Re: [asterisk-users] Fring and Asterisk
2009/1/20 D Tucny d...@tucny.com 2009/1/20 Olivier oza-4...@myamail.com Hi, Is anyone using Fring as a SIP client to an Asterisk server ? Yes, testing it... A prospective customer of mine is asking to integrate its iphones with an Asterisk server and after googling, I still have some unanswered questions : 1. Which codecs are available when calling from fring ? I believe it offers GSM, ilbc, ulaw and alaw... 2. Is it easy and natural to change your presence status (available, busy, ...) with Fring or will users prefer to use another software (bundled with iPhones) or to do nothing at all ? 3. Is it possible to add custom presence status in Fring client ? On the version running on my nokia phone, there seems to be no obvious way to change status... 4. Is it possible and recommended to limit Fring usage to WiFi presence ? The options on the version I have are: Wifi first, 3G/GPRS first, Wifi only, 3G/GPRS only, Always ask... So, it is possible... It does work over GPRS, but quality is noticably lower than over Wifi... 5. Would fring replies to Qualify messages ? Yes One thing to note about fring, the device establishes a connection using fring's proprietary protocols to fring servers, fring then establishes SIP connections from those servers... So, even if connected to the office Wifi connection, you could experience connectivity issues or high latency as a result of a potentially long path involved for the traffic to travel... d ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Thanks for these helpful answers. Something is bothering me about Fring : what is Fing's business model ? Software is free. Connections are made using Fring servers and seem to be free. So, where's the benefit for Fring ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fring and Asterisk
One thing to note about fring, the device establishes a connection using fring's proprietary protocols to fring servers, fring then establishes SIP connections from those servers... So, even if connected to the office Wifi connection, you could experience connectivity issues or high latency as a result of a potentially long path involved for the traffic to travel... d So, when in the office, whenever I'm calling someone using SIP and WiFi, for both signalling and media, data would travel from mobile phone to WiFi access point, then to switch, router, Fring server (at the other end of the world), then back to my routeur, and Asterisk server, right ? If think I'll try to compare this with a XMPP/Jingle-enabled client that can be installed in mobile phone (I don't know if such software exist) ... ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Skype beta news ?
Hi, Has anyone any return to share about Skype-Digium beta program ? I would be very curious to know how things are going on this. Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fring and Asterisk
2009/1/20 Olivier oza-4...@myamail.com One thing to note about fring, the device establishes a connection using fring's proprietary protocols to fring servers, fring then establishes SIP connections from those servers... So, even if connected to the office Wifi connection, you could experience connectivity issues or high latency as a result of a potentially long path involved for the traffic to travel... d So, when in the office, whenever I'm calling someone using SIP and WiFi, for both signalling and media, data would travel from mobile phone to WiFi access point, then to switch, router, Fring server (at the other end of the world), then back to my routeur, and Asterisk server, right ? If think I'll try to compare this with a XMPP/Jingle-enabled client that can be installed in mobile phone (I don't know if such software exist) ... GTalk seems to fill the bill of requirements, though, I don't think it's available on Nokia mobile phones .. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-users Digest, Vol 54, Issue 53
Hi Steve; Do u mean by the Iaxy2 is that IAX digium gateway adaptor? If yes, then it has a codec limitation and it does not take ddns name (it needs IP address), also it is gateway and not IP Phone. Or u mean something else? Do u have a link for it so I can see it? Regards Bilal Anyone knows an IAX IP Phone works fine and tested? How about IAX2 adapter from digium? I've been uing it and it works very well. Wow, that has NOT been my experience, though it has been a few years (2005) since I used them. The ones I purchased were first of all expensive. They overheated and froze up often. Only a single port. No dual ethernet option. Provisioning is a PITA. Codec support was minimal. I thought at the time that being IAX I wouldn't have to worry about NAT issues and that was worth the extra difficulties, but I have been using Linksys PAP2Ts ever since and never looked back. And it has been over a year since I had any NAT issue to deal with, though have now installed them in hundreds of different configurations. Perhaps these things have been rectified since... I've had an Iaxy2 (s101i) for several years. It's always worked fine for me. It does generate a very slight amount of heat. Just enough so you know it's plugged in -- as if the overly-bright blue registration LED wasn't a clue. The original iaxprov command line tool was a bit of a bother, but the iaxprov Asterisk command is better since it uses a centralized iaxprov.conf file to provision the devices. I prefer devices that request (via TFTP) configuration, but once configured you're done. It gets my vote for a just works device and it's great to travel with as long as you remember to use a transformer instead of an adapter in countries (England) that insist on delivering excessive voltage :) Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium TE220 supported protocol
Laurent a écrit : Le 19.01.2009 08:50, Benoit a écrit : Laurent a écrit : Well, the telcos techs said a straight cable should do the trick, but since i didn't get any isdn link up with the straight, i built a crossover like what you described, with no luck either. Did you check (like with a multimeter or something similar) the connectivity of your cable ? the first E1 crossover cable I made had a problem (entirely my own fault) and I thought it didn't work. The way I checked was by connecting the two ports of the Digium card with the crossover cable, and when I saw the LEDs turn red I knew my cable was not good. Then of course perhaps your gateway has not been setup yet, as you mentionned. Good luck! By the way, do you/someone know if their is a need for termination resistor like in this BRI crossover cable that was posted a few time ago: http://lists.digium.com/pipermail/asterisk-users/2006-August/162227.html http://lists.digium.com/pipermail/asterisk-users/attachments/20060809/113c5579/isdn-bri-crossover.gif ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] the FXS ports of Digium and damaging if connected to Tel Line
Hi All; I am facing a problem that always the users confused and connect the telephone line coming from the telephone service provider to the FXS port and cause it to be damaged, specially if the card was 2 fxs and 2 fxo, so they make mistake and connect the line to fxs while it should be connected to fxo. What is the solution for this disaster? Regards Bilal ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to cancel new recorded message from voicemail menu?
Philipp Kempgen schrieb: Klaus Darilion schrieb: If a user has recorded a new voicemail message (e.g. unavailable message) then it is prompted with 3 choices. 1. accept recording 2. listen to the recorded message 3. rerecord the message Isn't it possible to cancel the recording? You could hang up. But users might not be aware of this simple solution. Stupid me :-) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium TE220 supported protocol
Benoit a écrit : Laurent a écrit : Did you check (like with a multimeter or something similar) the connectivity of your cable ? the first E1 crossover cable I made had a problem (entirely my own fault) and I thought it didn't work. The way I checked was by connecting the two ports of the Digium card with the crossover cable, and when I saw the LEDs turn red I knew my cable was not good. Then of course perhaps your gateway has not been setup yet, as you mentionned. Good luck! By the way, do you/someone know if their is a need for termination resistor like in this BRI crossover cable that was posted a few time ago: http://lists.digium.com/pipermail/asterisk-users/2006-August/162227.html http://lists.digium.com/pipermail/asterisk-users/attachments/20060809/113c5579/isdn-bri-crossover.gif ? Also, while now the ISDN link is up, and we are able to make some calls thru the new line, we have two error messages going on the asterisk console: When asterisk start, or when the PRI reset, every channel trigger this message: [Jan 19 21:01:17] ERROR[13512]: chan_dahdi.c:8413 dahdi_pri_error: !! Unexpected Channel selection 3 -- B-channel 0/1 successfully restarted on span 2 [Jan 19 21:01:17] ERROR[13512]: chan_dahdi.c:8413 dahdi_pri_error: !! Unexpected Channel selection 3 -- B-channel 0/2 successfully restarted on span 2 ... And this one: Unable to handle return result on switchtype 5! happen on a regular basis, i'm not sure but it look like jsute before the outgoing call is bridged to the internal extension. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 T38 to G711 transcoding is this possible?
What you need is a so called T38Gateway application. there is a patch o the tracker which you might want to try: http://bugs.digium.com/view.php?id=13405 klaus Steve Gladden schrieb: The scenario we have is fax send/recieve software that ONLY talks T38 and an asterisk box. We have ITSP providers that do NOT talk T38 but G711 only. Does asterisk have the capability to take the T38 call from an ATA or T38 software then bridge/transcode it and do G711 out to the PSTN providers? If not is there another product PAID or FREE software or hardware that can do this easily and reliably? Thanks very much! Steve Gladden ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR problems -- two call legs create only one CDR. Using ForkCDR() not even working.
hello When I bridge an incoming and outgoing call (attempting to simulate call-forwarding) I'm only getting one CDR -- that of the outgoing call. A (PSTN) calls B (residing on Asterisk) and the Asterisk calls C (cell phone on PSTN) and bridges the call. The only CDR created is from B to C. I have even tried using Answer() and ForkCDR() to get two CDRs, but to no avail. I am starting to wonder if there's a bug in the CDR generation in general, because I set up an extension to do only that: exten = 5822558,1,Answer() exten = 5822558,n,ForkCDR() exten = 5822558,n,Playback(tt-monkeys) exten = 5822558,n,Hangup() I have the same problem, try older version like 1.4.21 for example.. And look at: http://bugs.digium.com/view.php?id=13797, there is some patch that might help. For me the only possible option to play with forkcdr as I expected is using older versions of Asterisk best regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] the FXS ports of Digium and damaging if connected to Tel Line
On 20 Jan 2009, at 10:18, bilal ghayyad wrote: Hi All; I am facing a problem that always the users confused and connect the telephone line coming from the telephone service provider to the FXS port and cause it to be damaged, specially if the card was 2 fxs and 2 fxo, so they make mistake and connect the line to fxs while it should be connected to fxo. What is the solution for this disaster? I have done this (very briefly) by accident. It survived. Is it definitely dead? Do the other ports work? If its just that port thats toasted then you might get away with just replacing that module. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP DTMF problem with SNOM
