Re: [asterisk-users] MixMonitor
none ? 2010/11/5 Mickael MONSIEUR mickael.monsi...@gmail.com Hi, Have you noticed a marked increase in CPU load when using MixMonitor? I use PHPAgi and Asterisk 1.6.2.9-2. Mickael. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MixMonitor
On 5 Nov 2010, at 01:22, Mickael MONSIEUR wrote: Have you noticed a marked increase in CPU load when using MixMonitor? Since when? 1.6.2.9-1? 1.6.2.8? 1.0? S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mobile Phones and Asterisk
Hi, one way to solve the problem with Mailbox or that Message that get's played when busy/not available (same happens with Orange in Austria and other providers) you can implement something similar to what Elastix/FreePBX has. Confirm call - this will let the caller think it's still ringing while you will have to confirm the call after picking it up by dialing 1#. I use this when traveling through more then one country. Since I don't want to always change the GSM Number that is dialed when not in the office I simply send the call to ALL GSM Numbers with this option activated. Whichever I answer and press 1# gets the call. Cris On 2 Nov, 2010, at 04:30 , GBR Icasiano, Ryan A. wrote: Yup, that's exactly what is happening. If there is only a way to override the response(busy tone) by a ringing tone from asterisk, then the caller will not hang up since after the busy status interpreted by asterisk as NOANSWER, there will be a fallback which it will either transfer to another extension or go directly to the callee's voicemail. regards, RYAN ICASIANO From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Sebastian [s...@open-t.co.uk] Sent: Sunday, October 31, 2010 9:24 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Mobile Phones and Asterisk On 10/29/2010 04:40 AM, jon pounder wrote: On 10/28/2010 11:18 PM, GBR Icasiano, Ryan A. wrote: Here is what I do today and it works fine: - asterisk/trixbox - Dext/android phone - Bell Canada cell provider - call comes in, to an extension with voicemail - rings a bunch of sip devices (real phones, and the android via linphone if it happens to be near wifi and registered (set to only use wifi not 3g to register) - if not answered call is forwarded back out a pots line and dials the cell number (cell is not subscribed to provider voicemail) This is an advantage over my situation. Here (UK) - if you don't configure voicemail on your mobile - the mobile operator just plays a message along the lines The phone number is not available right now. Please try again later (or something similar). Which screws things up - as Asterisk can't tell that the mobile is not available. To Asterisk, that message is the same as somebody answering the line. Same in France and Spain - as far as I've seen. Sebastian - still no answer that pots line is hung up and call drops back into the original extension's vm. (I have not run into a problem with answer detection, only that people don't stay on the line long enough for me to answer on the second set of ringing, but if they are that impatient the call was probably not important anyway) outgoing calls if registered I have a choice once I dial of linphone or dialer to make the call. checking vm is just *98ext from linphone as the dialing app, or dial in and navigate to vm. linphone is a little less polished gui but seems to work the best for me to reliably register when it should. (tried about 5 different sip clients) Hi, Thanks for your very informative response. This is really helpful. I wouldn't be pushing it though since it isn't possible as of now. Kudos! RYAN ICASIANO From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Sebastian [s...@open-t.co.uk] Sent: Friday, October 29, 2010 5:50 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Mobile Phones and Asterisk Hi, On 10/28/2010 11:20 AM, GBR Icasiano, Ryan A. wrote: Hi, I can actually place a successful call using that configuration. The telco i'm currently working requires the prefix. What I'm trying to do is to capture the status of the mobile phone, if it is currently engaged in a call or not. Maybe others who know better will jump in - but I seriously doubt you will be able to do this. From my limited knowledge, I believe mobile phone networks use different signalling then regular terrestrial based providers. I don't really think that the engaged tone sent back by the mobile operator will be decoded correctly by Asterisk. Not to mention that, I don't what happens where you are - but in UK for example - you don't even get an engaged tone from a mobile phone. You just get either sent to the user's voice mail, or you are played a message from the mobile phone operator which essentially tells you that the user is engaged or unavailable. Operators in many other European countries do the same. So from the point of what you are trying to achieve - this is useless in Asterisk. I would have liked to do the same thing - as I have line divert in Asterisk to my mobile phone - and I would have liked for Asterisk to just skip along to my Asterisk voice mail when my mobile is either out of coverage, or when I'm in a conversation on
Re: [asterisk-users] upgrade 1.6 - 1.8: wrong password!
Dear Paul, I submitted the issue to the tracker. ID 0018263 Thanks pepesz On Thu, Nov 4, 2010 at 8:46 PM, Paul Belanger paul.belan...@polybeacon.comwrote: On Thu, Nov 4, 2010 at 3:24 PM, pepesz pep...@gmail.com wrote: snip WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=5fcd5fa1 /snip I'm surprised to see the extra whitespaces in the nonce value. What can be the problem? If your working configuration worked with 1.6.2 but not 1.8, please created a new issue on the tracker and we will triage it. Also include a debug log [1]. [1] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) | Blog: http://blog.polybeacon.com | Twitter: http://twitter.com/pabelanger -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MoH streamers for asterisk
El 04/11/10 17:14, Tzafrir Cohen escribió: In the 'files' mode Asterisk plays the music separately for each channel. If you use mpg123 or any other streamer, there is a single stream per class. A single stream per class sounds like good efficiency. Could you please tell me what streamers can be used with asterisk to stream say, gsm or wav files for MoH? I'd appreciate this info, we usually use 'files' mode and changing that could lower the load on asterisk. Thanks, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Short rings for extensions when part of the Queue
On Fri, Nov 5, 2010 at 1:18 AM, Bruce B bruceb...@gmail.com wrote: Chad, You are absolutely right on this one. I had setup the Queue time out for agent set to 15 seconds and retry to 2 seconds. So, I think during those two seconds Asterisk for some crazy reason hits another extension and then comes back to the same extension to ring again. So, I have setup the agents to ring for ever for this call center since their agents always have to present or logout if not present. I will see the behavior tomorrow as they test it. My issue might be solved but for those call centers where you want the Queue to move onto the next agent or if you don't want to ring for ever and take a Retry break then it will still remain an issue. I will report back if setting to ring Unlimited doesn't work. Warren, The CLI shows the regular stuff. Nothing out of the ordinary but that it move on to the next agent because the first agent has timed-out for two seconds. Regards, Bruce Have you considered setting the queue timeout to 14 or 16 seconds and retry to 2 seconds? This way the timeout and the retry should line up better. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MixMonitor
Hi, marked - noticed. I do not know where it comes from, my CPU goes from 2% to 60-70% at a command Dial (sip) + MixMonitor. I have an Intel (R) Core (TM) 2 Duo CPU e4...@2.40ghz 2010/11/5 Norbert Zawodsky norb...@zawodsky.at Am 05.11.2010 10:16, schrieb Mickael MONSIEUR: none ? 2010/11/5 Mickael MONSIEUR mickael.monsi...@gmail.com mailto:mickael.monsi...@gmail.com Hi, Have you noticed a marked increase in CPU load when using MixMonitor? I use PHPAgi and Asterisk 1.6.2.9-2. Mickael. Obviously, if the box has more to do, CPU load will increase. What do you mean with marked ?? Norbet -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] VoIP Uses Conference: Friday November 5th: Cloud Computing
This is just a brief reminder that today's VUC call will be about cloud computing with some emphasis on voice applciations: We have assembled a small panel of experienced people to discuss the mattering, including: Eric Chamberlain, Founder of RF.com, Presenter to Astricon 2009 on running Asterisk in the Amazon EC2 cloud. Jason Goecke, Tropo Asterisk developer, founder of Adhearsion, now with Voxeo Greg Weidenhammer A VUC regular who works on massive scale systems for HP. Tim Higgins, Publisher of Small Net Builder Also publisher of the new site Small Cloud builder It starts at 12 noon EDT. For more info or connect details see http://vuc.me Michael -- Michael Graves mgravesatmstvp.com http://www.mgraves.org o713-861-4005 c713-201-1262 sip:mgra...@mstvp.onsip.com skype mjgraves Twitter mjgraves -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] GROUP_COUNT not counting correctly
Hello,
this is a test to add a channel to multiple GROUPs.
