Re: [asterisk-users] pattern base call routing
Than you Hey I have 100 SIP phone with 2 E1 card and IVR feature but i am not happy with my configuration so have u any configuration for advance level Rgd satish patel Al lists [EMAIL PROTECTED] wrote: exten = _98XX,1,Dial(ZAP/(your preferred E1) exten = _,1,Dial(ZAP/(second E1) On 7/20/07, satish patel [EMAIL PROTECTED] wrote:Dear all I have 2 E1 card on my asterisk and i want to route call with fix pattern like if anyone dial mobile number like 9818875535 so it will use PRI 1 and rest of the world goes through PRI 2 means whn number prefix 98XX then call goes through specified E1 is it possible ??? satish patel - Get the free Yahoo! toolbar and rest assured with the added security of spyware protection. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - Ready for the edge of your seat? Check out tonight's top picks on Yahoo! TV. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Idefisk softphone - official 2.0 release - Zoiper
On 20:24, Thu 19 Jul 07, Zoiper wrote: Hello guys, The so expected 2.0 release of Idefisk 2.0 softphone is a fact. Idefisk and Zoiper became one - Zoiper 2.06. Any indication when the linux and osx builds will be 2.0 ? -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk IVR Performance
I am also planning for IVR so u have any kind of script plz suggest me David Ruggles [EMAIL PROTECTED] wrote: I have written a script that is executed using ExternalIVR(). I am running in to performance issues when I have four or more simultaneous calls running this script. I'm running on a P4 2.8 with 512M, all calls are GSM coming in over IAX from an asterisk box that acts as a switch and handles all PSTN interfaces. My question are these: Are there ways of optimizing ExternalIVRs? (maybe something like FastAGI) Right now I'm writing in a scripting language, would there be a performance gain from writing in a compiled language? I don't see any serious memory utilization and normally processor utilization is below 50% with spikes to 70% under load with four or five ExternalIVRs running. I will gladly provide any additional information that would aid in answering these questions. Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network Engineer Safe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - Luggage? GPS? Comic books? Check out fitting gifts for grads at Yahoo! Search.___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Best and easiest soft phone for my Dad..
Hi, Here is the situation.. My Dad is working on contract in overseas.. He has internet access in his hotel.. He wants to be able to talk to my Mum but the calls are expensive.. I have an asterisk box setup for my business and it has a public IP etc.. My Mum has access to a working phone extension on this box.. I got my Dad to install X-Lite but for some reason it won't register and trying to talk him through working out whats wrong is proving to be difficult.. Also I haven't used a softphone in years.. It could be the NAT in the hotel, it could be a firewall or any number of things that can cause these issues.. It could even be X-Lite or something running on his PC.. So I am looking for a softphone thats really simple to setup and as foolproof as possible.. If SIP is likely to be problematic to setup then I have no problem getting him to use IAX but will need suggestions of which IAX softphone to use and also how to configure it in the iax.conf (haven't done this before).. Any suggestions welcome and appreciated.. Thanks.. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Freeze
Noah Miller wrote: Hi Arun - Asterisk - 1.2.10 with zaptel 1.2.7, 10 queues with 7 sip agents this asterisk box is connected to another asterisk box using 5 IAX trunk to load balance no of calls on each IAX trunk (g729 over trunk). Suddenly my cli start flooding with message: Maximum trunk data space exceeded even I've only 3 calls on my asterisk system. asterisk restart option don't work, my agents are not able to hear any audio only solution is to restart the whole box. Please advise soon. You really need to update to a later version of asterisk (and zaptel). There have probably been somewhere close to a thousand bug fixes since 1.2.10. If you still have this issue with the latest version, please collect as much information as possible (exact cli messages, turn on logging, your config files, etc) and post that information to this list. I am very wary of upgrading -- some versions of Asterisk do not work well in my environment. Thursday night I upgraded one of my 6 production system to Asterisk 1.2.22 and Zaptel 1.2.19. So far I have not had any reported problems. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Upgrade Procedure
Noah Miller wrote: You have to first uninstall your Asterisk1.2 like this-- First you have to stop your asterisk...using-- 1. killall -9 asterisk or killall -9 safe_asterisk, whichever you are using. In my experience, you don't need to do this step. In fact, you can keep the old asterisk running, compile and install asterisk 1.4 on top of it. Then issue a restart when convenient command from the asterisk 1.2 prompt, and Asterisk 1.4 will come up after the restart. The problem with this is that the upgraded Zaptel will not be active. Compile and install Zaptel, LibPRI and Asterisk (in the order), then stop asterisk, unload the zaptel drivers, then load everything. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pattern base call routing
On 7/21/07, satish patel [EMAIL PROTECTED] wrote: Than you Hey I have 100 SIP phone with 2 E1 card and IVR feature but i am not happy with my configuration so have u any configuration for advance level Rgd what kind of advanced level Asterisk side or IP phone side ram ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best and easiest soft phone for my Dad..