Hi! I have two identical SIP accounts on Asterisk 1.4.22. One account is registered with eyebeam, the other one is registered with a SNOM phone. When using the eyebeam client DMTF detection works fine, when using the SNOM phone many digits are missing in the DTMF detection. I analyzed with wireshark and both phones uses RFC 2833 and the trace looks pretty the same. Also the rtp debug log looks fine (see below). What could be the reason? thanks klaus trace: I have entered 1234#, but voicemail received as secret just 123. Got RTP packet from83.136.33.3:64118 (type 00, seq 042765, ts 4066332168, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042766, ts 4066332328, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042770, ts 4066332968, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042771, ts 4066333128, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042774, ts 4066333608, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042775, ts 4066333768, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042777, ts 4066334088, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042778, ts 4066334248, len 000160) Got RTP packet from83.136.33.3:64118 (type 101, seq 042780, ts 4066334648, len 04) Got RTP RFC2833 from 83.136.33.3:64118 (type 101, seq 042780, ts 4066334648, len 04, mark 0, event 0001, end 0, duration 00320) Got RTP packet from83.136.33.3:64118 (type 101, seq 042781, ts 4066334648, len 04) Got RTP RFC2833 from 83.136.33.3:64118 (type 101, seq 042781, ts 4066334648, len 04, mark 0, event 0001, end 0, duration 00480) Got RTP packet from83.136.33.3:64118 (type 101, seq 042782, ts 4066334648, len 04) Got RTP RFC2833 from 83.136.33.3:64118 (type 101, seq 042782, ts 4066334648, len 04, mark 0, event 0001, end 0, duration 00640) Got RTP packet from83.136.33.3:64118 (type 101, seq 042785, ts 4066334648, len 04) Got RTP RFC2833 from 83.136.33.3:64118 (type 101, seq 042785, ts 4066334648, len 04, mark 0, event 0001, end 0, duration 01120) Got RTP packet from83.136.33.3:64118 (type 101, seq 042786, ts 4066334648, len 04) Got RTP RFC2833 from 83.136.33.3:64118 (type 101, seq 042786, ts 4066334648, len 04, mark 0, event 0001, end 0, duration 01280) Got RTP packet from83.136.33.3:64118 (type 101, seq 042788, ts 4066334648, len 04) Got RTP RFC2833 from 83.136.33.3:64118 (type 101, seq 042788, ts 4066334648, len 04, mark 0, event 0001, end 0, duration 01600) Got RTP packet from83.136.33.3:64118 (type 101, seq 042789, ts 4066334648, len 04) Got RTP RFC2833 from 83.136.33.3:64118 (type 101, seq 042789, ts 4066334648, len 04, mark 0, event 0001, end 1, duration 01760) Got RTP packet from83.136.33.3:64118 (type 101, seq 042789, ts 4066334648, len 04) Got RTP RFC2833 from 83.136.33.3:64118 (type 101, seq 042789, ts 4066334648, len 04, mark 0, event 0001, end 1, duration 01760) Got RTP packet from83.136.33.3:64118 (type 101, seq 042789, ts 4066334648, len 04) Got RTP RFC2833 from 83.136.33.3:64118 (type 101, seq 042789, ts 4066334648, len 04, mark 0, event 0001, end 1, duration 01760) Got RTP packet from83.136.33.3:64118 (type 00, seq 042790, ts 4066336648, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042791, ts 4066336808, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042792, ts 4066336968, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042793, ts 4066337128, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042794, ts 4066337288, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042795, ts 4066337448, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042798, ts 4066337928, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042801, ts 4066338408, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042802, ts 4066338568, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042805, ts 4066339048, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042806, ts 4066339208, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042809, ts 4066339688, len 000160) Got RTP packet from83.136.33.3:64118 (type 101, seq 042812, ts 4066340168, len 04) Got RTP RFC2833 from 83.136.33.3:64118 (type 101, seq 042812, ts 4066340168, len 04, mark 0, event 0002, end 0, duration 00480) Got RTP packet from83.136.33.3:64118 (type 101, seq 042813, ts 4066340168, len 04) Got RTP RFC2833 from 83.136.33.3:64118 (type 101, seq 042813, ts 4066340168, len 04, mark 0, event 0002, end 0, duration 00640) Got RTP packet from83.136.33.3:64118 (type 101, seq
Re: [asterisk-users] SIP DTMF problem with SNOM
How are you testing DTMF detection with the Snom UA? Klaus Darilion wrote: Hi! I have two identical SIP accounts on Asterisk 1.4.22. One account is registered with eyebeam, the other one is registered with a SNOM phone. When using the eyebeam client DMTF detection works fine, when using the SNOM phone many digits are missing in the DTMF detection. I analyzed with wireshark and both phones uses RFC 2833 and the trace looks pretty the same. Also the rtp debug log looks fine (see below). What could be the reason? thanks klaus trace: I have entered 1234#, but voicemail received as secret just 123. Got RTP packet from83.136.33.3:64118 (type 00, seq 042765, ts 4066332168, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042766, ts 4066332328, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042770, ts 4066332968, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042771, ts 4066333128, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042774, ts 4066333608, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042775, ts 4066333768, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042777, ts 4066334088, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042778, ts 4066334248, len 000160) Got RTP packet from83.136.33.3:64118 (type 101, seq 042780, ts 4066334648, len 04) Got RTP RFC2833 from 83.136.33.3:64118 (type 101, seq 042780, ts 4066334648, len 04, mark 0, event 0001, end 0, duration 00320) Got RTP packet from83.136.33.3:64118 (type 101, seq 042781, ts 4066334648, len 04) Got RTP RFC2833 from 83.136.33.3:64118 (type 101, seq 042781, ts 4066334648, len 04, mark 0, event 0001, end 0, duration 00480) Got RTP packet from83.136.33.3:64118 (type 101, seq 042782, ts 4066334648, len 04) Got RTP RFC2833 from 83.136.33.3:64118 (type 101, seq 042782, ts 4066334648, len 04, mark 0, event 0001, end 0, duration 00640) Got RTP packet from83.136.33.3:64118 (type 101, seq 042785, ts 4066334648, len 04) Got RTP RFC2833 from 83.136.33.3:64118 (type 101, seq 042785, ts 4066334648, len 04, mark 0, event 0001, end 0, duration 01120) Got RTP packet from83.136.33.3:64118 (type 101, seq 042786, ts 4066334648, len 04) Got RTP RFC2833 from 83.136.33.3:64118 (type 101, seq 042786, ts 4066334648, len 04, mark 0, event 0001, end 0, duration 01280) Got RTP packet from83.136.33.3:64118 (type 101, seq 042788, ts 4066334648, len 04) Got RTP RFC2833 from 83.136.33.3:64118 (type 101, seq 042788, ts 4066334648, len 04, mark 0, event 0001, end 0, duration 01600) Got RTP packet from83.136.33.3:64118 (type 101, seq 042789, ts 4066334648, len 04) Got RTP RFC2833 from 83.136.33.3:64118 (type 101, seq 042789, ts 4066334648, len 04, mark 0, event 0001, end 1, duration 01760) Got RTP packet from83.136.33.3:64118 (type 101, seq 042789, ts 4066334648, len 04) Got RTP RFC2833 from 83.136.33.3:64118 (type 101, seq 042789, ts 4066334648, len 04, mark 0, event 0001, end 1, duration 01760) Got RTP packet from83.136.33.3:64118 (type 101, seq 042789, ts 4066334648, len 04) Got RTP RFC2833 from 83.136.33.3:64118 (type 101, seq 042789, ts 4066334648, len 04, mark 0, event 0001, end 1, duration 01760) Got RTP packet from83.136.33.3:64118 (type 00, seq 042790, ts 4066336648, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042791, ts 4066336808, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042792, ts 4066336968, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042793, ts 4066337128, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042794, ts 4066337288, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042795, ts 4066337448, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042798, ts 4066337928, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042801, ts 4066338408, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042802, ts 4066338568, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042805, ts 4066339048, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042806, ts 4066339208, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042809, ts 4066339688, len 000160) Got RTP packet from83.136.33.3:64118 (type 101, seq 042812, ts 4066340168, len 04) Got RTP RFC2833 from 83.136.33.3:64118 (type 101, seq 042812, ts 4066340168, len 04, mark 0, event 0002, end 0, duration 00480) Got RTP packet from83.136.33.3:64118 (type 101, seq 042813, ts 4066340168, len 04) Got RTP RFC2833
[asterisk-users] Forwarding calls and trasfer calls
Hi How do i set up so that everyone can dial, for example *21* to forward all calls to a cellphone or another extension and how do I enable so that cals can be transferd between extentions. I use asterisk 1.6 and have my phones in unistim.conf and my extensions in extensions.conf. Regards /ralf Ralf Träskman, IT AdLibris AB, Odengatan 106, 113 22 Stockholm, Sweden Dir: +46-(0)707548074, vxl: +46-(0)85460 60 00, fax: +46-(0)85460 60 99 r...@adlibris.commailto:r...@adlibris.com www.adlibris.comhttp://www.adlibris.com/ P Please consider the environment before printing this e-mail ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fring and Asterisk
On Tue, 20 Jan 2009, Olivier wrote: GTalk seems to fill the bill of requirements, though, I don't think it's available on Nokia mobile phones .. My Nokia mobile phone has a SIP client built-in which uses Wi-Fi... And while it's not perfect, it's actually very usable and works well with asterisk. Upgrade the phone? Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fring and Asterisk
2009/1/20 Gordon Henderson gordon+aster...@drogon.netgordon%2baster...@drogon.net On Tue, 20 Jan 2009, Olivier wrote: GTalk seems to fill the bill of requirements, though, I don't think it's available on Nokia mobile phones .. My Nokia mobile phone has a SIP client built-in which uses Wi-Fi... And while it's not perfect, it's actually very usable and works well with asterisk. Upgrade the phone? I agree that native Nokia SIP client is very usable and should be the preferred way to call over a WiFi connection. I mentioned GTalk as someone mentioned Fring on Nokia. Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Siemens S685IP registration problems
Hi folks, I wonder if any of you out there are using Siemens S685IP base station(s) (with S68H handsets) on Asterisk and experiencing problems with SIP registrations where the SIP extensions do not ring and peers become unreachable after a period of time. Symptoms are rather sporadic, but as described, SIP extensions being unreachable from Asterisk perspective. Also experienced 'not possible' messages trying to dial using the handsets. In Asterisk - 'sip show peers' shows the SIP device status as 'UNKNOWN' and console output says chan_sip.c Peer 'xxx' is unreachable. (as one would expect if it can't see it!!) So I'm thinking it's a problem with the base.. or.. some issue with qualifications and possibly the base station not responding (guessing here). I'm finding that the Siemens web GUI reports messages similar to 'server not accessible' or 'registration failed'. These messages appear randomly throughout the day following successful previous registration. Have enabled 'sip set debug ip xxx' for one of the bases and on the Siemens web GUI disabling the SIP account and re-enabling it doesn't work generally, and no SIP messages are being directed from the base to Asterisk. Leads me to think it's a base/firmware issue. Some times the phones are contactable for a day without fault, other times they're problematic at random intervals. It's not always all of the SIP accounts assigned on the Siemens base, sometimes it's just one account, other times it's all accounts. (What a horrible situation to debug/fault find! Glad my Aastra's are reliable!) The only resolve I've found which is rather unacceptable is to reboot the Siemens base station. Upon doing so, the base re-registers all of the accounts to the Asterisk server and calls to/from handsets work for 'a period of time'... My setup is as follows: - Asterisk 1.4.22 - Base 1 has 3x handsets and 3x SIP accounts (providers) and the SIP accounts are individually assigned to each handset. - Base 2 has 5 handsets and 6 SIP accounts. 5 SIP accounts for each handset, then the 6th SIP account is a 'group' extension which rings all of the phones on the base station. MWI for VM is also used and works. Call waiting disabled on handsets. - Both bases on latest available as of Jan 09 - 0214 / 043.00 The bases are set up with static IPs, info services off, etc. Am not using SIP domains, no NAT, all communicating on a LAN on same network so no routing or latency issues. Registrar server defined and refresh time set to 180 seconds (Siemens default). SIP.conf has nat=no, qualify=yes. host= is currently dynamic.. maybe I should set this to the IP of the base as they're using static IPs, but reading the specs of this setting describe set to 'dynamic' if the phone should register itself... hmm. I’ve seen similar posts from other users on the exact same subject. Sources as follows: Voip-Info http://www.voip-info.org/wiki/view/Siemens+Gigaset+S675IP (see 'Discussions' tab). The Open Sourcerer: http://www.theopensourcerer.com/2008/04/27/siemens-gigaset-685ip-phones/comment-page-3/ Siemens Forums: http://www.siemens-gigaset-forum.com/de/posts/list/13788.page Siemens Customer care: http://www.siemens-gigaset-forum.com/de/posts/list/13788.page (people with aged open support calls) Siemens support via phone were rather unhelpful and didn't grasp the technicalities of the issue I was conveying so drew a blank (have I checked my router.. hmm!) I'm guessing this is a firmware issue but intrigued to know if others are experiencing the same. Anyone else experiencing similar problems? Or indeed successes with a similar setup? Can anyone recommend a stable working DECT SIP phone for enterprise use with Asterisk? (The Snom M3 looks good but read about issues with transfers which concern me) May have to resort to ditching the S685s and go with Aastra desk sets all round - shame to lose the flexibility of cordless though. Many thanks in advance. _ Cut through the jargon: find a PC for your needs. http://clk.atdmt.com/UKM/go/130777504/direct/01/___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] the FXS ports of Digium and damaging if connected to Tel Line
On Tuesday, January 20, 2009, bilal ghayyad wrote: What is the solution for this disaster? I live in UK, where we don't use RJ11 for telephones and so need to use adapters, which I just leave hanging out of the FXO ports. With the adapters in place, it's difficult to plug the phones into the wrong ports. I've also stuck a large label on the computer cabinet to show which ports are for what. For info, I have a TDM400P. I suspect that the label idea might help but the other would depend on whether your country uses RJ11 for standard phone connections. HTH, -- Geoff ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Called's channel
Hi,
I have a question...
With the variable ${CHANNEL} I can get the channel whose made the call, or the
caller. How can I get the channel of the called side?