this is my dialplan :
exten = s,n,NoOp(groepcount = ${GROUP_COUNT(40)})
exten = s,n,Set(GROUP(40)=40)
exten = s,n,NoOp(This channel is member of : ${GROUP_LIST()})
exten = s,n,NoOp(groepcount = ${GROUP_COUNT(40)})
exten = s,n,NoOp(groepcount = ${GROUP_COUNT(40L)})
exten = s,n,Set(GROUP(40)=40L)
exten = s,n,NoOp(This channel is member of : ${GROUP_LIST()})
exten = s,n,NoOp(groepcount = ${GROUP_COUNT(40L)})
this is the output on a first call :
[Nov 5 17:17:31] -- Executing [...@sub-checkchannels:17]
NoOp(SIP/testcorp7-0036, groepcount = 0) in new stack
[Nov 5 17:17:31] -- Executing [...@sub-checkchannels:18]
Set(SIP/testcorp7-0036, GROUP(40)=40) in new stack
[Nov 5 17:17:31] -- Executing [...@sub-checkchannels:19]
NoOp(SIP/testcorp7-0036, This channel is member of : 4...@40) in
new stack
[Nov 5 17:17:31] -- Executing [...@sub-checkchannels:20]
NoOp(SIP/testcorp7-0036, groepcount = 1) in new stack
[Nov 5 17:17:31] -- Executing [...@sub-checkchannels:24]
NoOp(SIP/testcorp7-0036, groepcount = 0) in new stack
[Nov 5 17:17:31] -- Executing [...@sub-checkchannels:25]
Set(SIP/testcorp7-0036, GROUP(40)=40L) in new stack
[Nov 5 17:17:31] -- Executing [...@sub-checkchannels:26]
NoOp(SIP/testcorp7-0036, This channel is member of : 4...@40) in
new stack
[Nov 5 17:17:31] -- Executing [...@sub-checkchannels:27]
NoOp(SIP/testcorp7-0036, groepcount = 1) in new stack
this is the output on a second call :
[Nov 5 17:17:43] -- Executing [...@sub-checkchannels:17]
NoOp(SIP/testcorp6-0037, groepcount = 0) in new stack
[Nov 5 17:17:43] -- Executing [...@sub-checkchannels:18]
Set(SIP/testcorp6-0037, GROUP(40)=40) in new stack
[Nov 5 17:17:43] -- Executing [...@sub-checkchannels:19]
NoOp(SIP/testcorp6-0037, This channel is member of : 4...@40) in
new stack
[Nov 5 17:17:43] -- Executing [...@sub-checkchannels:20]
NoOp(SIP/testcorp6-0037, groepcount = 1) in new stack
[Nov 5 17:17:43] -- Executing [...@sub-checkchannels:24]
NoOp(SIP/testcorp6-0037, groepcount = 1) in new stack
[Nov 5 17:17:43] -- Executing [...@sub-checkchannels:25]
Set(SIP/testcorp6-0037, GROUP(40)=40L) in new stack
[Nov 5 17:17:43] -- Executing [...@sub-checkchannels:26]
NoOp(SIP/testcorp6-0037, This channel is member of : 4...@40) in
new stack
[Nov 5 17:17:43] -- Executing [...@sub-checkchannels:27]
NoOp(SIP/testcorp6-0037, groepcount = 2) in new stack
Notice that the GROUP_COUNT for 4...@40 in the second call is reset to 0
(zero) to be added by 1 again.
Notice that the GROUP_COUNT for 4...@40 in the second call is added by 1
to result in a total of 2.
Why is the GROUP_COUNT of 4...@40 not 2 also ??
Kind regards,
Jonas.
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Re: [asterisk-users] Short rings for extensions when part of the Queue
Yeah, I think I had it set to 2 seconds and that creates that short ring on another extension. Thanks, On Fri, Nov 5, 2010 at 9:47 AM, Mark Deneen mden...@gmail.com wrote: On Fri, Nov 5, 2010 at 1:18 AM, Bruce B bruceb...@gmail.com wrote: Chad, You are absolutely right on this one. I had setup the Queue time out for agent set to 15 seconds and retry to 2 seconds. So, I think during those two seconds Asterisk for some crazy reason hits another extension and then comes back to the same extension to ring again. So, I have setup the agents to ring for ever for this call center since their agents always have to present or logout if not present. I will see the behavior tomorrow as they test it. My issue might be solved but for those call centers where you want the Queue to move onto the next agent or if you don't want to ring for ever and take a Retry break then it will still remain an issue. I will report back if setting to ring Unlimited doesn't work. Warren, The CLI shows the regular stuff. Nothing out of the ordinary but that it move on to the next agent because the first agent has timed-out for two seconds. Regards, Bruce Have you considered setting the queue timeout to 14 or 16 seconds and retry to 2 seconds? This way the timeout and the retry should line up better. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk in the third world - Astricon 2010 keynote follow-up
Friends, After listening to Mark Summer's keynote at Astricon (hopefully soon on the Astricon web site) I think we should come back to the discussion he started. Mark talked about using Open Source in general and Asterisk in particular in third world projects as well as in emergencies in other countries. He and Inveneo help groups of people to get a better understanding of how to build network, IP and voice infrastructures. One part is of course learning and managing Asterisk. I do believe many of us wants to help his efforts, but lack the understanding and channels to reach out. I had a very brief discussion with Mark after the keynote and promised to get back to him. My thoughts are that if anyone from these countries try to reach us, we fail to listen and help. Could be language, could be attitude or something else. We can't expect them to have full understanding of net etiquette, the rules of Open Source project management or how to find information themselves (in a language they might not understand fully). The climate in our mailing lists and chat rooms are not always one of understanding, especially if someone copies their english language and attitude from Miami Vice ;-) Do you have any ideas of what could be done from our community? Can we create special forums where we have a different climate, more languages and better understanding? I also think we should copy ISOCs efforts and have a pre-astricon training/workshop for people that Inveneo locate and then invite them to Astricon, funded by grants form community or from somewhere else (since we lack an Asterisk foundation that could help here). I'm sure we can find resources to get them to Astricon and that we can find teachers in the community that are willing to help with this project. I would not hesitate in donating a few days myself. We have enormous powers in our community. If we can gather a small part of that and point it towards these people, we can change the situation for many more, just by doing what we do each day - enjoy building voice solutions and sharing our knowledge. Let's brainstorm for a while! The floor is open. /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to append custom option to Contact: header on outgoing SIP INVITE msgs?