Get portsip ( www.portsip.com ) its realtively easy to configure ( just push in user/password and server name at startup ) .. there might be NAT issue so make sure you have nat=yes in ur asterisk's sip.conf for the peer definition . If it still doesnt work then you need to find a iax phone like zoiper ( http://www.zoiper.com/ previously idefisk ). On 21/07/07, WipeOut [EMAIL PROTECTED] wrote: Hi, Here is the situation.. My Dad is working on contract in overseas.. He has internet access in his hotel.. He wants to be able to talk to my Mum but the calls are expensive.. I have an asterisk box setup for my business and it has a public IP etc.. My Mum has access to a working phone extension on this box.. I got my Dad to install X-Lite but for some reason it won't register and trying to talk him through working out whats wrong is proving to be difficult.. Also I haven't used a softphone in years.. It could be the NAT in the hotel, it could be a firewall or any number of things that can cause these issues.. It could even be X-Lite or something running on his PC.. So I am looking for a softphone thats really simple to setup and as foolproof as possible.. If SIP is likely to be problematic to setup then I have no problem getting him to use IAX but will need suggestions of which IAX softphone to use and also how to configure it in the iax.conf (haven't done this before).. Any suggestions welcome and appreciated.. Thanks.. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] New tutorial: compiling Asterisk 1.4 with zaptel and H323 support
Hello list, I have prepared a new tutorial for Astrecipes on how to compile the latest Asterisk 1.4 with H323 support, Google Talk and Zaptel support, starting from a stock TrixBox system. You can find it here: http://www.astrecipes.net/index.php?n=286 I hope somebody will find it useful :-) Thanks l. -- Home of QueueMetrics - http://queuemetrics.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best and easiest soft phone for my Dad..
SIP has a lot of issues with NAT, I can only get it to work correctly on my LAN with a softphone. IDEFISK, now known as Zoiper, is IAX based and I have tested it from all kinds of hotel rooms, even the free version supports 6 simultaneous calls : http://www.asteriskguru.com/idefisk/ good luck ! -baji. -- On 7/21/07, WipeOut wrote: Hi, Here is the situation.. My Dad is working on contract in overseas.. He has internet access in his hotel.. He wants to be able to talk to my Mum but the calls are expensive.. I have an asterisk box setup for my business and it has a public IP etc.. My Mum has access to a working phone extension on this box.. I got my Dad to install X-Lite but for some reason it won't register and trying to talk him through working out whats wrong is proving to be difficult.. Also I haven't used a softphone in years.. It could be the NAT in the hotel, it could be a firewall or any number of things that can cause these issues.. It could even be X-Lite or something running on his PC.. So I am looking for a softphone thats really simple to setup and as foolproof as possible.. If SIP is likely to be problematic to setup then I have no problem getting him to use IAX but will need suggestions of which IAX softphone to use and also how to configure it in the iax.conf (haven't done this before).. Any suggestions welcome and appreciated.. Thanks.. ___ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pattern base call routing
i want asterisk extention.conf IVR plan so i want idea of IVR means how other users use IVR in dialplan on asterisk ram [EMAIL PROTECTED] wrote: On 7/21/07, satish patel [EMAIL PROTECTED] wrote: Than you Hey I have 100 SIP phone with 2 E1 card and IVR feature but i am not happy with my configuration so have u any configuration for advance level Rgd what kind of advanced level Asterisk side or IP phone side ram ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - Choose the right car based on your needs. Check out Yahoo! Autos new Car Finder tool.___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pattern base call routing