Veja quais são os assuntos do momento no Yahoo! +Buscados
http://br.maisbuscados.yahoo.com
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Re: [asterisk-users] Forwarding calls and trasfer calls
features.conf 2009/1/20 Ralf Träskman r...@adlibris.com Hi How do i set up so that everyone can dial, for example **21** to forward all calls to a cellphone or another extension and how do I enable so that cals can be transferd between extentions. I use asterisk 1.6 and have my phones in unistim.conf and my extensions in extensions.conf. Regards /ralf Ralf Träskman, IT AdLibris AB, Odengatan 106, 113 22 Stockholm, Sweden Dir: +46-(0)707548074, vxl: +46-(0)85460 60 00, fax: +46-(0)85460 60 99 r...@adlibris.com www.adlibris.com P *Please consider the environment before printing this e-mail* ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Description of Zaptel/DAHDI E1 alarms
Message: 18 Date: Mon, 19 Jan 2009 19:56:14 +0200 From: Tzafrir Cohen tzafrir.co...@xorcom.com Subject: Re: [asterisk-users] Description of Zaptel/DAHDI E1 alarms To: asterisk-users@lists.digium.com Message-ID: On Mon, 2009-01-19 at 11:10 +0100, Lukas Rypl wrote: I am missing any description of zaptel/DAHDI alarms. The TE200 series user manual contains only a description of LEDs states. These alarms states are visible in zttool/dahditool or in astersick CLI (zap show status) and I wonder what is the real meaning of these alarms for E1 channel. I can't speak for all the possible states in the T1/E1 card driver, but I can state that typically in T1s and E1s you have three different general alarm states: RED alarms, YELLOW alarms, and BLUE alarms. (This is a brief synopsis of the information we cover in the Asterisk Advanced training class.) [snip] I hope the explanation helps. You can also find it in the README of DAHDI: http://docs.tzafrir.org.il/dahdi-linux/#_alarm_types Commments would be welcomed -- Tzafrir Cohen Thanks for this detailed reply, amount of information exceeded my expectation :) In dahdi_tool, there are three more indicators of error: IRQ misses Bipolar violation CRC error As I understand it now, these should be error counters and they provide additional information in case of RED alarm state. Regards Lukas Rypl ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fring and Asterisk
2009/1/20 Olivier oza-4...@myamail.com 2009/1/20 Gordon Henderson gordon+aster...@drogon.netgordon%2baster...@drogon.net On Tue, 20 Jan 2009, Olivier wrote: GTalk seems to fill the bill of requirements, though, I don't think it's available on Nokia mobile phones .. My Nokia mobile phone has a SIP client built-in which uses Wi-Fi... And while it's not perfect, it's actually very usable and works well with asterisk. Upgrade the phone? I agree that native Nokia SIP client is very usable and should be the preferred way to call over a WiFi connection. I mentioned GTalk as someone mentioned Fring on Nokia. Unfortunately, I upgraded my phone to one of these new fangled S60 3rd edition Feature Pack 2 phones (N78 in this case, but N96 has the same software) where Nokia have removed the SIP client... They've included a SIP stack and stated that this way third parties can write their own clients... Options are somewhat limited at the moment... d ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fring and Asterisk
2009/1/20 Olivier oza-4...@myamail.com One thing to note about fring, the device establishes a connection using fring's proprietary protocols to fring servers, fring then establishes SIP connections from those servers... So, even if connected to the office Wifi connection, you could experience connectivity issues or high latency as a result of a potentially long path involved for the traffic to travel... d So, when in the office, whenever I'm calling someone using SIP and WiFi, for both signalling and media, data would travel from mobile phone to WiFi access point, then to switch, router, Fring server (at the other end of the world), then back to my routeur, and Asterisk server, right ? Correct... If think I'll try to compare this with a XMPP/Jingle-enabled client that can be installed in mobile phone (I don't know if such software exist) ... Not seen anything, though I've mostly been looking for Nokia phones, my boss has a iPhone though, so something for that wouldn't be bad either... d ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems With Playback of Audio On SIP Only System
Mark Michelson wrote: If you are using gsm prompts and gcc version 4.2 or higher, then you may be experiencing the optimizer bug that gcc has with gsm audio. The workarounds for this are to use a different format for sounds or to set the DONT_OPTIMIZE flag in menuselect. If you want an optimized build and gsm formatted sounds, then you could always attempt downgrading your gcc version to 4.1 or earlier. This is affecting users frequently enough that we probably need to engineer some sort of configure-script test to check for this problem at build time. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] the FXS ports of Digium and damaging if connected to Tel Line
On Tue, 20 Jan 2009, Geoff Lane wrote: On Tuesday, January 20, 2009, bilal ghayyad wrote: What is the solution for this disaster? I live in UK, where we don't use RJ11 for telephones and so need to use adapters, which I just leave hanging out of the FXO ports. With the adapters in place, it's difficult to plug the phones into the wrong ports. I've also stuck a large label on the computer cabinet to show which ports are for what. For info, I have a TDM400P. If you're carefull, you can clip the tang of the RJ11 plug on the end of the BT adapter so that it needs a small screwdriver to remove it from the socket on the board... I found this out after swapping a Panasonic unit out for an Asterisk box of my own and having a devil of a time removing the connectors going into the Panasonic box - I'm told they're supplied like that and they were punched-down directly into the DP! (not a BT 'master' socket in sight!) I suspect that the label idea might help but the other would depend on whether your country uses RJ11 for standard phone connections. Never underestimate the stupity of sheeple... Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP DTMF problem with SNOM
I have a similar problem with Snom. Since I've upgraded from version 6 to version 7 I cannot call IVR systems. The first DTMF goes ok, but after that others are not accepted nor I am heard by the operator at the other side. Since I am the only one who has Snom here I didn't bother to debug it... __Yehavi: ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Siemens S685IP registration problems
2009/1/20 Simon Dixey simon_...@hotmail.co.uk Hi folks, I wonder if any of you out there are using Siemens S685IP base station(s) (with S68H handsets) on Asterisk and experiencing problems with SIP registrations where the SIP extensions do not ring and peers become unreachable after a period of time. Symptoms are rather sporadic, but as described, SIP extensions being unreachable from Asterisk perspective. Also experienced 'not possible' messages trying to dial using the handsets. In Asterisk - 'sip show peers' shows the SIP device status as 'UNKNOWN' and console output says chan_sip.c Peer 'xxx' is unreachable. (as one would expect if it can't see it!!) So I'm thinking it's a problem with the base.. or.. some issue with qualifications and possibly the base station not responding (guessing here). I'm finding that the Siemens web GUI reports messages similar to 'server not accessible' or 'registration failed'. These messages appear randomly throughout the day following successful previous registration. Have enabled 'sip set debug ip xxx' for one of the bases and on the Siemens web GUI disabling the SIP account and re-enabling it doesn't work generally, and no SIP messages are being directed from the base to Asterisk. Leads me to think it's a base/firmware issue. Some times the phones are contactable for a day without fault, other times they're problematic at random intervals. It's not always all of the SIP accounts assigned on the Siemens base, sometimes it's just one account, other times it's all accounts. (What a horrible situation to debug/fault find! Glad my Aastra's are reliable!) The only resolve I've found which is rather unacceptable is to reboot the Siemens base station. Upon doing so, the base re-registers all of the accounts to the Asterisk server and calls to/from handsets work for 'a period of time'... My setup is as follows: - Asterisk 1.4.22 - Base 1 has 3x handsets and 3x SIP accounts (providers) and the SIP accounts are individually assigned to each handset. - Base 2 has 5 handsets and 6 SIP accounts. 5 SIP accounts for each handset, then the 6th SIP account is a 'group' extension which rings all of the phones on the base station. MWI for VM is also used and works. Call waiting disabled on handsets. - Both bases on latest available as of Jan 09 - 0214 / 043.00 The bases are set up with static IPs, info services off, etc. Am not using SIP domains, no NAT, all communicating on a LAN on same network so no routing or latency issues. Registrar server defined and refresh time set to 180 seconds (Siemens default). SIP.conf has nat=no, qualify=yes what if setting qualify=1 (or any very long value) for every handset but one for which qualify=1000 is used ? maybe, the base takes too long to reply to qualify messages for an unknown reason and changing from qualify=yes to qualify=something might help to confirm parts of diagnosis ... . host= is currently dynamic.. maybe I should set this to the IP of the base as they're using static IPs, but reading the specs of this setting describe set to 'dynamic' if the phone should register itself... hmm. I've seen similar posts from other users on the exact same subject. Sources as follows: Voip-Info http://www.voip-info.org/wiki/view/Siemens+Gigaset+S675IP (see 'Discussions' tab). The Open Sourcerer: http://www.theopensourcerer.com/2008/04/27/siemens-gigaset-685ip-phones/comment-page-3/ Siemens Forums: http://www.siemens-gigaset-forum.com/de/posts/list/13788.page Siemens Customer care: http://www.siemens-gigaset-forum.com/de/posts/list/13788.page (people with aged open support calls) Siemens support via phone were rather unhelpful and didn't grasp the technicalities of the issue I was conveying so drew a blank (have I checked my router.. hmm!) I'm guessing this is a firmware issue but intrigued to know if others are experiencing the same. Anyone else experiencing similar problems? Or indeed successes with a similar setup? Can anyone recommend a stable working DECT SIP phone for enterprise use with Asterisk? (The Snom M3 looks good but read about issues with transfers which concern me) May have to resort to ditching the S685s and go with Aastra desk sets all round - shame to lose the flexibility of cordless though. Many thanks in advance. -- See all the ways you can stay connected to friends and familyhttp://www.microsoft.com/windows/windowslive/default.aspx ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:
Re: [asterisk-users] the FXS ports of Digium and damaging if connected to Tel Line
Dear Steve; But it is not logical to keep having damaging ports, till when these mistakes will be able to keep replace it? Regards Bilal Hi All; I am facing a problem that always the users confused and connect the telephone line coming from the telephone service provider to the FXS port and cause it to be damaged, specially if the card was 2 fxs and 2 fxo, so they make mistake and connect the line to fxs while it should be connected to fxo. What is the solution for this disaster? I have done this (very briefly) by accident. It survived. Is it definitely dead? Do the other ports work? If its just that port thats toasted then you might get away with just replacing that module. -- Message: 19 Date: Tue, 20 Jan 2009 11:28:17 +0100 From: sh0t s...@sh0t.org Subject: Re: [asterisk-users] CDR problems -- two call legs create only one CDR. Using ForkCDR() not even working. To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: 4975a741.5060...@sh0t.org Content-Type: text/plain; charset=UTF-8; format=flowed hello When I bridge an incoming and outgoing call (attempting to simulate call-forwarding) I'm only getting one CDR -- that of the outgoing call. A (PSTN) calls B (residing on Asterisk) and the Asterisk calls C (cell phone on PSTN) and bridges the call. The only CDR created is from B to C. I have even tried using Answer() and ForkCDR() to get two CDRs, but to no avail. I am starting to wonder if there's a bug in the CDR generation in general, because I set up an extension to do only that: exten = 5822558,1,Answer() exten = 5822558,n,ForkCDR() exten = 5822558,n,Playback(tt-monkeys) exten = 5822558,n,Hangup() I have the same problem, try older version like 1.4.21 for example.. And look at: http://bugs.digium.com/view.php?id=13797, there is some patch that might help. For me the only possible option to play with forkcdr as I expected is using older versions of Asterisk best regards -- Message: 20 Date: Tue, 20 Jan 2009 11:31:46 +0100 From: Klaus Darilion klaus.mailingli...@pernau.at Subject: Re: [asterisk-users] how to cancel new recorded message from voicemail menu? To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: 4975a812.8070...@pernau.at Content-Type: text/plain; charset=ISO-8859-1; format=flowed Philipp Kempgen schrieb: Klaus Darilion schrieb: If a user has recorded a new voicemail message (e.g. unavailable message) then it is prompted with 3 choices. 1. accept recording 2. listen to the recorded message 3. rerecord the message Isn't it possible to cancel the recording? You could hang up. But users might not be aware of this simple solution. Stupid me :-) -- Message: 21 Date: Tue, 20 Jan 2009 11:34:38 +0100 From: Klaus Darilion klaus.mailingli...@pernau.at Subject: Re: [asterisk-users] Asterisk 1.6 T38 to G711 transcoding is this possible? To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: 4975a8be.3020...@pernau.at Content-Type: text/plain; charset=ISO-8859-1; format=flowed What you need is a so called T38Gateway application. there is a patch o the tracker which you might want to try: http://bugs.digium.com/view.php?id=13405 klaus Steve Gladden schrieb: The scenario we have is fax send/recieve software that ONLY talks T38 and an asterisk box. We have ITSP providers that do NOT talk T38 but G711 only. Does asterisk have the capability to take the T38 call from an ATA or T38 software then bridge/transcode it and do G711 out to the PSTN providers? If not is there another product PAID or FREE software or hardware that can do this easily and reliably? Thanks very much! Steve Gladden -- Message: 22 Date: Tue, 20 Jan 2009 11:57:49 +0100 From: Klaus Darilion klaus.mailingli...@pernau.at Subject: [asterisk-users] SIP DTMF problem with SNOM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: 4975ae2d.9050...@pernau.at Content-Type: text/plain; charset=ISO-8859-15; format=flowed Hi! I have two identical SIP accounts on Asterisk 1.4.22. One account is registered with eyebeam, the other one is registered with a SNOM phone. When using the eyebeam client DMTF detection works fine, when using the SNOM phone many digits are missing in the DTMF detection. I analyzed with wireshark and both phones uses RFC 2833 and the trace looks pretty the same. Also the rtp debug log looks fine (see below). What could be the reason? thanks klaus trace: I have entered 1234#, but
[asterisk-users] X-Lite and Asterisk RTP cutting out
Hi, I am running Asterisk 1.4.22. If in X-Lite I have set to hang-up after 0 seconds of RTP the call gets cut off between 1 1/2 to 2 minutes. I have tried to connect to another server and the call stayed up. If I take out the (RTP) setting then it works fine. What would cause the RTP to stop for a brief moment or the phone to see that the RTP cut out ? Thanks___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP DTMF problem with SNOM
