan_bluetooth source is in /usr/src/asterisk-test/bluetooth/chan_bluetooth (I have two older versions in other directories). Steps taken: Followed the instructions here to a T: http://www.crazygreek.co.uk/content/chan_bluetooth. Basically, edit /usr/src/asterisk/channels/Makefile adding chan_bluetooth.so to CHANNEL_LIBS, and at the very bottom adding include /usr/src/asterisk-test/bluetooth/chan_bluetooth/Makefile. First tried the version by David Woodhouse, exact command used to download was cvs -d :pserver:anoncvs at cvs.infradead.org:/home/cvs co chan_bluetooth. Also tried the version at http://www.crazygreek.co.uk/data/pages/chan_bluetooth/latest.tar.gz. Lastly, wanted to try a newer version of Theo's code on the SVN server, which was down. Google helped me find r40 at http://rock.inode.at/ROCK-2.1/c/chan_bluetooth-r40.tar.bz2. Using the newest version (by David Woodhouse) gives me this error: make[1]: Entering directory `/usr/src/asterisk/channels' gcc -shared -Xlinker -x -o chan_bluetooth.so /usr/src/asterisk-test/bluetooth/chan_bluetooth/chan_bluetooth.o -lbluetooth gcc: /usr/src/asterisk-test/bluetooth/chan_bluetooth/chan_bluetooth.o: No such file or directory make[1]: *** [chan_bluetooth.so] Error 1 make[1]: Leaving directory `/usr/src/asterisk/channels' make: *** [subdirs] Error 1 Using an older version will at least try to compile, but giving many errors. I found by using the Makefile from an older version with the newest, it also tries to compile but with errors as well. The only difference I see in the Makefile is using a .tmp directory in the chan_bluetooth directory in order to compile. Here's the end of the error using that Makefile (I'd post the entire error, but it fills up the buffer and I can't copy it all, let me know if you need more than I posted): /usr/src/asterisk-test/bluetooth/chan_bluetooth/chan_bluetooth.c: In function `load_module': /usr/src/asterisk-test/bluetooth/chan_bluetooth/chan_bluetooth.c:3210: error: `sdp_session_t' undeclared (first use in this function) /usr/src/asterisk-test/bluetooth/chan_bluetooth/chan_bluetooth.c:3210: error: `sess' undeclared (first use in this function) /usr/src/asterisk-test/bluetooth/chan_bluetooth/chan_bluetooth.c:3221: warning: implicit declaration of function `hci_open_dev' /usr/src/asterisk-test/bluetooth/chan_bluetooth/chan_bluetooth.c:3227: warning: implicit declaration of function `hci_read_voice_setting' /usr/src/asterisk-test/bluetooth/chan_bluetooth/chan_bluetooth.c:3228: warning: implicit declaration of function `htobs' /usr/src/asterisk-test/bluetooth/chan_bluetooth/chan_bluetooth.c:3244: warning: implicit declaration of function `hci_devba' /usr/src/asterisk-test/bluetooth/chan_bluetooth/chan_bluetooth.c:3248: error: `BDADDR_LOCAL' undeclared (first use in this function) /usr/src/asterisk-test/bluetooth/chan_bluetooth/chan_bluetooth.c:3248: error: `SDP_RETRY_IF_BUSY' undeclared (first use in this function) make[1]: *** [/usr/src/asterisk-test/bluetooth/chan_bluetooth/.tmp/chan_bluetooth.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk/channels' make: *** [subdirs] Error 1 [EMAIL PROTECTED] asterisk]# I have also tried a few things, moving the include statement up in the Makefile, adding #define ASTERISK_VERSION_NUM 010201 to the top of chan_bluetooth.c (also used 010200, and 00). In /usr/src/asterisk/include/asterisk/version.h, it kept being set to 00, I had to edit the Makefile in /usr/src/asterisk to force it to 010201 (after trying it with the 00 value first, of course). When compiling Asterisk, I will do a make clean then make. When making minor changes I would just do a make clean in /usr/src/asterisk/channels then a make in /usr/src/asterisk. The two errors above were after doing a complete make clean in /usr/src/asterisk, then a make. Hopefully I gave enough information, if I missed anything let me know. Thank you. ___ --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] Welltech FXO: initial tests
