RE : [Asterisk-Users] codecs order and so on
Just have a lok at this config : [general] Disallow=all Allow=g729 Allow=ulaw [pstn] Disallow=all Allow=g729 [zap] Disallow=all Allow=ulaw In extensions.conf, I change the context for each call, Asterisk doesn't care of codecs in contexts, it uses the order of general... Could be good to have Ssterisk making a match between codecs in General and the context used to make a call But definitiely, Asterisk choose g729 even if I am in the zap context Any idea, help is welcome. Olivier -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Moises Silva Envoyé : mardi 10 janvier 2006 22:51 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [Asterisk-Users] codecs order and so on Doing in the console show translation i can see that it seems not be possible to translate from any to g729 codec, or from g729 to any. So, let me try to find a reason for this. When you have first allow=g729 (preferred codec) all the calls to pstn providers work because the phones and asterisk agree to use g729, so no codec translation is done. all the calls to and from fxo fails because no translation can be made from ULAW to g729, and from g729 (phones) to ulaw. then asterisk is not smart enough to realize that can ask the phones to use ulaw (i assume the phones support ulaw) to not use translation to call the fxo??? When you have first allow=ulaw (prefered codec) all the calls to and from fxo works because the prefered codec is ulaw, then from fxo to phones using ulaw, no codec translation is made all the calls to pstn providers fails, again, because it seems asterisk gives preference to ulaw codec (the first list codec) so, the phones use ulaw, and is not possible to translate ulaw to g729 and viceversa?? im interested in knowing the reason too, any guidelines? regards On 1/10/06, Olivier Taylor [EMAIL PROTECTED] wrote: The problem : an asterisk box with 2 fxo First fxo just receive calls from pstn (ulaw) Second fxo receive and send call to mobile network thru a sipbox(ulaw) Calls to pstn are sent to a pstn provider accepting only g729 Internal calls doesn't care of codecs All Uas have g729 (g729 is then pass-thru when needed) All Uas have ulaw(of course) If I have in [general] disallow=all allow=g729 allow=ulaw In this case: all calls to pstn providers works all calls to and from fxo fails because of : No translator path exists for If I have in [general] disallow= all allow= ulaw allow= g729 In this case: all calls to and from fxo works all calls to pstn providers fails because of : No translator path exists for ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VizuFon CIP-4500 with Asterisk through SIP
Ian White wrote: Make sure you have a recent copy of the firmware. There was a bug preventing registrations from succeeding until Nov 08 2005 and newer firmwares. Where can I find the firmware? -- Best regards, Bartosz Piec ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SOLVED: Hung Zap channels connected to old key system
Philip Edelbrock wrote: We've got a Toshiba DK system w/ analog ports that went to a voicemail server. I swapped in an Asterisk box with a Digium 4-port fxo card. It /almost/ worked perfectly. The problem is that Zap channels never hang up. They have to time out. I set up MeetMe, but all Zap channels hung forever. Very annoying. Same thing for FXO-to-FXO bridges. I figured out today why and fixed it. Some proprietary voicemail systems (and probably tie-lines, too) like to use DTMF tones instead of standard ground/loop/kewl whatever signaling. Our key system was programmed to use such DTMF tones instead of the usual analog signaling on those ports. (I think it was program 31 on our Toshiba DK40i) Asterisk of course ignored those, but the other systems used those for line signaling (including our previous 3rd party system). Amusingly, I know now why for years we kept hearing loud DTMF tones when our branch office picked up their phones. Their system, too, was configured to have those analog lines be connected to a voicemail system and not to a FXO port on a T1 CSU. I've just come across a similar problem with a more modern SpliceCom hybrid PBX. We have an * system connected to two analog (FXS) ports via a couple of Sipura SPA3000 ATAs, and we thought the Sipuras were failing to detect call termination. Turns out that the default behaviour of an FXS port on this PBX is *never* to hang up. jd -- John Daragon [EMAIL PROTECTED] argv[0] limited (Asterisk implementation consultancy) Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CHAN_CAPI problem
Thank you very much for your attention; Here is what you asked for: *** asteriskge03*CLI set verbose 15 Verbosity is at least 15 asteriskge03*CLI capi debug CAPI Debugging Enabled asteriskge03*CLI capi info Contr1: 2 B channels total, 2 B channels free. CONNECT_IND ID=002 #0x2011 LEN=0047 Controller/PLCI/NCCI= 0x101 CIPValue= 0x1 CalledPartyNumber = a1104695467 CallingPartyNumber = 21 81108680550 CalledPartySubaddress = default CallingPartySubaddress = default BC = 80 90 a3 LLC = default HLC = default AdditionalInfo = default -- CONNECT_IND (PLCI=0x101,DID=104695467,CID=108680550,CIP=0x1,CONTROLLER=0x1) BRI1: msn='*' DNID='104695467' MSN == BRI1: Incoming call '0108680550' - '104695467' INFO_IND ID=002 #0x2012 LEN=0017 Controller/PLCI/NCCI= 0x101 InfoNumber = 0x1e InfoElement = 82 81 INFO_RESP ID=002 #0x2012 LEN=0012 Controller/PLCI/NCCI= 0x101 -- BRI1: info element PI 82 81 BRI1: Not end-to-end ISDN INFO_IND ID=002 #0x2013 LEN=0025 Controller/PLCI/NCCI= 0x101 InfoNumber = 0x70 InfoElement = a1104695467 INFO_RESP ID=002 #0x2013 LEN=0012 Controller/PLCI/NCCI= 0x101 -- BRI1: info element CALLED PARTY NUMBER BRI1: INFO_IND DID digits not used in this state. INFO_IND ID=002 #0x2014 LEN=0016 Controller/PLCI/NCCI= 0x101 InfoNumber = 0x18 InfoElement = 89 INFO_RESP ID=002 #0x2014 LEN=0012 Controller/PLCI/NCCI= 0x101 -- BRI1: info element CHANNEL IDENTIFICATION 89 DISCONNECT_IND ID=002 #0x2017 LEN=0014 Controller/PLCI/NCCI= 0x101 Reason = 0x0 DISCONNECT_RESP ID=002 #0x2017 LEN=0012 Controller/PLCI/NCCI= 0x101 -- BRI1: DISCONNECT_IND on incoming without pbx, doing hangup. == BRI1: CAPI Hangingup == BRI1: Interface cleanup PLCI=0x101 CAPI devicestate requested for BRI1/104695467 *** The lines: BRI1: DISCONNECT_IND on incoming without pbx, doing hangup. == BRI1: CAPI Hangingup == BRI1: Interface cleanup PLCI=0x101 CAPI devicestate requested for BRI1/104695467 appeared on the console WHILE I was still earing the ring tone on the calling phone. When I , at last, after other 4 rings, hangup the calling phone, nothing changed on the console Andrea Armin Schindler [EMAIL PROTECTED] To Sent by: Asterisk Users Mailing List - asterisk-users-bo Non-Commercial Discussion [EMAIL PROTECTED] asterisk-users@lists.digium.com m.com cc Subject 10/01/2006 19.16 Re: [Asterisk-Users] CHAN_CAPI problem Please respond to Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] ists.digium.com On Tue, 10 Jan 2006 [EMAIL PROTECTED] wrote: Thank you. I already downloaded and installed it (they are dated 07-01-2006, version 0.6.3, correct ?) Yes. I maked clean, make and make install. Nothing changed, dial out perfect, dial in: (capi debug on) asteriskge03*CLI capi info Contr1: 2 B channels total, 2 B channels free. asteriskge03*CLI asteriskge03*CLI -- CONNECT_IND (PLCI=0x101,DID=104695467,CID=108680550,CIP=0x1,CONTROLLER=0x1) == BRI1: Interface cleanup PLCI=0x101 BRI1 is the name of my
Re: [Asterisk-Users] Eid Mubarak
Carlos Alperin a écrit : No, I never said that. I'm only not joking with another people believes. Well, I *am*. Believe it or not, it wasn't even disrespectful. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] video development
I would like to develop a video file player tool inside Asterisk. When calling to an extension answer and Play a video file (H264). With the applications PlayBack is not possible to give a video extension (only sound file extension). is it posible? How do u start in this development? With AGI scripts is not possible to send a video stream...(i tried to send images but with SIP channel doesnt work. I am testing with SEREyeBeam ) greetings and thanks in advance. Fran ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Eid Mubarak
Mark Phillips a écrit : It has to be said that Eid is a funny and possibly suspect celebration though. As I understand it (from one of my Muslim underlings) 3 Mad Mulahs have to look for a particular phase of the moon. When they see this phase they declare the start of Eid. They apparently get 3 nights in which to look for this moon phase. I guess my question is what happens if its cloudy on all 3 nights? You know, usually south mediteranean and middle east regions are pretty dry... so it looks like there is a good chance that you will have your kebab. I kinda like the idea of having a huge BBQ and getting your friends and neighbor to come and it eat. The only downside is that muslims can't drink alcohol, and how good would that be without a few beers? You know, the kinda stuff you need to get the goat to go down nicely. Another thing I thought about is this; If we could get the Faithfull whom are attending the Haaj this week to suddenly apply their brakes do you think they could stop the world from turning? Better yet if they all jumped into the air at once would the resultant landing knock us off off our regular orbit? No it wouldn't. Even if it was a billion people jumping simultaneously. Even if they are big 100kg guys, that's 10^11kg. The mass of the earth is around 6 x 10^24 kg. The effect of a billion people jumping would be as efficient as a fly trying to crash into a brick to move it... Talk about death to Ifidels! They could do it in one fell swoop! I wonder if Al Quaeda has spent any research money on this? Oh yeah let them do that. I could live with that form of peaceful protest. rant Al Quaeda was *mostly* bin laden, and he's probably safe and home in his rich royal Saudi family by now. They probably think of him as a little teenage troublemaker or something... he's gonna have to learn to be a rich prince of petrol like the rest of the family, it must be sooo annoying. /rant ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: New Freelance Site for Asterisk Consultants and Those who Need Projects Done
In article [EMAIL PROTECTED], Steve Totaro [EMAIL PROTECTED] wrote: Sorry if this is slightly off topic but it does pertain to Asterisk Users as well as the biz list. Also, sorry if it is a double post but the first one never made it to the list for some reason. Please test it out and let me know what you think. http://www.asteriskhelpdesk.com Steve, a couple of things I noticed: - on the contact page, you have used the domain asterisk-helpdesk.com, instead of asteriskhelpdesk.com. The version with the hyphen doesn't show up in whois. - I tried to vote, and it came back with Cookies must be enabled!. But I never disable cookies, and other cookie-using sites work fine. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CHAN_CAPI problem