I have a Snom 320 and both inband and RFC2833 OOB work fine for me. Yehavi Bourvine wrote: I have a similar problem with Snom. Since I've upgraded from version 6 to version 7 I cannot call IVR systems. The first DTMF goes ok, but after that others are not accepted nor I am heard by the operator at the other side. Since I am the only one who has Snom here I didn't bother to debug it... __Yehavi: ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] the FXS ports of Digium and damaging if connected to Tel Line
Dear Gordon; I did not understand the idea of the following: If you're carefull, you can clip the tang of the RJ11 plug on the end of the BT adapter so that it needs a small screwdriver to remove it from the socket on the board... Do u have any link? What do u mean by clibing the tang of the RJ11 plug on the end of the BT adaptor? Regards Bilal What is the solution for this disaster? I live in UK, where we don't use RJ11 for telephones and so need to use adapters, which I just leave hanging out of the FXO ports. With the adapters in place, it's difficult to plug the phones into the wrong ports. I've also stuck a large label on the computer cabinet to show which ports are for what. For info, I have a TDM400P. If you're carefull, you can clip the tang of the RJ11 plug on the end of the BT adapter so that it needs a small screwdriver to remove it from the socket on the board... I found this out after swapping a Panasonic unit out for an Asterisk box of my own and having a devil of a time removing the connectors going into the Panasonic box - I'm told they're supplied like that and they were punched-down directly into the DP! (not a BT 'master' socket in sight!) I suspect that the label idea might help but the other would depend on whether your country uses RJ11 for standard phone connections. Never underestimate the stupity of sheeple... Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hang up detection problems
hi i have a E1 connected to the PSTN when the remote site hang up asterisk dont detect it i found this in the pri debug Ext: 1 Progress Description: Call is not end-to-end ISDN; further call progress information may be available inband. (1) ] asterisk version is 1.4 any ideas? thanks! -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] the FXS ports of Digium and damaging if connected to Tel Line
On Tuesday, January 20, 2009, bilal ghayyad wrote: What do u mean by clibing the tang of the RJ11 plug on the end of the BT adaptor? On an RJ11 plug, the casing includes a springy piece that locks the plug into an RJ11 socket. When plugged in, the end of the springy piece sticks out of the socket and is the bit you need to squeeze towards the plug body to release the plug from the socket. That springy piece is called the tang. If you take care, it's possible to cut off the end of the tang that sticks out of the socket while leaving enough of the tang to lock the plug in place. If you do that, you can't remove the plug by hand. You need to insert a small screwdriver into the socket to depress the tang in order to release the plug from the socket. HTH, -- Geoff ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] the FXS ports of Digium and damaging if connected to Tel Line
On Tue, 20 Jan 2009, bilal ghayyad wrote: Dear Gordon; I did not understand the idea of the following: If you're carefull, you can clip the tang of the RJ11 plug on the end of the BT adapter so that it needs a small screwdriver to remove it from the socket on the board... Do u have any link? What do u mean by clibing the tang of the RJ11 plug on the end of the BT adaptor? The tang is the thin bit of plastic that makes the RJ11 plug latch into the socket. You push it down to un-latch the plug so you can pull it out of the socket. If you cut it about a third of the way down, you'll not be able to easilly remove the plug once inserted into the socket. Maybe it doesn't translate well: http://www.merriam-webster.com/dictionary/tang Maybe there's another word for it, but it's what I've always used... Gordon Regards Bilal What is the solution for this disaster? I live in UK, where we don't use RJ11 for telephones and so need to use adapters, which I just leave hanging out of the FXO ports. With the adapters in place, it's difficult to plug the phones into the wrong ports. I've also stuck a large label on the computer cabinet to show which ports are for what. For info, I have a TDM400P. If you're carefull, you can clip the tang of the RJ11 plug on the end of the BT adapter so that it needs a small screwdriver to remove it from the socket on the board... I found this out after swapping a Panasonic unit out for an Asterisk box of my own and having a devil of a time removing the connectors going into the Panasonic box - I'm told they're supplied like that and they were punched-down directly into the DP! (not a BT 'master' socket in sight!) I suspect that the label idea might help but the other would depend on whether your country uses RJ11 for standard phone connections. Never underestimate the stupity of sheeple... Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] the FXS ports of Digium and damaging if connected to Tel Line
Geoff Lane wrote: If you take care, it's possible to cut off the end of the tang that sticks out of the socket while leaving enough of the tang to lock the plug in place. You may not even need to clip the end off, with the last lot of RJ-11 plugs I ordered the tangs were short enough to snap in the socket on a TDM400P. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-users Digest, Vol 54, Issue 53
On Tue, 20 Jan 2009, bilal ghayyad wrote: Do u mean by the Iaxy2 is that IAX digium gateway adaptor? Yes. Your request was for a hard phone, but I was replying to the reply about the Iaxy2. Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CallerID ANI issues
Hello, We're having some issues with CallerID and I thought someone here might be able to shed some light as none of our carriers seem to know what I'm talking about. The issues is this: A client of ours uses an after-hours voicemail service as mandated by their corporate office. We have a Day/Night setting that lets them turn this on and off. A call comes in from one of their customers and gets forwarded back out to the voicemail service with the CallerID set to the clients DID. The voicemail service checks the callerID of the incoming call to determine which agency is calling. The problem is this: The voicemail service (who uses verizon), looks at the ANI field in the CallerID which shows up as something other than our clients DID (notably our BILLING number) We've called two of our carriers (one a SIP provider, the other our PRI provider) and they both say they make no distinction between 'regular' CallerID and the ANI field. The PRI provider said if we have 'station-level' callerID (which we do) the number should show up fine. I've contacted the voicemail service and they say there's nothing they can do on their end. I've played around with setting the ANI fields on our asterisk servers and as far as I can tell the ANI is correctly set to the same as the callerID (Tried Set(CALLERID(all) and Set(CALLERID(ANI) ). Does anyone have any idea what we might look at next to get this resolved? I'm pretty eager to figure this out as we potentially have a dozen clients that are interested in signing with us, provided we have this working. Thanks a lot! Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CallerID ANI issues
Most voicemail/answering service dont' care about callerid or ani, they instead use the DID that the call comes in on to decide how to answer the call. Get a different voicemail/answering service. On Tue, Jan 20, 2009 at 9:32 AM, mail-lists mail-li...@peachnet.com wrote: Hello, We're having some issues with CallerID and I thought someone here might be able to shed some light as none of our carriers seem to know what I'm talking about. The issues is this: A client of ours uses an after-hours voicemail service as mandated by their corporate office. We have a Day/Night setting that lets them turn this on and off. A call comes in from one of their customers and gets forwarded back out to the voicemail service with the CallerID set to the clients DID. The voicemail service checks the callerID of the incoming call to determine which agency is calling. The problem is this: The voicemail service (who uses verizon), looks at the ANI field in the CallerID which shows up as something other than our clients DID (notably our BILLING number) We've called two of our carriers (one a SIP provider, the other our PRI provider) and they both say they make no distinction between 'regular' CallerID and the ANI field. The PRI provider said if we have 'station-level' callerID (which we do) the number should show up fine. I've contacted the voicemail service and they say there's nothing they can do on their end. I've played around with setting the ANI fields on our asterisk servers and as far as I can tell the ANI is correctly set to the same as the callerID (Tried Set(CALLERID(all) and Set(CALLERID(ANI) ). Does anyone have any idea what we might look at next to get this resolved? I'm pretty eager to figure this out as we potentially have a dozen clients that are interested in signing with us, provided we have this working. Thanks a lot! Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] channel var for Call on hold?
Hi all, Does asterisk (I'm using 1.4.19) sets any channel variable with the holded chan when it does an atxfer? I tried to see if it does on the source, but i didn't find any clue, neither enabling console debug. Thanks, Gabriel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP DTMF problem with SNOM
Alex Balashov schrieb: How are you testing DTMF detection with the Snom UA? The Voicemail(u...@context) application asks the user for the voicemail password. Using eyebeam everyhing works fine. Using SNOM (320 FW 7.3.10a) it works almost never. regards klaus Klaus Darilion wrote: Hi! I have two identical SIP accounts on Asterisk 1.4.22. One account is registered with eyebeam, the other one is registered with a SNOM phone. When using the eyebeam client DMTF detection works fine, when using the SNOM phone many digits are missing in the DTMF detection. I analyzed with wireshark and both phones uses RFC 2833 and the trace looks pretty the same. Also the rtp debug log looks fine (see below). What could be the reason? thanks klaus trace: I have entered 1234#, but voicemail received as secret just 123. Got RTP packet from83.136.33.3:64118 (type 00, seq 042765, ts 4066332168, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042766, ts 4066332328, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042770, ts 4066332968, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042771, ts 4066333128, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042774, ts 4066333608, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042775, ts 4066333768, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042777, ts 4066334088, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042778, ts 4066334248, len 000160) Got RTP packet from83.136.33.3:64118 (type 101, seq 042780, ts 4066334648, len 04) Got RTP RFC2833 from 83.136.33.3:64118 (type 101, seq 042780, ts 4066334648, len 04, mark 0, event 0001, end 0, duration 00320) Got RTP packet from83.136.33.3:64118 (type 101, seq 042781, ts 4066334648, len 04) Got RTP RFC2833 from 83.136.33.3:64118 (type 101, seq 042781, ts 4066334648, len 04, mark 0, event 0001, end 0, duration 00480) Got RTP packet from83.136.33.3:64118 (type 101, seq 042782, ts 4066334648, len 04) Got RTP RFC2833 from 83.136.33.3:64118 (type 101, seq 042782, ts 4066334648, len 04, mark 0, event 0001, end 0, duration 00640) Got RTP packet from83.136.33.3:64118 (type 101, seq 042785, ts 4066334648, len 04) Got RTP RFC2833 from 83.136.33.3:64118 (type 101, seq 042785, ts 4066334648, len 04, mark 0, event 0001, end 0, duration 01120) Got RTP packet from83.136.33.3:64118 (type 101, seq 042786, ts 4066334648, len 04) Got RTP RFC2833 from 83.136.33.3:64118 (type 101, seq 042786, ts 4066334648, len 04, mark 0, event 0001, end 0, duration 01280) Got RTP packet from83.136.33.3:64118 (type 101, seq 042788, ts 4066334648, len 04) Got RTP RFC2833 from 83.136.33.3:64118 (type 101, seq 042788, ts 4066334648, len 04, mark 0, event 0001, end 0, duration 01600) Got RTP packet from83.136.33.3:64118 (type 101, seq 042789, ts 4066334648, len 04) Got RTP RFC2833 from 83.136.33.3:64118 (type 101, seq 042789, ts 4066334648, len 04, mark 0, event 0001, end 1, duration 01760) Got RTP packet from83.136.33.3:64118 (type 101, seq 042789, ts 4066334648, len 04) Got RTP RFC2833 from 83.136.33.3:64118 (type 101, seq 042789, ts 4066334648, len 04, mark 0, event 0001, end 1, duration 01760) Got RTP packet from83.136.33.3:64118 (type 101, seq 042789, ts 4066334648, len 04) Got RTP RFC2833 from 83.136.33.3:64118 (type 101, seq 042789, ts 4066334648, len 04, mark 0, event 0001, end 1, duration 01760) Got RTP packet from83.136.33.3:64118 (type 00, seq 042790, ts 4066336648, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042791, ts 4066336808, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042792, ts 4066336968, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042793, ts 4066337128, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042794, ts 4066337288, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042795, ts 4066337448, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042798, ts 4066337928, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042801, ts 4066338408, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042802, ts 4066338568, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042805, ts 4066339048, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042806, ts 4066339208, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042809, ts 4066339688, len 000160) Got RTP packet from83.136.33.3:64118 (type 101, seq 042812, ts 4066340168, len 04) Got RTP RFC2833 from 83.136.33.3:64118
Re: [asterisk-users] CallerID ANI issues
Not a possibility I'm afraid. Our client is an insurance agent and the voicemail/answering service is mandated by corporate. There also are not various DID's to call in on. All voicemail calls go to an 800 number Thanks for your advice though. Most voicemail/answering service dont' care about callerid or ani, they instead use the DID that the call comes in on to decide how to answer the call. Get a different voicemail/answering service. On Tue, Jan 20, 2009 at 9:32 AM, mail-lists mail-li...@peachnet.com wrote: Hello, We're having some issues with CallerID and I thought someone here might be able to shed some light as none of our carriers seem to know what I'm talking about. The issues is this: A client of ours uses an after-hours voicemail service as mandated by their corporate office. We have a Day/Night setting that lets them turn this on and off. A call comes in from one of their customers and gets forwarded back out to the voicemail service with the CallerID set to the clients DID. The voicemail service checks the callerID of the incoming call to determine which agency is calling. The problem is this: The voicemail service (who uses verizon), looks at the ANI field in the CallerID which shows up as something other than our clients DID (notably our BILLING number) We've called two of our carriers (one a SIP provider, the other our PRI provider) and they both say they make no distinction between 'regular' CallerID and the ANI field. The PRI provider said if we have 'station-level' callerID (which we do) the number should show up fine. I've contacted the voicemail service and they say there's nothing they can do on their end. I've played around with setting the ANI fields on our asterisk servers and as far as I can tell the ANI is correctly set to the same as the callerID (Tried Set(CALLERID(all) and Set(CALLERID(ANI) ). Does anyone have any idea what we might look at next to get this resolved? I'm pretty eager to figure this out as we potentially have a dozen clients that are interested in signing with us, provided we have this working. Thanks a lot! Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Why does Asterisk not hangup?
