[asterisk-users] Force asterisk to re-resolve dns names?
Is there really no way to have asterisk re-resolve domain names from iax or sip providers if this failed or timed out the first time? When asterisk boots on every box i have asterisk is t impatient trrying to resolve the domain names for a first time. This results in asterisk thinking the provider is unreachable and only trying again in one week or so. This results (depending on the dial plan) on either not being able to make calls or to see all calls flow out via (extremely expensive) PSTN. This 'feature' of asterisk really pisses me off, why can't it just re-resolve the few host names again within a reasonable amount of time??? ___ --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] Music on hold problem
Hi, Thanks for the answer. Actually I noticed that several things dont work properly and I think the ooh323 channel driver is the reason. For instance, when I configure my extensions.conf to answer the phone and playback a sound (for instance nobodyavailable message after 10secs), only the last part of the sound is played (i hear sometimes only the last word, sometimes i hear nothing!). In the CLI I see that asterisk is playing the sound file, but in the h323 phone i do not hear the complete message. I solved this issue putting a wait(3) just before the Playback function. It seems that somehow ohh323 or asterisk needs some time to setup the channel or something, and the sound has been already played in the meantime. I also have the same sound problem using the Meetme function to join a conference bridge. I should hear After the tone say your name and then press the pound key. I normally do not hear the first 4 or 5 words. I tried to put some wait functions, but here it does not work. In the CLI I see again those strange messages (Don't know how to indicate condition -xxx on ooh323c_1...), and asterisk says that he is playing several sound files, but in the phones The same problem using Voicemail. I should hear a message asking for my mailbox number, but normally I do not hear anything. Do you think the patch I will fix the problems? I will try later, thanks 2007/7/19, Russell Bryant [EMAIL PROTECTED]: yonoko molomo wrote: [Jul 17 11:19:22] WARNING[23645]: src/chan_h323.c:1044 ooh323_indicate: Don't know how to indicate condition -1 on ooh323c_1 [Jul 17 11:19:25] NOTICE[23645]: rtp.c:783 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 10.4.0.116 [Jul 17 11:19:25] WARNING[23645]: src/chan_h323.c:1044 ooh323_indicate: Don't know how to indicate condition 16 on ooh323c_1 This would be a bug in the ooh323 channel driver. Feel free to report it to bugs.digium.com. I think it's an easy fix. ... In fact, here is a patch that should fix it. Feel free to go ahead and give it a try and let me know if it fixes the problem for you. However, please still report it to bugs.digium.com, or I will forget to merge the change. Index: asterisk-ooh323c/src/chan_h323.c === --- asterisk-ooh323c/src/chan_h323.c(revision 413) +++ asterisk-ooh323c/src/chan_h323.c(working copy) @@ -1036,9 +1036,16 @@ ast_set_flag(p, H323_ALREADYGONE); } break; + case AST_CONTROL_HOLD: + ast_moh_start(ast, data, NULL); + break; + case AST_CONTROL_UNHOLD: +ast_moh_stop(ast); + break; case AST_CONTROL_PROCEEDING: case AST_CONTROL_RINGING: case AST_CONTROL_PROGRESS: + case -1; break; default: ast_log(LOG_WARNING,Don't know how to indicate condition %d on %s\n, -- 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 ___ --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] Gtalk/Jabber connect issues in 1.4.8
Hi Bruce, [EMAIL PROTECTED] Google's server is expecting you to provide a valid gmail address here, suffixed with @gmail.com Cheers, Philippe ___ --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] Gtalk/Jabber connect issues in 1.4.8
Hi, Are you connecting to talk.google.com as a C2S or a S2S? BR, Dems Hi Bruce, [EMAIL PROTECTED] Google's server is expecting you to provide a valid gmail address here, suffixed with @gmail.com Cheers, Philippe ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Gtalk/Jabber connect issues in 1.4.8
Hi, That should be: [EMAIL PROTECTED] Got it? BR, Dems I've included my jabber.conf below. I'm betting the following errors: [Jul 18 21:05:22] ERROR[28166]: res_jabber.c:609 aji_act_hook: JABBER: Node Error [Jul 18 21:05:22] WARNING[28166]: res_jabber.c:1537 aji_recv_loop: JABBER: Got hook event. jabber test [Jul 18 21:04:16] WARNING[32691]: res_jabber.c:1421 ast_aji_send: JABBER: Not connected can't send User: [EMAIL PROTECTED] Anyone have a hint?? Thanks [general] debug=no;;Turn on debugging by default. autoprune=yes ;;Auto remove users from buddy list. autoregister=yes;;Auto register users from buddy list. [asterisk] ;;label type=client ;;Client or Component connection ;;Route to server for example, serverhost=talk.google.com ;; talk.google.com ;[EMAIL PROTECTED]/asterisk ;;Username with optional roster. [EMAIL PROTECTED] secret= ;;Password port=5222 ;;Port to use defaults to 5222 usetls=yes ;;Use tls or not usesasl=yes ;;Use sasl or not [EMAIL PROTECTED];;Manual addition of buddy statusmessage=I am available ;;Have custom status message for ;;Asterisk. timeout=100 ;;Timeout on the message stack. ___ --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] Ncurses dependencies
Hello, I'm deploying asterisk on a comtrend mips adsl router, I'm aware of the dependence of libncurses, so I compiled ncurses 5.6 for that platform, As you must Know this devices are not resource wide and flash memory especially, after ncurses compilation I have a /usr/share/terminfo with 1,6 MB space and some other libraries, libmenu libform llibpanel etc, I would like to know if some developers or other person with asterisk knowledge point me to the exact resources needed fom ncurses in order to delete everything else given that asterisk is the only soft is going to use ncurses here... thanks -- Francisco J. Pérez Botella ___ --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 with 2 Public IP-Is it possible?
Good morning, we actually have 2 Asterisks 1.2 running on one server each of them in a XEN Dom and connected together with a IAX trunk. This setup allow us to use our both public IP (different ISP's) and to have failover solution in case of a problem on one of the ISP's line. Is it a way in 1.4 to do the same but with only one Asterisk instance? First problem we see is that externalip in sip.conf can be fixed for only one IP. Second is, if we could put 2 external IP's, if one packet is going out with, let's say PublicIP#1, and at this time the link from this ISP is down, packets will -helped by external routing- go out through the second line but never come back. Does someone knows an Asterisk solution for this? -- Daniel ___ --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 libraries dependecies
Hello, I'm deploying asterisk on a comtrend mips adsl router, I'm aware of the dependence of libncurses, so I compiled ncurses 5.6 for that platform, As you must Know this devices are not resource wide and flash memory especially, after ncurses compilation I have a /usr/share/terminfo with 1,6 MB space and some other libraries, libmenu libform llibpanel etc, I would like to know if some developers or other person with asterisk knowledge point me to the exact resources needed fom ncurses in order to delete everything else given that asterisk is the only soft is going to use ncurses here... thanks -- Francisco J. Pérez Botella ___ --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 libraries dependecies
On Thu, Jul 19, 2007 at 12:58:47PM +0200, Francisco Pérez Botella wrote: Hello, I'm deploying asterisk on a comtrend mips adsl router, I'm aware of the dependence of libncurses, so I compiled ncurses 5.6 for that platform, As you must Know this devices are not resource wide and flash memory especially, after ncurses compilation I have a /usr/share/terminfo with 1,6 MB space /usr/share/terminfo has the terminal types database. You really don't need mot if not all of them. Maybe leave just the entries for 'linux' and two or three other terminals you actually use. -- 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
Re: [asterisk-users] New book Asterisk Cookbook any good?
On Thu, 2007-07-19 at 00:30 -0500, Larry Alkoff wrote: I have received mail from Amazon touting this book that will soon be available. Know anything about the book or it's authors? It's a little pricey. I have it on good authority that the book has been delayed, since two of the authors (and myself) have been busy working on the second edition of Asterisk: The Future of Telephony. Work is progressing on the Cookbook, it's just not coming along as quickly as they'd like. -- Jared Smith Community Relations Manager 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] Force asterisk to re-resolve dns names?
On Thu, 2007-07-19 at 09:44 +0200, Remco Barendse wrote: Is there really no way to have asterisk re-resolve domain names from iax or sip providers if this failed or timed out the first time? Have you tried the Asterisk DNS manager, as configured in dnsmgr.conf? It might be just the thing you're looking for. -- Jared Smith Community Relations Manager 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] Not hearing the caller after 2 x Flash
Dnia 2007-07-17, o godz. 14:06:42 Jakub Głazik [EMAIL PROTECTED] napisał(a): When client calls, I press Flash, client hears music on hold (only on voip line as said in previous post), when I get back and press Flash again to get back to my client I cannon hear him, but he hears me without problems. No ideas? I have switched voip gateways (grandstream), phones but the problem still exists. I phone from LAN to LAN, when I press Flash and then press againg - I cannon hear the caller, but he hears me. Please post suggestions what is wrong with my setup. I will buy SIP hardphones and test this issue again. -- .: Jakub Głazik, .: email jabber: zytekatnuxi.pl ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem building Asterisk 1.2.22
Hi John, On 7/18/07, John covici [EMAIL PROTECTED] wrote: I wonder what version of Zaptel you are using -- sounds like you have not installed a new version or you are using an older one. Actually I just made and installed zaptel 1.2.19 then made asterisk 1.2.22 and I am getting this same error ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem building Asterisk 1.2.22
On 7/19/07, randulo [EMAIL PROTECTED] wrote: Actually I just made and installed zaptel 1.2.19 then made asterisk 1.2.22 and I am getting this same error Fixed! I just noticed that I asked about this same problem in March and got a workaround (edit makefile) from Tzafrir. Could someone explain why this codec_zap line is in Makefile has to be manually commented out? THere must be a reason why this happens on my box that I can fix? ___ --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] Pass Dialed number to a script
I'm in the process of writing a simple autodialer to dial a list of numbers
and play a message. One of the options I want to give them is a way to
dial X to have a customer service representative call you
Looking for a simple way to pass the number that I dialed to a script in
extensions.conf... something like this:
[serviceinterruption]
exten = s,1,Set(TIMEOUT(response)=10)
exten = s,2,Answer
exten = s,3,Playback(outboundmsgs/serviceinterrupt)
exten = s,4,Playback(outboundmsgs/choice)
exten = s,5,wait(3)
exten = 1,1,Goto(s,3) ; replay message
exten = 2,1,Goto(msgack,s,1); acknowledge message
exten = 3,1,Goto(callback,s,1)
exten = t,1,Playback(vm-goodbye)
exten = t,2,Hangup
[callback]
exten = s,1,Playback(outboundmsgs/customerrepwillcall)
-- exten = s,n,system(${SCRIPTS_DIR}/rep_callback.sh ${} )
exten = s,n,Playback(vm-goodbye)
exten = s,n,Hangup
how do I pass the number that was dialed (from the creation of a .call file)
to the rep_callback.sh script ?