I have a Welltech 3502 (2 FXS ports) and callerid will not work in SIP mode. I contacted Welltech support and they informed me that callerid is only working with the H.323 firmware. Once I flashed it with the H.323 firmware and figured out how to get it to work with asterisk, callerid did indeed start working. Joseph Tanner [EMAIL PROTECTED] Message: 15 Date: Fri, 02 Apr 2004 11:13:35 -0500 From: Jorge Mendoza [EMAIL PROTECTED] Organization: TCC S.A. To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Welltech FXO: initial tests Reply-To: [EMAIL PROTECTED] Hi, After a long way of problems (shipping, customs, etc) finally I got Welltech working. Here below my comments. - The documentation is poor and have errors - The web configuration is not complete. However is useful for the basic configuration parameters. The command line is necessary for modify all parameters. - The software upgrade is easy. Initially the gw came with H323, we upgrade to SIP. - We have tested only one port, it works well, audio quality is good (alaw). - Outgoing and incoming calls are working ok. - The Caller ID (from PSTN side) does not work - Answer supervision (reversal polarity detection) seems to work fine. This feature is very important to us, is the first time that we found this feature in a analog CO trunk. In a test application where we play a voice message to the called user, the message start to play just after answer. Tested with wire phone and cell phones. - Disconnect tone seems reliable (although the default configuration was not adjusted). We have done dozen of test in order to get the gw working. During the tests two issues came up, they need further analysis and tests: - Two times a UDP packages loop between the gw and * saturated the bandwidth after a hung up. Rebooting the gw does not stop the loop. Even with the gw turn off, * was sending the packages.Only rebooting * turn the system normal. - The gw port stay locked after a hung up. Apparently due to a no detection of the disconnect tone (in this case the tests were carried out with a PABX without disconnect tone). But the * user (SIP) was hung up and it seems that there are not a release timer. We will continue the tests and test the Welltech technical support as well (no required until now). Jorge --__--__-- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users End of Asterisk-Users Digest ___ 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] External and internal SIP do not work together with nat
Here's the main problem I've run into. I'm trying to use FWD with Asterisk, and am behind a nat device (dsl modem with nat built-in, no way to bind the IP directly to a server/PC). I also have a SIP gateway, a Welltech 3502 (it goes by many other names, always see it with the 3502 model number). I am unable to get Asterisk to work with both FWD and the 3502 at the same time. It will work perfectly with one or the other, just not both. Since I'm using NAT, in my sip.conf I have to specify the external IP to get FWD to work. I also have a dynamic IP if that matters, but I found that using a domain name in place of an IP works (i.e., I use externip = myserver.gotdns.com and it works fine). When I comment this out, FWD stops working but the 3502 starts working fine. I ran sip debug on the Asterisk console, and it appears that with the externip value set, it's returning that IP to the 3502 instead of the internal IP. If I could get it to return the internal IP for the 3502, and the external IP for FWD, I think it'd work. Below is my sip.conf, with a few minor changes (edited the dynamic dns domain and callerid numbers, and took out actual FWD username/password). This is the current working configuration; I comment out externip and the 3502 gateway works, and if I uncomment it FWD works. For kicks I have set nat=yes and nat=no for both FWD and the gateway ports (had it set to yes for fwd and no for the gateway, then reversed, then both set to yes, and both set to no...no change). I also changed canreinvite to yes for the gateway ports, with no change. ; ; SIP Configuration for Asterisk ; [general] port = 5061 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to externip = mydomain.gotdns.com localnet = 192.168.1.18 ; Internal NETWORK address localmask = 255.255.255.0 ; Internal netmask context = biz ; Default for incoming calls register = 255:[EMAIL PROTECTED] [fwd] type=friend secret=mypassword username=55 host=fwd.pulver.com dtmfmode=inband context=biz nat=yes canreinvite=no callerid=Business Line (800) 555-1212 [1001] type=friend username=1001 host=dynamic context=main canreinvite=no txgain=3.5 rxgain=2.5 nat=no [1002] type=friend username=1002 host=dynamic context=main canreinvite=no txgain=3.5 rxgain=2.5 nat=no ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Extensions do not display CallerID
How do I configure Asterisk to send CallerID info to an extension? I'm using three Quicknet Phonejack ISA cards with cordless phones. The phones receive CallerID info fine when plugged directly into the incoming lines. Asterisk is recognizing the correct CallerID info according to /var/log/asterisk/cdr-csv/Master.csv (CallerID via PSTN is hit and miss, but when dialing an extension directly it's always 100%). I simply cannot get CallerID to display on the phones, when they're plugged into the Quicknet Phonejack cards. I hope this isn't a stupid question, but I've searched google for about five hours and while I've learned a lot about Asterisk and CID, I haven't found anything relating to this. ___ 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] Cannot call extensions or make outgoing calls