Hi! I have the following scenario: Asterisk INVITE- | --200,ACK-- | Playback(Foo) | Dial(..) | -INVITE- | -404. ACK-- | As my extension configuration stops after the Dial command I expect Asterisk to hang up the call. Instead I see on the console: | | == Auto fallthrough, channel 'SIP/81.16.153.183-0883dea0' status is 'CONGESTION' | | Then I hear the congestion tone for 10 secondes, then Asterisk sends BYE | -BYE| Why does Asterisk not hangup immediately? Why 10 seconds congestion tone? Is this configurable? thanks klaus ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why does Asterisk not hangup?
On Tue, 2009-01-20 at 16:08 +0100, Klaus Darilion wrote: As my extension configuration stops after the Dial command I expect Asterisk to hang up the call. Instead I see on the console: | | == Auto fallthrough, channel 'SIP/81.16.153.183-0883dea0' status is 'CONGESTION' | Auto-fallthrough means that Asterisk couldn't find any more priority numbers for the current extension, so it had to guess what the correct course of action was (whether to hang up, play congestion, etc.). If you want it to hangup instead of playing congestion, simply add another priority to your extension (right after the call to the Dial() application) which calls the Hangup() application explicitly. -- Jared Smith Digium, Inc. | Training Manager ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Outgoing CallerID w. DAHDI on ISDN BRI
Hi List, i've set up an Asterisk 1.6.0.3 Server equipped with an Xorcom Astribank BRI XR0013 (2 Port BRI) and Dahdi 2.1.0. In- and outgoing calls are no problem. But I can't get Asterisk/DAHDI to use the second Number of the BRI. All calls set the primary number as callerid. It wouldn't be a problem, if we would use this one as id for our company, but it's already in use (private). I tried with Set(Callerid(num)=12345)/Set(Callerid(number) in extensions.conf and callerid=12345 in chan_dahdi.conf and in dahdi-channels.conf. In Dahdi i activated all options for transmitting a callerid, including the ones for PRI (for Testing). Has anybody an idea? Thanks in advance for any hint/help/tip. Mit freundlichen Grüßen / Kind Regards Peter Müller___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems With Playback of Audio On SIP Only System
On Tuesday 20 January 2009 06:31:51 Kevin P. Fleming wrote: Mark Michelson wrote: If you are using gsm prompts and gcc version 4.2 or higher, then you may be experiencing the optimizer bug that gcc has with gsm audio. The workarounds for this are to use a different format for sounds or to set the DONT_OPTIMIZE flag in menuselect. If you want an optimized build and gsm formatted sounds, then you could always attempt downgrading your gcc version to 4.1 or earlier. This is affecting users frequently enough that we probably need to engineer some sort of configure-script test to check for this problem at build time. Actually, this problem should already be fixed. It was a case of optimization with assembly code. Adding the noclobber option to the assembly should have fixed this. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Called's channel
with {$BRIDGEPEER}
2009/1/20 Jose Enes Mateus jemat...@yahoo.com.br
Hi,
I have a question...
With the variable ${CHANNEL} I can get the channel whose made the call, or
the caller. How can I get the channel of the called side?
Veja quais são os assuntos do momento no Yahoo! +Buscados
http://br.maisbuscados.yahoo.com
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[asterisk-users] open-source ZRTP implementation (was: Re: [somewhat OT] seeking ideas/input for my thesis)
John Todd schrieb: So here are some projects you might look into: - open-source ZRTP implementation libzrtpcpp is open source. GPL. http://www.gnu.org/software/ccrtp/ Philipp Kempgen -- AMOOCON 2009, May 4-5, Rostock / Germany - http://www.amoocon.de Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Stutter/chopoff first audio played
I've noticed on a few installations that the very first audio played after a call in answered (eg: Greeting), the first part of the audio is cutoff/stuttered. Is this because Asterisk needs some RTP to create a sync for audio - and the first 1 second is lost? Should one play 1 sec of silence first? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Interesting observation
At 12:48 PM 1/19/2009, you wrote: I do not know if they use analog or digital signals for the phones but if we use the cell phone system as an example they took down all analog towers because they could service more phones on the same bandwidth with digital. I would assume that would hold true for the spectrum on a cable as well. I would also find it hard to believe that they would not use off the shelf technology. There used to be is a modem of sorts on the wall where they connect into the house wiring. It looked like a analog phone line on the house side and connected to the cable on the other. Supported 4 analog lines. Then a year or 2 ago they changed the service so that it now uses a cable modem with a built in backup battery and 2 analog phone ports next to the ethernet port. I have to assume it's digital as I can't imagine there's any analog in a modern cable modem. Ira ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Forwarding calls and trasfer calls
features.conf for transfers
for call forwardin you need some application logic.
e.g.
_**21**. = {
Set(NUM=${EXTEN:6}); // contains the new target
// now store this number somewhere, e.g. astdb, odbc ...
...
}
context fromPstn {
1234 = {
// check if user has actived forwarding
// retrieve NUM from astdb or ODBC
if(${EXISTS(${NUM})}) {
Dial(DAHDI/g1/${NUM});
} else {
Dial(SIP/${EXTEN});
}
}
}
regards
klaus
Ralf Träskman schrieb:
Hi
How do i set up so that everyone can dial, for example **21** to forward
all calls to a cellphone or another extension and how do I enable so
that cals can be transferd between extentions.
I use asterisk 1.6 and have my phones in unistim.conf and my extensions
in extensions.conf.
Regards
/ralf
Ralf Träskman, IT
AdLibris AB, Odengatan 106, 113 22 Stockholm, Sweden
Dir: +46-(0)707548074, vxl: +46-(0)85460 60 00, fax: +46-(0)85460 60 99
r...@adlibris.com mailto:r...@adlibris.com www.adlibris.com
http://www.adlibris.com/
P *Please consider the environment before printing this e-mail*
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Re: [asterisk-users] Why does Asterisk not hangup?
Jared Smith schrieb: On Tue, 2009-01-20 at 16:08 +0100, Klaus Darilion wrote: As my extension configuration stops after the Dial command I expect Asterisk to hang up the call. Instead I see on the console: | | == Auto fallthrough, channel 'SIP/81.16.153.183-0883dea0' status is 'CONGESTION' | Auto-fallthrough means that Asterisk couldn't find any more priority numbers for the current extension, so it had to guess what the correct course of action was (whether to hang up, play congestion, etc.). If you want it to hangup instead of playing congestion, simply add another priority to your extension (right after the call to the Dial() application) which calls the Hangup() application explicitly. Ok. Just for the info to others: the 10 seconds are hardcoded in pbx.c regards klaus ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems With Playback of Audio On SIP Only System
Tilghman Lesher wrote: Actually, this problem should already be fixed. It was a case of optimization with assembly code. Adding the noclobber option to the assembly should have fixed this. Well, people are still running into it as evidenced by this thread, and it's a very subtle issue to debug and resolve. What releases of Asterisk contain the fix for this, and can we get the OP of this thread to confirm that using the newer version resolved the issue for him? If so, maybe we should make an explicit announcement with more details about this specific issue so people will be more aware of it. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Setting up an outgoing trunk group
Hi All,
I'm confused! My Asterisk system has a Zap trunk and three SIP trunks.
I'd like to configure the dialplan to route via the first trunk in a
list and if that's not available or it's busy, fall over to the
second, then to the third, etc.
AIUI Dial(Zap/1SIP/out1SIP/out2/${EXTEN}) rings all the trunks in
the list and bridges to the first to answer. Unfortunately, that's not
what I want, which is (in pseudocode):
if Zap/1 is available then
Dial(Zap/1/${EXTEN})
elseif SIP/out1 is available then
Dial(SIP/out1/${EXTEN})
else
Dial(SIP/out2/${EXTEN})
end if
Also, to make it easier to reconfigure quickly, I've got a variable
defined in [globals] thus:
MainOutbound=Zap/1SIP/out1SIP/out2
so the Dial statement above would be written in the dialplan thus:
Dial(${MainOutbound}/${EXTEN})
So if I can't find the syntax to get the Dial application to do what I
want I guess I'd need to use a dialplan function or AGI.
Can anyone help?
TIA,
--
Geoff
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Re: [asterisk-users] Setting up an outgoing trunk group
I would use the ${DIALSTATUS} variable. In your dialplan dial the first trunk
you wish, then afterwards examine the ${DIALSTATUS} variable. If that is not
equal to ANSWER then dial your second trunk and so on.
For example:
exten = s,1,Dial(ZAP/1/${EXTEN})
exten = s,n,ExecIf($[${DIALSTATUS} != ANSWER]|Dial|(SIP/out1/${EXTEN})
exten = s,n,ExecIf($[${DIALSTATUS} != ANSWER]|Dial|(SIP/out2/${EXTEN})
That's kind of rough but it should work.
Cheers,
Darrin Henshaw | IT Administrator | MCTS: Exchange 2007 | MCSE 2003 | LPIC
Ignition Support Center | www.ignition.bm
Bermuda (441) 496-4319 | Cayman (345) 947-4357 | Halifax (902) 482-1288
Atlanta | Bermuda | Cayman | Halifax
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Geoff Lane
Sent: Tuesday, January 20, 2009 13:31
To: Asterisk Users
Subject: [asterisk-users] Setting up an outgoing trunk group
Hi All,
I'm confused! My Asterisk system has a Zap trunk and three SIP trunks.
I'd like to configure the dialplan to route via the first trunk in a
list and if that's not available or it's busy, fall over to the
second, then to the third, etc.
AIUI Dial(Zap/1SIP/out1SIP/out2/${EXTEN}) rings all the trunks in
the list and bridges to the first to answer. Unfortunately, that's not
what I want, which is (in pseudocode):
if Zap/1 is available then
Dial(Zap/1/${EXTEN})
elseif SIP/out1 is available then
Dial(SIP/out1/${EXTEN})
else
Dial(SIP/out2/${EXTEN})
end if
Also, to make it easier to reconfigure quickly, I've got a variable
defined in [globals] thus:
MainOutbound=Zap/1SIP/out1SIP/out2
so the Dial statement above would be written in the dialplan thus:
Dial(${MainOutbound}/${EXTEN})
So if I can't find the syntax to get the Dial application to do what I
want I guess I'd need to use a dialplan function or AGI.