thanks
Shawn
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[asterisk-users] Upgrade Procedure
Hello All, I would like to upgrade my recently installed Asterisk 1.2.21.1 to Asterisk 1.4.8? My OS is CentOS 4.5 with Linux 2.6.9-55.0.2.plus.c4smp #1 SMP Fri Jul 6 05:25:07 EDT 2007 i686 i686 i386 GNU/Linux Is there any detail step by step procedure to uninstall the current version and install Asterisk 1.4.8, Zaptel 1.4.4, Libpri 1.4.1, Addons 1.4.2? Cheers, Nitesh ___ --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] Parsing IAXPeers from Asterisk Manager (PHP API)
Hi all, Trying to do what should be a basic info retrieval from my asterisk network -- get the list of IAXPeers and break down by IP, name, and Status etc. Asterisk 1.0 Manager unfortunately. Using PHP. Easy to get the response -- but parsing it is impossible! That table format throws me for a loop ... Any idea of an easy way to seperate out the columns of that horrid 'table' reponse layout? I really want to get the value of the Status ping time ... hope to produce a neat little google maps mashup out of it. Parsing ideas appreciated ... go lightly on the regex if possible! ;-) -- Chris ___ --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] G729 copy protection
Hi All, I have been trying to get the Solaris version of the G729 codec to work with asterisk 1.2.17 and 1.2.22. However, I come up against the very same error every time I try to install it. Has anyone out there seen this error, taken from the asterisk console straight from startup: [codec_g729a.so] = (Annex A/B (floating point) G.729 Codec (optimized for i386)) Jul 19 14:11:23 NOTICE[28243]: codec_g729.c:465 load_module: G.729 transcoding module version 32, Copyright (C) 1999-2007 Digium, Inc. Jul 19 14:11:23 NOTICE[28243]: codec_g729.c:469 load_module: This module is supplied under a commercial license granted by Digium, Inc. Jul 19 14:11:23 NOTICE[28243]: codec_g729.c:470 load_module: Please see the full license text supplied by the accompanying Jul 19 14:11:23 NOTICE[28243]: codec_g729.c:471 load_module: register utility, or ask for a copy from Digium. Jul 19 14:11:23 NOTICE[28243]: codec_g729.c:472 load_module: This product includes software developed by the OpenSSL Project Jul 19 14:11:23 NOTICE[28243]: codec_g729.c:473 load_module: for use in the OpenSSL Toolkit. (http://www.openssl.org/) Jul 19 14:11:23 NOTICE[28243]: codec_g729.c:474 load_module: Copyright (C) 1998-2006 The OpenSSL Project Jul 19 14:11:23 WARNING[28243]: codec_g729.c:481 load_module: Failed to initialize G.729 copy protection! I have tried it with all the available v32 architectures and every one of them comes back with the very same error. I have done a search to see if anyone else came accross this error, there was one reference to the FreeBSD codec doing this, but apparently a new version of the codec came out that fixed it, the link for the FreeBSD error reference is here: http://lists.digium.com/pipermail/asterisk-bsd/2005-March/000699.html I'm at wits end at the moment, so, if anyone has any suggestions whatsoever, please feel free to put them forth, I'm willing to try anything at the moment. Oh, and the hardware we're running it with is: Solaris 10 Update 3 The CPU's are Opterons, but I have forced Solaris to boot in 32bit mode as the target server for the asterisk package I'm making is 32bit Solaris. Hopefully the i386 version of the codec should work on Opteron processors? 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
[asterisk-users] Does anyone have the file: pickup-mgernoth-2006-07-28.patch.txt
Hi, Does anyone have this file? It seems to not be available on bugs.digium.com any more. Cheers Alex -- This message is intended only for the addressee and may contain confidential information. Unless you are that person, you may not disclose its contents or use it in any way and are requested to delete the message along with any attachments and notify us immediately. Transact is operated by Integrated Financial Arrangements plc Domain House, 5-7 Singer Street, London EC2A 4BQ Tel: (020) 7608 4900 Fax: (020) 7608 1200 (Registered office: as above; Registered in England and Wales under number: 3727592) Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sip Providers
[EMAIL PROTECTED] wrote: Hi John, Try ... carriers.icall.com - No minimum, unlimited concurrent calls, great price, some areas US 0,009. Only USA voipjet.com teliax.com - Not so cheap, and they do one-minute rounding ... not good at all. But they hold a very good quality Teliax does 60/6 rounding. You only pay for the first full minute, then fractionally there after. I've been using them for over 2 years with only a few issues that were quickly resolved. -- 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] Sip Providers
Hi John, Try ... carriers.icall.com - No minimum, unlimited concurrent calls, great price, some areas US 0,009. Only USA voipjet.com teliax.com - Not so cheap, and they do one-minute rounding ... not good at all. But they hold a very good quality Hope it helps -- Original message -- From: John Meksavan [EMAIL PROTECTED] Asterisk Users, I have Asterisk PBX System running at my work. The system is working great. Currently, I have Broadvoice as my sip provider and I am not completely satisfy with their service. Broadvoice only allows 2 simultaneous calls, which hinders my company's communications ability. I am looking for a sip provider that would work with Asterisk and allow at least 6 simultaneous calls, locally and internationally. Of course the voice quality, pricing, number portability are the main determining factors. I will have a T1 connection at the office, so bandwidth would not be an issue. Any thoughts on this matter would be greatly appreciated. Thanks. Best Regards, John _ Need a brain boost? Recharge with a stimulating game. Play now! http://club.live.com/home.aspx?icid=club_hotmailtextlink1 ---BeginMessage--- ___ --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---End Message--- ___ --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] Pass Dialed number to a script
Hi,
add new line in
[serviceinterruption]
exten = s,1,Set(TIMEOUT(response)=10)
exten = s,2,Set(dialedno=${EXTEN}) //Add This Line
and change
[callback]
exten = s,1,Playback(outboundmsgs/customerrepwillcall)
exten = s,n,System(${SCRIPTS_DIR}/rep_callback.sh ${dialedno}) //This Line
Changed
I think that you know how to get arg from shell script.
cheers
Nasir Iqbal
ICT Innovations
On Thu, 2007-07-19 at 08:43 -0400, [EMAIL PROTECTED] wrote:
I'm in the process of writing a simple autodialer to dial a list of numbers
and play a message. One of the options I want to give them is a way to
dial X to have a customer service representative call you
Looking for a simple way to pass the number that I dialed to a script in
extensions.conf... something like this:
[serviceinterruption]
exten = s,1,Set(TIMEOUT(response)=10)
exten = s,2,Answer
exten = s,3,Playback(outboundmsgs/serviceinterrupt)
exten = s,4,Playback(outboundmsgs/choice)
exten = s,5,wait(3)
exten = 1,1,Goto(s,3) ; replay message
exten = 2,1,Goto(msgack,s,1); acknowledge message
exten = 3,1,Goto(callback,s,1)
exten = t,1,Playback(vm-goodbye)
exten = t,2,Hangup
[callback]
exten = s,1,Playback(outboundmsgs/customerrepwillcall)
-- exten = s,n,system(${SCRIPTS_DIR}/rep_callback.sh ${} )
exten = s,n,Playback(vm-goodbye)
exten = s,n,Hangup
how do I pass the number that was dialed (from the creation of a .call file)
to the rep_callback.sh script ?
thanks
Shawn
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[asterisk-users] open up firewall ports for Asterisk - safe?
Right now I've been working on setting up an Trixbox server on our internal network. Its behind the firewall, but I'd like to open up the firewall to it because we sometimes have developers working off site and I'd like them to be able to connect. Is this safe to do? I've got the Allow Anonymous Inbound SIP Calls box unchecked in freePBX. Is there anything else I need to do? Isn't there an issue with the extension/secret being passed in clear text? It looks like I need to open port 5060, and whatever ports are inbetween the rtpstart/rtpend values in /etc/asterisk/rtp.conf. Is that right? Right now thats ports, I've read that you can chop that down to 20 ports for just a few calls. We want to have 5-6 simultaneous calls, so if I set rtpstart to 10001 and rtpend to 10100, then open up those ports, is that adequate? Thanks for any help. -Ryan ___ --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
I have snom SI 120 sip phone and there is transfer button but id there any configuration in asterisk part for call transfer feature ??? Andrew Joakimsen [EMAIL PROTECTED] wrote: On 7/18/07, satish patel 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 There should be some sort of transfer button on your phone. You don't specify which model phone is being used, but both the SNOM 360 and 190 have a button labeled as Transfer. You should read the manual for more details :) ___ --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 - Shape Yahoo! in your own image. Join our Network Research Panel today!___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] open up firewall ports for Asterisk - safe?
Also the, the firewall does NAT for the server, it sounds like this may cause some issues for my SIP clients? -Ryan Ryan Stille wrote: Right now I've been working on setting up an Trixbox server on our internal network. Its behind the firewall, but I'd like to open up the firewall to it because we sometimes have developers working off site and I'd like them to be able to connect. Is this safe to do? I've got the Allow Anonymous Inbound SIP Calls box unchecked in freePBX. Is there anything else I need to do? Isn't there an issue with the extension/secret being passed in clear text? It looks like I need to open port 5060, and whatever ports are inbetween the rtpstart/rtpend values in /etc/asterisk/rtp.conf. Is that right? Right now thats ports, I've read that you can chop that down to 20 ports for just a few calls. We want to have 5-6 simultaneous calls, so if I set rtpstart to 10001 and rtpend to 10100, then open up those ports, is that adequate? Thanks for any help. -Ryan ___ --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] 1.4.X howto disable able xpp with ./configure
How do I (from command line) disable xpp in 1.4? Sure I can run menuselect but I want to do that from the ./configure script. Thanks, jerry ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zap channels unavailable?
Hi, I was talking to a technican at our telco yesterday and he told me that this problem was most likely caused by our PBX sending channel identification Exclusive when we dial out. If there's a heavy load and someone is dialing in on the same time on the same channel that we try to dial out from - it causes a deadlock. He said some Cisco PBXs have the same problem. Now, I'm no asterisk expert and I don't quite understand what this means. I've emailed the list asking if this can be changed to Preferred or Negotiation as the technican told me to. But I got no response yet. I did however solve the problem by reversing the channels that we dial out from (so now it tries the last channel first and then backwards to the first). Since all of our incoming calls come from the first to the last this minimizes the risk of a collision of the incoming/outgoing calls. This is of cource no long-term solution but anyway. I need to know if it's possible to change channel identification (whatever that is) to preferred or negotiation. Regards, Jan Martin Smith wrote: Hello Jan, We have also been seeing this issue, and we are running Asterisk 1.2.17/Zaptel 1.2.16/LibPRI 1.2.4-r2. We have been informed by our PRI provider that a 3rd party vendor has applied firmware to some hardware along our path, and that it has an unfortunate bug of hanging B-channels in the PRI flags resetting state. We have been assured that the vendor has been given a 30-day deadline (starting maybe 2 weeks ago) to fix the problem, and that it will go away soon. In the mean time, we've also had to restart Asterisk to free our B-channels for use, and any link-level event potentially re-hangs them. Keep us posted if you find out anything! Martin Smith, Systems Developer martins at bebr.ufl.edu Bureau of Economic and Business Research University of Florida (352) 392-0171 Ext. 221 -Original Message- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of jan.sarin at securia.se Sent: Tuesday, July 17, 2007 9:44 AM To: asterisk-users at lists.digium.com Subject: [asterisk-users] Zap channels unavailable? Hi, Lately we've noticed that some Zap channels on one of our PRIs are unavailable. We have 2 PRI lines with 60 channels in total. On the first PRI there are currently 20 channels that are not being used for some reason. I tried googling around and found some similar problems but there really was no solution (?). I'm not sure if this problem has occured now because of more load on the pbx but the machine should take it just fine (2x3,0 ghz xeon with 1 gb ram etc). Restarting asterisk makes the zaps' available again but they get locked later again. It seems it's always the same channels that are unavailable too? This one is unavailable and not being used... It's been in PRI Flags state resetting for hours now. Channel: 1 File Descriptor: 11 Span: 1 Extension: Dialing: no Context: from-pstn Caller ID: 702821667 Calling TON: 33 Caller ID name: Destroy: 0 InAlarm: 0 Signalling Type: PRI Signalling Radio: 0 Owner: None Real: None Callwait: None Threeway: None Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: alaw Fax Handled: no Pulse phone: no Echo Cancellation: 128 taps unless TDM bridged, currently OFF PRI Flags: Resetting PRI Logical Span: Implicit Hookstate (FXS only): Onhook If anyone can help me with this I'd be really glad. Thanks. Regards, Jan ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Upgrade Procedure
X-ECN Telecoms-MailScanner-Information: Please contact ECN Telecoms for more information X-ECN Telecoms-MailScanner: Found to be clean X-ECN Telecoms-MailScanner-From: [EMAIL PROTECTED] X-Spam-Status: No Nitesh Divecha wrote: Hello All, I would like to upgrade my recently installed Asterisk 1.2.21.1 to Asterisk 1.4.8? My OS is CentOS 4.5 with Linux 2.6.9-55.0.2.plus.c4smp #1 SMP Fri Jul 6 05:25:07 EDT 2007 i686 i686 i386 GNU/Linux Is there any detail step by step procedure to uninstall the current version and install Asterisk 1.4.8, Zaptel 1.4.4, Libpri 1.4.1, Addons 1.4.2? Cheers, Nitesh Hi, there is an UPGRADE.txt file in each folder of asterisk, zaptel, etc. You now need to './configure' before 'make'. Also check out 'make menuselect' to select which modules you need or don't. Please check out the default configs first, look in asterisk-1.4.8/configs/ -- thanks, Yusuf ___ --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] New book Asterisk Cookbook any good?