On 7/21/07, satish patel [EMAIL PROTECTED] wrote: i want asterisk extention.conf IVR plan so i want idea of IVR means how other users use IVR in dialplan on asterisk Hi Hint is Look at Agi Scripts you can write small agi scripts to do your job ram ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call Initiation with Asterisk
Hi List; I need help for the following senario: Initiating a call from Asterisk to an extension and after it answers, IVR prompts will be played Mohammad Mirzaee +989121750530___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk IVR Performance
satish patel wrote: I am also planning for IVR so u have any kind of script plz suggest me */David Ruggles [EMAIL PROTECTED]/* wrote: I have written a script that is executed using ExternalIVR(). I am running in to performance issues when I have four or more simultaneous calls running this script. I'm running on a P4 2.8 with 512M, all calls are GSM coming in over IAX from an asterisk box that acts as a switch and handles all PSTN interfaces. My question are these: Are there ways of optimizing ExternalIVRs? (maybe something like FastAGI) Right now I'm writing in a scripting language, would there be a performance gain from writing in a compiled language? I don't see any serious memory utilization and normally processor utilization is below 50% with spikes to 70% under load with four or five ExternalIVRs running. I will gladly provide any additional information that would aid in answering these questions. Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network Engineer Safe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] We do IVR swith standard dial plan syntax or AEL and do agi calls for database lookups/transactions. This works well for us. Mike Clark ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729 copy protection
Does anyone out there know what version/release of solaris the g729 (v32) codec is built on? Is it built on Solaris 10 GA, Solaris 10 U1, Solaris 10 U2, Solaris 10 U3, OpenSolaris (Nevada), which build? I'm just trying to find out if my problem with the codec may be due to a release difference, possibly a version of a library that the codec requires is not there? I will give it a test with the same version that the codec is built on just to see if it works. Thanks Bruce ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] * core file not recognized
Jay Wilton wrote: gdb /usr/sbin/asterisk -c /tmp/core.4545 GNU gdb 6.3-debian ...CUT This GDB was configured as i386-linux...Using host libthread_db library /lib/libthread_db.so.1. /tmp/core.4545 is not a core dump: File format not recognized Does the user running gdb have proper permissions to the core file? -- Russell Bryant Software Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] priorityjumping not working, Dial goes to n+1 not n+101
I think n+101 worked in Asterisk 1.2.x but it doesn't work in Asterisk 1.4.x
use ${DIALSTATUS} if you want Asterisk to act depending the result of Dial()
I read that the variable has been disabled in SVN to be replaced by the
DEVSTATE function, I need to confirm that.
Well... an example:
exten = 1337,1,Dial(SIP/zytek,5,Ttj)
exten = 1337,n,Goto(${DIALSTATUS},1)
exten = CONGESTION,1,Congestion
exten = CANCEL,1,Hangup
exten = BUSY,1,Busy
exten = CHANUNAVAIL,1,NoOp(I can't find it)
exten = CHANUNAVAIL,n,Busy
Although it would look a lot nicer if you create a macro that acts upon
the result of Dial.
Perssy Llamosas
Original Message
Subject: [asterisk-users] priorityjumping not working,Dial goes to
n+1 not n+101
From: Jakub Głazik [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: 20/07/2007 04:45 a.m.