I compiled Asterisk last night from the stable cvs branch. I have two X101P cards, and three Quicknet Phonejack ISA cards. Asterisk is able to receive calls on both lines, and all three extensions are at least partly working. Here's basically what it's doing: I can pick up any extension and get a psuedo dial-tone. I can call ext 500 for the demo, or 600 for the echo test. Both work fine (I hear myself on the echo test, so audio's working both ways). When I dial an actual extension (whether from another extension or after calling into asterisk), the phone on that extension will ring. When I pick up the extension that's ringing though, both lines give a fast-busy and I get this error in the CLI: -- Called phone2 -- Phone/phone2 is ringing -- Phone/phone2 answered Zap/2-1 Mar 12 15:56:14 WARNING[245776]: chan_phone.c:417 phone_read: Error reading: Input/output error -- Hungup 'Phone/phone2' == Spawn extension (default, 421, 1) exited non-zero on 'Zap/2-1' -- Hungup 'Zap/2-1' That example was when I dialed into asterisk from an outside line, then called one of the extensions; but the same phone_read: Error reading: Input/output error error comes up when dialing from extension to extension. Also, it throws up a similar error when I try to make an outgoing call. Here's what I get: -- Executing Dial(Phone/phone2, Zap/1/1611212) in new stack -- Called 1/1611212 -- Zap/1-1 answered Phone/phone2 Mar 12 16:13:03 WARNING[245776]: chan_phone.c:417 phone_read: Error reading: Input/output error -- Hungup 'Zap/1-1' == Spawn extension (default, 99317216871, 1) exited non-zero on 'Phone/phone2' -- Hungup 'Phone/phone2' I edited out the actual number and replaced it with 1611212. The number I'm dialing on the extension is 9611212 (in the dialing rules I have it add a 1). FYI, the number I'm calling never rings. Hopefully it's just something stupid I'm doing on my end, but I've gone over all the config files and don't see what I could have done wrong, that would give these results. ___ 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] Cannot call extensions or make outgoing calls
Well, I did some more playing around and uncommented the format=slinear line in phone.conf. This has resolved my problems. I hope this helps someone else out. I compiled Asterisk last night from the stable cvs branch. I have two X101P cards, and three Quicknet Phonejack ISA cards. Asterisk is able to receive calls on both lines, and all three extensions are at least partly working. Here's basically what it's doing: I can pick up any extension and get a psuedo dial-tone. I can call ext 500 for the demo, or 600 for the echo test. Both work fine (I hear myself on the echo test, so audio's working both ways). When I dial an actual extension (whether from another extension or after calling into asterisk), the phone on that extension will ring. When I pick up the extension that's ringing though, both lines give a fast-busy and I get this error in the CLI: -- Called phone2 -- Phone/phone2 is ringing -- Phone/phone2 answered Zap/2-1 Mar 12 15:56:14 WARNING[245776]: chan_phone.c:417 phone_read: Error reading: Input/output error -- Hungup 'Phone/phone2' == Spawn extension (default, 421, 1) exited non-zero on 'Zap/2-1' -- Hungup 'Zap/2-1' That example was when I dialed into asterisk from an outside line, then called one of the extensions; but the same phone_read: Error reading: Input/output error error comes up when dialing from extension to extension. Also, it throws up a similar error when I try to make an outgoing call. Here's what I get: -- Executing Dial(Phone/phone2, Zap/1/1611212) in new stack -- Called 1/1611212 -- Zap/1-1 answered Phone/phone2 Mar 12 16:13:03 WARNING[245776]: chan_phone.c:417 phone_read: Error reading: Input/output error -- Hungup 'Zap/1-1' == Spawn extension (default, 99317216871, 1) exited non-zero on 'Phone/phone2' -- Hungup 'Phone/phone2' I edited out the actual number and replaced it with 1611212. The number I'm dialing on the extension is 9611212 (in the dialing rules I have it add a 1). FYI, the number I'm calling never rings. Hopefully it's just something stupid I'm doing on my end, but I've gone over all the config files and don't see what I could have done wrong, that would give these results. ___ 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 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