You can find the frame work of the Asterisk Cookbook here http://etel.wiki.oreilly.com/wiki/index.php/Main_Page I have begun inserting content from my personal wiki to the Cook book, slowly but surely... On 7/19/07, Jared Smith [EMAIL PROTECTED] wrote: On Thu, 2007-07-19 at 00:30 -0500, Larry Alkoff wrote: I have received mail from Amazon touting this book that will soon be available. Know anything about the book or it's authors? It's a little pricey. I have it on good authority that the book has been delayed, since two of the authors (and myself) have been busy working on the second edition of Asterisk: The Future of Telephony. Work is progressing on the Cookbook, it's just not coming along as quickly as they'd like. -- Jared Smith Community Relations Manager 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 -- /* Andrew Latham LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] [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] how to use call transfer
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] Sip Providers
Darrick Hartman (lists) wrote: [EMAIL PROTECTED] wrote: Hi John, Try ... carriers.icall.com - No minimum, unlimited concurrent calls, great price, some areas US 0,009. Only USA voipjet.com teliax.com - Not so cheap, and they do one-minute rounding ... not good at all. But they hold a very good quality Teliax does 60/6 rounding. You only pay for the first full minute, then fractionally there after. I've been using them for over 2 years with only a few issues that were quickly resolved. I also vouch for Teliax as I send overflow LD through their trunks. I know the people there and they are great guys. ___ --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
you are right but can u explain me i have SNOM SI 120 phone with transfer button on it but what entry i will do on asterisk feature.conf and what configuration and button will use for transfer call Gordon Henderson [EMAIL PROTECTED] wrote: On Wed, 18 Jul 2007, satish patel 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 You need to read your phone manual, not the asterisk manual. Every (SIP) phone has it's own ways and means (in addition to the generic features offered by asterisk detailled in features.conf) Gordon ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - 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] open up firewall ports for Asterisk - safe?
On 7/19/07, Ryan Stille [EMAIL PROTECTED] wrote: Right now I've been working on setting up an Trixbox server on our internal network. Its behind the firewall, but I'd like to open up the firewall to it because we sometimes have developers working off site and I'd like them to be able to connect. How many developers? And what kind of developers? If they're developing things for your phone system, then you may want them on their own development boxes instead. If you're a software shop and they're just users, then that's different. Is this safe to do? I've got the Allow Anonymous Inbound SIP Calls box unchecked in freePBX. Is there anything else I need to do? Isn't there an issue with the extension/secret being passed in clear text? I'm not the most knowledgable on what freePBX does, as far as the check box. My guess is that it's just tweaking the SIP users/peers in the sip.conffile. This gives only a minimal level of security, in my opinion. It looks like I need to open port 5060, and whatever ports are inbetween the rtpstart/rtpend values in /etc/asterisk/rtp.conf. Is that right? Right now thats ports, I've read that you can chop that down to 20 ports for just a few calls. We want to have 5-6 simultaneous calls, so if I set rtpstart to 10001 and rtpend to 10100, then open up those ports, is that adequate? If it were me, and I had 20 remote users or less, I would create a VPN and have them join my network that way. Then, no SIP ports would be open to the world. And the NAT problems would pretty much disappear. You may have a slight reduction in sound quality, depending on how you set up the VPN. I really haven't had major problems with it, but again, it depends on your type of VPN. We're using a site-to-site hardware-accelerated IPSec VPN for each of our remote sites (including my house), and I have not had any problems. Except when the underlying medium (the Intarweb) has latency/jitter problems. But then, straight SIP would have issues too... ___ --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 Hosting (Dedicated Servers)
On Tue, 17 Jul 2007, [EMAIL PROTECTED] wrote: Hello guys, Does anyone has an Asterisk server hosted off-site ? Like in those data centers that do web hosting in dedicated servers ? Is there a hosting company that has a special plan to host voip services like this, or usually is hosted in those dedicated servers like the ones I asked above ? What about QoS ? I know that most (if not all) are connected to their switch through a 10Mbps/100Mbps port ? But ... without a QoS rule ... even with that speed doesn't it affect the quality of voice ? Can you please tell me your experience ? Or point me some good hosting companies ? It can be a bit of a minefield - especially if it's an area you've not looked into before. I've been doing this (in a very minor way) for over 10 years now. So I run what could be described as a small hosting company, however, my hosts are currently inside another ISPs data centre rather than in a neutral data centre, so I get 100% of my Internet connectivity from my upstream ISP, and I am relying on them to do the right thing with having multiple transit providers and redundant network routing, UPSs and generators, all of which they have to my satisfaction. The next step for me would be to host in some neutral facility, get my own IP address space, my own AS number, then connect into multiple transit providers and arrange peering through the various neutral connection points that exist in the UK (LINX, MaNAP, etc.) I'm not big enough for that ... yet ;-) So I have routers and switches and connect into the ISP via a redundant mechanism (VRRP). I can apply QoS in my own routers, so that traffic from the Asterisk servers can be prioritised over the traffic from the LAMPy type servers, however, without the co-operation of the upstream ISP(s), you can't effectively apply QoS to the incoming traffic. (Fortunately in my instance, incoming is so much lighter than outgoing, and their network in not oversubscribed, so it's not really an issue) The easiest way to start, would be to simply place hosts inside another ISPs network, and rely on them for quality transit - ie. make sure they have multiple transit providers themselves, good power supplies, UPS, generators, etc. and if they are good and don't oversell their bandwidth then for the most part you'll be just fine. Once you have several hosts you might want to look at having your own router(s) and switch(es), but this will depend on how the hosting company operates (and your budget!) Finding that hosting company where you live is the challenge though! You need to ask lots of questions - can you get physical access to the servers? is it within driving distance? do you have remote screen keyboard facilities? (or if they offer remote hands and if so, how much do they charge?) How well do they connect to the world in general, and do they charge separate for power or bandwidth (and is bandwidth in terms of speed, or is it per bit pricing, or some combination of the 2?) Start phoning emailling - how fast do they answer the phone, or return email will be a good metric too... In the UK, London appears to be power starved right now - it would appear that bandwidth is virtually free, but you'll get charged per amp used! Outside London you rarely have this restriction, but then bandwidth is costly as it's got to be back-hauled to London (or Manchester), so local knowledge is something you'll need to find out about. Good luck! Gordon ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Blank Voicemails
Hi, we're running Asterisk 1.2.10 and have been randomly being left blank voicemails with long messages that we can't hear. I've searched and searched but cannot find a solution. This is what happens: Internal Server runs Asterisk 1.2.10 where our mailboxes are Incoming Server (behind a firewall) runs Asterisk 1.2.13 and calls are bridged between this server and our internal server. I have not heard any complaints from users on the .13 server, but it's happening too frequently to call a fluke on the .10 server. Caller gets voicemail, leaves a message, hangs up. Voicemail message is emailed to user saying the correct length (0:32, 1:12, etc.), tries to play it and player says 0 seconds long. Tries to access via phone, and the message again is blank, even though the text file specifies correct length. Voicemail is being saved in .WAV (wav49). I tried adding in [options] transmit_silence_during_record = yes into asterisk.conf and it seemed to help for a bit, but then we started getting the odd behavior again. Here is a capture of a failed message: //DIDN'T WORK Jul 6 11:57:07 DEBUG[9601] app.c: Locked path '/var/spool/asterisk/voicemail/default/116/INBOX' Jul 6 11:57:07 DEBUG[9601] app.c: Unlocked path '/var/spool/asterisk/voicemail/default/116/INBOX' Jul 6 11:57:07 DEBUG[9601] app.c: play_and_record: None, /var/spool/asterisk/voicemail/default/116/tmp/IVnRHt, 'wav49' Jul 6 11:57:55 DEBUG[9601] app.c: Locked path '/var/spool/asterisk/voicemail/default/116/INBOX' Jul 6 11:57:55 DEBUG[9601] app.c: Unlocked path '/var/spool/asterisk/voicemail/default/116/INBOX' Jul 6 11:57:55 DEBUG[9601] app_voicemail.c: Attaching file '/var/spool/asterisk/voicemail/default/116/INBOX/msg', format 'WAV', uservm is '2048', global is 2048 Jul 6 11:57:55 DEBUG[9601] app_voicemail.c: Sent mail to [EMAIL PROTECTED] with command '/usr/sbin/sendmail -t' //THIS WORKED/WORKED Jul 6 12:11:24 DEBUG[10184] app.c: Locked path '/var/spool/asterisk/voicemail/default/116/INBOX' Jul 6 12:11:24 DEBUG[10184] app.c: Unlocked path '/var/spool/asterisk/voicemail/default/116/INBOX' Jul 6 12:11:24 DEBUG[10184] app.c: play_and_record: None, /var/spool/asterisk/voicemail/default/116/tmp/rGc1XJ, 'wav49' Jul 6 12:11:51 DEBUG[10184] app.c: Locked path '/var/spool/asterisk/voicemail/default/116/INBOX' Jul 6 12:11:51 DEBUG[10184] app.c: Unlocked path '/var/spool/asterisk/voicemail/default/116/INBOX' Jul 6 12:11:51 DEBUG[10184] app_voicemail.c: Attaching file '/var/spool/asterisk/voicemail/default/116/INBOX/msg', format 'WAV', uservm is '2048', global is 2048 Jul 6 12:11:51 DEBUG[10184] app_voicemail.c: Sent mail to [EMAIL PROTECTED] with command '/usr/sbin/sendmail -t' They look identical! Same mailbox, same debug output, different behavior. I was noticing a pattern of certain callers (which made me turn on the record silence option), but my users tell me it's not only those callers, and sometimes those callers do successfully leave messages; I only hear when it doesn't work. What can I do?! I'm stumped, and the situation is intolerable. Thanks! Leah Newmark Capalon VoIP ___ --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] Pass Dialed number to a script
On Thu, 2007-07-19 at 08:43 -0400, [EMAIL PROTECTED] wrote: Looking for a simple way to pass the number that I dialed to a script in extensions.conf... something like this: [snip] how do I pass the number that was dialed (from the creation of a .call file) to the rep_callback.sh script ? You can define a channel variable in your call file, and that channel variable will be exposed to the dialplan. Simply put a line in your call file that looks like: Set: variablename=somevalue -- Jared Smith Community Relations Manager 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] Parsing IAXPeers from Asterisk Manager (PHP API)
On Thu, 2007-07-19 at 09:16 -0400, Chris Earle wrote: Trying to do what should be a basic info retrieval from my asterisk network -- get the list of IAXPeers and break down by IP, name, and Status etc. Unfortunately, this is one of those cases where the AMI output is the same as the CLI output, which makes it difficult to parse. (Regular expressions can help, but you obviously have an aversion to them, so we'll use another method.) As long as your IAX peer names are shorter than 16 characters, you can use a trick I've used for a while... first get the list of iax2 peer names: Action: IAXPeers ActionID: 987325918659161 ActionID: 987325918659161 Name/UsernameHost Mask Port Status flibbertygibbit (Unspecified) (D) 255.255.255.255 0 Unmonitored demo/asterisk216.207.245.47 (S) 255.255.255.255 4569 Unmonitored 2 iax2 peers [0 online, 0 offline, 2 unmonitored] and break on the first space or forward slash. Once I have an array of IAX peer names, I use the Command action to run iax2 show peer peername on each of the IAX peers, like this: Action: Command Command: iax2 show peer flibbertygibbit ActionID: 762315185916555 Response: Follows Privilege: Command ActionID: 762315185916555 * Name : flibbertygibbit Secret : Not set Context : blah Mailbox : Dynamic : Yes Callerid : Expire : -1 ACL : No Addr-IP : (Unspecified) Port 0 Defaddr-IP : 0.0.0.0 Port 4569 Username : hullabaloo Codecs : 0xe703 (g723|gsm|g729|speex|ilbc) Codec Order : (none) Status : Unmonitored Qualify : every 6ms when OK, every 1ms when UNREACHABLE (sample smoothing Off) --END COMMAND-- That output is obviously somewhat easier to parse, although it's not perfect either. Hopefully the Asterisk development community will eventually get around to rewriting much of the AMI actions to make their output easier for programs to parse. -- Jared Smith Community Relations Manager 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] G729 copy protection