Priorityjumping is totally ignored by my asterisk (tested 1.4.4 and
1.4.7.1 on FreeBSD 6.2)
[general]
priorityjumping=yes
With n+101:
exten = 1337,1,Dial(SIP/zytek,5,Ttj)
exten = 1337,102,Dial(SIP/zytek,${RINGTIME},${OPTIONS})
exten = 1337,n,Hangup
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/113-087a3000,
SIP/zytek|5|Ttj) in new stack
-- Called zytek
-- SIP/zytek-087b9000 is ringing
-- Nobody picked up in 5000 ms
== Auto fallthrough, channel 'SIP/113-087a3000' status is 'NOANSWER'
With n+1:
exten = 1337,1,Dial(SIP/zytek,5,Ttj)
exten = 1337,2,Dial(SIP/zytek,${RINGTIME},${OPTIONS})
exten = 1337,n,Hangup
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/113-087c8000,
SIP/zytek|5|Ttj) in new stack
-- Called zytek
-- SIP/zytek-087da000 is ringing
-- Nobody picked up in 5000 ms
-- Executing [EMAIL PROTECTED]:2] Dial(SIP/113-087c8000,
SIP/zytek|720|Ttm) in new stack
-- Called zytek
-- Started music on hold, class 'default', on channel
'SIP/113-087c8000'
-- SIP/zytek-087b6000 is ringing
Why?
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Re: [asterisk-users] Call Initiation with Asterisk
Hi Mohammad, The best way is tu use .call files, check here : http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out Cheers, Yves. On Sat, 2007-07-21 at 18:08 +0430, mohammad mirzaee wrote: Hi List; I need help for the following senario: Initiating a call from Asterisk to an extension and after it answers, IVR prompts will be played Mohammad Mirzaee +989121750530 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Yves Räber [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best and easiest soft phone for my Dad..
One I really like is the idefsk version that was a zip file, you could extract the file configure the softphone, zip it up and email it out. Saved the headache of walking someone through the process and even ran of thumb drives. On 7/21/07, WipeOut [EMAIL PROTECTED] wrote: Hi, Here is the situation.. My Dad is working on contract in overseas.. He has internet access in his hotel.. He wants to be able to talk to my Mum but the calls are expensive.. I have an asterisk box setup for my business and it has a public IP etc.. My Mum has access to a working phone extension on this box.. I got my Dad to install X-Lite but for some reason it won't register and trying to talk him through working out whats wrong is proving to be difficult.. Also I haven't used a softphone in years.. It could be the NAT in the hotel, it could be a firewall or any number of things that can cause these issues.. It could even be X-Lite or something running on his PC.. So I am looking for a softphone thats really simple to setup and as foolproof as possible.. If SIP is likely to be problematic to setup then I have no problem getting him to use IAX but will need suggestions of which IAX softphone to use and also how to configure it in the iax.conf (haven't done this before).. Any suggestions welcome and appreciated.. Thanks.. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 700Mhz Spectrum
Great article here about the concept of open spectrum I posted a few weeks ago. Could be very very interesting. http://machinist.salon.com/blog/2007/07/20/google_fcc/ Maybe an Asterisk/OpenMoko tie-in? Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best and easiest soft phone for my Dad..
So I am looking for a softphone thats really simple to setup and as foolproof as possible.. If SIP is likely to be problematic to setup then I have no problem getting him to use IAX but will need suggestions of which IAX softphone to use and also how to configure it in the iax.conf (haven't done this before).. You don't specify if he's on Windows, Linux or OSX. But if he is on Windows, you can try my softphone : http://www.marccharbonneau.com/asterisk/mediaxphone.php There is a version using INI file, so you can put all the settings then zip it and send it to him already configured. hth ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best and easiest soft phone for my Dad..
On 7/21/07, Time Bandit [EMAIL PROTECTED] wrote: You don't specify if he's on Windows, Linux or OSX. But if he is on Windows, you can try my softphone : http://www.marccharbonneau.com/asterisk/mediaxphone.php There is a version using INI file, so you can put all the settings then zip it and send it to him already configured. looks very interesting, I will try it out when I get a chance. -baji. -- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Has anybody used fanless computers of logic supply with asterisk?