Hi list, My need is to append a site specific parameter to the Contact: header on all INVITEs exiting * via a SIP trunk. I'd like it to look something like this: Contact: bob:3125551...@10.10.10.10;SITE-ID=us.here where SITE-ID=us.here is set in a config file that * parses on startup. Or in a Dial() command option? Or I don't care exactly how. :-) It is possible to affect the Contact: header via a line in sip.conf: register = toronto:welc...@192.168.1.101/contact but I can't get it to also accept any ;X=Y params for the contact. I can add a custom Contact header in the dialplan with SipAddHeader, but then I have two. SipRemoveHeader only removes headers previously added by SipAddHeader, so no luck there. I have googled, and searched the asterisk-users mailing list archives and not yet found a solution. I did see some work back in 2004 (issues 732 and 777) which mentioned not stripping contact header parameters from arriving requests/registrations, but nothing about creating any such parameters. Thanks for any help/hints, -- -Bob Beers -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Funky IAX behavior between 1.4 and 1.8
Hi Gang, My production box with my DAHDI cards is a 1.4.26 build. I have 3 test machines that I do IAX communication with. Machine 1 is a real Dell POWEREDGE 1500 running CENTOS running 1.4.30. Machine 2 is a SUSE 11.1 VM running 1.4.30. Machine 3 is another SUSE 11.1 VM running 1.8.0. I can SIP into all 4 machines and life is great. When I try to IAX from the live machine to Machine 3, I get lags/pauses on Background/Playback commands. I play files and groups of files that last from 1-45 seconds, so I can press keys and proceed, but I don't expect my end-users to know to do this. Any clues? Do I need to open a tracker issue on this one? Thanks Danny Nicholas -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Elementary question - accessing feature codes from cell phone
Hi, please forgive me for this (hopefully) simple question. I cannot seem to find an answer or solution while searching around. I want to be able to call in to my server using my cell phone and be able to set call forwarding for my extension and enter a phone number and also be able to call in to that extension and disable the call forwarding. I see I can do this through the ARI web interface but wish to do this with my cell phone. Currently, it seems I can only get into my voicemail and attempting to run feature codes like *72 don't get recognized. I currently have a DID assigned to my extension if this helps. I am not sure if it is required. Thanks much. jellydog -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Elementary question - accessing feature codes fromcell phone
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Regal Sent: Friday, November 05, 2010 10:11 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Elementary question - accessing feature codes fromcell phone Hi, please forgive me for this (hopefully) simple question. I cannot seem to find an answer or solution while searching around. I want to be able to call in to my server using my cell phone and be able to set call forwarding for my extension and enter a phone number and also be able to call in to that extension and disable the call forwarding. I see I can do this through the ARI web interface but wish to do this with my cell phone. Currently, it seems I can only get into my voicemail and attempting to run feature codes like *72 don't get recognized. I currently have a DID assigned to my extension if this helps. I am not sure if it is required. Thanks much. jellydog Hope this answer is more helpful than harmful, but in my experience and reading, feature codes and cell phones don't play well together. DTMF processing is usually way less than 100% reliable in this setup. Your best bet is probably to replicate the feature function you want into an extension (1234 instead of *72) and dialing that from your cell or using the web interface on your cell to do the ARI function. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk default sound files
Hi, I installed BigBlueButton and I want to change default conference playbacks. Is it possible ? if Yes, how :) Thank you. Bu elektronik postada bulunan tum fikir ve gorusler ve ekindeki dosyalar sadece adres sahip/sahiplerine ait olup, Yasar Toplulugu Sirketleri bu mesajin icerigi ile ilgili olarak hic bir hukuksal sorumlulugu kabul etmez. Eger gonderilmesi dusunulen kisi veya kurulus degilseniz, lutfen gonderen kisiyi derhal haberdar ediniz ve mesaji sisteminizden siliniz.The information contained in this e-mail and any files transmitted with it are intended solely for the use of the individual or entity to whom they are addressed and Yasar Group Companies do not accept legal responsibility for the contents. If you are not the intended recipient, please immediately notify the sender and delete it from your system.-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.8 Installation Problem
Hi, We want to upgrade both our servers to asterisk 1.8, the one from Romania and the one from Chicago, but for the moment I`m trying to install Asterisk 1.8 on a test machine running CentOS 5.5 with the kernel: Linux asterisk3 2.6.18-194.17.4.el5PAE #1 SMP Mon Oct 25 16:35:27 EDT 2010 i686 i686 i386 GNU/Linux . I`ve tried many things from the forums and mailing lists but none seemed to help me. Our problem is that when we want to compile asterisk 1.8 we get this error: /packages/asterisk-1.8.0/addons/chan_ooh323.c:3888: multiple definition of `configure_local_rtp' ../addons/chan_ooh323.eo:(.text+0xd100): first defined here ../addons/chan_ooh323.o: In function `ooh323_update_capPrefsOrderForCall': /packages/asterisk-1.8.0/addons/chan_ooh323.c:3803: multiple definition of `ooh323_update_capPrefsOrderForCall' ../addons/chan_ooh323.eo:(.text+0xe1f0): first defined here /usr/bin/ld: Dwarf Error: Abbrev offset (13856) greater than or equal to .debug_abbrev size (1228). ../channels/chan_mgcp.eo: In function `mgcp_hangup': chan_mgcp.c:(.text+0xaf04): undefined reference to `ast_pktccops_gate_alloc' ../channels/chan_mgcp.eo: In function `start_rtp': chan_mgcp.c:(.text+0xbdb2): undefined reference to `ast_pktccops_gate_alloc' collect2: ld returned 1 exit status make[1]: *** [asterisk] Error 1 make: *** [main] Error 2 I don’t know if this helps but I`ve also copied a part of the config.log file which contains the ld word: [r...