There is no 'sending-complete'/'setup' info-element, please use immediate=yes in capi.conf Armin On Wed, 11 Jan 2006 [EMAIL PROTECTED] wrote: Thank you very much for your attention; Here is what you asked for: *** asteriskge03*CLI set verbose 15 Verbosity is at least 15 asteriskge03*CLI capi debug CAPI Debugging Enabled asteriskge03*CLI capi info Contr1: 2 B channels total, 2 B channels free. CONNECT_IND ID=002 #0x2011 LEN=0047 Controller/PLCI/NCCI= 0x101 CIPValue= 0x1 CalledPartyNumber = a1104695467 CallingPartyNumber = 21 81108680550 CalledPartySubaddress = default CallingPartySubaddress = default BC = 80 90 a3 LLC = default HLC = default AdditionalInfo = default -- CONNECT_IND (PLCI=0x101,DID=104695467,CID=108680550,CIP=0x1,CONTROLLER=0x1) BRI1: msn='*' DNID='104695467' MSN == BRI1: Incoming call '0108680550' - '104695467' INFO_IND ID=002 #0x2012 LEN=0017 Controller/PLCI/NCCI= 0x101 InfoNumber = 0x1e InfoElement = 82 81 INFO_RESP ID=002 #0x2012 LEN=0012 Controller/PLCI/NCCI= 0x101 -- BRI1: info element PI 82 81 BRI1: Not end-to-end ISDN INFO_IND ID=002 #0x2013 LEN=0025 Controller/PLCI/NCCI= 0x101 InfoNumber = 0x70 InfoElement = a1104695467 INFO_RESP ID=002 #0x2013 LEN=0012 Controller/PLCI/NCCI= 0x101 -- BRI1: info element CALLED PARTY NUMBER BRI1: INFO_IND DID digits not used in this state. INFO_IND ID=002 #0x2014 LEN=0016 Controller/PLCI/NCCI= 0x101 InfoNumber = 0x18 InfoElement = 89 INFO_RESP ID=002 #0x2014 LEN=0012 Controller/PLCI/NCCI= 0x101 -- BRI1: info element CHANNEL IDENTIFICATION 89 DISCONNECT_IND ID=002 #0x2017 LEN=0014 Controller/PLCI/NCCI= 0x101 Reason = 0x0 DISCONNECT_RESP ID=002 #0x2017 LEN=0012 Controller/PLCI/NCCI= 0x101 -- BRI1: DISCONNECT_IND on incoming without pbx, doing hangup. == BRI1: CAPI Hangingup == BRI1: Interface cleanup PLCI=0x101 CAPI devicestate requested for BRI1/104695467 *** The lines: BRI1: DISCONNECT_IND on incoming without pbx, doing hangup. == BRI1: CAPI Hangingup == BRI1: Interface cleanup PLCI=0x101 CAPI devicestate requested for BRI1/104695467 appeared on the console WHILE I was still earing the ring tone on the calling phone. When I , at last, after other 4 rings, hangup the calling phone, nothing changed on the console Andrea Armin Schindler [EMAIL PROTECTED] To Sent by: Asterisk Users Mailing List - asterisk-users-bo Non-Commercial Discussion [EMAIL PROTECTED] asterisk-users@lists.digium.com m.com cc Subject 10/01/2006 19.16 Re: [Asterisk-Users] CHAN_CAPI problem Please respond to Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] ists.digium.com On Tue, 10 Jan 2006 [EMAIL PROTECTED] wrote: Thank you. I already downloaded and installed it (they are dated 07-01-2006, version 0.6.3, correct ?) Yes. I maked clean, make and make install. Nothing changed, dial out perfect, dial in: (capi debug on)
[Asterisk-Users] Asterisk REGISTERs
Hi List, Is there a way to have Asterisk remember which agents are registered to it using a MySQL database rather than in memory? It would help with high availability / clustering scenarios. It also means you could restart the server without loosing this information... Cheers, Jean-Michel. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Transfer to meetme on different server
Title: Transfer to meetme on different server Hi there I am using IAX2 based phones and am wondering if the following is possible: 1. User registers with Server 1 2. User dials an extension on Server 1 3. Extension transfers call to an extension on Server 2, which transfers the call to a Meetme conference. If this is possible, would anyone be able to give me an idea how this can be done? Many thanks Steven ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CHAN_CAPI problem
Ok it solved my problem (immediate=yes in capi.conf) !!! Here is the console log *** CONNECT_IND ID=002 #0x201f LEN=0047 Controller/PLCI/NCCI= 0x101 CIPValue= 0x1 CalledPartyNumber = a1104695467 CallingPartyNumber = 21 81108680550 CalledPartySubaddress = default CallingPartySubaddress = default BC = 80 90 a3 LLC = default HLC = default AdditionalInfo = default -- CONNECT_IND (PLCI=0x101,DID=104695467,CID=108680550,CIP=0x1,CONTROLLER=0x1) BRI1: msn='*' DNID='104695467' MSN == BRI1: Incoming call '0108680550' - '104695467' -- BRI1: CAPI/BRI1/104695467-1: 104695467 matches in context from-pstn CAPI devicestate requested for BRI1/104695467 == Started pbx on channel CAPI/BRI1/104695467-1 INFO_IND ID=002 #0x2020 LEN=0017 Controller/PLCI/NCCI= 0x101 InfoNumber = 0x1e InfoElement = 81 83 INFO_RESP ID=002 #0x2020 LEN=0012 Controller/PLCI/NCCI= 0x101 -- BRI1: info element PI 81 83 BRI1: Origination is non ISDN INFO_IND ID=002 #0x2021 LEN=0025 Controller/PLCI/NCCI= 0x101 InfoNumber = 0x70 InfoElement = a1104695467 INFO_RESP ID=002 #0x2021 LEN=0012 Controller/PLCI/NCCI= 0x101 -- BRI1: info element CALLED PARTY NUMBER BRI1: INFO_IND DID digits not used in this state. INFO_IND ID=002 #0x2022 LEN=0016 Controller/PLCI/NCCI= 0x101 InfoNumber = 0x18 InfoElement = 89 INFO_RESP ID=002 #0x2022 LEN=0012 Controller/PLCI/NCCI= 0x101 -- BRI1: info element CHANNEL IDENTIFICATION 89 -- Executing Goto(CAPI/BRI1/104695467-1, s) in new stack == Spawn extension (from-pstn, 104695467, 1) exited non-zero on 'CAPI/BRI1/104695467-1' -- Executing Goto(CAPI/BRI1/104695467-1, s|1) in new stack -- Goto (from-pstn,s,1) -- Executing SetVar(CAPI/BRI1/104695467-1, FROM_DID=s) in new stack -- Executing SetVar(CAPI/BRI1/104695467-1, FAX_RX=disabled) in new stack -- Executing Goto(CAPI/BRI1/104695467-1, ext-local|577|1) in new stack -- Goto (ext-local,577,1) -- Executing Macro(CAPI/BRI1/104695467-1, exten-vm|577|577) in new stack -- Executing Macro(CAPI/BRI1/104695467-1, user-callerid) in new stack -- Executing DBget(CAPI/BRI1/104695467-1, AMPUSER=DEVICE/0108680550/user) in new stack -- DBget: varname=AMPUSER, family=DEVICE, key=0108680550/user -- DBget: Value not found in database. -- Executing DBget(CAPI/BRI1/104695467-1, AMPUSERCIDNAME=AMPUSER//cidname) in new stack -- DBget: varname=AMPUSERCIDNAME, family=AMPUSER, key=/cidname -- DBget: Value not found in database. -- Executing GotoIf(CAPI/BRI1/104695467-1, 1?5) in new stack -- Goto (macro-user-callerid,s,5) -- Executing NoOp(CAPI/BRI1/104695467-1, Using CallerID 0108680550) in new stack -- Executing SetVar(CAPI/BRI1/104695467-1, FROMCONTEXT=exten-vm) in new stack -- Executing Macro(CAPI/BRI1/104695467-1, record-enable|577|IN) in new stack -- Executing GotoIf(CAPI/BRI1/104695467-1, 0 0?2:4) in new stack -- Goto (macro-record-enable,s,4) -- Executing AGI(CAPI/BRI1/104695467-1, recordingcheck|20060111-103127|1136971887.1) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck recordingcheck|20060111-103127|1136971887.1: Inbound recording not enabled -- AGI Script recordingcheck completed, returning 0 -- Executing NoOp(CAPI/BRI1/104695467-1, No recording needed) in new stack -- Executing Macro(CAPI/BRI1/104695467-1, dial|15|tr|577) in new stack -- Executing GotoIf(CAPI/BRI1/104695467-1, 0?4:2) in new stack -- Goto (macro-dial,s,2) -- Executing GotoIf(CAPI/BRI1/104695467-1, 0?5:4) in new stack -- Goto (macro-dial,s,4) -- Executing AGI(CAPI/BRI1/104695467-1, dialparties.agi) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi -- dialparties.agi: priority = 4 -- dialparties.agi: callingani2 = 0 -- dialparties.agi: accountcode = -- dialparties.agi: channel = CAPI/BRI1/104695467-1 -- dialparties.agi: callerid = 0108680550 -- dialparties.agi: context = macro-dial -- dialparties.agi: callington = 33 -- dialparties.agi: dnid = 104695467 -- dialparties.agi: request = dialparties.agi -- dialparties.agi: calleridname = unknown -- dialparties.agi: extension = s -- dialparties.agi: language = -- dialparties.agi: uniqueid = 1136971887.1 -- dialparties.agi: callingpres = 1 -- dialparties.agi: type = CAPI -- dialparties.agi: rdnis = unknown -- dialparties.agi: callingtns = 0 -- dialparties.agi: enhanced = 0.0
[Asterisk-Users] RE: Wake-Up Call
In article [EMAIL PROTECTED] ny.censys.net, [EMAIL PROTECTED] says... Something to think about is this too, when completed scheduling, ask would you like to notify another extension, so if the first does not answer in two attempts, ring a cell phone or such. But I cannot complain, I use the wakeup call function every day, and it is definitely better than any alarm clock or pbx reminder available. Yes, I like it. It could have more features, but I won't complain ;)) -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: mpg123 removal
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... How can you convert mp3 to gsm? mencoder? Do you have an example? You can use this page. http://www.asteriskguru.com/tools/audio_conversion.php -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Re: Re: Remotely reboot SIP Phones ?
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Figured it out :) Basically, you have to have a file called syncinfo.xml in the tftp root directory, with the following contents: SYNCINFO IMAGE VERSION=* SYNC=1/ /SYNCINFO Also, in SIPDefault.cnf or the phone's configuration file, stick: sync: 0 somewhere so the phone's sync value doesn't match the value in syncinfo.xml. If you make a change of sorts, just run sip notify reboot-cisco username at any time in asterisk and it'll send the notify to the phone. If the phone is in use, it waits until it's idle, once it is, it waits 20 seconds and then checks the syncinfo.xml file, and if the values of sync are different, it reboots :) Hi Aron! What Cisco phone do you use? I use 7940 with SIP firmware version POS3- 07-5-00. For me it works but on wery strange, I shuld say wrong way. I have put syncinfo.xml in tftp root and when I enter this in * CLI pbx*CLI sip notify reboot-cisco 201 202 Unable to find notify type 'reboot-cisco' pbx*CLI sip notify cisco-check-cfg 201 202 Sending NOTIFY of type 'cisco-check-cfg' to '201' Sending NOTIFY of type 'cisco-check-cfg' to '202' Like you said, after 20s it looks for two files in tftp root dir - dialplan.xml (why?) and syncinfo.xml. Then Cisco waits. I have wait for more then 12 min and nothing happened. Then when I decaided to pick up handset, then it started to reboot. He reboots for 2-3 min. If my boss needs to make a important phone call I'll get fired :)) Why he vaits that I pick up handset (or press any bottun)? Anyway, thank you for this one (if I don't get fired :)) -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Latency in Asterisk
hi, What is the typical delay (latency and latency variance) in Asterisk when you use rtp/rtcp between 2 endpoint's? Has anyone measured this? Also, how much better is the TDMoE on this? Jan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Incoming PSTN Calls - Can't interrupt Main Menu
Hi Kokmeng,
Unfortunately that's wasn't it. WaitExten was executed but then I still
get the timeout error -
Timeout, but no rule 't' in context 'incomingpstn'
I am totally stuck...I have been googling and searching the archives and
testing different things for days to no avail. I thought at some stage
it might be an issue with the priorities and all different priorities
but that didn't work either.
I see the Asterisk console play the MainMenu (i.e. the Background rule),
I press an option and absolutely nothing appears on the console, the
menu carries on regardless. Its only at the end I see this timeout
error.
Thanks,
Aisling.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of KokMeng
Loh
Sent: 11 January 2006 01:20
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Incoming PSTN Calls - Can't interrupt Main
Menu
Hi Aisling,
You're missing the 'WaitExten' directive after playing the background
sound file. Your lines should be like this:
[incomingpstn]
exten = s,1,Wait(1)
exten = s,n,Background(MainMenu)
exten = s,n,WaitExten(10)
exten = 1,1,Goto(internalExt,s,1)
exten = 2,1,Goto(mainconfmenu,s,1)
-kokmeng.
Aisling wrote:
Hi,
Thanks to both Iqbal and Kokmeng for the replies.
Kokmeng I tried what you suggested however no luck...
What I have done which is currently working(kind of) is that in my
sip.conf in the [general] section I have set context=incomingpstn. My
register line looks like:
register = username:[EMAIL PROTECTED]/
In my extensions.conf I then have
[incomingpstn]
exten = s,1,Wait(1)
exten = s,n,Background(MainMenu)
exten = 1,1,Goto(internalExt,s,1)
exten = 2,1,Goto(mainconfmenu,s,1)
[internalExt]
exten = s,n,Background(InternalExtension)
[mainconfmenu]
exten = s,n,Background(MainConfMenu)
I can hear the MainMenu sound file being played. What's strange is that
when I press '1' to interrupt, which in my logic should invoke the
internalExt context, nothing happens. The MainMenu sound file continues
to play and finally I get the error:
Warning: pbx.c:2405 __ast_pbx_run: Timeout, but no rule 't' in context
'incomingpstn'
I used the 'Goto' as Iqbal suggested instead of includes...
Has anyone ever experienced this kind of behaviour before?
Many thanks,
Aisling.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of KokMeng
Loh
Sent: 09 January 2006 08:53
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Incoming PSTN Calls - Stumped
Hi,
The hostname that you used in your register directive ('provider.ie')
must have a corresponding section in sip.conf. In your example, you
used
'[provider-in]'. If that is what you actually used, then this might
explain why your incoming goes to the default context because it
couldn't find its own section. Try renaming '[provider-in]' to
'[provider.ie]'.
-kokmeng.
Aisling O'Driscoll wrote:
Hi,
Yes InternalExtension is the context and 2093 the extension.