I want my freedom to setup and configure PBX hardware and software how i want, not how Digium or anybody else wants, so not interested in Asterisk Appliances. So anybody with experience with Supply Logics computers. Or any other recommendations for asterisk pbx casings? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Has anybody used fanless computers of logic supply with asterisk?
On 7/21/07, Zeeshan Zakaria wrote: I want my freedom to setup and configure PBX hardware and software how i want, not how Digium or anybody else wants, so not interested in Asterisk Appliances. So anybody with experience with Supply Logics computers. Or any other recommendations for asterisk pbx casings? ---( info from a friend's email ) I'm still running Asterisk on a Soekris Net4801. http://www.soekris.com/net4801.htm It's fine for 3-4 calls using g726. Runs off of Compact Flash and I use the Astlinux distribution on it. Very stable, very low power. Heck, I think I'm running a pre-1.2 release on it. :) -- -- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Has anybody used fanless computers of logic supply with asterisk?
Zeeshan Zakaria wrote: I want my freedom to setup and configure PBX hardware and software how i want, not how Digium or anybody else wants, so not interested in Asterisk Appliances. So anybody with experience with Supply Logics computers. Or any other recommendations for asterisk pbx casings? The fanless mini-itx boards should be just fine. There are too many factors to give you a definite answer, but I currently use one with a TDM400 card. A majority of the calls on the board are sip with no transcoding so there is a very small load on the system (hardly noticeable). If you are doing a ton of transcoding or recording calls, your results may be different. Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] * core file not recognized
Jay Wilton wrote: gdb /usr/sbin/asterisk -c /tmp/core.4545 /tmp/core.4545 is not a core dump: File format not recognized Does the user running gdb have proper permissions to the core file? I was running gdb as root. Asterisk is running as the asterisk user. I chmod 777'd the core file. The box has not been stable since some poor mp3's were uploaded and the machine started seg faulting. We removed the bad mp3's but it still dumps core a few times a week. A recompile didn't help. JJ Take the Internet to Go: Yahoo!Go puts the Internet in your pocket: mail, news, photos more. http://mobile.yahoo.com/go?refer=1GNXIC ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk and COS bits
Is there any way to change COS bits for packets? There is a tos option on sip.conf, does asterisk change COS bits considering tos value? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4, Unicall and Nextel...
On Thu, 19 Jul 2007 12:14:53 -0500, Alvaro Parres wrote Yes Moises, i was looking for it. The main problem is only on the files for version 1.4... it give that error when no CallerID is recive or a private caller id is recive. The change i made is to add to Mexico variant on mfcr2.c this line mfcr2-group_i_end_of_ANI_restricted = R2_SIGI_12; This works for nextel or phones that send private caller id.. But doesn't work when no CallerID is recive. I have al ready check diff files from 1.2 files and 1.4 files and i didn't find any big difference between both version. I patched mfcr2.c but I still cannot receive calls from Nextel phones unless I put ANI to 0 on unicall.conf -- Carlos Chavez Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best and easiest soft phone for my Dad..
Check this out: http://www.kapanga.net/IP/home.cfm Very easy to create a self-installing pre-configured soft phone. On 7/21/07, WipeOut [EMAIL PROTECTED] wrote: Hi, Here is the situation.. My Dad is working on contract in overseas.. He has internet access in his hotel.. He wants to be able to talk to my Mum but the calls are expensive.. I have an asterisk box setup for my business and it has a public IP etc.. My Mum has access to a working phone extension on this box.. I got my Dad to install X-Lite but for some reason it won't register and trying to talk him through working out whats wrong is proving to be difficult.. Also I haven't used a softphone in years.. It could be the NAT in the hotel, it could be a firewall or any number of things that can cause these issues.. It could even be X-Lite or something running on his PC.. So I am looking for a softphone thats really simple to setup and as foolproof as possible.. If SIP is likely to be problematic to setup then I have no problem getting him to use IAX but will need suggestions of which IAX softphone to use and also how to configure it in the iax.conf (haven't done this before).. Any suggestions welcome and appreciated.. Thanks.. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AstLinux 0.4.7
Hello Everyone, I have released AstLinux 0.4.7. This release includes Asterisk 1.2.22. More here: http://www.astlinux.org/node/26 -- Kristian Kielhofner ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Has anybody used fanless computers of logic supply with asterisk?