On Thu, 2007-07-19 at 14:20 +0100, Bruce McAlister wrote: I have been trying to get the Solaris version of the G729 codec to work with asterisk 1.2.17 and 1.2.22. However, I come up against the very same error every time I try to install it. Has anyone out there seen this error, taken from the asterisk console straight from startup: I'm probably asking the obvious here, but were you able to successfully register your codec with the Digium registration server? Hase your ethernet MAC address changed since you registered the codec? -- Jared Smith Community Relations Manager 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] Upgrade Procedure
Thanks Yusuf, But is it possible to upgrade from Asterisk 1.2 to Asterisk 1.4? I went over the UPGRADE.txt but it didn't explain much about uninstalling the old version and then install a new version. Cheers, Nitesh Yusuf wrote: X-ECN Telecoms-MailScanner-Information: Please contact ECN Telecoms for more information X-ECN Telecoms-MailScanner: Found to be clean X-ECN Telecoms-MailScanner-From: [EMAIL PROTECTED] X-Spam-Status: No Nitesh Divecha wrote: Hello All, I would like to upgrade my recently installed Asterisk 1.2.21.1 to Asterisk 1.4.8? My OS is CentOS 4.5 with Linux 2.6.9-55.0.2.plus.c4smp #1 SMP Fri Jul 6 05:25:07 EDT 2007 i686 i686 i386 GNU/Linux Is there any detail step by step procedure to uninstall the current version and install Asterisk 1.4.8, Zaptel 1.4.4, Libpri 1.4.1, Addons 1.4.2? Cheers, Nitesh Hi, there is an UPGRADE.txt file in each folder of asterisk, zaptel, etc. You now need to './configure' before 'make'. Also check out 'make menuselect' to select which modules you need or don't. Please check out the default configs first, look in asterisk-1.4.8/configs/ ___ --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] Gtalk/Jabber connect issues in 1.4.8
DOH! Thanks [EMAIL PROTECTED] wrote: Hi, That should be: [EMAIL PROTECTED] Got it? BR, Dems I've included my jabber.conf below. I'm betting the following errors: [Jul 18 21:05:22] ERROR[28166]: res_jabber.c:609 aji_act_hook: JABBER: Node Error [Jul 18 21:05:22] WARNING[28166]: res_jabber.c:1537 aji_recv_loop: JABBER: Got hook event. jabber test [Jul 18 21:04:16] WARNING[32691]: res_jabber.c:1421 ast_aji_send: JABBER: Not connected can't send User: [EMAIL PROTECTED] Anyone have a hint?? Thanks [general] debug=no;;Turn on debugging by default. autoprune=yes ;;Auto remove users from buddy list. autoregister=yes;;Auto register users from buddy list. [asterisk] ;;label type=client ;;Client or Component connection ;;Route to server for example, serverhost=talk.google.com ;; talk.google.com ;[EMAIL PROTECTED]/asterisk ;;Username with optional roster. [EMAIL PROTECTED] secret= ;;Password port=5222 ;;Port to use defaults to 5222 usetls=yes ;;Use tls or not usesasl=yes ;;Use sasl or not [EMAIL PROTECTED];;Manual addition of buddy statusmessage=I am available ;;Have custom status message for ;;Asterisk. timeout=100 ;;Timeout on the message stack. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Upgrade Procedure
On Thu, 2007-07-19 at 10:53 -0400, Nitesh Divecha wrote: But is it possible to upgrade from Asterisk 1.2 to Asterisk 1.4? I went over the UPGRADE.txt but it didn't explain much about uninstalling the old version and then install a new version. For the most part, you should simply be able to install Asterisk 1.4 on top of Asterisk 1.2. The one place this won't work to well is with the Asterisk modules (usually located in /usr/lib/asterisk/modules). I typically move those modules to a new location, then install Asterisk 1.4 over the top of Asterisk 1.2, and change my configuration files to match the new Asterisk 1.4 settings. Another common problem is that a couple of new items have been added to asterisk.conf, so I typically renamed asterisk.conf before installing 1.4, so that I get the new version of asterisk.conf as well. -- Jared Smith Community Relations Manager 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] 1.4.X howto disable able xpp with ./configure
On Thu, Jul 19, 2007 at 10:05:26AM -0400, Jerry Geis wrote: How do I (from command line) disable xpp in 1.4? Sure I can run menuselect but I want to do that from the ./configure script. You need to do that through menuselect. But why would you disable xpp? If there's any problem building it, I'd like to know. And as you asked: in the branch /zaptel/team/tzafrir/kernelmove_conf you would do a: ./configure --with-modules=all,-xpp or even simpler: ./configure --with-modules=ztdummy (in the latter case not even ztcfg would have been built, as it is not needed) -- 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
Re: [asterisk-users] G729 copy protection
Jared Smith wrote: I'm probably asking the obvious here, but were you able to successfully register your codec with the Digium registration server? Hase your ethernet MAC address changed since you registered the codec? Hi Jared, I tried to run the register utility and I get as far as this (entering the Key-ID): # ./register Digium Product Registration - Version Copyright (C) 2004-2007, Digium, Inc. Use the '-l' option to see license information for software included in this program. Please select a product category. 1 - Digium Products 2 - Cepstral Products 0 - Quit Your Choice: 1 Please select a Product. 1 - Asterisk Business Edition 2 - G.729 Codec 3 - High Performance Echo Can 0 - Quit Your Choice: 2 Please enter the Key-ID: How do I know what the Key-ID is that it's asking for? If I run the asthostid app that accompanies the register utility and enter that ID in the above questions, then I get the following: The license key for this product should begin with G729! Am I doing something wrong? The README files dont quite explain how to get the Key-ID? 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
[asterisk-users] PRI Card
Hello, We're in the process of moving to a PRI circuit for our asterisk switch. Can anyone point me in the right direction as far as PRI Cards are concerned? 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] G729 copy protection
On Thu, 2007-07-19 at 16:20 +0100, Bruce McAlister wrote: Am I doing something wrong? The README files dont quite explain how to get the Key-ID? You should have received a key from Digium when you bought your license to use the G.729 codec. If you haven't yet bought any G.729 licenses, you can buy them from Digium's website at http://www.digium.com/en/products/voice/g729codec.php -- Jared Smith Community Relations Manager 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] how to use call transfer
On Thu, 19 Jul 2007, satish patel wrote: you are right but can u explain me i have SNOM SI 120 phone with transfer button on it but what entry i will do on asterisk feature.conf and what configuration and button will use for transfer call I'd need to read the manual (and I'm sure you're in a better position to do this than I am, as you have the phones and I don't!) You'd normally not need to do anything to the features.conf file to make phone transfers work using the phone features. Gordon Gordon Henderson [EMAIL PROTECTED] wrote: On Wed, 18 Jul 2007, satish patel 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 You need to read your phone manual, not the asterisk manual. Every (SIP) phone has it's own ways and means (in addition to the generic features offered by asterisk detailled in features.conf) Gordon ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - 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] New book Asterisk Cookbook any good?
On 7/19/07, Jared Smith [EMAIL PROTECTED] wrote: On Thu, 2007-07-19 at 00:30 -0500, Larry Alkoff wrote: I have received mail from Amazon touting this book that will soon be available. Know anything about the book or it's authors? It's a little pricey. I have it on good authority that the book has been delayed, since two of the authors (and myself) have been busy working on the second edition of Asterisk: The Future of Telephony. Work is progressing on the Cookbook, it's just not coming along as quickly as they'd like. -- Jared Smith Community Relations Manager Digium, Inc. Jared, As you know books never come as quickly as they (publishers, etc) like because what they like is RIGHT NOW! :) While I can't speak for everyone I can say that at least a couple of us have been waiting for some other big projects to finish up before the big push on The Cookbook. I'd expect to see activity on the wiki pick up a bit in the new few weeks: http://etel.wiki.oreilly.com/wiki/index.php/Main_Page -- 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] PRI Card
On Thu, 2007-07-19 at 11:28 -0400, mail-lists wrote: We're in the process of moving to a PRI circuit for our asterisk switch. Can anyone point me in the right direction as far as PRI Cards are concerned? You haven't given us many details on your setup, but I'll take a stab at answering your question anyway. For a single-port PRI card, I recommend the Digium TE120P card[1]. It can be configured for either T1 (United States and Canada), E1 (Europe, South America, and most of the rest of the world), or J1 (Japan). It will work with both channelized T1s as well as PRI circuits. This is a PCI card, and will work in either a 3.3 volt or 5 volt PCI slot. [1] http://www.digium.com/en/products/hardware/te120p.php -- Jared Smith Community Relations Manager 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] Sip Providers
Anthony, So you know all 4 that work at teliax.com I only know what others have told me about teliax.com Most of what I know was told to me from someone that worked there. Best regards, Al Bochter http://www.BochterServices.com --- Take a look at our online store http://www.bochterservices.com/onlinestore/ --- Join our forum. This is where you can talk about VOIP You can overview some providers others have used. http://bochterservices.com/phpbb/ --- Anthony Francis wrote: Darrick Hartman (lists) wrote: [EMAIL PROTECTED] wrote: Hi John, Try ... carriers.icall.com - No minimum, unlimited concurrent calls, great price, some areas US 0,009. Only USA voipjet.com teliax.com - Not so cheap, and they do one-minute rounding ... not good at all. But they hold a very good quality Teliax does 60/6 rounding. You only pay for the first full minute, then fractionally there after. I've been using them for over 2 years with only a few issues that were quickly resolved. I also vouch for Teliax as I send overflow LD through their trunks. I know the people there and they are great guys. ___ --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 Inbound (clean). Database: 000757-4, 07/18/2007 - 7/19/2007 10:35:00 AM ___ --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
w/ snom u can use the snom transfer and do nothing in asterisk. Or u can use the asterisk transfer (or bind transfer) changing the features.conf (see example) 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 Gordon Henderson wrote: On Thu, 19 Jul 2007, satish patel wrote: you are right but can u explain me i have SNOM SI 120 phone with transfer button on it but what entry i will do on asterisk feature.conf and what configuration and button will use for transfer call I'd need to read the manual (and I'm sure you're in a better position to do this than I am, as you have the phones and I don't!) You'd normally not need to do anything to the features.conf file to make phone transfers work using the phone features. Gordon Gordon Henderson [EMAIL PROTECTED] wrote: On Wed, 18 Jul 2007, satish patel 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 You need to read your phone manual, not the asterisk manual. Every (SIP) phone has it's own ways and means (in addition to the generic features offered by asterisk detailled in features.conf) Gordon ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - 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 -- Bruno De Luca, mailto:[EMAIL PROTECTED] FGA srl - http://www.fgasoftware.com - [EMAIL PROTECTED] - The Agile PBX http://www.voiceatwork.eu Tel: 02 997663.12, Fax: 02 91390172 ___ --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] iaxtel.com down?