@asterisk3 asterisk-1.8.0]# less config.log | grep ld configure:3302: checking build system type configure:6461: checking for ld used by gcc configure:6528: result: /usr/bin/ld configure:6535: checking if the linker (/usr/bin/ld) is GNU ld /usr/bin/ld: cannot find -lpthreads collect2: ld returned 1 exit status collect2: ld returned 1 exit status collect2: ld returned 1 exit status collect2: ld returned 1 exit status /usr/bin/ld: cannot find -llthread collect2: ld returned 1 exit status configure:9680: gcc -o conftest -g -O2conftest.c -L/usr/kerberos/lib -lcurl -ldl -lgssapi_krb5 -lkrb5 -lk5crypto -lcom_err -lidn -lssl -lcrypto -lz5 configure:9707: gcc -o conftest -g -O2conftest.c -L/usr/kerberos/lib -lcurl -ldl -lgssapi_krb5 -lkrb5 -lk5crypto -lcom_err -lidn -lssl -lcrypto -lz 5 collect2: ld returned 1 exit status |builtin and then its argument prototype would still apply. */ collect2: ld returned 1 exit status |builtin and then its argument prototype would still apply. */ collect2: ld returned 1 exit status |builtin and then its argument prototype would still apply. */ collect2: ld returned 1 exit status |builtin and then its argument prototype would still apply. */ collect2: ld returned 1 exit status |builtin and then its argument prototype would still apply. */ collect2: ld returned 1 exit status |builtin and then its argument prototype would still apply. */ configure:14123: checking for strtold collect2: ld returned 1 exit status |builtin and then its argument prototype would still apply. */ collect2: ld returned 1 exit status |builtin and then its argument prototype would still apply. */ configure:14268: checking for a version of GNU ld that supports the --dynamic-list flag collect2: ld returned 1 exit status |builtin and then its argument prototype would still apply. */ collect2: ld returned 1 exit status collect2: ld returned 1 exit status |builtin and then its argument prototype would still apply. */ collect2: ld returned 1 exit status |builtin and then its argument prototype would still apply. */ collect2: ld returned 1 exit status |builtin and then its argument prototype would still apply. */ collect2: ld returned 1 exit status |builtin and then its argument prototype would still apply. */ collect2: ld returned 1 exit status collect2: ld returned 1 exit status |builtin and then its argument prototype would still apply. */ collect2: ld returned 1 exit status |builtin and then its argument prototype would still apply. */ collect2: ld returned 1 exit status configure:16677: gcc -o conftest -g -O2conftest.c -lasound -lm -ldl -lm 5 /usr/bin/ld: cannot find -lgsm collect2: ld returned 1 exit status |builtin and then its argument prototype would still apply. */ /usr/bin/ld: cannot find -liconv collect2: ld returned 1 exit status |builtin and then its argument prototype would still apply. */ /usr/bin/ld: cannot find -lical collect2: ld returned 1 exit status |builtin and then its argument prototype would still apply. */ collect2: ld returned 1 exit status /usr/bin/ld: cannot find -liodbc collect2: ld returned 1 exit status |builtin and then its argument prototype would still apply. */ /usr/bin/ld: cannot find -ljack collect2: ld returned 1 exit status |builtin and then its argument prototype would still apply. */ collect2: ld returned 1 exit status |builtin and then its argument prototype would still apply. */ collect2: ld returned 1 exit status
Re: [asterisk-users] Asterisk default sound files
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Erol Demir Sent: Friday, November 05, 2010 10:22 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk default sound files Hi, I installed BigBlueButton and I want to change default conference playbacks. Is it possible ? if Yes, how :-) Thank you. All Asterisk sound files (unless you have tinkered with configs) will be in /var/lib/asterisk/sounds. To replace a file, simply look at CLI output for the name and copy a new file over that one or use Record command to put your own file there. Bu elektronik postada bulunan tum fikir ve gorusler ve ekindeki dosyalar sadece adres sahip/sahiplerine ait olup, Yasar Toplulugu Sirketleri bu mesajin icerigi ile ilgili olarak hic bir hukuksal sorumlulugu kabul etmez. Eger gonderilmesi dusunulen kisi veya kurulus degilseniz, lutfen gonderen kisiyi derhal haberdar ediniz ve mesaji sisteminizden siliniz.The information contained in this e-mail and any files transmitted with it are intended solely for the use of the individual or entity to whom they are addressed and Yasar Group Companies do not accept legal responsibility for the contents. If you are not the intended recipient, please immediately notify the sender and delete it from your system. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Short rings for extensions when part of the Queue
On Fri, Nov 5, 2010 at 10:38 AM, Bruce B bruceb...@gmail.com wrote: Yeah, I think I had it set to 2 seconds and that creates that short ring on another extension. Thanks, The point was that 14 and 16 are divisible by 2 (evenly) while 15 is not. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Funky IAX behavior between 1.4 and 1.8
On Fri, Nov 5, 2010 at 11:04 AM, Danny Nicholas da...@debsinc.com wrote: Hi Gang, My production box with my DAHDI cards is a 1.4.26 build. I have 3 test machines that I do IAX communication with. Machine 1 is a real Dell POWEREDGE 1500 running CENTOS running 1.4.30. Machine 2 is a SUSE 11.1 VM running 1.4.30. Machine 3 is another SUSE 11.1 VM running 1.8.0. I can SIP into all 4 machines and life is great. When I try to IAX from the live machine to Machine 3, I get lags/pauses on Background/Playback commands. I play files and groups of files that last from 1-45 seconds, so I can press keys and proceed, but I don’t expect my end-users to know to do this. Any clues? Do I need to open a tracker issue on this one? Thanks Danny Nicholas https://issues.asterisk.org/view.php?id=18105 From my testing, I have seen it happen in both inside a VM and outside a VM... but it happens far more reliably in a VM. I added some debug code to chan_iax2.c, but I haven't been able to find anything useful yet besides a better idea of how IAX2 works. :) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Elementary question - accessing feature codes fromcell phone
DTMF sent from cell phones are usually not well recognized at the asterisk end. The main reason for this is that cell phones transmit out-of-band DTMF, which by the time reaches an asterisk server traveling through cell towers, their equipment, various VoIP carriers etc. is usually drifted away from its acceptable frequency threshhold. Or if a carrier is converting it into inband, it might not be right at this carrier's end, meaningful it'll have no tone at all. Receiving out-of-band DTMF over physical lines like T1s is usually much more reliable than SIP, because the expensive equipment at big telcos is better at fixing up bad tones and send you the correct tones. Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com On 2010-11-05 11:24 AM, Danny Nicholas da...@debsinc.com wrote: -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *John Regal *Sent:* Friday, November 05, 2010 10:11 AM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] Elementary question - accessing feature codes fromcell phone Hi, please forgive me for this (hopefully) simple question. I cannot seem to find an answer or ... Hope this answer is more helpful than harmful, but in my experience and reading, feature codes and cell phones don’t play well together. DTMF processing is usually way less than 100% reliable in this setup. Your best bet is probably to replicate the feature function you want into an extension (1234 instead of *72) and dialing that from your cell or using the web interface on your cell to do the ARI function. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Phones slow to ring