Just to explain something odd that's happening (and I'm very stumped
with this)..I think my contexts are definately the reason that I
can't interrupt the menu for incoming pstn calls to choose a submenu:
My users register with my sip proxy (SER). Therefore when I create an
entry for them in sip.conf I set only one context. Also to allow for
incoming calls from my provider it seems I must direct the calls
firstly to a 'dummy' extension.
sip.conf
register = username:[EMAIL PROTECTED]/2093
[provider-in]
type=peer
host=sip.provider.ie
context=onecontext
[2092]
type=peer
other stuff
context=onecontext
So the dummy extension here is '2093' and 2092 is a phone who
registers with SER and when SER redirects to Asterisk uses the
'onecontext' context.
Now in my extensions.conf 'onecontext' includes other contexts. This
is how I get access to conference calls, creating IVR menus etc. Also
the main purpose of 'onecontext' is to allow outgoing access to the
PSTN.
[onecontext]
include = createmenu //creating an IVR menu
include = createconf //creating a conf call
etc
include = default//used for voicemail
[createmenu]
;does something
[createconf]
;does something
;outgoing calls - main purpose of onecontext
exten = _X.,1,Dial(SIP/[EMAIL PROTECTED])
exten = _X.,2,Hangup
[default]
;mailbox for 2092 and other users
Now this is where the problems start! For incoming calls I tried to
do include = incomingpstn in 'onecontext' which I thought would
call a new context called 'incomingpstn' which would have an entry
for the dummy user. i.e.
[incomingpstn]
exten = 2093,1,Wait(1)
exten = 2093,n,Background(MainMenu)
exten = 1,1,Goto(InternalExtension,2093,1)//directs to another
context called Internal Extension
I also changed the [provider-in] for context=incomingpstn in my
sip.conf. However this didn't work and I kept getting directed to the
voicemail of my pstn provider. The ONLY way I could get the
Re: [Asterisk-Users] Incoming PSTN Calls - Can't interrupt Main Menu
Just another bit of info which might help solve this:
Looking at the Asterisk log messages - I notice when I start up
Asterisk, I see the error:
pbx_config.c: Can't use 'next' priority on the first entry!
Could I be right that its something got to do with priorities? I changed
the incomingpstn context to the following eliminating the 'n' field and
still the same errors were display in the log file on startup and it
didn't allow me to interrupt the menu.
[incomingpstn]
exten = s,1,Wait(1)
exten = s,2,Background(MainMenu)
;exten = s,3,WaitExten(10)
exten = 1,1,Goto(internalExt,s,1)
exten = 2,1,Goto(mainconfmenu,s,1)
Many Thanks,
Aisling.
-Original Message-
From: Aisling [mailto:[EMAIL PROTECTED]
Sent: 11 January 2006 10:14
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Incoming PSTN Calls - Can't interrupt Main
Menu
Hi Kokmeng,
Unfortunately that's wasn't it. WaitExten was executed but then I still
get the timeout error -
Timeout, but no rule 't' in context 'incomingpstn'
I am totally stuck...I have been googling and searching the archives and
testing different things for days to no avail. I thought at some stage
it might be an issue with the priorities and all different priorities
but that didn't work either.
I see the Asterisk console play the MainMenu (i.e. the Background rule),
I press an option and absolutely nothing appears on the console, the
menu carries on regardless. Its only at the end I see this timeout
error.
Thanks,
Aisling.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of KokMeng
Loh
Sent: 11 January 2006 01:20
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Incoming PSTN Calls - Can't interrupt Main
Menu
Hi Aisling,
You're missing the 'WaitExten' directive after playing the background
sound file. Your lines should be like this:
[incomingpstn]
exten = s,1,Wait(1)
exten = s,n,Background(MainMenu)
exten = s,n,WaitExten(10)
exten = 1,1,Goto(internalExt,s,1)
exten = 2,1,Goto(mainconfmenu,s,1)
-kokmeng.
Aisling wrote:
Hi,
Thanks to both Iqbal and Kokmeng for the replies.
Kokmeng I tried what you suggested however no luck...
What I have done which is currently working(kind of) is that in my
sip.conf in the [general] section I have set context=incomingpstn. My
register line looks like:
register = username:[EMAIL PROTECTED]/
In my extensions.conf I then have
[incomingpstn]
exten = s,1,Wait(1)
exten = s,n,Background(MainMenu)
exten = 1,1,Goto(internalExt,s,1)
exten = 2,1,Goto(mainconfmenu,s,1)
[internalExt]
exten = s,n,Background(InternalExtension)
[mainconfmenu]
exten = s,n,Background(MainConfMenu)
I can hear the MainMenu sound file being played. What's strange is that
when I press '1' to interrupt, which in my logic should invoke the
internalExt context, nothing happens. The MainMenu sound file continues
to play and finally I get the error:
Warning: pbx.c:2405 __ast_pbx_run: Timeout, but no rule 't' in context
'incomingpstn'
I used the 'Goto' as Iqbal suggested instead of includes...
Has anyone ever experienced this kind of behaviour before?
Many thanks,
Aisling.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of KokMeng
Loh
Sent: 09 January 2006 08:53
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Incoming PSTN Calls - Stumped
Hi,
The hostname that you used in your register directive ('provider.ie')
must have a corresponding section in sip.conf. In your example, you
used
'[provider-in]'. If that is what you actually used, then this might
explain why your incoming goes to the default context because it
couldn't find its own section. Try renaming '[provider-in]' to
'[provider.ie]'.
-kokmeng.
Aisling O'Driscoll wrote:
Hi,
Yes InternalExtension is the context and 2093 the extension.
Just to explain something odd that's happening (and I'm very stumped
with this)..I think my contexts are definately the reason that I
can't interrupt the menu for incoming pstn calls to choose a submenu:
My users register with my sip proxy (SER). Therefore when I create an
entry for them in sip.conf I set only one context. Also to allow for
incoming calls from my provider it seems I must direct the calls
firstly to a 'dummy' extension.
sip.conf
register = username:[EMAIL PROTECTED]/2093
[provider-in]
type=peer
host=sip.provider.ie
context=onecontext
[2092]
type=peer
other stuff
context=onecontext
So the dummy extension here is '2093' and 2092 is a phone who
registers with SER and when SER redirects to Asterisk uses the
'onecontext' context.
Now in my extensions.conf 'onecontext' includes other contexts. This
is how I get access to conference calls, creating IVR menus etc. Also
the main purpose of 'onecontext' is to allow outgoing access to the
PSTN.
[onecontext]
include = createmenu //creating an IVR menu
include = createconf
Re: [Asterisk-Users] iax2 wireless and Multicast
El Jueves, 5 de Enero de 2006 01:12, Alexander Lopez escribió: Asterisk dows not currently support MultiCast. You may want to look at some applications that where written for Mbone http://ntrg.cs.tcd.ie/undergrad/4ba2/multicast/bryan/index.html If you can incorporate them into an Asterisk Channel Driver, These tools would allow you to: Use Multicast to 'broadcast' from one Asterisk server. Your Clients could then be either a dedicated application sittng on a PC or other Device, or you could have an Aserisk server convert from MultiCast back into Unitcast and then to any device that Asterisk supports. Let Me know if you need any help. Alex I'm thinkig about a more network approach than software... one say in the Access Point with address 192.168.1.120 iptables -A PREROUTING -t nat -d 192.168.1.120 -p (iax2port) -j DNAT 224.224.0.1 ---this is the multicast default group adrress and at the client router iptables -A PREROUTING -t nat -d 239.xxx.yyy.zzz -j DNAT 192.168..1.220--this is the UNIcast address of the client station. Then an asterisk server at the client station extracting the flow for let's say client 123 and 234 that are in the client subnet. The same for all clients hanging on the same AP. this would be the cleanest solution( if this fail I can find some proxy multicast-unicast solution based that deal with joint group address etc..), here the point is how is going to react the server that is in client router when he find that some or even none of the flows belongs to any client registered with it. I realize that I really need a server at the access point with ALL the clients of the wireless subnet the AP serves registered, I would like all the network domain (wired and wireles) trunked. ex: [EMAIL PROTECTED]call--[EMAIL PROTECTED](a)--[EMAIL PROTECTED] [EMAIL PROTECTED]call--[EMAIL PROTECTED](a)--[EMAIL PROTECTED] |trunk|multiplexed| AP(b)packet stolen/redirect multicastClientstation1:multi-uni-[EMAIL PROTECTED]' \--Clientstation2:multi-uni-[EMAIL PROTECTED]' [EMAIL PROTECTED]answer-[EMAIL PROTECTED](b)-[EMAIL PROTECTED] [EMAIL PROTECTED]answer-[EMAIL PROTECTED](b)-[EMAIL PROTECTED] |trunk|multiplexed| AP(a)packet stolen/redirect multicastClientstation3:multi-uni-[EMAIL PROTECTED]' \--Clientstation4:multi-uni-[EMAIL PROTECTED]' Now the question: Is posibble for an IAX client recieve a call from one server (serverB')and answer trought a diferent one(serverB)??? If I'm center on IAX is for the multiplex-demultiplex and signaling-media bonding. Alex, It's very interesting to incorporate a channel driver, I saw a little paper here http://jungla.dit.upm.es/~jmseyas/linux/mcast.lj/mcast-lj.html where exposes multicast programming fundamentals. I'm curious about what would happen with local calls beetwen wireles subnet members, I would be nice to insert this flows in the trunked flow. Even more if the packet size permits, it would be nice multiplex several trunks together at the AP before multicast them. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Francisco Pérez Botella Sent: Wednesday, January 04, 2006 7:05 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] iax2 wireless and Multicast El Miércoles, 4 de Enero de 2006 16:06, tim panton escribió: On 4 Jan 2006, at 13:28, Francisco Pérez Botella wrote: El Miércoles, 4 de Enero de 2006 12:28, tim panton escribió: On 3 Jan 2006, at 19:10, Francisco Pérez Botella wrote: Hi. I will have to manage From asterisk to clients IP-phones, so biefly the idea is to multiplex voip flows in large packets and multicast them from asterisk/AP to client stations. flows from client stations to asterisk gateway go unicast. I wonder how iax2 protocol will be good for multiplex (trunk) and multicast ?? Hmm, it won't be easy. The IAX protocol is not multicast aware, so it is expecting a single ack to each full frame. You will have to do quite a bit of work on the IAX implementation for it to do the right thing in that area. I see, maybe I could redirect at network layer unicast--multicast addresses/group and give back a false single ack at that point. On the other side (client side). I need some like a virtual trunk where each station recieves the full frame and stealth the payload it needs for the user/phone(s) it serves. I could at client station redirect traffic from multicast to unicast interface address and serve the full frame to iax2 at client station, silently dropping the acks they give back. yes, but you need to ensure that only one client station sends an ack, or that the server station can cope with multiple acks. explained before... send back the ack from AP
Re: [Asterisk-Users] Cisco 801 and rcapi
James Harper wrote: Okay then... next question... if I were to come up with a driver for asterisk (either as hack in chan_capi, an extension to libcapi20, or a driver for the kernel) to use the rcapi functionality of the cisco (and other) isdn ta's, would anyone care to try it? Thanks James (ps. Would I get flamed if I crossposted to asterisk-dev?) -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of James Harper Sent: Sunday, 8 January 2006 23:23 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Cisco 801 and rcapi (This is an extension of an email I sent earlier, but I'm not sure if it made it to the list or not I never saw it!) We seem to be accumulating Cisco 8XX series ISDN routers as DSL becomes more and more available in Australia and our clients upgrade. Does anyone know if those routers can make the ISDN channels available in a way that can be used by Asterisk? Preferably in a fairly raw form, eg not SIP. Further investigation reveals that the 801 can be a server for something called rcapi, net-capi, or ISDN-DCP, from RCS-COM (which I think is a company or product that uses it). This doesn't appear compatible with any of the existing remote capi solutions available for Linux. Can anyone elaborate? Details of the ISDN-DCP protocol seem a bit hard to find... Thanks James ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I Have a lot of 8xx ciscos too, i would try it too. Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 32 E1's in one Asterisk 'box'
Well, We built a site that runs about 30 E1 PRIs. Heavy load, about a million call attempts per day. We built it using 10 Asterisk servers. Integration is achieved through the application design. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: RE : codecs order and so on
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... In extensions.conf, I change the context for each call, Asterisk doesn't care of codecs in contexts, it uses the order of general... Could be good to have Ssterisk making a match between codecs in General and the context used to make a call But definitiely, Asterisk choose g729 even if I am in the zap context Any idea, help is welcome. Phones usualy use only one prefered codec. So, if your phone supports ulaw and g729, it will use only one of those two to communicate with *. Once the phone is authenticated with * he allways use the same codec. So you have to get use that on that side is that specific codec. What is on another side (SIP, Zap, IAX2...) and what codec other side uses, determinates do you need codec translation in * box. If you need codec translation then you need to have licence (for g729). I hope I have make it clear for you. Solution: Count do you get more outside ulaw or g729 calls (at the same time). If you get more ulaw calls then use ulaw codec on SIP phones. Buy the same number of g729 licences as you need simultanius phone calls to that provider. -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Incoming PSTN Calls - Can't interrupt Main Menu
Hi, On 1/11/06, Aisling [EMAIL PROTECTED] wrote: Hi Kokmeng, Unfortunately that's wasn't it. WaitExten was executed but then I still get the timeout error - Timeout, but no rule 't' in context 'incomingpstn' You are still in context 'incomingpstn', this indicates that the Goto has not fired, which suggests to me that the DTMF tone is possibly never being seen by asterisk... Just my 2p. Cheers, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Transfer sounds - notifications