www.iaxtel.com seems to be down, does anyone know if there is another way to register new numbers with them? -- Kyle Sexton ___ --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] PRI Card
Jared Smith wrote: On Thu, 2007-07-19 at 11:28 -0400, mail-lists wrote: We're in the process of moving to a PRI circuit for our asterisk switch. Can anyone point me in the right direction as far as PRI Cards are concerned? You haven't given us many details on your setup, but I'll take a stab at answering your question anyway. For a single-port PRI card, I recommend the Digium TE120P card[1]. It can be configured for either T1 (United States and Canada), E1 (Europe, South America, and most of the rest of the world), or J1 (Japan). It will work with both channelized T1s as well as PRI circuits. This is a PCI card, and will work in either a 3.3 volt or 5 volt PCI slot. [1] http://www.digium.com/en/products/hardware/te120p.php Jared, thanks for your reply - Our setup isn't complicated at all - just a PRI coming into an asterisk box. Maybe you could answer another question for me - what disadvantages does a PRI have from a channelized T1? or vice versa. is the TE205 the double port version of the TE120P? Also, what's required as far as echo cancellation goes? Is that built into these cards or do you have to move up to a TE207P? What is the difference between a TE205 and a TE210? Sorry about all the qeustions - the info on digiums web site doesn't really make this clear. ___ --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
Jared Smith wrote: On Thu, 2007-07-19 at 16:20 +0100, Bruce McAlister wrote: Am I doing something wrong? The README files dont quite explain how to get the Key-ID? You should have received a key from Digium when you bought your license to use the G.729 codec. If you haven't yet bought any G.729 licenses, you can buy them from Digium's website at http://www.digium.com/en/products/voice/g729codec.php OK, I got hold of the G729 Key that was issued to us by digium recently and have now successfully registered the codec on the host. However, it still comes back with the following warning on the console after a restart: [codec_g729a.so] = (Annex A/B (floating point) G.729 Codec (optimized for i686)) Jul 19 17:07:27 NOTICE[20591]: codec_g729.c:465 load_module: G.729 transcoding module version 32, Copyright (C) 1999-2007 Digium, Inc. Jul 19 17:07:27 NOTICE[20591]: codec_g729.c:469 load_module: This module is supplied under a commercial license granted by Digium, Inc. Jul 19 17:07:27 NOTICE[20591]: codec_g729.c:470 load_module: Please see the full license text supplied by the accompanying Jul 19 17:07:27 NOTICE[20591]: codec_g729.c:471 load_module: register utility, or ask for a copy from Digium. Jul 19 17:07:27 NOTICE[20591]: codec_g729.c:472 load_module: This product includes software developed by the OpenSSL Project Jul 19 17:07:27 NOTICE[20591]: codec_g729.c:473 load_module: for use in the OpenSSL Toolkit. (http://www.openssl.org/) Jul 19 17:07:27 NOTICE[20591]: codec_g729.c:474 load_module: Copyright (C) 1998-2006 The OpenSSL Project Jul 19 17:07:27 WARNING[20591]: codec_g729.c:481 load_module: Failed to initialize G.729 copy protection! I can see the licence there (10 channel), but it looks like the codec does not want to inititalize properly. 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] Upgrade Procedure
Thanks Jared, Does the same procedure works for updating Zaptel, Libpri, and Asterisks-Addons? Cheers, Nitesh Jared Smith wrote: On Thu, 2007-07-19 at 10:53 -0400, Nitesh Divecha wrote: But is it possible to upgrade from Asterisk 1.2 to Asterisk 1.4? I went over the UPGRADE.txt but it didn't explain much about uninstalling the old version and then install a new version. For the most part, you should simply be able to install Asterisk 1.4 on top of Asterisk 1.2. The one place this won't work to well is with the Asterisk modules (usually located in /usr/lib/asterisk/modules). I typically move those modules to a new location, then install Asterisk 1.4 over the top of Asterisk 1.2, and change my configuration files to match the new Asterisk 1.4 settings. Another common problem is that a couple of new items have been added to asterisk.conf, so I typically renamed asterisk.conf before installing 1.4, so that I get the new version of asterisk.conf as well. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Why using usecallerid=no?
Hi everybody. I'm in a discussion and someone ask me in which situation we should use the zapata.conf usecallerid set to no. I didn't have the answer. I understand what the usecallerid keyword does but I'm talking about a actual situation that is interesting to to avoid receiving the caller id. Thanks in advance! -- Paulo Garcia Pika Technologies ___ --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] PRI Card
On Thu, 2007-07-19 at 12:11 -0400, mail-lists wrote: thanks for your reply - Our setup isn't complicated at all - just a PRI coming into an asterisk box. Maybe you could answer another question for me - what disadvantages does a PRI have from a channelized T1? or vice versa. A channelized T1 is 24 channels over a T1 -- basically little more than 24 POTS lines that happen to come across a 2-pair digital interface rather than 24 pair analog interface. PRI over T1, on the other hand, only gives you 23 voice channels (often call B or Bearer channels), and uses the 24th channel for call signalling between you and the upstream switch. (This channel is called a D or Delta channel and I've even heard it called a Data channel, although my friends in big telco say that the D doesn't stand for Data.) PRI gives you much more advanced call control, allows you to more easily do things like DIDs, and gives much quicker call setup and dialing. Depending on your location, the only major downside to PRI might be price... they're often more expensive than channelized T1s. is the TE205 the double port version of the TE120P? Yes, the TE205P is the 2-port version of the TE120P. Once you get to the 2-port versions and 4-port versions of the Digium cards, they come in three flavors: a card for 5-volt PCI slots (the TE205P, for example), a card for 3.3-volt PCI slots (the TE210P, for example), and a card for a PCI-Express slot (the TE220P). Also, what's required as far as echo cancellation goes? Is that built into these cards or do you have to move up to a TE207P? It depends... if you're OK with software echo cancellation, you don't need anything special. If you want the hardware echo cancellation (which many people do), you could move to the TE207P or TE212P cards (for PCI slots in 5v and 3.3v, respectively), or the TE220P plus a VPMOCT064 echo cancellation module. What is the difference between a TE205 and a TE210? Explained above... difference between the different types of PCI slots. Sorry about all the qeustions - the info on digiums web site doesn't really make this clear. I agree. Luckily, I know Digium's marketing department is working to improve the information on the website so that it's clearer which hardware is appropriate for different situations. -- Jared Smith Community Relations Manager 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] Why using usecallerid=no?
I would think if one didn't want caller ID they wouldn't pay the phone company the extra couple bucks for it.. but just coincidence maybe :) Paulo Garcia wrote: Hi everybody. I'm in a discussion and someone ask me in which situation we should use the zapata.conf usecallerid set to no. I didn't have the answer. I understand what the usecallerid keyword does but I'm talking about a actual situation that is interesting to to avoid receiving the caller id. Thanks in advance! -- Paulo Garcia Pika Technologies ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI Card
mail-lists wrote: Jared Smith wrote: On Thu, 2007-07-19 at 11:28 -0400, mail-lists wrote: We're in the process of moving to a PRI circuit for our asterisk switch. Can anyone point me in the right direction as far as PRI Cards are concerned? You haven't given us many details on your setup, but I'll take a stab at answering your question anyway. For a single-port PRI card, I recommend the Digium TE120P card[1]. It can be configured for either T1 (United States and Canada), E1 (Europe, South America, and most of the rest of the world), or J1 (Japan). It will work with both channelized T1s as well as PRI circuits. This is a PCI card, and will work in either a 3.3 volt or 5 volt PCI slot. [1] http://www.digium.com/en/products/hardware/te120p.php Jared, thanks for your reply - Our setup isn't complicated at all - just a PRI coming into an asterisk box. Maybe you could answer another question for me - what disadvantages does a PRI have from a channelized T1? or vice versa. is the TE205 the double port version of the TE120P? Also, what's required as far as echo cancellation goes? Is that built into these cards or do you have to move up to a TE207P? What is the difference between a TE205 and a TE210? Sorry about all the qeustions - the info on digiums web site doesn't really make this clear. The ONLY disadvantage of PRI is (maybe) cost. Channelized T-1 is basically a T-1 version of 24 analog ports. ___ --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] Idefisk softphone - official 2.0 release - Zoiper
Hello guys, The so expected 2.0 release of Idefisk 2.0 softphone is a fact. Idefisk and Zoiper became one - Zoiper 2.06. Here are some of the features: SIP and IAX, TCP, TLS support, Multi-language support, Automatic provisioning (XML), URL handling, Outlook Integration, Native conferencing, API, Changeable number of lines You could read the complete Press Release here: http://www.zoiper.com/press.php For more information please visit http://zoiper.com Do not hesitate to contact me, Mira t: + 32 2 400 73 69 (GMT+2) e: [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] Why using usecallerid=no?
Paulo Garcia wrote: Hi everybody. I'm in a discussion and someone ask me in which situation we should use the zapata.conf usecallerid set to no. I didn't have the answer. I understand what the usecallerid keyword does but I'm talking about a actual situation that is interesting to to avoid receiving the caller id. If you do not have Caller*ID service on your telephone lines then you would want to set usercallerid=no. If you leave it set to yes, Asterisk will wait for Caller*ID information to arrive. This will delay processing of calls by (typically) 6 seconds. ___ --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] iaxtel.com down?
On Thu, 2007-07-19 at 11:05 -0500, Kyle Sexton wrote: www.iaxtel.com seems to be down, does anyone know if there is another way to register new numbers with them? The server handling iaxtel.com is currently down and/or disconnected from the internet. Digium is working to restore the service, but I haven't received any kind of estimate on how long this might take. (As I understand it, the bandwidth for the service was being provided by a third party, so the box may not be readily accessible.) -- Jared Smith Community Relations Manager 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] Why using usecallerid=no?
If you dont' get callerid from your telco and usecallerid is set to yes, then it will take at least an extra ring till it gets passed down to the dialplan. On 7/19/07, Paulo Garcia [EMAIL PROTECTED] wrote: Hi everybody. I'm in a discussion and someone ask me in which situation we should use the zapata.conf usecallerid set to no. I didn't have the answer. I understand what the usecallerid keyword does but I'm talking about a actual situation that is interesting to to avoid receiving the caller id. Thanks in advance! -- Paulo Garcia Pika Technologies ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why using usecallerid=no?