It worked! I['ll have to figure out how to add the dial string to the phone. Thanks a bunch for your help On Thu, Nov 4, 2010 at 9:04 PM, Mark Phillips g7...@g7ltt.com wrote: I would second that. If you don't set a dial string in your handset then it waits for N seconds before submitting the call. Pressing # will force an immediate dial. Mark On 11/04/2010 07:19 PM, Cary Fitch wrote: Watch the console as you dial. Dial the number and “#”. The ring should be “instant”. Or if not, look at the console and report what you see. Cary Fitch *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *jy *Sent:* Thursday, November 04, 2010 5:32 PM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] Phones slow to ring I am new to asterisk and using it for a research project. Have set up an server (version 1.6.2.6) and 2 SIP phones (Linksys spa901) which are registering fine with the server. They are able to call one another, however, the problem is it takes roughly 8-10 seconds for the called phone to ring. I have a really simple dialplan using only 4 digit extensions and have turned off callerid. Both phones are on the same subnet and I have enabled nat and keepalive. Does anyone have an idea what could be wrong here or idea on how to debug this problem? Thanks, John -- /\/\ark Phillips -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Elementary question - accessing feature codes fromcell phone
Thanks for the quick response! I have had a lot of issues in the past with DTMF. Anyway, I think the idea of replicating the function into an extension will work. Any pointers on the best way to accomplish this? I created a new extension but am unsure what to do next. I thought about the FollowMe feature but I would have to hardcode the number and I want to be able to enter a forwarding phone number for the extension using my cell. Thanks again. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Friday, November 05, 2010 11:21 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Elementary question - accessing feature codes fromcell phone _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Regal Sent: Friday, November 05, 2010 10:11 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Elementary question - accessing feature codes fromcell phone Hi, please forgive me for this (hopefully) simple question. I cannot seem to find an answer or solution while searching around. I want to be able to call in to my server using my cell phone and be able to set call forwarding for my extension and enter a phone number and also be able to call in to that extension and disable the call forwarding. I see I can do this through the ARI web interface but wish to do this with my cell phone. Currently, it seems I can only get into my voicemail and attempting to run feature codes like *72 don't get recognized. I currently have a DID assigned to my extension if this helps. I am not sure if it is required. Thanks much. jellydog Hope this answer is more helpful than harmful, but in my experience and reading, feature codes and cell phones don't play well together. DTMF processing is usually way less than 100% reliable in this setup. Your best bet is probably to replicate the feature function you want into an extension (1234 instead of *72) and dialing that from your cell or using the web interface on your cell to do the ARI function. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] No audio with gtalk client behind http proxy
Hi all, I'm trying to establish jingle call in this network scenario: Asterisk - NAT - Internet - HTTP_PROXY - GTalk client The call is received and answered in gtalk but there is no audio in the call. I suppose it could be related to the support for relay candidates in asterisk jingle implementation.Anybody else has faced this problem? Notes: Asterisk 1.6.2. It works fine with natted gtalk clients not being behind proxies. G. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Elementary question - accessing feature codes fromcell phone
On Fri, Nov 05, 2010 at 12:49:45PM -0400, John Regal wrote:
Anyway, I think the idea of replicating the function into an extension will
work. Any pointers on the best way to accomplish this? I created a new
extension but am unsure what to do next. I thought about the FollowMe
feature but I would have to hardcode the number and I want to be able to
enter a forwarding phone number for the extension using my cell.
You could set up an extension match that triggers on
(feature ID)(access code)(extension)
as it might be, with an access code of 62889:
exten = _*7262889.,1,Set(FWDNUM=${EXTEN:8})
and then put FWDNUM into the astdb or however else you want to handle
it.
--
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Re: [asterisk-users] Asterisk 1.8 Installation Problem
Thanks for the answer. All of those libraries are already installed and it's still not working. Package libstdc++-devel-4.1.2-48.el5.i386 already installed and latest version Package matching libxml2-devel-2.6.26-2.1.2.8.i386 already installed. Checking for update. Package openssl-devel-0.9.8e-12.el5_4.6.i386 already installed and latest version Package slang-devel-2.0.6-4.el5.i386 already installed and latest version Package libsepol-devel-1.15.2-3.el5.i386 already installed and latest version Package ncurses-devel-5.5-24.20060715.i386 already installed and latest version Package libtool-ltdl-devel-1.5.22-7.el5_4.i386 already installed and latest version Package newt-devel-0.52.2-15.el5.i386 already installed and latest version Package matching glibc-devel-2.5-49.el5_5.4.i386 already installed. Checking for update. Package zlib-devel-1.2.3-3.i386 already installed and latest version Package libselinux-devel-1.33.4-5.5.el5.i386 already installed and latest version Package krb5-devel-1.6.1-36.el5_5.5.i386 already installed and latest version Package keyutils-libs-devel-1.2-1.el5.i386 already installed and latest version Package e2fsprogs-devel-1.39-23.el5.i386 already installed and latest version Package unixODBC-devel-2.2.11-7.1.i386 already installed and latest version Any other ideas ? Thanks, Bogdan Sarandan -Original Message- From: Tilghman Lesher Sent: Friday, November 05, 2010 12:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.8 Installation Problem On Friday 05 November 2010 10:21:32 Bogdan Sarandan wrote: We want to upgrade both our servers to asterisk 1.8, the one from Romania and the one from Chicago, but for the moment I`m trying to install Asterisk 1.8 on a test machine running CentOS 5.5 with the kernel: Linux asterisk3 2.6.18-194.17.4.el5PAE #1 SMP Mon Oct 25 16:35:27 EDT 2010 i686 i686 i386 GNU/Linux . I`ve tried many things from the forums and mailing lists but none seemed to help me. Our problem is that when we want to compile asterisk 1.8 we get this error: /packages/asterisk-1.8.0/addons/chan_ooh323.c:3888: multiple definition of `configure_local_rtp' ../addons/chan_ooh323.eo:(.text+0xd100): first defined here ../addons/chan_ooh323.o: In function `ooh323_update_capPrefsOrderForCall': /packages/asterisk-1.8.0/addons/chan_ooh323.c:3803: multiple definition of `ooh323_update_capPrefsOrderForCall' ../addons/chan_ooh323.eo:(.text+0xe1f0): first defined here /usr/bin/ld: Dwarf Error: Abbrev offset (13856) greater than or equal to .debug_abbrev size (1228). ../channels/chan_mgcp.eo: In function `mgcp_hangup': chan_mgcp.c:(.text+0xaf04): undefined reference to `ast_pktccops_gate_alloc' ../channels/chan_mgcp.eo: In function `start_rtp': chan_mgcp.c:(.text+0xbdb2): undefined reference to `ast_pktccops_gate_alloc' collect2: ld returned 1 exit status make[1]: *** [asterisk] Error 1 make: *** [main] Error 2 Does anyone have any idea what should we do in order to get it working ? From what I know the library libgpgme-pthread11.i386 was needed in order to have lpthread but with no luck , still doesn’t work. This isn't correct. The pthreads library should already be installed by default. The package you cite above is an encryption library. I have a barebones CentOS 5.5 machine here, which I use for testing, and the devel packages which I have installed are: libstdc++-devel-4.1.2-48.el5 libxml2-devel-2.6.26-2.1.2.8 openssl-devel-0.9.8e-12.el5_4.6 slang-devel-2.0.6-4.el5 libsepol-devel-1.15.2-3.el5 ncurses-devel-5.5-24.20060715 libtool-ltdl-devel-1.5.22-7.el5_4 newt-devel-0.52.2-15.el5 glibc-devel-2.5-49.el5_5.4 zlib-devel-1.2.3-3 libselinux-devel-1.33.4-5.5.el5 krb5-devel-1.6.1-36.el5_5.5 keyutils-libs-devel-1.2-1.el5 e2fsprogs-devel-1.39-23.el5 unixODBC-devel-2.2.11-7.1 Given that I do not have any problems compiling Asterisk 1.8.0 (including ooh323, which I had to enable, and chan_mgcp), I would suggest that you verify that each of these packages is installed on your system. If you install any of these, and that fixes the problem, please report back which package solved the problem. I'd love to make a configure test to verify that all required packages are installed before configure will succeed. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by