When I try to make attendend transfer (*2) this what hapends. I press *2 other person goes on hold and I hear transfer. I press extension number and that extension starts to ring but I don't hear anything. If nobody picks up that phone call in few seconds I get back to the person I was talking to (the person I triesd to transfer). The problem is that again, I don't hear anything (that person waits for me to say something) and I don't know that I'm back to transfered person. I hope that I have make it clear enough. Anyway, how can I solve this one? I would like to hear that the phone of extension is ringing, and I would like to konw when I'm speaking again with my caller. -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco 801 and rcapi
I Have a lot of 8xx ciscos too, i would try it too. ISDN-DCP, which looked pretty straightforward at first glance, isn't. Rather than a simple wrapper around the CAPI messages it seems to provide a similar but not even closely compatible message structure, such that my libcapi20 code is going to need to do a heap of manipulation. So I'm working on it, but slowly. James ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 32 E1's in one Asterisk 'box'
hi, Thanks - I was hoping someone who had done this would pop-in. Do you treat each Asterisk server as a separate entity or do you have a sentralized Asterisk that perform call-control for all etc? How do you make them behave as one, or is this not needed? Also, do you switch voice from B-channel's on one server to the B-channel's on another? In case how do you do this? SIP w/rtp/rtcp, TDMoE or ? Do you have any measurement of latency etc? (Sorry for all the questions) jan [EMAIL PROTECTED] wrote: Well, We built a site that runs about 30 E1 PRIs. Heavy load, about a million call attempts per day. We built it using 10 Asterisk servers. Integration is achieved through the application design. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Transfer sounds - notifications
On Wed, January 11, 2006 12:46, Tomislav Parcina said: When I try to make attendend transfer (*2) this what hapends. I press *2 other person goes on hold and I hear transfer. I press extension number and that extension starts to ring but I don't hear anything. If nobody picks up that phone call in few seconds I get back to the person I was talking to (the person I triesd to transfer). The problem is that again, I don't hear anything (that person waits for me to say something) and I don't know that I'm back to transfered person. I hope that I have make it clear enough. Anyway, how can I solve this one? I would like to hear that the phone of extension is ringing, and I would like to konw when I'm speaking again with my caller. On http://www.voip-info.org/wiki-Asterisk+config+features.conf: ;courtesytone = beep; Sound file to play to the parked caller ; when someone dials a parked call ;xfersound = beep ; to indicate an attended transfer is complete ;xferfailsound = beeperr; to indicate a failed transfer You could try these to see if that makes a difference?... Good luck! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Transfer sounds - notifications
Tomislav Parcina wrote: When I try to make attendend transfer (*2) this what hapends. I press *2 other person goes on hold and I hear transfer. I press extension number and that extension starts to ring but I don't hear anything. If nobody picks up that phone call in few seconds I get back to the person I was talking to (the person I triesd to transfer). The problem is that again, I don't hear anything (that person waits for me to say something) and I don't know that I'm back to transfered person. Edit your features.conf, you can adjust the sound effects from there. Doug ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX CallerID
Hi All Apologises if this has been disussed and I missed it. My SetUp I have a sip phone registered to an asterisk box (a1) in one location 1. This phone dials an extension which is in another location, so a1 passes the call via IAX to the other asterisk (a2) in location 2 which then dials the local phone. My Problem The caller ID setup in the sip.conf for the phone registered to a1 is not passed via the IAX to a2 and is therefor not being displayed on the phone in location2. The only way I can get the phone in location2 to display the caller ID is to set the callerid in the user part in the iax.conf on a2. Hope this makes sense Many thanks Scott Pinhorne ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MTU and Voice Delay (latency??)
Rich Adamson wrote: No. The reason is that if the phones are the only thing on this, the size of the sip packets will never be greater then 214 bytes. Given your table below, there are other devices on your network and 6% of those are sending packets of in the 512 to 1023 byte range. Actually these are the only devices, honestly. Looking at a packet capture from the SDSL network shows plenty of larger packets. The SIP Invite packets are 769 bytes, SIP Notify at 516 bytes, SIP Option packets at 481, Register packets between 430-609 bytes, Status 200 at 725 packets. They are minimal in number compared to the RTP packets though. Have you tried the previous suggestion relative to two simultaneous ftp sessions? Unfortunately not, I have no access to the remote site inside the LAN. The onsite tech is out of the office and it is difficult to walk others through this process. What city/state are you located in? The phones and the asterisk server are in London. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Recommend Fax Hardware for T1 PRI
I have posted this to the Asterisk Forums, but got no response yet. Sorry if you are reading this for the second time. What fax hardware do I need for a T1? Ideally, I will switch my T1 to a digital PRI (not CAS I'm told, which is not as good) coming into the building. My CLEC said I can do this switch no problem. I have an analog T1 coming in now. From the Asterisk box, I will connect IP phones, but I still need 2 analog ports for fax machines. I don't want to do any VOIP fax like T.38 or anything. I just want to use a standard fax machine so I can send outbound faxes reliably and so I don't confuse my users and more than they will be with a swtich from analog old ATT Merlin system to IP PBX. I assume I need a TDM400P (TDM20B flavor for 2 analog stations), but I am not sure. Down the line I may buy something like FaxFinder or try to figure out Hylafax or some other solution if this meets our needs, but I like the flexibility of outbound faxing of a paper document. You just can't send a paper document as easily with all electronic faxing. Thanks in advance! Sent by Go2net Mail! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Transfer to meetme on different server
add to iax.conf on server1 register = username:[EMAIL PROTECTED] on server1 lets say extension 1001 on server1 will transfer the call to extension 1002 on server2 exten = 1001,1,Dial(IAX2/[EMAIL PROTECTED]) ; replace server2 with ip/domain of server2 on server 2 extension 1002 will join a meetme conference room 999 exten = 1002,1,Meetme(999) to choose a dynamic generated room exten = 1002,1,Meetme(|d) Hope that helps Diyanat From: Steven Langley [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] Transfer to meetme on different server Date: Wed, 11 Jan 2006 11:31:54 +0200 Return-Path: [EMAIL PROTECTED] X-OriginalArrivalTime: 11 Jan 2006 09:36:19.0107 (UTC) FILETIME=[79CBBF30:01C61692] Hi there I am using IAX2 based phones and am wondering if the following is possible: 1. User registers with Server 1 2. User dials an extension on Server 1 3. Extension transfers call to an extension on Server 2, which transfers the call to a Meetme conference. If this is possible, would anyone be able to give me an idea how this can be done? Many thanks Steven ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MTU and Voice Delay (latency??)
On Tuesday 10 January 2006 17:28, Geoff Manning wrote: Just as an update, the users used to be on two 2mb down/512 up ADSL lines (PPPoE) (4 users on each) and they never reported a problem. Now that they are on one SDSL (PPPoA) line (2mb) is when they report the issues. Threre are *plenty* of cracked-out ADSL and SDSL modems out there. I suspect the hardware's seeing a ton of tiny packets and trying to be smart about handling them, likely by waiting until there is sufficient traffic to fill its output buffer and then sending the entire buffer. Also FWIW, at least in North America, ADSL (PPPoE) is actually PPPoA as well; any packet you send is being fragmented into dozens of ATM cells and sent out, travelling over a multitude of hardware devices which just don't exist on the IP layer. My suspect is the SDSL modem; what is it? We use ADC Megabit modems here and they work fairly well. We've had some issue with the old Flowpoint 5250s. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX CallerID
On Wed, January 11, 2006 7:52, scott said: Hi All Apologises if this has been disussed and I missed it. My SetUp I have a sip phone registered to an asterisk box (a1) in one location 1. This phone dials an extension which is in another location, so a1 passes the call via IAX to the other asterisk (a2) in location 2 which then dials the local phone. My Problem The caller ID setup in the sip.conf for the phone registered to a1 is not passed via the IAX to a2 and is therefor not being displayed on the phone in location2. The only way I can get the phone in location2 to display the caller ID is to set the callerid in the user part in the iax.conf on a2. Hope this makes sense Many thanks It sure does, as I am examining exactly the same issue for my set up... -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MTU and Voice Delay (latency??)
Andrew Kohlsmith wrote: My suspect is the SDSL modem; what is it? We use ADC Megabit modems here and they work fairly well. We've had some issue with the old Flowpoint 5250s. It is a Speedtouch 610s. Seems like a pretty robust small biz class modem but it could be the issue. We are just trying to determine what has changed since we moved from ADSL to SDSL (which is when the issue started) Here is what has changed 1) Contention on the internal network doubled to 8 users (used to be 4 users on 2 slower ADSL lines 1a) There used to be 3 VLANs, 2 for voice, 1 for data; now there is 1 for voice, one for data. 2) The modem is new with the SDSL line 3) We are using PPPoA vs. PPPoE Thanks, Geoff ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Transfer sounds - notifications
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... On http://www.voip-info.org/wiki-Asterisk+config+features.conf: ;courtesytone = beep; Sound file to play to the parked caller ; when someone dials a parked call ;xfersound = beep ; to indicate an attended transfer is complete ;xferfailsound = beeperr; to indicate a failed transfer You could try these to see if that makes a difference?... Thank you. I have uncommented those and restart asterisk but it is the same. I hear beep only when I establish att transfer and other party doesn't want to take over a call. So, other party hangs up before I do, and in that case I hear beep. In all other cases I don't hear any tone. I couldn't done anything wrong?!? Do I need to add any DYNAMIC_FEATURES in extensions.conf? This is my features.conf [general] parkext = 700 ; What ext. to dial to park parkpos = 701-720 ; What extensions to park calls on context = parkedcalls ; Which context parked calls are in ;parkingtime = 45 ; Number of seconds a call can be parked for ; (default is 45 seconds) ;transferdigittimeout = 3 ; Number of seconds to wait between digits when transfering a call courtesytone = beep ; Sound file to play to the parked caller ; when someone dials a parked call xfersound = beep; to indicate an attended transfer is complete xferfailsound = beeperr ; to indicate a failed transfer ;adsipark = yes ; if you want ADSI parking announcements ;findslot = next ; Continue to the 'next' parking space. Defaults to 'first' available ;pickupexten = *8 ; Configure the pickup extension. Default is *8 featuredigittimeout = 1000 ; Max time (ms) between digits for feature activation. Default is 500 [featuremap] blindxfer = #1 ; Blind transfer ;disconnect = *0 ; Disconnect automon = *1 ; One Touch Record atxfer = *2; Attended transfer [applicationmap] ;testfeature = #9,callee,Playback,tt-monkeys ;Play tt-monkeys to ;callee if #9 was pressed -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] video development
Hi Fran, you could do it using Adobe/Macromedia Flash Media Server 2, but I guess that's not the answer you are looking for. If you manage to do this and release it under GPL I'll kick in $50 for a bounty. Regards, Dean Collins [EMAIL PROTECTED] +1-212-203-4357 +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Fran Sent: Wednesday, 11 January 2006 3:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] video development I would like to develop a video file player tool inside Asterisk. When calling to an extension answer and Play a video file (H264). With the applications PlayBack is not possible to give a video extension (only sound file extension). is it posible? How do u start in this development? With AGI scripts is not possible to send a video stream...(i tried to send images but with SIP channel doesnt work. I am testing with SEREyeBeam ) greetings and thanks in advance. Fran ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX CallerID
Good to know its not just me then. Thanks Scott -Original message- From: Francesco Peeters (Asterisk) [EMAIL PROTECTED] Date: Wed, 11 Jan 2006 07:18:30 -0600 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] IAX CallerID On Wed, January 11, 2006 7:52, scott said: Hi All Apologises if this has been disussed and I missed it. My SetUp I have a sip phone registered to an asterisk box (a1) in one location 1. This phone dials an extension which is in another location, so a1 passes the call via IAX to the other asterisk (a2) in location 2 which then dials the local phone. My Problem The caller ID setup in the sip.conf for the phone registered to a1 is not passed via the IAX to a2 and is therefor not being displayed on the phone in location2. The only way I can get the phone in location2 to display the caller ID is to set the callerid in the user part in the iax.conf on a2. Hope this makes sense Many thanks It sure does, as I am examining exactly the same issue for my set up... -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MTU and Voice Delay (latency??)