Can anyone help?
TIA,
--
Geoff
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[asterisk-users] dead sip channel
I have ran into a case using 1.4.22 where a SIP call to an asterisk
client (running a slow PC) to ALSA
does not hangup the call when it is done. The server is using call files
to initiate the call, the client answers on
the ALSA port, the server plays the message and hangs up.
I found that SOMETIMES -its hard to recreate - that the slow pc keeps
the SIP channel active. further calls in
are getting a busy signal and the one call is NEVER hung up.
How can I detect this and hang up the channel on the slow PC. I verified
on the server that it thinks NO calls are active.
my context looks like this:
[mycontext]
exten = s,1,ChanIsAvail(Console/Dsp)
exten = s,n,GotoIf($[${AVAILCHAN} = ]?smvoice-busy,s,1)
exten = s,n,Playback(beep)
exten = s,n,Dial(Console/dsp)
exten = s,n,Hangup
[smvoice-busy]
exten = s,1,playtones(busy)
exten = s,1,wait(10)
exten = s,1,Hangup
Thanks,
Jerry
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[asterisk-users] Cisco 7941G-GE with Asterisk and CTPSEP odyssee
I contacted Christophorus directly and he was able to point me to the correct solution. Apparently, not all TFTP servers are created equal. I uninstalled tftpd (running Ubuntu server Hardy Heron) and installed atftpd instead. As soon as this was configured, the phone quit searching for the .tlv file and I am now able to start to work on the .xml file. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call Dropped in Voicemail / No Reply to Our Critical Packet w/ SIP Debug
Hi All, A long time ago I posted about an issue where calls on one of our Asterisk boxes were being dropped in Voicemail (and only in voicemail) after about 20 seconds with the error logged [Jan 19 14:33:26] WARNING[27458]: chan_sip.c:1980 retrans_pkt: Hanging up call 001d45b6-1d490088-46ef7ed6-adbeb...@10.2.0.203 - no reply to our critical packet (see doc/sip-retransmit.txt).. I thought the issue had cleared itself up, but it has resurfaced. We're running 1.4.22 on built from source, using Cisco 7961G phones with the SIP firmware image (and I've tried most of the recent firmware versions for the Cisco phones to see if that would change things at all). I would appreciate any assistance since I'm stumped. The output of SIP DEBUG for the extension most frequently affected by the issue is below; starting with one call to voicemail that was successfully completed, followed by a 2nd call that was dropped after approximately 18 seconds. The issue is consistently inconsistent - it doesn't happen on every call to Voicemail, but those that it does happen on it's always within the first approximately 20 seconds of the call; once you pass the 25 second mark you're free and clear for that call-it will not be dropped. It also seems like it's possible to reproduce the issue by making several calls to Voicemail in short order, but this isn't the only trigger as sometimes the first call to voicemail in 12+ hours will also trigger it. I'm also baffled by the fact that this ONLY crops up on calls to Voicemail on the local box; SIP to SIP calls on the same Asterisk box, SIP to SIP calls from this Asterisk box to an Asterisk Appliance at a remote site, SIP to POTS, and POTS to SIP calls are completely unaffected. Again, any advice/suggestions/things to look at/etc are greatly appreciated! Thanks in advance, Lincoln --- Reliably Transmitting (no NAT) to 10.2.0.203:5060 --- SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.2.0.203:5060;branch=z9hG4bK1971acea;received=10.2.0.203 From: Jim Felderman sip:1...@10.2.0.2;tag=001d45b61d4906943a6bb290-9435a462 To: sip:voicem...@10.2.0.2;tag=as00aa5042 Call-ID: 001d45b6-1d490087-44d5cce8-d5996...@10.2.0.203 CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=1db91b00 Content-Length: 0 Scheduling destruction of SIP dialog '001d45b6-1d490087-44d5cce8-d5996...@10.2.0.203' in 32000 ms (Method: INVITE) Sending to 10.2.0.203 : 5060 (no NAT) Using INVITE request as basis request - 001d45b6-1d490087-44d5cce8-d5996...@10.2.0.203 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 10.2.0.203:29422 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 101 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.2.0.203:29422 Looking for Voicemail in internal (domain 10.2.0.2) list_route: hop: sip:1...@10.2.0.203:5060;transport=udp cworks-phones1*CLI --- Transmitting (no NAT) to 10.2.0.203:5060 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.2.0.203:5060;branch=z9hG4bKddc0a0b8;received=10.2.0.203 From: Jim Felderman sip:1...@10.2.0.2;tag=001d45b61d4906943a6bb290-9435a462 To: sip:voicem...@10.2.0.2 Call-ID: 001d45b6-1d490087-44d5cce8-d5996...@10.2.0.203 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:voicem...@10.2.0.2 Content-Length: 0 Audio is at 10.2.0.2 port 12088 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP --- Reliably Transmitting (no NAT) to 10.2.0.203:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 10.2.0.203:5060;branch=z9hG4bKddc0a0b8;received=10.2.0.203 From: Jim Felderman sip:1...@10.2.0.2;tag=001d45b61d4906943a6bb290-9435a462 To: sip:voicem...@10.2.0.2;tag=as58c8eec4 Call-ID: 001d45b6-1d490087-44d5cce8-d5996...@10.2.0.203 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:voicem...@10.2.0.2 Content-Type: application/sdp Content-Length: 256 v=0 o=root 27452 27452 IN IP4 10.2.0.2 s=session c=IN IP4 10.2.0.2 t=0 0 m=audio 12088 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv [Jan 19 14:33:01] NOTICE[15644]: sched.c:220 ast_sched_add_variable: Scheduled event in 0 ms? Sending to 10.2.0.203
Re: [asterisk-users] dead sip channel
hi,
try to set the rtptimeout value in sip.conf to a resonable value - so
asterisk will kill the channels if it does not receive rtp traffic for
the specified time
regards,
Wolfgang
Jerry Geis schrieb:
I have ran into a case using 1.4.22 where a SIP call to an asterisk
client (running a slow PC) to ALSA
does not hangup the call when it is done. The server is using call files
to initiate the call, the client answers on
the ALSA port, the server plays the message and hangs up.
I found that SOMETIMES -its hard to recreate - that the slow pc keeps
the SIP channel active. further calls in
are getting a busy signal and the one call is NEVER hung up.
How can I detect this and hang up the channel on the slow PC. I verified
on the server that it thinks NO calls are active.
my context looks like this:
[mycontext]
exten = s,1,ChanIsAvail(Console/Dsp)
exten = s,n,GotoIf($[${AVAILCHAN} = ]?smvoice-busy,s,1)
exten = s,n,Playback(beep)
exten = s,n,Dial(Console/dsp)
exten = s,n,Hangup
[smvoice-busy]
exten = s,1,playtones(busy)
exten = s,1,wait(10)
exten = s,1,Hangup
Thanks,
Jerry
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Re: [asterisk-users] dead sip channel
hi, try to set the rtptimeout value in sip.conf to a resonable value - so asterisk will kill the channels if it does not receive rtp traffic for the specified time regards, Wolfgang I uncommeted the rtptimeout=60 value in sip.conf and did a reload. It still hasnt dropped the dead call after a couple minutes now... Do I have to stop and start again? Was hoping it would just drop the call and continue on. Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dead sip channel
Jerry Geis wrote: hi, try to set the rtptimeout value in sip.conf to a resonable value - so asterisk will kill the channels if it does not receive rtp traffic for the specified time regards, Wolfgang I uncommeted the rtptimeout=60 value in sip.conf and did a reload. It still hasnt dropped the dead call after a couple minutes now... Do I have to stop and start again? Was hoping it would just drop the call and continue on. Jerry Sounds like the problem is that the slow computer is no longer accepting calls after the first. Is Asterisk running on that machine as well? If so, check to see what it says about the sip channels. If not, you will need to look into the software running on that machine and try to figure out why it is either not hanging up or why it is dieing after it gets a call. -Brent ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems With Playback of Audio On SIP Only System
On Tuesday 20 January 2009 11:13:43 Kevin P. Fleming wrote: Tilghman Lesher wrote: Actually, this problem should already be fixed. It was a case of optimization with assembly code. Adding the noclobber option to the assembly should have fixed this. Well, people are still running into it as evidenced by this thread, and it's a very subtle issue to debug and resolve. What releases of Asterisk contain the fix for this, and can we get the OP of this thread to confirm that using the newer version resolved the issue for him? If so, maybe we should make an explicit announcement with more details about this specific issue so people will be more aware of it. It may be another set of code entirely, then: http://svn.digium.com/view/asterisk/branches/1.4/codecs/gsm/inc/private.h?r1=43376r2=111856 -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with TDM808
Dear List, I have just installed a Digium TDM808 (8 fxo port) on an Asterisk 1.6.3. When I try making a call with a .call file, the call goes straight to the dialplan and start executing the dialplan even before the called party has pick up. Anybody knows why by any chance? Any help would be appreciated. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Using centos and kickstart to build a minimum installation
Using centos 5.2, I want to use a kickstart file to select packages in order to have an unattended install onto a bare-metal server. Is there any way of finding out which packages I need to compile, build and run asterisk ? I generally want to build all modules in asterisk and the wct4xxp zaptel module I want to be able to not select any groups in the kickstart file, but only select individual packages so as to minimise the footprint and installation time ;) Anyone done anything similar before that would care to share ? Julian __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using centos and kickstart to build a minimum installation
On 20 Jan 2009, at 20:58, Julian Lyndon-Smith wrote: Using centos 5.2, I want to use a kickstart file to select packages in order to have an unattended install onto a bare-metal server. Is there any way of finding out which packages I need to compile, build and run asterisk ? I generally want to build all modules in asterisk and the wct4xxp zaptel module I want to be able to not select any groups in the kickstart file, but only select individual packages so as to minimise the footprint and installation time ;) Anyone done anything similar before that would care to share ? Just strip down trixbox / pbx in a flash etc. Should give you a fairly good idea. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems With Playback of Audio On SIP Only System
Brian Alexander wrote: Mark, Thanks - that was the problem I was having. Is there somewhere I could have looked to have discovered the problem on my own? I would never have guessed that on my own and my searches had not found it either. Thanks again, -Brian In this particular case, I know of a bug report being reported on the Asterisk bug tracker (it has since been closed, though). I also think that it has been discussed on this mailing list before, too. The thing that makes it difficult to track is the fact that to you, it just sounded like garbled audio, so that's probably what you searched for. There have probably been hundreds of threads on that subject on this list, so filtering through it all is not easy. Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Timestamp on voice mail messages is based on wrong timezone
running asterisk 1.4.13... I've noticed with SOME email clients that the timestamp reported from voicemail is 5 hours off (difference of EST vs UTC). That is, a voicemail received at 15:17:59 EST is sent via email with a Date: header of 10:17:59 - 0500. An email sent through normal means (mail client) from the asterisk server reports the correct time, so it's definitely something inside Asterisk that is inserting the wrong Date stamp. Most email clients seem to display messages by the date RECEIVED, as opposed to date SENT, but naturally, I am the silly one who views email by date SENT, so my voicemails get dropped back 5 hours (which in many cases is no longer on the first screen of my email index). Following are some of the applicable email headers showing this situation Notice the time stamps highlighted. X-MimeOLE: Produced By Microsoft Exchange V6.5 Received: from chi.hcst.net ([192.168.254.230]) by hcst.com with Microsoft SMTPSVC(6.0.3790.3959); Tue, 20 Jan 2009 15:14:00 -0500 MIME-Version: 1.0 Content-Type: multipart/mixed; boundary=_=_NextPart_001_01C97B3B.A1867400 Received: from asterisk-bvr.hcst.com (asterisk-bvr.hcst.com [192.52.183.237]) by chi.hcst.net (8.13.6/8.13.6) with ESMTP id n0KKE0kn024700 for barry.hass...@hcst.com; Tue, 20 Jan 2009 15:14:00 -0500 Received: from asterisk-bvr.hcst.com (localhost [127.0.0.1]) by asterisk-bvr.hcst.com (8.14.0/8.14.0) with ESMTP id n0KKE1jd003654 for barry.hass...@hcst.com; Tue, 20 Jan 2009 15:14:01 -0500 Received: (from r...@localhost) by asterisk-bvr.hcst.com (8.14.0/8.13.6/Submit) id n0KKE0vX003643; Tue, 20 Jan 2009 15:14:00 -0500 Return-Path: r...@asterisk-bvr.hcst.com X-OriginalArrivalTime: 20 Jan 2009 20:14:00.0942 (UTC) FILETIME=[A21630E0:01C97B3B] X-Asterisk-CallerID: 2311 X-Asterisk-CallerIDName: Barry D. Hassler Content-class: urn:content-classes:message Subject: [PBX] New message 1 in mailbox 2302 Date: Tue, 20 Jan 2009 10:14:00 -0500 Message-ID: asterisk-1-745348593-2302-8...@asterisk-bvr X-MS-Has-Attach: yes X-MS-TNEF-Correlator: Thread-Topic: [PBX] New message 1 in mailbox 2302 thread-index: Acl7O6I+TAESdG+3RlmwhFhRW9rXEg== From: HCST Asterisk PBX voicem...@asterisk-bvr.hcst.com To: Barry D. Hassler barry.hass...@hcst.com X-Evolution-Source: exchange://hass...@zeta.hcst.com/ -- Barry D. Hassler President, HCST http://www.hcst.net/ 937-427-9000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why does Asterisk not hangup?