On Thu, 2007-07-19 at 13:55 -0300, Paulo Garcia wrote: I'm in a discussion and someone ask me in which situation we should use the zapata.conf usecallerid set to no. I didn't have the answer. In the United States, the CallerID information comes as a short modem tone (FSK data) between the first and second ring. If you have the usercallerid setting set to yes, Asterisk will wait until after the first ring to get the CallerID data, and then pass the call on to the dialplan. If you have usercallerid set to no, Asterisk won't wait until after the first ring to send the call to the dialplan. -- Jared Smith Community Relations Manager 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] Why using usecallerid=no?
Hi, You answered my question! Thanks Paulo On 7/19/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: Paulo Garcia wrote: Hi everybody. I'm in a discussion and someone ask me in which situation we should use the zapata.conf usecallerid set to no. I didn't have the answer. I understand what the usecallerid keyword does but I'm talking about a actual situation that is interesting to to avoid receiving the caller id. If you do not have Caller*ID service on your telephone lines then you would want to set usercallerid=no. If you leave it set to yes, Asterisk will wait for Caller*ID information to arrive. This will delay processing of calls by (typically) 6 seconds. ___ --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 -- Paulo Garcia Pika Technologies ___ --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] Blank Voicemails/Vonage Problem
Regarding this message, I've actually been told one caller who has consistently had this problem was using Vonage, but calling from his Verizon line, it worked. This skewed my survey. Therefore I do believe it's the same callers having the issue, and in which case, I think Vonage is to blame. I found this thread: http://forums.digium.com/viewtopic.php?p=49236highlight=sid=d3888f3bb90e5c96b5c0432bd632a2d4 but it doesn't help much. All incoming calls are using IAX. Did anyone have a similar problem and resolve it? Thank you. Leah Newmark Capalon VoIP [EMAIL PROTECTED] wrote: Message: 8 Date: Thu, 19 Jul 2007 10:41:44 -0400 From: Leah Newmark [EMAIL PROTECTED] Subject: [asterisk-users] Blank Voicemails To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1 Hi, we're running Asterisk 1.2.10 and have been randomly being left blank voicemails with long messages that we can't hear. I've searched and searched but cannot find a solution. This is what happens: Internal Server runs Asterisk 1.2.10 where our mailboxes are Incoming Server (behind a firewall) runs Asterisk 1.2.13 and calls are bridged between this server and our internal server. I have not heard any complaints from users on the .13 server, but it's happening too frequently to call a fluke on the .10 server. Caller gets voicemail, leaves a message, hangs up. Voicemail message is emailed to user saying the correct length (0:32, 1:12, etc.), tries to play it and player says 0 seconds long. Tries to access via phone, and the message again is blank, even though the text file specifies correct length. Voicemail is being saved in .WAV (wav49). I tried adding in [options] transmit_silence_during_record = yes into asterisk.conf and it seemed to help for a bit, but then we started getting the odd behavior again. Here is a capture of a failed message: //DIDN'T WORK Jul 6 11:57:07 DEBUG[9601] app.c: Locked path '/var/spool/asterisk/voicemail/default/116/INBOX' Jul 6 11:57:07 DEBUG[9601] app.c: Unlocked path '/var/spool/asterisk/voicemail/default/116/INBOX' Jul 6 11:57:07 DEBUG[9601] app.c: play_and_record: None, /var/spool/asterisk/voicemail/default/116/tmp/IVnRHt, 'wav49' Jul 6 11:57:55 DEBUG[9601] app.c: Locked path '/var/spool/asterisk/voicemail/default/116/INBOX' Jul 6 11:57:55 DEBUG[9601] app.c: Unlocked path '/var/spool/asterisk/voicemail/default/116/INBOX' Jul 6 11:57:55 DEBUG[9601] app_voicemail.c: Attaching file '/var/spool/asterisk/voicemail/default/116/INBOX/msg', format 'WAV', uservm is '2048', global is 2048 Jul 6 11:57:55 DEBUG[9601] app_voicemail.c: Sent mail to [EMAIL PROTECTED] with command '/usr/sbin/sendmail -t' //THIS WORKED/WORKED Jul 6 12:11:24 DEBUG[10184] app.c: Locked path '/var/spool/asterisk/voicemail/default/116/INBOX' Jul 6 12:11:24 DEBUG[10184] app.c: Unlocked path '/var/spool/asterisk/voicemail/default/116/INBOX' Jul 6 12:11:24 DEBUG[10184] app.c: play_and_record: None, /var/spool/asterisk/voicemail/default/116/tmp/rGc1XJ, 'wav49' Jul 6 12:11:51 DEBUG[10184] app.c: Locked path '/var/spool/asterisk/voicemail/default/116/INBOX' Jul 6 12:11:51 DEBUG[10184] app.c: Unlocked path '/var/spool/asterisk/voicemail/default/116/INBOX' Jul 6 12:11:51 DEBUG[10184] app_voicemail.c: Attaching file '/var/spool/asterisk/voicemail/default/116/INBOX/msg', format 'WAV', uservm is '2048', global is 2048 Jul 6 12:11:51 DEBUG[10184] app_voicemail.c: Sent mail to [EMAIL PROTECTED] with command '/usr/sbin/sendmail -t' They look identical! Same mailbox, same debug output, different behavior. I was noticing a pattern of certain callers (which made me turn on the record silence option), but my users tell me it's not only those callers, and sometimes those callers do successfully leave messages; I only hear when it doesn't work. What can I do?! I'm stumped, and the situation is intolerable. Thanks! Leah Newmark Capalon VoIP -- ade Procedure (Yusuf) 3. Re: New book Asterisk Cookbook any good? (Andrew Latham) 4. Re: how to use call transfer (satish patel) 5. Re: Sip Providers (Anthony Francis) 6. Re: how to use call transfer (satish patel) 7. Re: open up firewall ports for Asterisk - safe? (David Gomillion) 8. Blank Voicemails (Leah Newmark) 9. Re: Pass Dialed number to a script (Jared Smith) 10. Re: Parsing IAXPeers from Asterisk Manager (PHPAPI) (Jared Smith) 11. Re: G729 copy protection (Jared Smith) 12. Re: Upgrade Procedure (Nitesh Divecha) 13. Re: Gtalk/Jabber connect issues in 1.4.8 (Bruce Ferrell) 14. Re: Upgrade Procedure (Jared Smith) 15. Re: 1.4.X howto disable able xpp with ./configure (Tzafrir Cohen) 16. Re: G729 copy protection (Bruce McAlister) 17. Re: G729 copy protection (Bruce McAlister) 18. PRI Card (mail-lists) 19. Re: G729 copy protection (Jared Smith) 20. Re: how to use call transfer (Gordon Henderson) 21. Re: New book Asterisk Cookbook any good? (Kristian
[asterisk-users] MultiParking
Does anyone have the multiparking feature enabled in asterisk 1.4? or suggest multiple parking lots? ___ --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
On Thu, 2007-07-19 at 12:15 -0400, Nitesh Divecha wrote: Does the same procedure works for updating Zaptel, Libpri, and Asterisks-Addons? Yes, you should be able to install the new version over top of the old version without any problems. -- Jared Smith Community Relations Manager 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] Asterisk and Mitel 3300 ICP
Joseph, I have been completely unsucessful in doing the same thing as you - do you use a Mitel LX or MXe controller? We have a few probs with the LX internally too... Cheers Alex On Wed, 2007-07-18 at 06:42 +0100, Joesph O wrote: Good morning, it now works, failure was due to a misconfigured/misunderstood Class of Restriction Group Assignment for the SIP Trunk Routes on the 3300ICP. Now Asterisk can call the world through the Mitel and incoming calls (DID, operator transfers etc) to Asterisk via the 3300ICP, all work. Interesting side note - both phone systems have same range of extensions e.g 100 - 299 (just an example)and we created routes on both to point to the other for the range, of course, an extension should only exist on one at any time. therefore, if an extension does not exist locally, it is routed to the other and vice versa, this way, we keep same range of extensions and this has helped with migrating users who do not want to trade their loved Extension DID number for anything. will continue to test and share results. Joesph O. On 7/9/07, Joesph O [EMAIL PROTECTED] wrote: Good day everyone, I have Asterisk and Mitel 3300 ICP communicating via SIP. Calls to and from extensions on both sides are completing successfully (details on config coming soon). Problem is that calls from Asterisk to PSTN via E1 on Mitel 3300 ICP are rejected. What do I need to do to get Asterisk to route PSTN calls through it successfully? Here is an extract of the log on Asterisk whenever I try to call PSTN through 3300ICP, in this case, Extension 2540 on Asterisk called 2345678, 9 is a leading digit - Jul 7 16:48:07 DEBUG[6860] chan_sip.c: Outgoing Call for 92345678 Jul 7 16:48:07 VERBOSE[6860] logger.c: -- Called Mitel3300ICP/92345678 Jul 7 16:48:07 DEBUG[6860] channel.c: Prodding channel 'SIP/2540-b7904a98' Jul 7 16:48:07 VERBOSE[6860] logger.c: -- Started music on hold, class '24', on SIP/2540-b7904a98 Jul 7 16:48:07 DEBUG[6860] channel.c: Scheduling timer at 160 sample intervals Jul 7 16:48:07 DEBUG[6270] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 102: Found Jul 7 16:48:07 DEBUG[6270] chan_sip.c: Acked pending invite 102 Jul 7 16:48:07 DEBUG[6270] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Jul 7 16:48:07 DEBUG[6860] channel.c: Generator got voice, switching to phase locked mode Jul 7 16:48:07 DEBUG[6860] channel.c: Scheduling timer at 0 sample intervals Jul 7 16:48:07 DEBUG[6270] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 103: Found Jul 7 16:48:08 DEBUG[6270] chan_sip.c: Acked pending invite 103 Jul 7 16:48:08 DEBUG[6270] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 103: Match Found Jul 7 16:48:08 WARNING[6270] chan_sip.c: Forbidden - wrong password on authentication for INVITE to 'Tester sip:[EMAIL PROTECTED] ;tag=as07fef065' Jul 7 16:48:08 VERBOSE[6860] logger.c: -- SIP/Mitel3300ICP-0832de50 is circuit-busy ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message is intended only for the addressee and may contain confidential information. Unless you are that person, you may not disclose its contents or use it in any way and are requested to delete the message along with any attachments and notify us immediately. Transact is operated by Integrated Financial Arrangements plc Domain House, 5-7 Singer Street, London EC2A 4BQ Tel: (020) 7608 4900 Fax: (020) 7608 1200 (Registered office: as above; Registered in England and Wales under number: 3727592) Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem after upgrading from 1.2.21.1 to 1.2.22