On Wed, 2006-01-11 at 08:19 -0500, Geoff Manning wrote: Andrew Kohlsmith wrote: My suspect is the SDSL modem; what is it? We use ADC Megabit modems here and they work fairly well. We've had some issue with the old Flowpoint 5250s. It is a Speedtouch 610s. Seems like a pretty robust small biz class modem but it could be the issue. We are just trying to determine what has changed since we moved from ADSL to SDSL (which is when the issue started) Here is what has changed 1) Contention on the internal network doubled to 8 users (used to be 4 users on 2 slower ADSL lines 1a) There used to be 3 VLANs, 2 for voice, 1 for data; now there is 1 for voice, one for data. 2) The modem is new with the SDSL line 3) We are using PPPoA vs. PPPoE Are you sure about that? Most ADSL in the UK is on PPPoA (BT supplied - it may be different for LLU providers), not PPPoE so I wouldn't think this has actually changed. Rgds Pete ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Errors with bristuff-0.3.0-PRE-1e and asterisk cores
Hi, can anybody tell me what the errors mean and why my asterisk server falls from time to time. From time to time means several hours, not regularly. I also can provide a core if someone can debug? Thanks and regards Jan 11 14:34:59 NOTICE[13573] chan_zap.c: Hangup, did not find cref 83, tei 64 Jan 11 14:34:59 WARNING[13573] chan_zap.c: Hangup on bad channel 0/2 on span 8 Jan 11 14:35:03 NOTICE[13573] chan_zap.c: Hangup, did not find cref 83, tei 64 Jan 11 14:35:03 WARNING[13573] chan_zap.c: Hangup on bad channel 0/2 on span 8 Jan 11 14:35:20 NOTICE[13573] chan_zap.c: Hangup, did not find cref 84, tei 64 Jan 11 14:35:20 WARNING[13573] chan_zap.c: Hangup on bad channel 0/2 on span 8 Jan 11 14:35:24 NOTICE[13573] chan_zap.c: Hangup, did not find cref 84, tei 64 Jan 11 14:35:24 WARNING[13573] chan_zap.c: Hangup on bad channel 0/2 on span 8 Jan 11 14:35:44 WARNING[13573] chan_zap.c: Whoa, there's no owner, and we're having to fix up channel 22 to channel 23 Jan 11 14:37:44 WARNING[13568] chan_zap.c: 3 received SETUP message for call that is not a new call (retransmission). Jan 11 14:37:54 WARNING[13568] chan_zap.c: 3 received SETUP message for call that is not a new call (retransmission). Jan 11 14:38:04 WARNING[13568] chan_zap.c: 3 received SETUP message for call that is not a new call (retransmission). Jan 11 14:38:14 WARNING[13568] chan_zap.c: 3 received SETUP message for call that is not a new call (retransmission). Jan 11 14:38:24 WARNING[13568] chan_zap.c: 3 received SETUP message for call that is not a new call (retransmission). Jan 11 14:38:34 WARNING[13568] chan_zap.c: 3 received SETUP message for call that is not a new call (retransmission). Jan 11 14:38:44 WARNING[13568] chan_zap.c: 3 received SETUP message for call that is not a new call (retransmission). Jan 11 14:38:54 WARNING[13568] chan_zap.c: 3 received SETUP message for call that is not a new call (retransmission). ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Failover Device?
First, Something seems to be wrong with the list. I'm not the only person who has expressed seeing their messages either arrive late, or not at all. With that out of the way.. Is anyone aware of any type of failover device for PRI on asterisk? I've found the ISDNGuard, however it is currently not made in the U.S., nor does it run on U.S. power. Is anyone aware of a device that will detect (heartbeat?) if Asterisk is running, and if not, failover to a backup server? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MTU and Voice Delay (latency??)
Pete Barnwell wrote: Are you sure about that? Most ADSL in the UK is on PPPoA (BT supplied - it may be different for LLU providers), not PPPoE so I wouldn't think this has actually changed. Correction, you are right. The old ADSL we were running was indeed PPPoA. That has not changed. Thanks, Geoff ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Major Problems UTStarcom F1000 registering -- pls help
Hi there, I am trying desperatly to register my WiFi Phone UTStarcomm F1000 with my asterisk server. I already changed the name of the user to anonymous since it looks like the phone sends that name. The WiFi Phone's IP is 192.168.1.217, the asterisk server's IP is 192.168.1.200 What is it that I am missing? Any help very much appreciated!!! The error message I get is: Jan 11 13:49:30 NOTICE[24024]: chan_sip.c:10817 handle_request_register: Registration from 'anonymous sip:[EMAIL PROTECTED]' failed for '192.168.1.217' - Username/auth name mismatch extract of [sip.conf]: ... [UTStarcomF1000] type=friend bindport=5060 username=anonymous ;fromuser=anonymous secret=welcome mailbox=1000 canreinvite=yes context=sipout insecure=very defaultip=192.168.1.217 host=dynamic qualify=yes nat=no ;auth=anonymous:[EMAIL PROTECTED] dtmfmode=rcfa2833 *CLI sip show peers Name/username HostDyn Nat ACL Port Status UTStarcomF1000/anonymous (Unspecified)D 0UNKNOWN omp-out-4321/419941x 212.117.200.148 N 5060 OK (64 ms) omp-out-5211/419941x 212.117.200.148 N 5060 OK (64 ms) omp-out-5200/419941x 212.117.200.148 N 5060 OK (64 ms) web-de/x 217.72.200.89N 5060 OK (64 ms) sipgate-out/19x217.10.79.9 N 5060 OK (68 ms) 8 sip peers [5 online , 3 offline] *CLI sip debug ip 192.168.1.217 SIP Debugging Enabled for IP: 192.168.1.217 *CLI sip show registry HostUsername Refresh State sip.web.de:5060 x 105 Registered sipgate.de:5060 19x105 Registered And here the debug message: . Jan 11 14:28:38 NOTICE[24049]: chan_sip.c:10817 handle_request_register: Registration from 'anonymous sip:[EMAIL PROTECTED] ' failed for '192.168.1.217' - Username/auth name mismatch Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms -- SIP read from 192.168.1.217:5060: REGISTER sip:192.168.1.200:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.217:5060;rport;branch=z9hG4bK3499846672 From: anonymous sip:[EMAIL PROTECTED];tag=787472657 To: anonymous sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 90 REGISTER Contact: sip:[EMAIL PROTECTED]:5060;action=proxy max-forwards: 70 expires: 60 user-agent: UTSTARCOM F1000/Device ID-0007ba253307 Content-Length: 0 --- (11 headers 0 lines)--- Using latest REGISTER request as basis request Sending to 192.168.1.217 : 5060 (non-NAT) Transmitting (no NAT) to 192.168.1.217:5060: SIP/2.0 404 Not found Via: SIP/2.0/UDP 192.168.1.217:5060;rport;branch=z9hG4bK3499846672;received=192.168.1.217 From: anonymous sip:[EMAIL PROTECTED];tag=787472657 To: anonymous sip:[EMAIL PROTECTED];tag=as750293ee Call-ID: [EMAIL PROTECTED] CSeq: 90 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: sip:[EMAIL PROTECTED] Content-Length: 0 and here is the SIP and RTP Configuration of the phone: (STUN is turned off) (I hope this will be transmitted to the list as well since it is a paste from the Web Interfrace. In short it says: Sip Terminal Use Outbound Proxy yes sip terminal use register yes sip outbound server domain name server.x.y sip outbound server ip address 192.168.1.200 sip outbound server port 5060 sip rigister server domain name server.x.y sip register server ip address 192.168.1.200 sip register server port 5060 sip authentication string anonymous sip user name anonymous sip password welcome sip terminal port 5060 sip terminal use null packet no both sip proxy and regisister server use IP yes dns query type yes set registration duration 60 sec terminal audio rtp port 10120 terminal audio packetize time 20 milliseconds *SIP Terminal Use Outbound Proxy:* No Yes *SIP Terminal Use Register: * No Yes *SIP Outbound Server Domain Name:* *SIP Outbound Server IP Address:* *SIP Outbound Server Port:* *SIP Register Server Domain Name:* *SIP Register Server IP Address:* *SIP Register Server Port:* *SIP Authentication String:* *SIP User Name:* *SIP Password:* *SIP Terminal Port:* *SIP Terminal Use Null Packet:* No Yes *SIP Terminal Use DNS:* Both SIP Proxy And Register Servers Use IP Register Server Uses DNS And SIP Proxy Uses IP Register Server Uses IP And SIP Proxy Server Uses DNS Both Register And SIP Proxy Servers Use DNS *DNS Query Type: * None SRV SRV *Set Registration Duration:* (sec) *Terminal Audio RTP Port:* *Terminal Audio Packetize Time:*
Re: [Asterisk-Users] video development
This is a great idea! You could have an IVR presented by a computer generated figure. You could play viewzak to folks on hold. Or how about the company promo reel when waiting for you turn in the call center queue? I'm loving this idea!! In a previous life I used to be a video editor for the BBC. If you want me to knock up some video stuff for you lemme know! Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com Fran wrote: I would like to develop a video file player tool inside Asterisk. When calling to an extension answer and Play a video file (H264). With the applications PlayBack is not possible to give a video extension (only sound file extension). is it posible? How do u start in this development? With AGI scripts is not possible to send a video stream...(i tried to send images but with SIP channel doesnt work. I am testing with SEREyeBeam ) greetings and thanks in advance. Fran ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Connecting to a legacy PBX extension
Hello, I am have trouble figuring out how to connect my [EMAIL PROTECTED] system (2.2) to a legacy PBX extension. I have FXO ports available to use, and I am able to dial in to Asterisk from any extension via port 1, and I want to use port 2 for dial from an Asterisk extension (SIP, IAX, etc) to any PBX extension, or even the outside world. Being that it is a PBX extension that I am hooking into, the AMP has a specific setting for dialing a 9 to get an outside line. Is there a good example of doing this on the web anywhere? I am not having much luck googling this, but I got to beleive I am not the only one trying to use a legacy PBX extension. Thanks for your help - Tom C _ Is your PC infected? Get a FREE online computer virus scan from McAfee® Security. http://clinic.mcafee.com/clinic/ibuy/campaign.asp?cid=3963 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Remotely reboot SIP Phones ?
I've tried the Sipura and it doesn't work. It says it's sending a notify but the SPA-2002 doesn't reboot. On 1/5/06, Jian Hong GUAN [EMAIL PROTECTED] wrote: Hi, Can you give me some councils of remotely rebooting sip phones in asterisk server? How to configure sip_notify.conf and sip.conf? Kind regards, Guan ; Reboot Polycom Phone Event=check-sync Content-Length=0 ; Untested (Reboot Sipura Phone) Event=resync Content-Length=0 ; Untested (Reboot GrandStream Phone) Event=sys-control ; Untested (Reboot Cisco Phone) Event=check-sync Content-Length=0 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Web based SIP client
Hi Someone knows a free web based SIP client for use with any provider ? Thanks roberto-- Ing. Roberto PereyraContenidosOnlineServidores BSD, Solaris y LinuxSoporte técnico ISPsJabber ID: [EMAIL PROTECTED] For reliable and professional DNS, use DNS Made Easy!http://www.dnsmadeeasy.com/u/14989 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP standard for flash
Are there a SIP standard to transmit flash? For instance I would like to send a SIP message indicating to a FXO gateway to apply a flash for transfer. In RFC 2833 page 11, in DTMF Events, the table show that DTMF 16 (decimal) is used for flash. Can I use this? Jorge Mendoza ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Pri Gateway Hardware
As far as I know, you define the interface to TDMoE when you choose the zaptel driver to work with. One of the options is Zaptel over Ethernet. After that everything belongs to a PtP Ethernet connection between the box with the TDMoE the Interface to T1, FXO or what ever you has and your asterisk box. So, echo cancellation has to be taken care on both sides. Regards, Carlos -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Kim Sent: Tuesday, January 10, 2006 9:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Pri Gateway Hardware Does TDMoE supports kernel 2.6? Where should I do echo cancellation? --- Carlos Alperin [EMAIL PROTECTED] wrote: Low level requeriment, just you transfer everything using level 2. So you don't need to the overhead to have Asterisk running to route that traffic. Carlos Alperin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jean-Michel Hiver Sent: Tuesday, January 10, 2006 1:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Pri Gateway Hardware Alexander Lopez a ?rit : TDMoE is stable and stale, There is no more development planed or needed as it only opens up a pipe between two ethernet points using Layer 2. OK... What would be in the advantage in running TDMoE rather than using IAX or SIP? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Failover Device?