Klaus Darilion wrote: Ok. Just for the info to others: the 10 seconds are hardcoded in pbx.c What line in which version? -- Sean Bright sean.bri...@gmail.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with TDM808
On Tue, 2009-01-20 at 15:30 -0500, Pascal Bruno wrote: I have just installed a Digium TDM808 (8 fxo port) on an Asterisk 1.6.3. When I try making a call with a .call file, the call goes straight to the dialplan and start executing the dialplan even before the called party has pick up. Anybody knows why by any chance? That's not a problem with the TDM800 card... it's just a side-effect of analog signaling. For analog calls, the central office doesn't give any type of signal when the far end has answered the call, so Asterisk has no way of knowing when that happens. For that reason, Asterisk immediately treats any outgoing analog call as having been answered. -- Jared Smith Digium, Inc. | Training Manager ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using centos and kickstart to build a minimum installation
Steve Howes wrote: On 20 Jan 2009, at 20:58, Julian Lyndon-Smith wrote: Using centos 5.2, I want to use a kickstart file to select packages in order to have an unattended install onto a bare-metal server. Is there any way of finding out which packages I need to compile, build and run asterisk ? I generally want to build all modules in asterisk and the wct4xxp zaptel module I want to be able to not select any groups in the kickstart file, but only select individual packages so as to minimise the footprint and installation time ;) Anyone done anything similar before that would care to share ? Just strip down trixbox / pbx in a flash etc. Don't they have pre-built asterisk binaries ? I need to be able to compile asterisk, as there are a couple of custom programs Should give you a fairly good idea. Thanks for the tip - I will look into that idea. Julian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with TDM808
Is there any way of going around this??? Any tricks, configuration hacks?? On Tue, Jan 20, 2009 at 4:39 PM, Jared Smith jsm...@digium.com wrote: On Tue, 2009-01-20 at 15:30 -0500, Pascal Bruno wrote: I have just installed a Digium TDM808 (8 fxo port) on an Asterisk 1.6.3. When I try making a call with a .call file, the call goes straight to the dialplan and start executing the dialplan even before the called party has pick up. Anybody knows why by any chance? That's not a problem with the TDM800 card... it's just a side-effect of analog signaling. For analog calls, the central office doesn't give any type of signal when the far end has answered the call, so Asterisk has no way of knowing when that happens. For that reason, Asterisk immediately treats any outgoing analog call as having been answered. -- Jared Smith Digium, Inc. | Training Manager ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] extensions.conf -- what to do when command throws errors?
Hi, all. I've got app_rxfax going and nicely receiving a fax, which I
then throw to a script, and have it convert it to a PDF and mail it.
Works great... a lot of the time. But a fair bit of the time, rxfax
throws errors, and extensions.conf seems never to invoke my script. Here
are the pertinent lines:
exten = _6403,n,rxfax(${FAXFILE})
exten = _6403,n,System(/usr/local/bin/fax-sender.py ${FAXFILE})
Now the problem here is that the .TIF file is received just fine, so,
errors or no, I'd like to get to the script. Instead, I get this:
...
[5410] /tmp/spandsp/agx-ast-addons/trunk/app_rxfax.c: Channel T30 DONE 0.
[5410] /tmp/spandsp/agx-ast-addons/trunk/app_rxfax.c: Channel T30 DONE 0.
[5410] logger.c: == Auto fallthrough, channel 'Zap/4-1' status is 'UNKNOWN'
[5410] chan_zap.c: Set option AUDIO MODE, value: ON(1) on Zap/4-1
[5410] chan_zap.c: Not yet hungup... Calling hangup once with icause, and
clearing call
[5410] chan_zap.c: Set option AUDIO MODE, value: OFF(0) on Zap/4-1
[5410] logger.c: -- Hungup 'Zap/4-1'
...
In an ideal world, getting rid of the Channel T30 DONE 0 errors would
be great, but I'll take run-the-script-when-it's-done-regardless, instead.
Note, however, that I can't just call it from extension i, because I need
to pass it information, and don't want it executing on errant voice calls.
Suggestions?
Thanks!
-Ken
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Re: [asterisk-users] extensions.conf -- what to do when command throwserrors?
It seems to me that $FAXFILE lives in the 6403 context, not the n, so it
would still be useable. If this isn't the case, I'd call an AGI script in
the I context.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ken D'Ambrosio
Sent: Tuesday, January 20, 2009 4:03 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] extensions.conf -- what to do when command
throwserrors?
Hi, all. I've got app_rxfax going and nicely receiving a fax, which I
then throw to a script, and have it convert it to a PDF and mail it.
Works great... a lot of the time. But a fair bit of the time, rxfax
throws errors, and extensions.conf seems never to invoke my script. Here
are the pertinent lines:
exten = _6403,n,rxfax(${FAXFILE})
exten = _6403,n,System(/usr/local/bin/fax-sender.py ${FAXFILE})
Now the problem here is that the .TIF file is received just fine, so,
errors or no, I'd like to get to the script. Instead, I get this:
...
[5410] /tmp/spandsp/agx-ast-addons/trunk/app_rxfax.c: Channel T30 DONE 0.
[5410] /tmp/spandsp/agx-ast-addons/trunk/app_rxfax.c: Channel T30 DONE 0.
[5410] logger.c: == Auto fallthrough, channel 'Zap/4-1' status is 'UNKNOWN'
[5410] chan_zap.c: Set option AUDIO MODE, value: ON(1) on Zap/4-1
[5410] chan_zap.c: Not yet hungup... Calling hangup once with icause, and
clearing call
[5410] chan_zap.c: Set option AUDIO MODE, value: OFF(0) on Zap/4-1
[5410] logger.c: -- Hungup 'Zap/4-1'
...
In an ideal world, getting rid of the Channel T30 DONE 0 errors would
be great, but I'll take run-the-script-when-it's-done-regardless, instead.
Note, however, that I can't just call it from extension i, because I need
to pass it information, and don't want it executing on errant voice calls.
Suggestions?
Thanks!
-Ken
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Re: [asterisk-users] Using centos and kickstart to build a minimum installation
On Tue, 20 Jan 2009, Julian Lyndon-Smith wrote:
Using centos 5.2,
I want to use a kickstart file to select packages in order to have an
unattended install onto a bare-metal server.
I started (but have not finished) a live pen drive install. I've snipped out
some bits specific to this project, but you're welcome to crib off it.
#
# Filename: example.ks
#
# Version:000
#
# Edit date: 2008-06-11
#
# Facility: example.com
#
# Abstract: This kickstart script creates the CD image.
#
# Environment:Unix, sh
#
# Author: Steven L. Edwards
#
# Modified by
#
# 000 2008-06-11 SLE Create.
langen_US.UTF-8
keyboardus
network --bootproto=dhcp --device=eth0
network --bootproto=dhcp --device=eth1
rootpw example
firewall--enabled --port=22:tcp
authconfig --enablemd5 --enableshadow
selinux --disabled
timezone--utc America/Los_Angeles
# US english
# US keyboard
# use DHCP to set up networking
# set the password for root
# disable the firewall
# use MD5 and shadow passwords
# list the repositories
repo --name=base --baseurl=http://localhost/yum/base
repo --name=extras
--baseurl=http://ftp.telus.net/pub/centos/5/extras/$basearch
repo --name=live
--baseurl=http://www.nanotechnologies.qc.ca/propos/linux/centos-live/$basear
ch/live
repo --name=updates --baseurl=http://localhost/yum/updates
# repo --name=base
--baseurl=http://ftp.telus.net/pub/centos/5/os/$basearch
# repo --name=base
--baseurl=http://isoredirect.centos.org/centos/5/os/$basearch
# repo --name=extras
--baseurl=http://isoredirect.centos.org/centos/5/extras/$basearch
# repo --name=updates
--baseurl=http://ftp.telus.net/pub/centos/5/updates/$basearch
# repo --name=updates
--baseurl=http://isoredirect.centos.org/centos/5/updates/$basearch
# packages
%packages
authconfig
bash
chkconfig
comps-extras
kernel
passwd
policycoreutils
rootfiles
syslinux
xkeyboard-config
# more packages
dhcdbd
dhclient
dhcp
emacs-nox
lynx
mailx
memtest86+
ntp
openssh
openssh-clients
openssh-server
wget
%post
echo Post package customization in chroot
PATH=$PATH:/sbin/
# housekeeping
TIMESTAMP=$(date +Created by example.ks on %F at %T)
# clean up install
rm --force /root/.cshrc
rm --force /root/.tcshrc
echo Configuring the timezone
(
printf # %s\n\n $TIMESTAMP
printf \tARC=false\n
printf \tUTC=true\n
printf \tZONE=\America/Los_Angeles\\n
printf \n# (end of /etc/sysconfig/clock)\n
) /etc/sysconfig/clock
cp /usr/share/zoneinfo/America/Los_Angeles /etc/localtime
chmod u=rw,g=r,o=r /etc/localtime
echo Configuring syslogging
(
printf # %s\n\n $TIMESTAMP
printf # log everything to messages\n
printf \t*.*\t\t\t\t/var/log/messages\n
printf \n# (end of /etc/syslog.conf)\n
) /etc/syslog.conf
echo Configure networking
(
printf # %s\n\n $TIMESTAMP
# printf \tDOMAIN=${DOMAINNAME}\n
# printf \tGATEWAY=${GATEWAY}\n
printf \tHOSTNAME=localhost.localdomain\n
printf \tNETWORKING=yes\n
printf \tNETWORKING_IPV6=no\n
printf \n# (end of /etc/sysconfig/network)\n
) /etc/sysconfig/network
echo Configure ntp
(
printf # %s\n\n $TIMESTAMP
printf \t0.pool.ntp.org\n
printf \t1.pool.ntp.org\n
printf \t2.pool.ntp.org\n
printf \t3.pool.ntp.org\n
printf \n# (end of /etc/ntp/step-tickers)\n
) /etc/ntp/step-tickers
chkconfig --add ntpd
chkconfig --level 2345 ntpd on
echo Link for using emacs as edit
rm --force /usr/local/bin/edit
ln --symbolic /usr/bin/emacs-nox /usr/local/bin/edit
%post --nochroot
echo Post package customization not in chroot
PATH=$PATH:/sbin/
PROJECT_DIR=/home/sedwards/example.com/
cp ${PROJECT_DIR}/bashrc ${INSTALL_ROOT}/etc/
cp ${PROJECT_DIR}/emacs ${INSTALL_ROOT}/root/.emacs
cp ${PROJECT_DIR}/asterisk.tar.gz ${INSTALL_ROOT}/root/
%post
echo Post package customization in chroot
PATH=$PATH:/sbin/
tar\
--gzip\
--directory /\
--extract\
--file /root/asterisk.tar.gz\
${END_OF_LIST}
chkconfig --add asterisk
chkconfig --add zaptel
# this is where --shell kicks in -- at the end
Re: [asterisk-users] extensions.conf -- what to do when command throws errors?