Darrick, can you tell which mini-itx board you have and what processor it has on it? I don't them with Pentium processors, instead they have some VIA C3 and C7 processors, which are completely new to me and I have no idea how will they perform with Asterisk. On 7/21/07, Darrick Hartman (lists) [EMAIL PROTECTED] wrote: Zeeshan Zakaria wrote: I want my freedom to setup and configure PBX hardware and software how i want, not how Digium or anybody else wants, so not interested in Asterisk Appliances. So anybody with experience with Supply Logics computers. Or any other recommendations for asterisk pbx casings? The fanless mini-itx boards should be just fine. There are too many factors to give you a definite answer, but I currently use one with a TDM400 card. A majority of the calls on the board are sip with no transcoding so there is a very small load on the system (hardly noticeable). If you are doing a ton of transcoding or recording calls, your results may be different. Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Zeeshan A Zakaria ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to use call transfer
the best way attended transfer. See my feature.conf:
example:
[general]
; Call parking configuration
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, need to
INCLUDE this in extensions.conf
parkingtime = 45 ; Number of seconds a call can be parked for (default
is 45)
pickupexten = *8
; Max time (ms) between digits for feature activation. Default is 500
featuredigittimeout = 1500
[featuremap]
; Blind transfer, default is pound sign (#)
blindxfer = #
; Attended transfer
atxfer = *7
--END--
Bruno De Luca
Keshav K. wrote:
There is one thing,
just forget that your phone is a snom phone or whatever...
simply to make a blind call transfer press #8, according to the my
feature.conf, default it is #, or you can assign it any, then after
pressing that you will listen a IVR transfer and dial the desired
number followed by the # sign, then you will connect to the new
number, now hangup your phone, and the other two will be connected.
But make sure, that in your extensions.conf you should have the entry
for t, as I have showed in the entry..
Regards,
Keshav
*/satish patel [EMAIL PROTECTED]/* wrote:
but what buttons i will use for call transfer ??? I have SNOM SI
120 phon with transfer button so how to do it ?
*/Keshav K. [EMAIL PROTECTED]/* wrote:
Hi,
To use call tranfer you have to make entry in extension.conf...
exten = _7.,1,Dial(SIP/${EXTEN},20,Ttr)
then in feature.conf
[featuremap]
blindxfer = #8 ; Blind transfer (default is #)
;disconnect = *0 ; Disconnect (default is *)
;automon = *1 ; One Touch Record a.k.a.
Touch Monitor
atxfer = #9; Attended transfer
parkcall = #72; Park call (one step parking)
I'm using this...end its working wonderfully.
--Keshav
*/satish patel [EMAIL PROTECTED]/* wrote:
Dear all
I have beginer in Voip and i have
configured Asterisk server with 100 IP SIP phone ( SNOM )
everything is fine but problem is how to transfer call
from one user to other means i call to some one and then
someone want to transfer call to another person how it is
possible i have also try with feartue.conf but it is now
working i have also read document on voip-info website but
now clear yet can anyone explain me how to asterisk
transfer call from one user to other and what
extention.conf look like is there any change in sip.conf
or extention.conf
Rgd
Satish patel
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Re: [asterisk-users] Best and easiest soft phone for my Dad..
On 7/21/07, Andrew Joakimsen wrote: Check this out: http://www.kapanga.net/IP/home.cfm Very easy to create a self-installing pre-configured soft phone. I don't see IAX listed as one of the features, do you know if it is supported ? http://www.kapanga.net/IP/specs.cfm -- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4, Unicall and Nextel...