Do you need failover on wich side? PRI or Asterisk? Both? Straight to the last option: PRI: the best if you have more than one PRI is to do hunt on the provider side, so when one is full or down, all calls are going to be directed to the second one. Asterisk: Do redundancy, so you need to have a second Asterisk box ready for failover, taken all the traffic of the first one in such case. You can do Hearthbeat, or DNS handling for this. I never try to run asterisk in a Cluster, that can be a third option. Any experience on that direction??? Regards, Carlos -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Sent: Wednesday, January 11, 2006 8:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Failover Device? First, Something seems to be wrong with the list. I'm not the only person who has expressed seeing their messages either arrive late, or not at all. With that out of the way.. Is anyone aware of any type of failover device for PRI on asterisk? I've found the ISDNGuard, however it is currently not made in the U.S., nor does it run on U.S. power. Is anyone aware of a device that will detect (heartbeat?) if Asterisk is running, and if not, failover to a backup server? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] video development
I would like to develop a video file player tool inside Asterisk. When calling to an extension answer and Play a video file (H264). With the applications PlayBack is not possible to give a video extension (only sound file extension). is it posible? How do u start in this development? With AGI scripts is not possible to send a video stream...(i tried to send images but with SIP channel doesnt work. I am testing with SEREyeBeam ) greetings and thanks in advance. Asterisk already does this. We provide Video IVR creation for customers. All you have to do is have an audio file and video file that are the same length and then play the audio file, the video file will be played with the audio. H264 support was added to Asterisk about 3 days ago. H263+ has been in for a while. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Recommend Fax Hardware for T1 PRI
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... I assume I need a TDM400P (TDM20B flavor for 2 analog stations), but I am not sure. You can buy ATA (analog terminal adapter) or the card you mention. Bouth of them shuld work just fine. -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Why remotely reboot SIP phones?
Over the last couple of weeks I have seen a thread about remotely rebooting SIP phones from Asterisk. Is there something inherent in Asterisk that *requires* that SIP phones to be rebooted in a particular scenario, or is it just so that phones can pickup new firmware and/or configuration from their boot server? TIA. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] video development
On Wed, 11 Jan 2006 15:38:04 +0100 Matt Riddell (IT) [EMAIL PROTECTED] wrote: I would like to develop a video file player tool inside Asterisk. When calling to an extension answer and Play a video file (H264). With the applications PlayBack is not possible to give a video extension (only sound file extension). is it posible? How do u start in this development? With AGI scripts is not possible to send a video stream...(i tried to send images but with SIP channel doesnt work. I am testing with SEREyeBeam ) greetings and thanks in advance. Asterisk already does this. We provide Video IVR creation for customers. All you have to do is have an audio file and video file that are the same length and then play the audio file, the video file will be played with the audio. H264 support was added to Asterisk about 3 days ago. H263+ has been in for a while. -- Cheers, Matt Riddell As a noob that might be interested in this also, how well does this work with the seperate audio and video files and keeping them in sync? I just keep flashing back to the old days of trying to do stereo with music using two C64's.. :-) Robert ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Why remotely reboot SIP phones?
We have to reboot our phones sometimes when we do something server side, mainly because the cisco firmware doesn't seem to handle everything very well. Usually it's just to pull new configs though, as we test more features and roll them out. Aaron Steve Langstaff wrote: Over the last couple of weeks I have seen a thread about remotely rebooting SIP phones from Asterisk. Is there something inherent in Asterisk that *requires* that SIP phones to be rebooted in a particular scenario, or is it just so that phones can pickup new firmware and/or configuration from their boot server? TIA. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Better solution to mysql reconnect timeout
vmail*CLI realtime mysql status Jan 11 09:53:04 ERROR[3597]: res_config_mysql.c:623 mysql_reconnect: MySQL RealTime: Failed to reconnect. Check debug for more info. vmail*CLI realtime mysql status Connected to [EMAIL PROTECTED], port 3306 with username voicemail for 1 days, 5 hours, 32 minutes, 7 seconds. vmail*CLI realtime mysql status Connected to [EMAIL PROTECTED], port 3306 with username voicemail for 0 seconds. - There seems to be a problem with the way mysql is reconnected, apparantly it seems to be calling mysql_reconnect() but I find that hard to believe since it doesn't connect. Then the second time I run the command, it says it's been connected for 1 days, 5 hours, etc. That doesn't make any sense since the connection already failed to reconnect. There is no connection. Running the command again assures in fact, finally I do have a connection. My particular purpose is a voicemail system, which isn't taking voicemail messages all the time, it is important the system always be connected to mysql. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAX CallerID
As a rule of thumb, I always explicitly set CallerID in my dialplan before making a call through IAX, SIP or PSTN. If you make it part of a generic dialout routine then it isn't a hassle. It always works. -Original Message- From: scott [mailto:[EMAIL PROTECTED] Sent: Wednesday, January 11, 2006 12:28 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] IAX CallerID Good to know its not just me then. Thanks Scott -Original message- From: Francesco Peeters (Asterisk) [EMAIL PROTECTED] Date: Wed, 11 Jan 2006 07:18:30 -0600 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] IAX CallerID On Wed, January 11, 2006 7:52, scott said: Hi All Apologises if this has been disussed and I missed it. My SetUp I have a sip phone registered to an asterisk box (a1) in one location 1. This phone dials an extension which is in another location, so a1 passes the call via IAX to the other asterisk (a2) in location 2 which then dials the local phone. My Problem The caller ID setup in the sip.conf for the phone registered to a1 is not passed via the IAX to a2 and is therefor not being displayed on the phone in location2. The only way I can get the phone in location2 to display the caller ID is to set the callerid in the user part in the iax.conf on a2. Hope this makes sense Many thanks It sure does, as I am examining exactly the same issue for my set up... -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: using a Gigaset SX440isdn on a Diva 4BRI ?
On Tue, Jan 10, 2006 at 06:52:43PM +0100, Louis-David Mitterrand wrote: On Tue, Jan 10, 2006 at 05:43:12PM +0100, Armin Schindler wrote: On Tue, 10 Jan 2006, Louis-David Mitterrand wrote: [C:4] 22:0188:202 - D-X(003) 02 01 7F [C:4] 22:0189:202 - D-X(003) 02 01 7F [C:4] 22:0190:202 - D-X(003) 02 01 7F [C:4] 22:0191:201 - MDL-ERROR(G) [C:4] 22:0191:202 - SIG-EVENT 0A The diva card is sending (D-X), but does not receive anything (D-R). It looks like either the cross connection still isn't working or the protocol is wrong. OK, making some progress here: I removed -u (ptp mode) from the divactrl init string and now I can call in and out with my Gigaset handset! Calling and receiving calls works but I get no call progress indications at all until the call is connected. Even when using immediate=yes and landing directly in exten = s,1,Dial(CAPI/g2//bo) I get no dial tone. Here are my capi.conf settings: [DIVA2] ntmode=yes isdnmode=did incomingmsn=* controller=3 group=3 prefix=0 context=international echocancel=yes bridge=yes devices=2 Is there some setting I forgot about? Thanks, -- [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Why remotely reboot SIP phones?
Also, the old grandstreams would lose their registrations periodically. I have not played with a grandtream in quite a while so I would assume they fixed this in firmware but that was another reason for regular reboots. -Original Message- From: Aaron Daniel [mailto:[EMAIL PROTECTED] Sent: Wednesday, January 11, 2006 9:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Why remotely reboot SIP phones? We have to reboot our phones sometimes when we do something server side, mainly because the cisco firmware doesn't seem to handle everything very well. Usually it's just to pull new configs though, as we test more features and roll them out. Aaron Steve Langstaff wrote: Over the last couple of weeks I have seen a thread about remotely rebooting SIP phones from Asterisk. Is there something inherent in Asterisk that *requires* that SIP phones to be rebooted in a particular scenario, or is it just so that phones can pickup new firmware and/or configuration from their boot server? TIA. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] video development
Thank u Matt!! I will try it!!! and what about the extensions supported? file.gsm and file.h264 is possible? how do u create both files? would it be possible to create both files from an AVI or a MPEG? may i use MPEG4IP?? Thank u in advance!!! Fran -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de Matt Riddell (IT) Enviado el: miércoles, 11 de enero de 2006 15:38 Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [Asterisk-Users] video development I would like to develop a video file player tool inside Asterisk. When calling to an extension answer and Play a video file (H264). With the applications PlayBack is not possible to give a video extension (only sound file extension). is it posible? How do u start in this development? With AGI scripts is not possible to send a video stream...(i tried to send images but with SIP channel doesnt work. I am testing with SEREyeBeam ) greetings and thanks in advance. Asterisk already does this. We provide Video IVR creation for customers. All you have to do is have an audio file and video file that are the same length and then play the audio file, the video file will be played with the audio. H264 support was added to Asterisk about 3 days ago. H263+ has been in for a while. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Why remotely reboot SIP phones?
Polycom phones need a reboot after making configuration changes. -Original Message- From: Aaron Daniel [mailto:[EMAIL PROTECTED] Sent: Wednesday, January 11, 2006 7:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Why remotely reboot SIP phones? We have to reboot our phones sometimes when we do something server side, mainly because the cisco firmware doesn't seem to handle everything very well. Usually it's just to pull new configs though, as we test more features and roll them out. Aaron Steve Langstaff wrote: Over the last couple of weeks I have seen a thread about remotely rebooting SIP phones from Asterisk. Is there something inherent in Asterisk that *requires* that SIP phones to be rebooted in a particular scenario, or is it just so that phones can pickup new firmware and/or configuration from their boot server? TIA. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Better solution to mysql reconnect timeout
Sig Lange wrote: vmail*CLI realtime mysql status Jan 11 09:53:04 ERROR[3597]: res_config_mysql.c:623 mysql_reconnect: MySQL RealTime: Failed to reconnect. Check debug for more info. vmail*CLI realtime mysql status Connected to [EMAIL PROTECTED], port 3306 with username voicemail for 1 days, 5 hours, 32 minutes, 7 seconds. vmail*CLI realtime mysql status Connected to [EMAIL PROTECTED], port 3306 with username voicemail for 0 seconds. - There seems to be a problem with the way mysql is reconnected, apparantly it seems to be calling mysql_reconnect() but I find that hard to believe since it doesn't connect. Then the second time I run the command, it says it's been connected for 1 days, 5 hours, etc. That doesn't make any sense since the connection already failed to reconnect. There is no connection. Running the command again assures in fact, finally I do have a connection. My particular purpose is a voicemail system, which isn't taking voicemail messages all the time, it is important the system always be connected to mysql. Have you had a look at the source? Sounds like the connection stats are not refreshed on a reconnect. If you are not versed in C, you can still just look through the source at the plaintext and comments. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] video development
Robert Webb wrote: As a noob that might be interested in this also, how well does this work with the seperate audio and video files and keeping them in sync? I just keep flashing back to the old days of trying to do stereo with music using two C64's.. :-) Heh, my nick is ZX81! :) The thing is that you can record the two calls together (video and voice) and Asterisk will make sure that they are the correct length. You can't however use the record application :) You can however leave someone a voicemail message with video and audio :) -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Signaling the status of the line on the phone
Hello everybody, Do you know if it's possible to push the status of an extension (a phone) to a phone like blinking a light on the phone ? And do you know wich brand of phone can do this ? I'd like to make the same as the secretary phones that can see the status of lines before putting a call on it or transfering someone to. As i know that the Flash Operator Panel get the global status of Asterisk, it should be possible. If you have some pointers about that feature ... Thanks a lot. Cem ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Web based SIP client
Roberto Pereyra wrote: Hi Someone knows a free web based SIP client for use with any provider ? Thanks roberto -- Ing. Roberto Pereyra ContenidosOnline Servidores BSD, Solaris y Linux Soporte técnico ISPs Jabber ID: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Hi Roberto, im looking for a similar solution,i found this on the archives http://www.microappliances.com/site/html/index.php It seems very complete to me (look at the customers page), does anyone here have it in production? Any comment? thanks in advance --- Miguel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Parking...
Call parking... I can park a call that was received on a particular phone. But I can not park a call from the phone that initiated a call. The DTMF are just sent out to audio channel. Any hints anyone? Thanks, Andre ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Signaling the status of the line on the phone
We use Snom phones for the BLF function as you are suggesting and it works great. The Grandstream GXP-2000 with the beta firmware supports this as well but I hear its a bit buggy. The snom phones are nice because depending on the size of the office you can add an additional side cart with many more lights as well. On 1/11/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hello everybody, Do you know if it's possible to push the status of an extension (a phone) to a phone like blinking a light on the phone ? And do you know wich brand of phone can do this ? I'd like to make the same as the secretary phones that can see the status of lines before putting a call on it or transfering someone to. As i know that the Flash Operator Panel get the global status of Asterisk, it should be possible. If you have some pointers about that feature ... Thanks a lot. Cem ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] The second edition of my Asterisk book is now available
[EMAIL PROTECTED] wrote: The second edition of my book VoIP Telephony with Asterisk is now in print and available. You can find out more about it at our web site http://www.signate.com/products.php You've posted this every week for the past three or four weeks now; please stop. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE405p -- loopback for the phone company?