Re: [asterisk-users] Asterisk 1.8 Installation Problem
I`ve disabled chan_ooh323 and res_adsi and it worked . Bogdan -Original Message- From: Bogdan Sarandan Sent: Friday, November 05, 2010 12:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.8 Installation Problem Thanks for the answer. All of those libraries are already installed and it's still not working. Package libstdc++-devel-4.1.2-48.el5.i386 already installed and latest version Package matching libxml2-devel-2.6.26-2.1.2.8.i386 already installed. Checking for update. Package openssl-devel-0.9.8e-12.el5_4.6.i386 already installed and latest version Package slang-devel-2.0.6-4.el5.i386 already installed and latest version Package libsepol-devel-1.15.2-3.el5.i386 already installed and latest version Package ncurses-devel-5.5-24.20060715.i386 already installed and latest version Package libtool-ltdl-devel-1.5.22-7.el5_4.i386 already installed and latest version Package newt-devel-0.52.2-15.el5.i386 already installed and latest version Package matching glibc-devel-2.5-49.el5_5.4.i386 already installed. Checking for update. Package zlib-devel-1.2.3-3.i386 already installed and latest version Package libselinux-devel-1.33.4-5.5.el5.i386 already installed and latest version Package krb5-devel-1.6.1-36.el5_5.5.i386 already installed and latest version Package keyutils-libs-devel-1.2-1.el5.i386 already installed and latest version Package e2fsprogs-devel-1.39-23.el5.i386 already installed and latest version Package unixODBC-devel-2.2.11-7.1.i386 already installed and latest version Any other ideas ? Thanks, Bogdan Sarandan -Original Message- From: Tilghman Lesher Sent: Friday, November 05, 2010 12:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.8 Installation Problem On Friday 05 November 2010 10:21:32 Bogdan Sarandan wrote: We want to upgrade both our servers to asterisk 1.8, the one from Romania and the one from Chicago, but for the moment I`m trying to install Asterisk 1.8 on a test machine running CentOS 5.5 with the kernel: Linux asterisk3 2.6.18-194.17.4.el5PAE #1 SMP Mon Oct 25 16:35:27 EDT 2010 i686 i686 i386 GNU/Linux . I`ve tried many things from the forums and mailing lists but none seemed to help me. Our problem is that when we want to compile asterisk 1.8 we get this error: /packages/asterisk-1.8.0/addons/chan_ooh323.c:3888: multiple definition of `configure_local_rtp' ../addons/chan_ooh323.eo:(.text+0xd100): first defined here ../addons/chan_ooh323.o: In function `ooh323_update_capPrefsOrderForCall': /packages/asterisk-1.8.0/addons/chan_ooh323.c:3803: multiple definition of `ooh323_update_capPrefsOrderForCall' ../addons/chan_ooh323.eo:(.text+0xe1f0): first defined here /usr/bin/ld: Dwarf Error: Abbrev offset (13856) greater than or equal to .debug_abbrev size (1228). ../channels/chan_mgcp.eo: In function `mgcp_hangup': chan_mgcp.c:(.text+0xaf04): undefined reference to `ast_pktccops_gate_alloc' ../channels/chan_mgcp.eo: In function `start_rtp': chan_mgcp.c:(.text+0xbdb2): undefined reference to `ast_pktccops_gate_alloc' collect2: ld returned 1 exit status make[1]: *** [asterisk] Error 1 make: *** [main] Error 2 Does anyone have any idea what should we do in order to get it working ? From what I know the library libgpgme-pthread11.i386 was needed in order to have lpthread but with no luck , still doesn’t work. This isn't correct. The pthreads library should already be installed by default. The package you cite above is an encryption library. I have a barebones CentOS 5.5 machine here, which I use for testing, and the devel packages which I have installed are: libstdc++-devel-4.1.2-48.el5 libxml2-devel-2.6.26-2.1.2.8 openssl-devel-0.9.8e-12.el5_4.6 slang-devel-2.0.6-4.el5 libsepol-devel-1.15.2-3.el5 ncurses-devel-5.5-24.20060715 libtool-ltdl-devel-1.5.22-7.el5_4 newt-devel-0.52.2-15.el5 glibc-devel-2.5-49.el5_5.4 zlib-devel-1.2.3-3 libselinux-devel-1.33.4-5.5.el5 krb5-devel-1.6.1-36.el5_5.5 keyutils-libs-devel-1.2-1.el5 e2fsprogs-devel-1.39-23.el5 unixODBC-devel-2.2.11-7.1 Given that I do not have any problems compiling Asterisk 1.8.0 (including ooh323, which I had to enable, and chan_mgcp), I would suggest that you verify that each of these packages is installed on your system. If you install any of these, and that fixes the problem, please report back which package solved the problem. I'd love to make a configure test to verify that all required packages are installed before configure will succeed. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello
Re: [asterisk-users] Short rings for extensions when part of the Queue
Sorry, I am not following. If an extension rings for 15 or 16 seconds and then waits for 2 or three seconds what difference does the being divisible make? Is there something internal to Asterisk that makes the Retry time dependent on Time Out (also known as Ring Time)? P.S. I think the 15 seconds is just three rings complete. Thanks, Bruce On Fri, Nov 5, 2010 at 11:31 AM, Mark Deneen mden...@gmail.com wrote: On Fri, Nov 5, 2010 at 10:38 AM, Bruce B bruceb...@gmail.com wrote: Yeah, I think I had it set to 2 seconds and that creates that short ring on another extension. Thanks, The point was that 14 and 16 are divisible by 2 (evenly) while 15 is not. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Elementary question - accessing feature codes from cell phone
We use DISA (http://www.voip-info.org/wiki/view/Asterisk+cmd+DISA) to access our entire [features] context from our cell phones. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Regal Sent: Friday, November 05, 2010 11:11 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Elementary question - accessing feature codes from cell phone Hi, please forgive me for this (hopefully) simple question. I cannot seem to find an answer or solution while searching around. I want to be able to call in to my server using my cell phone and be able to set call forwarding for my extension and enter a phone number and also be able to call in to that extension and disable the call forwarding. I see I can do this through the ARI web interface but wish to do this with my cell phone. Currently, it seems I can only get into my voicemail and attempting to run feature codes like *72 don't get recognized. I currently have a DID assigned to my extension if this helps. I am not sure if it is required. Thanks much. jellydog -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom WEB UI configuration - What needs to be put in for basic SIP registration?