Yesterday I upgraded from asterisk 1.2.21.1 to 1.2.22. We are running Zaptel-1.2.17.1. After upgrading all of our calls from the phone company (DIDs trunks with wink start on a channelized T1) were not coming in. Looking in the log file '/var/log/messages'. I saw the following error message. pbx.c: Cannot find extension '' in context '(null)' This was confusing since I had not changed my configuration files at all. After going back to 1.2.21, things worked fine again. Again last night I did a make clean; make install for 1.2.22. Again this error message appeared. Internal calls (Zap to Zap or Zap to SIP or SIP to SIP) worked fine. Outgoing calls worked fine. Has something changed in the way configuration files are parsed? Where does a context '(null)' come from. Don Pobanz ___ --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
Make sure there are no other files in the license path other than your valid license for this server. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce McAlister Sent: Thursday, July 19, 2007 09:13 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] G729 copy protection Jared Smith wrote: On Thu, 2007-07-19 at 16:20 +0100, Bruce McAlister wrote: Am I doing something wrong? The README files dont quite explain how to get the Key-ID? You should have received a key from Digium when you bought your license to use the G.729 codec. If you haven't yet bought any G.729 licenses, you can buy them from Digium's website at http://www.digium.com/en/products/voice/g729codec.php OK, I got hold of the G729 Key that was issued to us by digium recently and have now successfully registered the codec on the host. However, it still comes back with the following warning on the console after a restart: [codec_g729a.so] = (Annex A/B (floating point) G.729 Codec (optimized for i686)) Jul 19 17:07:27 NOTICE[20591]: codec_g729.c:465 load_module: G.729 transcoding module version 32, Copyright (C) 1999-2007 Digium, Inc. Jul 19 17:07:27 NOTICE[20591]: codec_g729.c:469 load_module: This module is supplied under a commercial license granted by Digium, Inc. Jul 19 17:07:27 NOTICE[20591]: codec_g729.c:470 load_module: Please see the full license text supplied by the accompanying Jul 19 17:07:27 NOTICE[20591]: codec_g729.c:471 load_module: register utility, or ask for a copy from Digium. Jul 19 17:07:27 NOTICE[20591]: codec_g729.c:472 load_module: This product includes software developed by the OpenSSL Project Jul 19 17:07:27 NOTICE[20591]: codec_g729.c:473 load_module: for use in the OpenSSL Toolkit. (http://www.openssl.org/) Jul 19 17:07:27 NOTICE[20591]: codec_g729.c:474 load_module: Copyright (C) 1998-2006 The OpenSSL Project Jul 19 17:07:27 WARNING[20591]: codec_g729.c:481 load_module: Failed to initialize G.729 copy protection! I can see the licence there (10 channel), but it looks like the codec does not want to inititalize properly. 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 ___ --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] PRI Card
Jared Smith wrote: On Thu, 2007-07-19 at 12:11 -0400, mail-lists wrote: thanks for your reply - Our setup isn't complicated at all - just a PRI coming into an asterisk box. Maybe you could answer another question for me - what disadvantages does a PRI have from a channelized T1? or vice versa. A channelized T1 is 24 channels over a T1 -- basically little more than 24 POTS lines that happen to come across a 2-pair digital interface rather than 24 pair analog interface. PRI over T1, on the other hand, only gives you 23 voice channels (often call B or Bearer channels), and uses the 24th channel for call signalling between you and the upstream switch. (This channel is called a D or Delta channel and I've even heard it called a Data channel, although my friends in big telco say that the D doesn't stand for Data.) PRI gives you much more advanced call control, allows you to more easily do things like DIDs, and gives much quicker call setup and dialing. Depending on your location, the only major downside to PRI might be price... they're often more expensive than channelized T1s. That's what I pretty much thought - We're about to sign a two year agreement so I wanted to make double sure a PRI was the best route. is the TE205 the double port version of the TE120P? Yes, the TE205P is the 2-port version of the TE120P. Once you get to the 2-port versions and 4-port versions of the Digium cards, they come in three flavors: a card for 5-volt PCI slots (the TE205P, for example), a card for 3.3-volt PCI slots (the TE210P, for example), and a card for a PCI-Express slot (the TE220P). Also, what's required as far as echo cancellation goes? Is that built into these cards or do you have to move up to a TE207P? It depends... if you're OK with software echo cancellation, you don't need anything special. If you want the hardware echo cancellation (which many people do), you could move to the TE207P or TE212P cards (for PCI slots in 5v and 3.3v, respectively), or the TE220P plus a VPMOCT064 echo cancellation module. I'm going to look into what sort of overhead software echo canceling incurs and go from there. Is it possible to get a TE205 and then add the echo cancellation separately or are they sold as a unit? I don't see any info about the TE220 and a google search for vpmocto64 didn't turn up much. What is the difference between a TE205 and a TE210? Explained above... difference between the different types of PCI slots. Sorry about all the qeustions - the info on digiums web site doesn't really make this clear. I agree. Luckily, I know Digium's marketing department is working to improve the information on the website so that it's clearer which hardware is appropriate for different situations. Thanks for all the advice/answers. ___ --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] PRI Card
On Thu, 2007-07-19 at 16:18 -0400, mail-lists wrote: That's what I pretty much thought - We're about to sign a two year agreement so I wanted to make double sure a PRI was the best route. Yes, in most circumstances PRI is the way to go. Is it possible to get a TE205 and then add the echo cancellation separately or are they sold as a unit? Yes, it's my understanding that you can add the echo can later, but you'd want to talk to someone in sales to make sure. (I'm new at Digium, and don't pretend to know everything yet.) I don't see any info about the TE220 and a google search for vpmocto64 didn't turn up much. The TE220 is a very new card, so I'm not sure there's a lot of info about it on Google yet. -- Jared Smith Community Relations Manager 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] Redundancy / Failover
Hello... I have been studying how to get Asterisk redundancy with Linux-HA. You can have 2 asterisk boxes, then install the linux-HA (Heartbeat) software and if one of the asterisk servers fails the other will take his IP so the clients registered on that server can re-register in the other. The redundancy can be hot stand by, in this case, both servers attend a part of the users. Also it's possible to have differents trunks connected in the servers and configure them in order to use the other's trunks in case of all lines busy. And of course, is possible configure the servers to have the same dial plan as there were only one server. I have been also studyng how to use Linux Virtual Server in order to get load balancing across a cluster of asterisk servers. On 7/18/07, Norman Franke [EMAIL PROTECTED] wrote: I've been evaluating Asterisk for a while, and things seem to be going very well. The issue of redundancy and automatic fail-over is now on my mind. I searched the archives and googled for solutions, but didn't really come up with much. We'll be using queues (modified), which precludes some of the standard redundancy solutions, since the queue needs to know all the agents and calls to effectively decide what to do. (Correct me if I'm wrong as to distributing queues.) As far as I can tell, I need to be able to fail over to a backup server. What's the best way to go about this? Obviously one can just have the clients re-connect to the new server, but with 50 machines, that can be a major pain. Both could have the same network, and just swap network cables and re-login. One could log into both and the second wouldn't send calls to the agent unless it got calls from the main switch in the event of failure. I'm sure there other solutions. I've come across other products like Ranch Networks (their web site is rather uninformative, although it seems promising) and Redfone's foneBRIDGE (which seems it may help.) I'll be using Sangoma cards on the Asterisk box, if that matters. Does anyone have experience they'd like to share on effective ways to do this? -Norman ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to use call transfer
Hi Satish - you are right but can u explain me i have SNOM SI 120 phone with transfer button on it but what entry i will do on asterisk feature.conf and what configuration and button will use for transfer call If you're using the Snom transfer button, you don't need to do anything in features.conf. In extensions.conf, just make sure that the dial() command used to call the snom phone uses the 't' flag. For example: exten = 100,1,Dial(SIP/100,20,t) This will allow the person using the '100' phone to transfer an incoming call. The 't' means the receiving party can transfer a call. If you use 'T' instead it means the calling party can transfer the call. - Noah ___ --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
Darryl Dunkin wrote: Make sure there are no other files in the license path other than your valid license for this server. Hi, I have just checked this, and there is only the 1 license file in the /var/lib/asterisk/licenses directory. Is that what you meant? 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] Redundancy / Failover
Hi Norman - On 7/18/07, Norman Franke [EMAIL PROTECTED] wrote: I've been evaluating Asterisk for a while, and things seem to be going very well. The issue of redundancy and automatic fail-over is now on my mind. I searched the archives and googled for solutions, but didn't really come up with much. To add to what Edgar said, yes, use linux-ha. It works nicely in combination with DRBD. DRBD uses a dedicated network interface on each box with a crossover cable between the two. It does a block level copy of the entire filesystem, so you have two machines that are identical. The you use the linux-ha heartbeat to monitor the OS and asterisk. If anything goes wrong, it can fail over to the second machine. This is pretty easy to set up with Analog lines. With PRI's you'd need the fonebridge or the FSV-4PFS from http://www.failsafevoip.com - Noah ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem after upgrading from 1.2.21.1 to 1.2.22
Hi Don - Yesterday I upgraded from asterisk 1.2.21.1 to 1.2.22. We are running Zaptel-1.2.17.1. After upgrading all of our calls from the phone company (DIDs trunks with wink start on a channelized T1) were not coming in. I believe you'll need to upgrade Zaptel to the latest version if you want to run asterisk 1.2.22. - Noah ___ --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] Any open source OSS system for asterisk?