Eric Lyons wrote: I got zttool running and selected loop on the interface, but it didn't seem to do what they wanted (nor could I tell that it did anything at all). Many googles for zaptel and loop didn't turn up anything useful. This is a bug that needs to be fixed; currently the dual-/quad-span drivers to not respond to remote loop-up requests, nor do they have any mode to loop data back towards the network. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Recommend Fax Hardware for T1 PRI
Tomislav Parcina wrote: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... I assume I need a TDM400P (TDM20B flavor for 2 analog stations), but I am not sure. You can buy ATA (analog terminal adapter) or the card you mention. Bouth of them shuld work just fine. Wonderful advice. Both of these solutions actually fail for most people. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Why remotely reboot SIP phones?
Do you mean changes to the phone's configuration, or changes to Asterisk's configuration? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Douglas Garstang Sent: 11 January 2006 15:07 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Why remotely reboot SIP phones? Polycom phones need a reboot after making configuration changes. -Original Message- From: Aaron Daniel [mailto:[EMAIL PROTECTED] Sent: Wednesday, January 11, 2006 7:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Why remotely reboot SIP phones? We have to reboot our phones sometimes when we do something server side, mainly because the cisco firmware doesn't seem to handle everything very well. Usually it's just to pull new configs though, as we test more features and roll them out. Aaron Steve Langstaff wrote: Over the last couple of weeks I have seen a thread about remotely rebooting SIP phones from Asterisk. Is there something inherent in Asterisk that *requires* that SIP phones to be rebooted in a particular scenario, or is it just so that phones can pickup new firmware and/or configuration from their boot server? TIA. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Web based SIP client
Miguel wrote: Roberto Pereyra wrote: Hi Someone knows a free web based SIP client for use with any provider ? Thanks roberto -- Ing. Roberto Pereyra ContenidosOnline Servidores BSD, Solaris y Linux Soporte técnico ISPs Jabber ID: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Hi Roberto, im looking for a similar solution,i found this on the archives http://www.microappliances.com/site/html/index.php It seems very complete to me (look at the customers page), does anyone here have it in production? Any comment? thanks in advance --- Miguel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users There was someone here on the lists a while ago that had a java based iax client.. might find it if you search the archives.. -- . -BEGIN GEEK CODE BLOCK- Version: 3.1 GCS/GIT d-@ s+:+ a? C+++ BLHIS$ U+++ P+ L+++ !E W+++$ N++ o+ K w-- PS+++ PE@ Y+ PGP++ t 5? X !R tv+ b- DI-- D G e+ h r+++ y --END GEEK CODE BLOCK-- . signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Failover Device?
Is there any documentation around for running Asterisk in a Cluster (I assume you mean a n+1 cluster as you list a failover cluster as a different option). I was under the impression that it can't be done.. Thanks. On 1/11/06, Carlos Alperin [EMAIL PROTECTED] wrote: Do you need failover on wich side? PRI or Asterisk? Both? Straight to the last option: PRI: the best if you have more than one PRI is to do hunt on the provider side, so when one is full or down, all calls are going to be directed to the second one. Asterisk: Do redundancy, so you need to have a second Asterisk box ready for failover, taken all the traffic of the first one in such case. You can do Hearthbeat, or DNS handling for this. I never try to run asterisk in a Cluster, that can be a third option. Any experience on that direction??? Regards, Carlos -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Sent: Wednesday, January 11, 2006 8:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Failover Device? First, Something seems to be wrong with the list. I'm not the only person who has expressed seeing their messages either arrive late, or not at all. With that out of the way.. Is anyone aware of any type of failover device for PRI on asterisk? I've found the ISDNGuard, however it is currently not made in the U.S., nor does it run on U.S. power. Is anyone aware of a device that will detect (heartbeat?) if Asterisk is running, and if not, failover to a backup server? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] patching asterisk with tzafrir patch for voicemail permission does not work
Hi, I tried to patch asterisk 1.2.1 on a Debian Sarge distro with the patch made by tzafrir but I still cannot set writing permission to directories. I tried to put umask 007 inside .bash_profile but it doesn't work. Is there anyone who can help me? TIA Giorgio Incantalupo ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Manager API and ZapBarge or ChanSpy
Am trying to monitor and record an in-process phone call using a
remote computer and the Asterisk Manager API. Have something that is
working, but the call recording volume is to low to be usable.
dialplan
exten = 8159,1,ZapBarge(Zap/1)
remote application with Asterisk Manager API
$telnet-print(Action: Originate\nChannel:
Local/[EMAIL PROTECTED]: ChanSpy\nData: |q\nPriority:
1\n\n);
$telnet-waitfor('/Response: Success/');
# get all the local channels and look for the extension in use
$telnet-print(Action: Command\nCommand: Local Show Channels\n\n);
$telnet-waitfor('/Response: Follows/');
while (($line = $telnet-getline) ($line !~ /END COMMAND/i)) {
push(@channels,$line);
}
# start the monitor
while ($line = pop(@channels)) {
$pattern = Local\/ . $exten;
if ($line =~ m/$pattern/i) {
print $line;
# start monitor
$recording = $timestamp . - . $uniqueid;
print $recording;
$telnet-print(Action: Monitor\nMix: 1\nFormat:
wav49\nChannel: . $currentChannel .\nFile: . $recording .
\n\n);
$telnet-waitfor('/Response: Success/');
}
}
What I think is happening is that a call is originated for the 8159
extension, which then executes the dialplan zapbarge on in process zap
channel call, then the chanspy listens in. This barely works, but the
call volume is just not usable. I am pretty sure I need to get rid of
the zapbarge or chanspy, but I am not sure how to go about originating
the call so it will work. Any advice would be appreciated.
Thanks;
Dan
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[Asterisk-Users] Asterisk doesn't detect answer for some numbers
Hi, we are running in difficulties with some (rare) numbers: Asterisk doesn't detect answer and rings indefinitely or drops call with NOANSWER. It seems that these numbers are automatic responders. I tried to debug with 'pri intense debug span 1' but no useful info. I'm using a Sangoma A102 card with wanpipe beta1-2.3.4, Asterisk is 1.2.1 and Linux is 2.6.9-22.Elsmp. Line is a PRI E1 in Italy. With a Digium card, I had not this problem. I'm looking also for a paid consultant but he/she sould be a really competent person. Thanks in advance Mimmus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk REGISTERs
Jean-Michel Hiver wrote: Is there a way to have Asterisk remember which agents are registered to it using a MySQL database rather than in memory? It would help with high availability / clustering scenarios. It also means you could restart the server without loosing this information... Check the sample config file docs... 'persistentagents=yes' will store then in the local Asterisk database. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] video development
Fran wrote: Thank u Matt!! I will try it!!! and what about the extensions supported? file.gsm and file.h264 is possible? how do u create both files? would it be possible to create both files from an AVI or a MPEG? may i use MPEG4IP?? I use VCDCutter to create a fake webcam which can be fed by audio and video files. Then I use Eyebeam to leave a voicemail message (using the fake webcam). This will deposit both the audio and video in the voicemail folder which you can then copy to the location of your choice. :) Make sense? -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE : [Asterisk-Users] Re: RE : codecs order and so on
Well, another try [general] Disallow=all Allow=ulaw Allow=g729 For the Uas, they are sets to have g729 first Calls to/from pstn needs g729 Calls to/from zap needs Ulaw ALL incoming calls works OK even if the caller is G729(I have made a caller using g729 only)... Calling zap = no problem, Ulaw is choosen Calling pstn provider =fail (I need g729 but Ulaw is choosen) Call from zap = no problem Ulaw is choosen Call from pstn = no problem g729 used... What does it mean? Strange isn't it? In fact Asterisk let the Uas negociates the codec for incoming calls and doesn't care for outgoing calls. In a context for incoming, no problems In a context for outgoing(I use goto context,extension,priority)Asterisk doesn't take care of the context codecs priority. It's then false to say that asterisk uses the prefered codec of Uas, I have here a Ua wich uses differents codecs for incoming calls. Question is : Why Asterisk doesn't care of codecs in an outgoing context? Any good idea is welcome. Ps: the solution is to have a g729 codec form Digium, ok, I have it and it works, but it takes a lot of cpu (50% of my Soekris box). -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Tomislav Parcina Envoyé : mercredi 11 janvier 2006 12:28 À : asterisk-users@lists.digium.com Objet : [Asterisk-Users] Re: RE : codecs order and so on In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... In extensions.conf, I change the context for each call, Asterisk doesn't care of codecs in contexts, it uses the order of general... Could be good to have Ssterisk making a match between codecs in General and the context used to make a call But definitiely, Asterisk choose g729 even if I am in the zap context Any idea, help is welcome. Phones usualy use only one prefered codec. So, if your phone supports ulaw and g729, it will use only one of those two to communicate with *. Once the phone is authenticated with * he allways use the same codec. So you have to get use that on that side is that specific codec. What is on another side (SIP, Zap, IAX2...) and what codec other side uses, determinates do you need codec translation in * box. If you need codec translation then you need to have licence (for g729). I hope I have make it clear for you. Solution: Count do you get more outside ulaw or g729 calls (at the same time). If you get more ulaw calls then use ulaw codec on SIP phones. Buy the same number of g729 licences as you need simultanius phone calls to that provider. -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP standard for flash
Jorge Mendoza wrote: Are there a SIP standard to transmit flash? For instance I would like to send a SIP message indicating to a FXO gateway to apply a flash for transfer. In RFC 2833 page 11, in DTMF Events, the table show that DTMF 16 (decimal) is used for flash. Can I use this? Yes. This is what Asterisk sends when requesting a flash-hook event over RTP. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Outbound routing
Hi all, I've 3 providers (A, B, and C) the A is giving me freecalls to USA, the B is giving my freecalls to Europe, and C is to call the otre destinations. My question is, how can I configure the outboud routing to select the right trunk for every destination? All the providers uses the dialing form 00 1 123 4567890 when 00 is the number dialed to call, 1 the country code, 123 the area code and 4567890 the phone number. I've the following outbound routing with AMP, but the calls are been started by the first provider in the trunk sequence list: Route Name: International Dial Patterns : 00. Trunk Sequence: A B C I want to make that the USA calls going with A, Europe calls with B and rest of the world with C. Is this possible ? Can you gime a little of help with this... Than you in advance. :) -- Guillermo Salas M. Telconet S.A. Manta Calle 15 y Av. 24 Esq. Phone : 593 5 262 8071 Mobile: 593 9 985 5138 SIP : [EMAIL PROTECTED] e-mail: [EMAIL PROTECTED] www : http://www.telconet.net http://www.telcocarrier.net Linux User: 255902 Soporte en Linea en http://www.manta.telconet.net Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Has anyone tried using flash() in features.conf (applicationmap) - RESOLVED
extensions.conf [context] exten = s,n,Set(DYNAMIC_FEATURES=zapflash) exten = s,n,Dial(SIP/,15,tw) features.conf [applicationmap] zapflash = *3,caller,flash,() needed a comma between flash an the () I Wonder (aloud) if there'd be a way to send the incoming call to another phone? IOW, Talking on phone1, call waiting beeps from FXO Flash send to phone2 Flash? recover original caller on phone1 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] patching asterisk with tzafrir patch for voicemail permission does not work
On Wed, Jan 11, 2006 at 04:59:00PM +0100, gincantalupo wrote: Hi, I tried to patch asterisk 1.2.1 on a Debian Sarge distro with the patch made by tzafrir from a deb or self built? If from a deb: dpkg -l asterisk but I still cannot set writing permission to directories. I tried to put umask 007 inside .bash_profile but it doesn't work. Is there anyone who can help me? I set umask in the asterisk init.d script . If you run asterisk from the shell, set umask manually. A umask of 007 is probably a bad default for root: all files that root creates are writeble to the group root: this is normally not the case. Also: no files is readable by others. Anyway, you can check your current umask in the shell with the command 'umask' . Note that you'll also see an extra 0 prefixed. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Music On Hold Dial(,m)
Hi, I have an issue, I can hear music on hold with MusicOnHold() but I cant hear anything with Dial(,m). (I did: make mpg123, cd mpg.., make, make install). Mi extensions.conf is: [incoming] exten =s,1,Answer() exten =s,n,Background(welcome) exten =s,n,WaitExten(20,m) ;at this point the debugger doesnt say anything about music on hold ; I dial 2 exten =2,1,MusicOnHold() ;the debugger says Started music on hold.. and I can hear the music Any ideas? Miguel Soto ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE : [Asterisk-Users] Outbound routing
Give me your providers and I give you the agi script to do that :) Olivier -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Guillermo Salas M Envoyé : mercredi 11 janvier 2006 17:17 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : [Asterisk-Users] Outbound routing Hi all, I've 3 providers (A, B, and C) the A is giving me freecalls to USA, the B is giving my freecalls to Europe, and C is to call the otre destinations. My question is, how can I configure the outboud routing to select the right trunk for every destination? All the providers uses the dialing form 00 1 123 4567890 when 00 is the number dialed to call, 1 the country code, 123 the area code and 4567890 the phone number. I've the following outbound routing with AMP, but the calls are been started by the first provider in the trunk sequence list: Route Name: International Dial Patterns : 00. Trunk Sequence: A B C I want to make that the USA calls going with A, Europe calls with B and rest of the world with C. Is this possible ? Can you gime a little of help with this... Than you in advance. :) -- Guillermo Salas M. Telconet S.A. Manta Calle 15 y Av. 24 Esq. Phone : 593 5 262 8071 Mobile: 593 9 985 5138 SIP : [EMAIL PROTECTED] e-mail: [EMAIL PROTECTED] www : http://www.telconet.net http://www.telcocarrier.net Linux User: 255902 Soporte en Linea en http://www.manta.telconet.net Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Recommend Fax Hardware for T1 PRI
You could use a 2 span T1 card from Digium and plug one span into a channel bank, and have FXS ports on the CB for the fax machines. With the latest firmwares from Digium these streams are bridged internaly on the card, and don't even come on to the PCI bus. On 1/11/06, John Crew [EMAIL PROTECTED] wrote: I have posted this to the Asterisk Forums, but got no response yet. Sorry if you are reading this for the second time. What fax hardware do I need for a T1? Ideally, I will switch my T1 to a digital PRI (not CAS I'm told, which is not as good) coming into the building. My CLEC said I can do this switch no problem. I have an analog T1 coming in now. From the Asterisk box, I will connect IP phones, but I still need 2 analog ports for fax machines. I don't want to do any VOIP fax like T.38 or anything. I just want to use a standard fax machine so I can send outbound faxes reliably and so I don't confuse my users and more than they will be with a swtich from analog old ATT Merlin system to IP PBX. I assume I need a TDM400P (TDM20B flavor for 2 analog stations), but I am not sure. Down the line I may buy something like FaxFinder or try to figure out Hylafax or some other solution if this meets our needs, but I like the flexibility of outbound faxing of a paper document. You just can't send a paper document as easily with all electronic faxing. Thanks in advance! Sent by Go2net Mail! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Echo on phones...