Hi Everyone, Configuring a Polycom conference bridge IP 5000 to connect to Asterisk. For some reason I don't see any SIP packets coming in to Asterisk at all. I don't want to use XML or ftp etc for now and just use the Web Interface to get it going with basic features. But the Web UI is a bit confusing with SIP and Line tabs. I have put this on the web interface: SIP Outbound Proxy: Address = 192.168.0.2 Port = 5060 Server 1: Address = 192.168.0.2 Port = 5060 Transport = DNSnaptr Expires = 300 Register = 1 Line: Display Name = 100 Address = 192.168.0.2 Authentication User ID = 100 Authentication Password = * Label = 100 Server 1: Address = 192.168.0.2 Port = 5060 Transport = DNSnaptr Expires = 300 Register = 1 I don't see any registration attempts but Snom phones on the same network can register to Asterisk just fine. Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Unable to place 2 or more calls to a DID
Hello, I'm having a problem trying to do this, and it used to work with Asterisk 1.4. Since Asterisk 1.6 series I have not been able to place more than one call to a DID. I get this message: Skipping dialing interface 'SIP/16034817...@flowroute' again since it has already been dialed I'm trying to do this because many of my clients have roll-over lines and they need to receive more than one call at a time. Any ideas? Thanks, -Mike Frager -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Unable to place 2 or more calls to a DID
Hello, I'm having a problem trying to do this, and it used to work with Asterisk 1.4. Since Asterisk 1.6 series I have not been able to place more than one call to a DID. I get this message: Skipping dialing interface 'SIP/16034817...@flowroute' again since it has already been dialed I'm trying to do this because many of my clients have roll-over lines and they need to receive more than one call at a time. Any ideas? Thanks, -Mike Frager -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unable to place 2 or more calls to a DID
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Frager Sent: Friday, November 05, 2010 2:33 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Unable to place 2 or more calls to a DID Hello, I'm having a problem trying to do this, and it used to work with Asterisk 1.4. Since Asterisk 1.6 series I have not been able to place more than one call to a DID. I get this message: Skipping dialing interface 'SIP/16034817...@flowroute' again since it has already been dialed I'm trying to do this because many of my clients have roll-over lines and they need to receive more than one call at a time. Any ideas? Thanks, -Mike Frager Maybe you need to use Originate instead of Dial? If you were using DAHDI and dialed DAHDI/1/1603. twice, it would give you a congestion or busy. If you did Dial(DAHDI/g1/1603.) twice it would roll over going out assuming you had more that one DAHDI line and 2 were available. With the literally thousands of folks who use SIP providers I'm surprised I haven't seen this one before now, but then again it's easy to miss something in nearly 2-3000 posts per week. Check the Issue Tracker as well. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Alternative to Proxmox
Hi Everyone, Is there other comparable products to Proxmox to be used for Asterisk instances? Ease of use, web interface, and Asterisk/CentOS support would be ideal. Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Alternative to Proxmox
- Bruce B bruceb...@gmail.com wrote: Hi Everyone, Is there other comparable products to Proxmox to be used for Asterisk instances? Ease of use, web interface, and Asterisk/CentOS support would be ideal. There is OpenNode: http://opennode.activesys.org/ I've heard good things thus far but have not had time nor need to test it myself. --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Alternative to Proxmox
- Tim Nelson tnel...@rockbochs.com wrote: Hi Everyone, Is there other comparable products to Proxmox to be used for Asterisk instances? Ease of use, web interface, and Asterisk/CentOS support would be ideal. There is OpenNode: http://opennode.activesys.org/ I've heard good things thus far but have not had time nor need to test it myself. Oh, and I meant to ask why you're looking for an alternative to Proxmox. Have you had problems with it that cannot be solved? --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] res_ais Error
Hi,
I'm trying distributed events with Openais but don't work.
I made the test with two asterisk box in the same LAN
box A: 192.168.142.246 asterisk 1.6.2.13
BoxB: 192.168.142.248 asterisk 1.8.0
openais.conf:
# Please read the openais.conf.5 manual page
totem {
version: 2
secauth: off
threads: 0
consensus: 4800
interface {
ringnumber: 0
bindnetaddr: 192.168.142.0
mcastaddr: 226.94.1.1
mcastport: 5405
}
}
logging {
to_stderr: yes
debug: on
timestamp: on
to_file: yes
to_syslog: no
syslog_facility: daemon
logfile: /var/log/openais.log
}
amf {
mode: disabled
}
When I load res_ais.so module, the pbx crash (boths)
Some time not crash but no clusters members are present.
What I'm doing wrong?
Thank's
Regards
--
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Re: [asterisk-users] res_ais Error
On Fri, Nov 5, 2010 at 4:44 PM, bakko asannu...@gmail.com wrote: When I load res_ais.so module, the pbx crash (boths) Generate a backtrace[1] and upload to this thread. [1] https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) | Blog: http://blog.polybeacon.com | Twitter: http://twitter.com/pabelanger -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Funky IAX behavior between 1.4 and 1.8
On Fri, Nov 5, 2010 at 10:34 AM, Mark Deneen mden...@gmail.com wrote: On Fri, Nov 5, 2010 at 11:04 AM, Danny Nicholas da...@debsinc.com wrote: Hi Gang, My production box with my DAHDI cards is a 1.4.26 build. I have 3 test machines that I do IAX communication with. Machine 1 is a real Dell POWEREDGE 1500 running CENTOS running 1.4.30. Machine 2 is a SUSE 11.1 VM running 1.4.30. Machine 3 is another SUSE 11.1 VM running 1.8.0. Danny, Do you have any kind of timing on the VM's, or are you running DAHDI_DUMMY? I'm pretty sure IAX requires some kind of internal timing to function properly... -- Thanks, --Warren Selby, dCAP http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Funky IAX behavior between 1.4 and 1.8
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby Sent: Friday, November 05, 2010 4:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Funky IAX behavior between 1.4 and 1.8 On Fri, Nov 5, 2010 at 10:34 AM, Mark Deneen mden...@gmail.com wrote: On Fri, Nov 5, 2010 at 11:04 AM, Danny Nicholas da...@debsinc.com wrote: Hi Gang, My production box with my DAHDI cards is a 1.4.26 build. I have 3 test machines that I do IAX communication with. Machine 1 is a real Dell POWEREDGE 1500 running CENTOS running 1.4.30. Machine 2 is a SUSE 11.1 VM running 1.4.30. Machine 3 is another SUSE 11.1 VM running 1.8.0. Danny, Do you have any kind of timing on the VM's, or are you running DAHDI_DUMMY? I'm pretty sure IAX requires some kind of internal timing to function properly... I added DAHDI so I would have DAHDI_DUMMY available, but the problem still occurs, just a little less. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Soundpoint IP 430 -- discontinued.