Yes Moises, i was looking for it. The main problem is only on the files for version 1.4... it give that error when no CallerID is recive or a private caller id is recive. The change i made is to add to Mexico variant on mfcr2.c this line mfcr2-group_i_end_of_ANI_restricted = R2_SIGI_12; This works for nextel or phones that send private caller id.. But doesn't work when no CallerID is recive. I have al ready check diff files from 1.2 files and 1.4 files and i didn't find any big difference between both version. On 7/18/07, Moises Silva [EMAIL PROTECTED] wrote: Alvaro, can you post the patch in a public place and post the URL here? It might be a good idea to contact steve underwood to see what he has to say about such a patch. Regards, On 7/18/07, Alvaro Parres [EMAIL PROTECTED] wrote: Carlos: Only for check do this change: protocolvariant=mx,10,4 for protocolvariant=mx,0,4 If it's works, contact me and i will send you a patch for libmfcr.c Thanks. Carlos: Has el cambio que te pido arriva, para revisar si es lo del caller ID. Casi estoy seguro nosotros en labortario y en el extrangero tuvimos esos problemas, si es asi te paso un parche solo que lo encuentre para la libmfcr.c Donde le digas como manejar la señal de private al recibir el ANI. Saludos. On 7/18/07, Carlos Chavez [EMAIL PROTECTED] wrote: On Wed, 2007-07-18 at 08:10 -0500, Alvaro Parres wrote: Could you send please your unicall.conf file Thanks. It appers to be a problem with de ANI digits you want to recive. Also Nextel never sends CallerID. When someone calls me from a Nextel phone to my cell or to my Asterisk server I always get Private Call. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alvaro I. Parres Peredo Director de IT Grupo Xmarts SA de CV Tel: +52 (33) 35 63 6261 Ext. 112 01 800 087 2260 Cel: +52 (33) 33 68 1087 [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alvaro I. Parres Peredo Director de IT Grupo Xmarts SA de CV Tel: +52 (33) 35 63 6261 Ext. 112 01 800 087 2260 Cel: +52 (33) 33 68 1087 [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best and easiest soft phone for my Dad..
It is possible that the hotel is only allowing certain incoming/outgoing ports as well (i.e. just allowing DNS and HTTP traffic). A VPN *might* help with that. On 7/21/07, WipeOut [EMAIL PROTECTED] wrote: Hi, Here is the situation.. My Dad is working on contract in overseas.. He has internet access in his hotel.. He wants to be able to talk to my Mum but the calls are expensive.. I have an asterisk box setup for my business and it has a public IP etc.. My Mum has access to a working phone extension on this box.. I got my Dad to install X-Lite but for some reason it won't register and trying to talk him through working out whats wrong is proving to be difficult.. Also I haven't used a softphone in years.. It could be the NAT in the hotel, it could be a firewall or any number of things that can cause these issues.. It could even be X-Lite or something running on his PC.. So I am looking for a softphone thats really simple to setup and as foolproof as possible.. If SIP is likely to be problematic to setup then I have no problem getting him to use IAX but will need suggestions of which IAX softphone to use and also how to configure it in the iax.conf (haven't done this before).. Any suggestions welcome and appreciated.. Thanks.. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mitchel Constantin Snap - A desktop user interface for Asterisk www.snapanumber.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] * core file not recognized
On Sat, Jul 21, 2007 at 04:35:21PM -0700, Jay Wilton wrote: Jay Wilton wrote: gdb /usr/sbin/asterisk -c /tmp/core.4545 /tmp/core.4545 is not a core dump: File format not recognized Does the user running gdb have proper permissions to the core file? I was running gdb as root. Asterisk is running as the asterisk user. I chmod 777'd the core file. The box has not been stable since some poor mp3's were uploaded and the machine started seg faulting. We removed the bad mp3's but it still dumps core a few times a week. A recompile didn't help. What do you see in the logs before a crash? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