I am having a bit of a problem with several phones (Polycom 601 and Aastra 9133i). I have a new installation in a brand new office. The office is bare and there is a lot of echo. This causes all the phones on the office to have a very audible echo. I know it is not really a hardware problem, but is there a setting on the phones to handle this kind of echo? -- Carlos Chavez Director de Tecnologa Telecomunicaciones Abiertas de Mxico S.A. de C.V. Tel: +52-55-91169161 Ext 2001 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Help with amportal: asterisk ended with exit status 127
Thanks so much for the suggestions. I'm having trouble getting the list emails so I just looked at the archives for yesterday. Funny, yesterday I did run into a broken pipe error while restarting using asterisk -v. It was wilcalu.so (or something like that). I've stopped and started asterisk several times over the past few weeks and had never gotten the broken pipe error, then all of a sudden I did. Anyway, I removed that yesterday and had never tried to restart amportal (or safe asterisk) at all until this morning--after we had also figured out that /usr/sbin was not in the PATH of the user I was trying to start amportal with. Anyway, amportal was able to start asterisk and FOP and it seems to be working, but now when I click to apply changes from the AMP admin page, I get this output on the CLI at which I had started amportal: /var/www/html/panel/safe_opserver: line 5: 12452 Terminated ./op_server.pl I'll be searching, but if anybody has a suggestion, lemme know. Thanks, Ben F ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] China DID Wanted
Looking for bulk DID's for the following location's in China (+86): Shanghai (021) Guangzhou (020) Shenzen (755) Also looking for bulk DID's in Hong Kong (+852). Thanks Steven Ducat. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ruby-agi-1.0.2 released !
I am happy to announce the release of ruby-agi-1.0.2 This is a stable release of ruby-agi. ruby-agi is available at http://rubyforge.org/projects/ruby-agi/ You can also install ruby-agi via gem. To install ruby-agi gem package, try % gem install ruby-agi Feel free to send me your feedback, feature request and bug report. Thank you, Mohammad Khan mail2web - Check your email from the web at http://mail2web.com/ . ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAX CallerID
On Wed, January 11, 2006 16:00, Colin Anderson said: As a rule of thumb, I always explicitly set CallerID in my dialplan before making a call through IAX, SIP or PSTN. If you make it part of a generic dialout routine then it isn't a hassle. It always works. It sometimes doesn't for my installation, but I'll check it later, it is not a big issue right now... -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Echo on phones...
As far as I know, the Polycom's don't have any kind of echo cancellation for this type of thing, however there is a technology called a FICUS PLANT, which inhabits many offices and can solve your bare office problem :) --TomOn 1/11/06, Carlos Chavez [EMAIL PROTECTED] wrote: I am having a bit of a problem with several phones (Polycom 601 and Aastra 9133i). I have a new installation in a brand new office. The office is bare and there is a lot of echo. This causes all the phones on the office to have a very audible echo. I know it is not really a hardware problem, but is there a setting on the phones to handle this kind of echo? -- Carlos ChavezDirector de TecnologíaTelecomunicaciones Abiertas de México S.A. de C.V.Tel: +52-55-91169161 Ext 2001 -BEGIN PGP SIGNATURE-Version: GnuPG v1.4.1 (GNU/Linux)iD8DBQBDxUAhVhw7eWImqUMRAt9fAJ9xe/8L3PmvXwxMi4AloiO4rSEg/wCgqUYB46L37C91W4DP+cwGpATOktk==lceK-END PGP SIGNATURE- ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Echo on phones...
Carlos Chavez wrote: I am having a bit of a problem with several phones (Polycom 601 and Aastra 9133i). I have a new installation in a brand new office. The office is bare and there is a lot of echo. This causes all the phones on the office to have a very audible echo. I know it is not really a hardware problem, but is there a setting on the phones to handle this kind of echo? -- Carlos Chavez Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 The echo are for internal calls, or for outside calls? We have Polycom phones and they have not echo on internal calls, some times a small echo on outside calls. Jorge ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Nested MySQL Commands
Is it possible to have nested MySQL queries in extensions.conf? Ie, perform a query, grab a value, and then jump to another location in the dialplan and do another query based on that original value. I'm having problems with the result and fetchid's and I'm not sure if it's even possible to do this or not. Thanks, Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Issue calling other PBX systems using VoIP with Polycom 501
I am having an issue using a Polycom 501 and VoIP for outgoing calls where if I call say my credit card company and try to follow their PBX menu, the key presses never register with their PBX. It's as if every key press I make absolutely nothing is being sent to them. Is there some setting in the phone or Asterisk that I need to change to fix this issue? Thanks for any help,Andrew ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Nested MySQL Commands
On 11/01/06, Douglas Garstang [EMAIL PROTECTED] wrote: Is it possible to have nested MySQL queries in extensions.conf? Ie, perform a query, grab a value, and then jump to another location in the dialplan and do another query based on that original value. I'm having problems with the result and fetchid's and I'm not sure if it's even possible to do this or not. When things start to get that complicated, I reckon it's time for AGI Peter -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] VoIP: [EMAIL PROTECTED] FWD: **275*5048707000 VoipTalk: **473*5048707000 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Why remotely reboot SIP phones?
Usually just phone changes, but if you reboot the server, or reload something, sometimes the phones need to re-register and it's just easier to send a remote reboot. Aaron Steve Langstaff wrote: Do you mean changes to the phone's configuration, or changes to Asterisk's configuration? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Douglas Garstang Sent: 11 January 2006 15:07 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Why remotely reboot SIP phones? Polycom phones need a reboot after making configuration changes. -Original Message- From: Aaron Daniel [mailto:[EMAIL PROTECTED] Sent: Wednesday, January 11, 2006 7:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Why remotely reboot SIP phones? We have to reboot our phones sometimes when we do something server side, mainly because the cisco firmware doesn't seem to handle everything very well. Usually it's just to pull new configs though, as we test more features and roll them out. Aaron Steve Langstaff wrote: Over the last couple of weeks I have seen a thread about remotely rebooting SIP phones from Asterisk. Is there something inherent in Asterisk that *requires* that SIP phones to be rebooted in a particular scenario, or is it just so that phones can pickup new firmware and/or configuration from their boot server? TIA. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zaptel modules load, but Asterisk fails at startup
I'm running Asterisk on a Gentoo box with the Zaptel 1.2.1 drivers. If I boot the machine without having the wcfxs module autoload, then install the module with modprobe, asterisk works just fine. If I boot the machine and autoload the wcfxs module, the module loads fine: Jan 11 11:06:55 asterisk Zapata Telephony Interface Registered on major 196 Jan 11 11:06:55 asterisk ACPI: PCI Interrupt Link [LNKC] enabled at IRQ 10 Jan 11 11:06:55 asterisk PCI: setting IRQ 10 as level-triggered Jan 11 11:06:55 asterisk ACPI: PCI Interrupt :00:0a.0[A] - Link [LNKC] - GSI 10 (level, low) - IRQ 10 Jan 11 11:06:55 asterisk Freshmaker version: 73 Jan 11 11:06:55 asterisk Freshmaker passed register test Jan 11 11:06:55 asterisk Module 0: Installed -- AUTO FXS/DPO Jan 11 11:06:55 asterisk Module 1: Not installed Jan 11 11:06:55 asterisk Module 2: Not installed Jan 11 11:06:55 asterisk Module 3: Installed -- AUTO FXO (FCC mode) Jan 11 11:06:55 asterisk Found a Wildcard TDM: Wildcard TDM400P REV I (2 modules) The module is running: asterisk sfbosch # lsmod Module Size Used by wctdm 39936 - zaptel226756 - asterisk sfbosch # But Asterisk behaves as though it were not: [chan_zap.so] = (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found Jan 11 11:32:53 WARNING[5778]: chan_zap.c:920 zt_open: Unable to specify channel 1: No such device or address Jan 11 11:32:53 ERROR[5778]: chan_zap.c:6847 mkintf: Unable to open channel 1: No such device or address here = 0, tmp-channel = 1, channel = 1 Jan 11 11:32:53 ERROR[5778]: chan_zap.c:10251 setup_zap: Unable to register channel '1' Jan 11 11:32:53 WARNING[5778]: loader.c:414 __load_resource: chan_zap.so: load_module failed, returning -1 Jan 11 11:32:53 WARNING[5778]: loader.c:554 load_modules: Loading module chan_zap.so failed! Warning, flexible rate not heavily tested! asterisk sfbosch # Ouch ... error while writing audio data: : Broken pipe Looking at this now as I write this, it seems that some module dependencies aren't loading, but I can't be sure. Does anybody have an idea what's going on here? -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Nested MySQL Commands
Peter, Too slow! We're going to potentially be doing several MySQL lookups for routing even the most basic of calls, and if every one of those queries has to make a call out to an AGI script, it would become a performance problem. Douglas. -Original Message- From: Peter Bowyer [mailto:[EMAIL PROTECTED] Sent: Wednesday, January 11, 2006 11:27 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Nested MySQL Commands On 11/01/06, Douglas Garstang [EMAIL PROTECTED] wrote: Is it possible to have nested MySQL queries in extensions.conf? Ie, perform a query, grab a value, and then jump to another location in the dialplan and do another query based on that original value. I'm having problems with the result and fetchid's and I'm not sure if it's even possible to do this or not. When things start to get that complicated, I reckon it's time for AGI Peter -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] VoIP: [EMAIL PROTECTED] FWD: **275*5048707000 VoipTalk: **473*5048707000 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Music On Hold WAITEXTEN(,m)
Scratch the last mail this is the right one Hi, I have an issue, I can hear music on hold with MusicOnHold() but I cant hear anything with WaitExten (,m). (I did: make mpg123, cd mpg.., make, make install). My extensions.conf is: [incoming] exten =s,1,Answer() exten =s,n,Background(welcome) exten =s,n,WaitExten(20,m) ;at this point the debugger doesnt say anything about music on hold ; I dial 2 exten =2,1,MusicOnHold() ;the debugger says Started music on hold.. and I can hear the music Any ideas? Miguel Soto ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