Hey, all. I'm in the middle of a rollout, and just learned that the SoundPoint IP 430 -- my favorite mid-range phone -- has been discontinued. The heir apparent is the SoundPoint IP 450 -- for a low, low, low $130 more/handset. AND it doesn't look as nice. Ouch. Does anyone have any recommendations -- Polycom or otherwise -- for a good-quality, mid-range, two-line SIP phone (with good speakerphone) for ~$150/ea.? I realize that there are still some 430's to be had, but they won't be around forever, and now might be the right time for me to be moving forward. Thanks, -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Soundpoint IP 430 -- discontinued.
On Nov 5, 2010, at 5:35 PM, Ken D'Ambrosio wrote: Hey, all. I'm in the middle of a rollout, and just learned that the SoundPoint IP 430 -- my favorite mid-range phone -- has been discontinued. The heir apparent is the SoundPoint IP 450 -- for a low, low, low $130 more/handset. AND it doesn't look as nice. Ouch. Does anyone have any recommendations -- Polycom or otherwise -- for a good-quality, mid-range, two-line SIP phone (with good speakerphone) for ~$150/ea.? I realize that there are still some 430's to be had, but they won't be around forever, and now might be the right time for me to be moving forward. Thanks, -Ken I have a 450 and it's got to be one of my favorite poly's. Great phone. But if you only want 2 line, go for the 3xx series. ---fred http://qxork.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] res_ais Error
Hi, after some test the system don't crash but no members: CLI ais show clm members = === Cluster Members = = === === - === Node Name: === == ID: 0x0 === == Address: === == Member: No === - === = In the openais.log i can see the connection but I don't know why not persistent: Nov 5 23:36:59.361077 [ipc.c:0799] connection received from libais client 5. Nov 5 23:36:59.363573 [ipc.c:0799] connection received from libais client 6. Nov 5 23:37:04.707123 [TOTEM] The consensus timeout expired. Nov 5 23:37:04.707207 [TOTEM] entering GATHER state from 3. Nov 5 23:37:10.526431 [TOTEM] The consensus timeout expired. Nov 5 23:37:10.526506 [TOTEM] entering GATHER state from 3. Nov 5 23:37:16.345764 [TOTEM] The consensus timeout expired. Nov 5 23:37:16.345829 [TOTEM] entering GATHER state from 3. Nov 5 23:37:22.165079 [TOTEM] The consensus timeout expired. Nov 5 23:37:22.165181 [TOTEM] entering GATHER state from 3. Nov 5 23:37:27.984413 [TOTEM] The consensus timeout expired. Nov 5 23:37:27.984534 [TOTEM] entering GATHER state from 3. Nov 5 23:37:33.803701 [TOTEM] The consensus timeout expired. Nov 5 23:37:33.803799 [TOTEM] entering GATHER state from 3. Nov 5 23:37:39.621992 [TOTEM] The consensus timeout expired. Nov 5 23:37:39.622067 [TOTEM] entering GATHER state from 3. Nov 5 23:37:45.441295 [TOTEM] The consensus timeout expired. Nov 5 23:37:45.441376 [TOTEM] entering GATHER state from 3. Any idea? Regards - bakko -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call using password
Hi, What is the easier way to make call using a password? I have A2billing but its authentication is too big, I would like four digits long. Something like that: In any extensons, the user dial the password and make call. Thanks in advanced! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Soundpoint IP 430 -- discontinued.
On Nov 5, 2010, at 10:45 PM, Michael Graves wrote: On Fri, 5 Nov 2010 19:02:43 -0400, Mike wrote: On Nov 5, 2010, at 5:35 PM, Ken D'Ambrosio wrote: Hey, all. I'm in the middle of a rollout, and just learned that the SoundPoint IP 430 -- my favorite mid-range phone -- has been discontinued. The heir apparent is the SoundPoint IP 450 -- for a low, low, low $130 more/handset. AND it doesn't look as nice. [snip] I have a 450 and it's got to be one of my favorite poly's. Great phone. But if you only want 2 line, go for the 3xx series. The problem with the Polycom 3xx series is that Polycom decided to be cheap on the phone UI, and it`s dramatically less-friendly than the other phones for anything else than simple calling/answering. Mike Yep. I love my IP650really like the IP550.have a bunch of users with IP450s...but simply won't subject our staff to the IP335. I regretted buying a bunch of refurb'd IP430s a while back. The larger display and added buttons are worth the price, IHMO. The memory of the bargain fades all too quickly. The sense that the phone is either cheap or annoying sticks with you forever. Life's just too short to use a cheap phone. Michael I really like my 450 (again). I had used the 650 and the 550... but the 450 covered everything I needed and has a special place as my first HD phone ;) Curious Michael... Why won't you subject people to the 335's? I love these phones for a call center deployment. The are a fantastic agent phone... let alone a great phone for kitchens, break rooms, lobby, etc. But I love them as an agent phone. Deploying them a few hundred at a time is great, and rarely hear a complaint. ---fred http://qxork.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Soundpoint IP 430 -- discontinued.
On Fri, 5 Nov 2010 23:09:19 -0400, Fred Posner wrote: Curious Michael... Why won't you subject people to the 335's? I love these phones for a call center deployment. The are a fantastic agent phone... let alone a great phone for kitchens, break rooms, lobby, etc. But I love them as an agent phone. Deploying them a few hundred at a time is great, and rarely hear a complaint. I'm not engaged in such activities. My users are more akin to executives. They want the bigger display and easier access to features. The IP335s sounds great, it's just the GUI that leaves them wanting more. As you suggest, in lesser roles...like the coffee room or lobby...they're perfectly suitable. Michael -- Michael Graves mgravesatmstvp.com http://www.mgraves.org o713-861-4005 c713-201-1262 sip:mgra...@mstvp.onsip.com skype mjgraves Twitter mjgraves -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Soundpoint IP 430 -- discontinued.
On Nov 5, 2010, at 11:13 PM, Michael Graves wrote: On Fri, 5 Nov 2010 23:09:19 -0400, Fred Posner wrote: Curious Michael... Why won't you subject people to the 335's? I love these phones for a call center deployment. The are a fantastic agent phone... let alone a great phone for kitchens, break rooms, lobby, etc. But I love them as an agent phone. Deploying them a few hundred at a time is great, and rarely hear a complaint. I'm not engaged in such activities. My users are more akin to executives. They want the bigger display and easier access to features. The IP335s sounds great, it's just the GUI that leaves them wanting more. As you suggest, in lesser roles...like the coffee room or lobby...they're perfectly suitable. Michael -- Michael Graves mgravesatmstvp.com http://www.mgraves.org o713-861-4005 c713-201-1262 sip:mgra...@mstvp.onsip.com skype mjgraves Twitter mjgraves Gotcha... thanks. The display sucks for execs. Sound is amazing though... and is just a little work horse. ---fred http://qxork.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Any way to stop Playtones(dial) when the user presses a key, emulating a CO's behavior?
The subject says it all. I'm betting there's a way to do it, but so far I haven't found the dialplan runestone via web searching. Thanks. b. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] gigasets A580IP Recall Button
Hi I am trying to get the recall button working for the gigasets What settings do i need to set in the advance settings? Zakir-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