On Tue, 20 Jan 2009, Ken D'Ambrosio wrote:
Hi, all. I've got app_rxfax going and nicely receiving a fax, which I
then throw to a script, and have it convert it to a PDF and mail it.
Works great... a lot of the time. But a fair bit of the time, rxfax
throws errors, and extensions.conf seems never to invoke my script. Here
are the pertinent lines:
exten = _6403,n,rxfax(${FAXFILE})
exten = _6403,n,System(/usr/local/bin/fax-sender.py ${FAXFILE})
What else can match _6403 other than 6403? Wouldn't exten = 6403,n,...
be better?
[5410] logger.c: == Auto fallthrough, channel 'Zap/4-1' status is 'UNKNOWN'
Auto fallthrough happens when Asterisk wants to go to the next priority
and there isn't one.
Does [show dialplan|dialplan show] context-containing-rxfax look like?
Any chance rxfax is trying to do a jump +101 on you?
Thanks in advance,
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] extensions.conf -- what to do when command throwserrors?
On Tue, 20 Jan 2009, Danny Nicholas wrote:
It seems to me that $FAXFILE lives in the 6403 context, not the n, so it
would still be useable. If this isn't the case, I'd call an AGI script
in the I context.
$FAXFILE is a channel variable. It is useable in any context, extension
and priority the channel visits. The example does not show what the
context is. 6403 is the extension. n is the priority (but we don't know
its value).
The error is not invalid, it is auto fallthrough.
A show dialplan may help.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ken D'Ambrosio
Sent: Tuesday, January 20, 2009 4:03 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] extensions.conf -- what to do when command
throwserrors?
Hi, all. I've got app_rxfax going and nicely receiving a fax, which I
then throw to a script, and have it convert it to a PDF and mail it.
Works great... a lot of the time. But a fair bit of the time, rxfax
throws errors, and extensions.conf seems never to invoke my script. Here
are the pertinent lines:
exten = _6403,n,rxfax(${FAXFILE})
exten = _6403,n,System(/usr/local/bin/fax-sender.py ${FAXFILE})
Now the problem here is that the .TIF file is received just fine, so,
errors or no, I'd like to get to the script. Instead, I get this:
...
[5410] /tmp/spandsp/agx-ast-addons/trunk/app_rxfax.c: Channel T30 DONE 0.
[5410] /tmp/spandsp/agx-ast-addons/trunk/app_rxfax.c: Channel T30 DONE 0.
[5410] logger.c: == Auto fallthrough, channel 'Zap/4-1' status is 'UNKNOWN'
[5410] chan_zap.c: Set option AUDIO MODE, value: ON(1) on Zap/4-1
[5410] chan_zap.c: Not yet hungup... Calling hangup once with icause, and
clearing call
[5410] chan_zap.c: Set option AUDIO MODE, value: OFF(0) on Zap/4-1
[5410] logger.c: -- Hungup 'Zap/4-1'
...
In an ideal world, getting rid of the Channel T30 DONE 0 errors would
be great, but I'll take run-the-script-when-it's-done-regardless, instead.
Note, however, that I can't just call it from extension i, because I need
to pass it information, and don't want it executing on errant voice calls.
Suggestions?
Thanks!
-Ken
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Thanks in advance,
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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[asterisk-users] PAP2T provisioning
Anyone have an example XML file for the PAP2T? Cheers, j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with TDM808
If your provider provides any signalling to indicate answer, such as a polarity reversal, this could be detected easily... ; Use a polarity reversal to mark when a outgoing call is answered by the ; remote party. ; ;answeronpolarityswitch=yes This isn't very common though... alternatively, there is the 'HIGHLY EXPERIMENTAL' call progress detection... ; On trunk interfaces (FXS) it can be useful to attempt to follow the progress ; of a call through RINGING, BUSY, and ANSWERING. If turned on, call ; progress attempts to determine answer, busy, and ringing on phone lines. ; This feature is HIGHLY EXPERIMENTAL and can easily detect false answers, ; so don't count on it being very accurate. ; ; Few zones are supported at the time of this writing, but may be selected ; with progzone. ; ; progzone also affects the pattern used for buzydetect (unless ; busypattern is set explicitly). The possible values are: ; us (default) ; ca (alias for 'us') ; cr (Costa Rica) ; br (Brazil, alias for 'cr') ; uk ; ; This feature can also easily detect false hangups. The symptoms of this is ; being disconnected in the middle of a call for no reason. ; ;callprogress=yes ;progzone=uk Obviously far from ideal, and at least, where I am, unworkable due to the way that all the telcos have got into providing musical ringing... The only real solution is to go digital... d 2009/1/21 Pascal Bruno tipas...@gmail.com Is there any way of going around this??? Any tricks, configuration hacks?? On Tue, Jan 20, 2009 at 4:39 PM, Jared Smith jsm...@digium.com wrote: On Tue, 2009-01-20 at 15:30 -0500, Pascal Bruno wrote: I have just installed a Digium TDM808 (8 fxo port) on an Asterisk 1.6.3. When I try making a call with a .call file, the call goes straight to the dialplan and start executing the dialplan even before the called party has pick up. Anybody knows why by any chance? That's not a problem with the TDM800 card... it's just a side-effect of analog signaling. For analog calls, the central office doesn't give any type of signal when the far end has answered the call, so Asterisk has no way of knowing when that happens. For that reason, Asterisk immediately treats any outgoing analog call as having been answered. -- Jared Smith Digium, Inc. | Training Manager ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PAP2T provisioning
I'm not sure if this trick will work with this device, but I was able to pull down a spa8000's config by connecting to: http://ipaddress/admin/spacfg.xml Tom -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Tuesday, January 20, 2009 6:04 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] PAP2T provisioning Anyone have an example XML file for the PAP2T? Cheers, j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] open-source ZRTP implementation (was: Re: [somewhat OT] seeking ideas/input for my thesis)
On Jan 20, 2009, at 8:58 AM, Philipp Kempgen wrote: John Todd schrieb: So here are some projects you might look into: - open-source ZRTP implementation libzrtpcpp is open source. GPL. http://www.gnu.org/software/ccrtp/ Philipp Kempgen Excellent! I didn't know that existed. Thanks for the link. JT --- John Todd email:jt...@digium.com Digium, Inc. | Asterisk Open Source Community Director 445 Jan Davis Drive NW - Huntsville AL 35806 - USA direct: +1-256-428-6083 http://www.digium.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with TDM808
With: callprogress=yes and progzone=us it works fine for not, not 100% percent because in some calls,it takes like 3-4 seconds before executing dialplan, which is not bad not to say normal. But most calls are ok. And when I tried with answeronpolarityswitch=yes it doesnt do dialplan at all, like the call was never picked up. Thanks for your help! On Tue, Jan 20, 2009 at 6:53 PM, D Tucny d...@tucny.com wrote: If your provider provides any signalling to indicate answer, such as a polarity reversal, this could be detected easily... ; Use a polarity reversal to mark when a outgoing call is answered by the ; remote party. ; ;answeronpolarityswitch=yes This isn't very common though... alternatively, there is the 'HIGHLY EXPERIMENTAL' call progress detection... ; On trunk interfaces (FXS) it can be useful to attempt to follow the progress ; of a call through RINGING, BUSY, and ANSWERING. If turned on, call ; progress attempts to determine answer, busy, and ringing on phone lines. ; This feature is HIGHLY EXPERIMENTAL and can easily detect false answers, ; so don't count on it being very accurate. ; ; Few zones are supported at the time of this writing, but may be selected ; with progzone. ; ; progzone also affects the pattern used for buzydetect (unless ; busypattern is set explicitly). The possible values are: ; us (default) ; ca (alias for 'us') ; cr (Costa Rica) ; br (Brazil, alias for 'cr') ; uk ; ; This feature can also easily detect false hangups. The symptoms of this is ; being disconnected in the middle of a call for no reason. ; ;callprogress=yes ;progzone=uk Obviously far from ideal, and at least, where I am, unworkable due to the way that all the telcos have got into providing musical ringing... The only real solution is to go digital... d 2009/1/21 Pascal Bruno tipas...@gmail.com Is there any way of going around this??? Any tricks, configuration hacks?? On Tue, Jan 20, 2009 at 4:39 PM, Jared Smith jsm...@digium.com wrote: On Tue, 2009-01-20 at 15:30 -0500, Pascal Bruno wrote: I have just installed a Digium TDM808 (8 fxo port) on an Asterisk 1.6.3. When I try making a call with a .call file, the call goes straight to the dialplan and start executing the dialplan even before the called party has pick up. Anybody knows why by any chance? That's not a problem with the TDM800 card... it's just a side-effect of analog signaling. For analog calls, the central office doesn't give any type of signal when the far end has answered the call, so Asterisk has no way of knowing when that happens. For that reason, Asterisk immediately treats any outgoing analog call as having been answered. -- Jared Smith Digium, Inc. | Training Manager ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk queues sending calls to members on the phone
Hello, We're using Asterisk to manage call queues. Queue members are connected via IAX2 using the Zoiper softphone, and Zoiper is configured with 2 lines. We're finding that calls are routed to queue members even when they are on the phone, on their softphone's other line. For example, if a queue member makes an outgoing call on line 1 or is handling a queue call on line 1, the queue will often route calls to them on line 2. When this happens, queue show shows the member's status in the queue as In use, so Asterisk seems to know that this member is busy. We are using Asterisk 1.4.18, configuring our IAX users and queues with the realtime extension. Is this expected behavior, and if so is there any way to turn it off? Or is this unusual behavior, and if so any tips on troubleshooting it? Does anybody happen to know how Asterisk determines whether a queue member is available to have a call routed their way? Thanks! Scott. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk queues sending calls to members on the phone
Hi, take a look at the rininuse setting - if set to yes than you have the behaviour you described. If set to no - then you will get nearly the behaviour you want. Try upgrading to 1.4.22 if using queues - some concerns regarding the member status have been fixed there. regards, Wolfgang Scott Gifford schrieb: Hello, We're using Asterisk to manage call queues. Queue members are connected via IAX2 using the Zoiper softphone, and Zoiper is configured with 2 lines. We're finding that calls are routed to queue members even when they are on the phone, on their softphone's other line. For example, if a queue member makes an outgoing call on line 1 or is handling a queue call on line 1, the queue will often route calls to them on line 2. When this happens, queue show shows the member's status in the queue as In use, so Asterisk seems to know that this member is busy. We are using Asterisk 1.4.18, configuring our IAX users and queues with the realtime extension. Is this expected behavior, and if so is there any way to turn it off? Or is this unusual behavior, and if so any tips on troubleshooting it? Does anybody happen to know how Asterisk determines whether a queue member is available to have a call routed their way? Thanks! Scott. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Job description
JOB TITLE: VOIP SOFTWARE DEVELOPER LOCATION: Zurich/Switzerland JOB TYPE: Full time COMPANY: Peoplefone AG is a Pan-European VoIP provider present in 4 countries and headquartered in Zurich/Switzerland ( www.peoplefone.com). JOB DESCRIPTION: - Manage together with CEO/CTO the geographical expansion on IT level - Develop and maintain the VoIP Platform (OPENSER, billing software,…) - Create new VoIP features for the clients - Ensure excellent quality of VoIP service and appropriate security level - Supervise interactions with webmasters for the website development - Provide technicians and customer service of affiliates with technical support REQUIREMENTS: - Graduation from college or university with a Bachelor's degree(preferably IT) - Practical knowledge of C, C++, PHP and JAVA - Practical knowledge of database like Mysql and Postgresql - Linux experience - Experience with tools for Network analysis like Ethereal - VoIP knowledge with experience on OPENSER/ASTERISK platform - Fluency in English - Ability to interact with individuals and groups at all levels - Detail oriented and analytical - Strong verbal and written communication skills +41 44 553 20 01 (Office CH) +43 1 266 00 50 (Office A) +48 22 311 78 35 (Offfice PL) Alfred Escher-Strasse 26 Pottendorfer Str. 25-27 Al. Jerozolimskie 179 8002 Zurich/Switzerland 1120 Wien/Austria 02-222 Warsaw/Poland www.peoplefone.ch www.peoplefone.at * www.plfon.pl* ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why does Asterisk not hangup?
Sean Bright schrieb: Klaus Darilion wrote: Ok. Just for the info to others: the 10 seconds are hardcoded in pbx.c What line in which version? at least in 1.4.22: grep -r -A 10 -B 10 Auto fallthrough * klaus ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