Like billing by CDR. 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
[asterisk-users] job opportunity
Senior Developer We are a fast growing service provider to the small and medium sized business community in NYC. The company is looking for a senior developer to lead the next stage of development for a web-based service application. The opportunity is significant for the right entrepreneurial person. * B.S./M.S. in Computer Science or equivalent work experience * Hands-on experience with Asterisk a definite plus * Understanding of telephony technologies such as VoIP, IVRs, audio formats helpful * Cisco knowledge a plus. * Experience with Databases and Server side programming * Linux experience essential * Familiarity with Open Source Environments * Self motivated, highly proactive and reliable. Team Player. Ability to excel under pressure. * Experience with software development methodology, source code control bug tracking system. What you need to apply: * Entrepreneurial Spirit * 2+ years of PHP programming experience (PHP5 beneficial) * 1+ year leading a team or mentoring other engineers * Linux, Unix expertise * Mysql experience Contact: Roger Gins at [EMAIL PROTECTED] No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.476 / Virus Database: 269.10.9/907 - Release Date: 7/18/2007 3:30 PM ___ --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
Correct, if you have multiple licenses in there (say a single storage location for a cluster of servers), it won't load. If you've tried other architectures of the codec and still had no luck, I'd say contact Digium support on it. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce McAlister Sent: Thursday, July 19, 2007 13:55 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] G729 copy protection Darryl Dunkin wrote: Make sure there are no other files in the license path other than your valid license for this server. Hi, I have just checked this, and there is only the 1 license file in the /var/lib/asterisk/licenses directory. Is that what you meant? 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 ___ --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
Darryl Dunkin wrote: Correct, if you have multiple licenses in there (say a single storage location for a cluster of servers), it won't load. If you've tried other architectures of the codec and still had no luck, I'd say contact Digium support on it. Hmm, one caveat tho', these are the Solaris 10 32bit g729 codecs, and according to the FTP directory structure, are unsupported. This is why i emailed the list, hoping to bounce some ideas of you lot, to see if someone could help out :) Thanks for all the suggestions thus far, any more would be greatly appreciated. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to use call transfer
Noah Miller wrote: Hi Satish - you are right but can u explain me i have SNOM SI 120 phone with transfer button on it but what entry i will do on asterisk feature.conf and what configuration and button will use for transfer call If you're using the Snom transfer button, you don't need to do anything in features.conf. In extensions.conf, just make sure that the dial() command used to call the snom phone uses the 't' flag. For example: exten = 100,1,Dial(SIP/100,20,t) This will allow the person using the '100' phone to transfer an incoming call. The 't' means the receiving party can transfer a call. If you use 'T' instead it means the calling party can transfer the call. THIS IS INCORRECT! The options t and T are for DTMF based transfers. You do not need any options to Dial() to do phone based transfers using the transfer button on your IP phone (or FLASH on your IP ATA). ___ --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] Polycom 650 freezing on Transfer
All, Has anyone experienced a problem with the Polycom 650 phone freezing when you try to do a transfer? I am running asterisk 1.2.20.1 with Polycom SIP version 2.1.1.0052 and boot rom version 3.2.3.0002. I have Polycom 501 phones that work perfectly with the same software versions. The 650 phone; when I hit transfer the caller is placed on hold and the phone is still operational. As soon as I hit a number the phone it will immediately hang and then reboot. Instead of trying to key in a number if I use the KEM for the transfer it works perfectly fine. The problem only occurs when I try to enter a number into the phone and it always freezes on the first digit. ___ --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
Sorry that this is unrelated but, Bruce, do you double-click to send your messages? Just curious. Bruce McAlister wrote: Darryl Dunkin wrote: Correct, if you have multiple licenses in there (say a single storage location for a cluster of servers), it won't load. If you've tried other architectures of the codec and still had no luck, I'd say contact Digium support on it. Hmm, one caveat tho', these are the Solaris 10 32bit g729 codecs, and according to the FTP directory structure, are unsupported. This is why i emailed the list, hoping to bounce some ideas of you lot, to see if someone could help out :) Thanks for all the suggestions thus far, any more would be greatly appreciated. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] PRI Card
On 7/19/07, mail-lists [EMAIL PROTECTED] wrote: Hello, We're in the process of moving to a PRI circuit for our asterisk switch. Can anyone point me in the right direction as far as PRI Cards are concerned? Thanks! www.sangoma.com those are the best. ___ --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] Redundancy / Failover
On Jul 19, 2007, at 5:16 PM, [EMAIL PROTECTED] wrote: On 7/18/07, Norman Franke [EMAIL PROTECTED] wrote: I've been evaluating Asterisk for a while, and things seem to be going very well. The issue of redundancy and automatic fail-over is now on my mind. I searched the archives and googled for solutions, but didn't really come up with much. To add to what Edgar said, yes, use linux-ha. It works nicely in combination with DRBD. DRBD uses a dedicated network interface on each box with a crossover cable between the two. It does a block level copy of the entire filesystem, so you have two machines that are identical. The you use the linux-ha heartbeat to monitor the OS and asterisk. If anything goes wrong, it can fail over to the second machine. This is pretty easy to set up with Analog lines. With PRI's you'd need the fonebridge or the FSV-4PFS from http://www.failsafevoip.com Thanks, I wasn't aware of the FSV-4PFS box. Can one switch it remotely (e.g. over the network?) All of my dynamic data is stored in a database (using Asterisk RT to read queue and agent settings.) So, that eases part of the problem. It would be nice to have a way to gracefully switch boxes, e.g. all new calls to the backup box, wait until all calls on the primary normally end, and then take server down for an upgrade. It's impossible to tell what the Ranch Networks box does from their web site. Anyone using it? -Norman Franke www.myasd.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] how to use call transfer
On 7/19/07, satish patel [EMAIL PROTECTED] wrote: I have snom SI 120 sip phone and there is transfer button but id there any configuration in asterisk part for call transfer feature ??? Nothing else is required. Since the phone has a transfer button there is no need to use features.conf. What happens is the call is placed just like a regular phone call and then the phone indicates to Asterisk how the call should be transfered. This is the normal behaviour of SIP and there is no configuration for it. Usually you press transfer, dial the number to transfer to and then press transfer again, but I've never used a SNOM phone so I wouldn't know how those work. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem after upgrading from 1.2.21.1 to 1.2.22
You should be running the latest Zaptel LibPRI both of which recently have been updated. We run a similar configuration and have not seen this problem with the upgrade. I do get a flood of: Jul 19 19:36:18 WARNING[7277]: pbx.c:815 pbx_find_extension: Maximum PBX stack exceeded Jul 19 19:36:18 WARNING[7277]: pbx.c:815 pbx_find_extension: Maximum PBX stack exceeded Jul 19 19:36:18 WARNING[7277]: pbx.c:815 pbx_find_extension: Maximum PBX stack exceeded Jul 19 19:36:18 WARNING[7277]: pbx.c:815 pbx_find_extension: Maximum PBX stack exceeded But that's probably because we updated the dialplan at the same time as the upgrade (bad bad bad idea) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom 650 freezing on Transfer
David, In my experience this is usually due to the 650 being provisioned with a config file that pre-dates SIP version 2.1.1.0052. There's all kinds of things that can lurk in older configs that will cause the newer phone to behave oddly in just the way you describe. New phones need new configs, without exceptions. Try provisioning the 650 with the stock configs supplied with the latest firmware, then go from there. -Darren -- Darren Nickerson Senior Sales Support Engineer Telephony Depot www.telephonydepot.com +1.215.825.8710 ext 8106 (office) +1.215.243.8335 (fax) - Original Message - From: Klaverstyn, David C To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Thursday, July 19, 2007 6:34 PM Subject: [asterisk-users] Polycom 650 freezing on Transfer All, Has anyone experienced a problem with the Polycom 650 phone freezing when you try to do a transfer? I am running asterisk 1.2.20.1 with Polycom SIP version 2.1.1.0052 and boot rom version 3.2.3.0002. I have Polycom 501 phones that work perfectly with the same software versions. The 650 phone; when I hit transfer the caller is placed on hold and the phone is still operational. As soon as I hit a number the phone it will immediately hang and then reboot. Instead of trying to key in a number if I use the KEM for the transfer it works perfectly fine. The problem only occurs when I try to enter a number into the phone and it always freezes on the first digit. -- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Blank Voicemails
I've got the exact same issue lately. Check the msg.txt file for blank lines or 2 line caller I'd info. That's causing my issue. Haven't figured out why yet but manually removing the blank line and consolidating the callerid to one line allows me to play it via the phone. D Dave Bour Desktop Solution Center 905.381.0077 [EMAIL PROTECTED] For those who just want it to work... Giving you complete IT peace of mind. (Sent via Blackberry - hence message may be shorter than my usual verbose responses) PIN 4cc364db (as of March 24, 2007) - Original Message - From: [EMAIL PROTECTED] [EMAIL PROTECTED] To: asterisk-users@lists.digium.com asterisk-users@lists.digium.com Sent: Thu Jul 19 10:41:44 2007 Subject: [asterisk-users] Blank Voicemails Hi, we're running Asterisk 1.2.10 and have been randomly being left blank voicemails with long messages that we can't hear. I've searched and searched but cannot find a solution. This is what happens: Internal Server runs Asterisk 1.2.10 where our mailboxes are Incoming Server (behind a firewall) runs Asterisk 1.2.13 and calls are bridged between this server and our internal server. I have not heard any complaints from users on the .13 server, but it's happening too frequently to call a fluke on the .10 server. Caller gets voicemail, leaves a message, hangs up. Voicemail message is emailed to user saying the correct length (0:32, 1:12, etc.), tries to play it and player says 0 seconds long. Tries to access via phone, and the message again is blank, even though the text file specifies correct length. Voicemail is being saved in .WAV (wav49). I tried adding in [options] transmit_silence_during_record = yes into asterisk.conf and it seemed to help for a bit, but then we started getting the odd behavior again. Here is a capture of a failed message: //DIDN'T WORK Jul 6 11:57:07 DEBUG[9601] app.c: Locked path '/var/spool/asterisk/voicemail/default/116/INBOX' Jul 6 11:57:07 DEBUG[9601] app.c: Unlocked path '/var/spool/asterisk/voicemail/default/116/INBOX' Jul 6 11:57:07 DEBUG[9601] app.c: play_and_record: None, /var/spool/asterisk/voicemail/default/116/tmp/IVnRHt, 'wav49' Jul 6 11:57:55 DEBUG[9601] app.c: Locked path '/var/spool/asterisk/voicemail/default/116/INBOX' Jul 6 11:57:55 DEBUG[9601] app.c: Unlocked path '/var/spool/asterisk/voicemail/default/116/INBOX' Jul 6 11:57:55 DEBUG[9601] app_voicemail.c: Attaching file '/var/spool/asterisk/voicemail/default/116/INBOX/msg', format 'WAV', uservm is '2048', global is 2048 Jul 6 11:57:55 DEBUG[9601] app_voicemail.c: Sent mail to [EMAIL PROTECTED] with command '/usr/sbin/sendmail -t' //THIS WORKED/WORKED Jul 6 12:11:24 DEBUG[10184] app.c: Locked path '/var/spool/asterisk/voicemail/default/116/INBOX' Jul 6 12:11:24 DEBUG[10184] app.c: Unlocked path '/var/spool/asterisk/voicemail/default/116/INBOX' Jul 6 12:11:24 DEBUG[10184] app.c: play_and_record: None, /var/spool/asterisk/voicemail/default/116/tmp/rGc1XJ, 'wav49' Jul 6 12:11:51 DEBUG[10184] app.c: Locked path '/var/spool/asterisk/voicemail/default/116/INBOX' Jul 6 12:11:51 DEBUG[10184] app.c: Unlocked path '/var/spool/asterisk/voicemail/default/116/INBOX' Jul 6 12:11:51 DEBUG[10184] app_voicemail.c: Attaching file '/var/spool/asterisk/voicemail/default/116/INBOX/msg', format 'WAV', uservm is '2048', global is 2048 Jul 6 12:11:51 DEBUG[10184] app_voicemail.c: Sent mail to [EMAIL PROTECTED] with command '/usr/sbin/sendmail -t' They look identical! Same mailbox, same debug output, different behavior. I was noticing a pattern of certain callers (which made me turn on the record silence option), but my users tell me it's not only those callers, and sometimes those callers do successfully leave messages; I only hear when it doesn't work. What can I do?! I'm stumped, and the situation is intolerable. Thanks! Leah Newmark Capalon VoIP ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem building Asterisk 1.2.22
On 7/19/07, randulo [EMAIL PROTECTED] wrote: I just noticed that I asked about this same problem in March and got a workaround (edit makefile) from Tzafrir. Could someone explain why this codec_zap line is in Makefile has to be manually commented out? THere must be a reason why this happens on my box that I can fix? Do you have any patches against your Asterisk, Zaptel or Kernel? Actually are you using anything but the factory Kernel? Besides that I just wouldn't advise on using Fedora for any production system. I can assure you Asterisk, Zaptel LibPRI compile correctly on other distributions. ___ --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] Polycom 650 freezing on Transfer
Hi Darren, Thanks for your reply. I have since downgraded to version 2.0.3.0131 and the problem has gone. I am waiting on a link for firmware 2.1.2 so I can try that. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Darren Nickerson Sent: Friday, 20 July 2007 8:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom 650 freezing on Transfer David, In my experience this is usually due to the 650 being provisioned with a config file that pre-dates SIP version 2.1.1.0052. There's all kinds of things that can lurk in older configs that will cause the newer phone to behave oddly in just the way you describe. New phones need new configs, without exceptions. Try provisioning the 650 with the stock configs supplied with the latest firmware, then go from there. -Darren -- Darren Nickerson Senior Sales Support Engineer Telephony Depot www.telephonydepot.com +1.215.825.8710 ext 8106 (office) +1.215.243.8335 (fax) - Original Message - From: Klaverstyn, David C mailto:[EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion mailto:asterisk-users@lists.digium.com Sent: Thursday, July 19, 2007 6:34 PM Subject: [asterisk-users] Polycom 650 freezing on Transfer All, Has anyone experienced a problem with the Polycom 650 phone freezing when you try to do a transfer? I am running asterisk 1.2.20.1 with Polycom SIP version 2.1.1.0052 and boot rom version 3.2.3.0002. I have Polycom 501 phones that work perfectly with the same software versions. The 650 phone; when I hit transfer the caller is placed on hold and the phone is still operational. As soon as I hit a number the phone it will immediately hang and then reboot. Instead of trying to key in a number if I use the KEM for the transfer it works perfectly fine. The problem only occurs when I try to enter a number into the phone and it always freezes on the first digit. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729 copy protection
I'd wager that you're using the wrong path for the licenses. I believe the correct path is something like /var/opt/asterisk/licenses/ - it's whatever Asterisk has ast_config_AST_VAR_DIR set to, with /licenses/ at the end. The easiest way to tell, is to find the sounds dir (usually at /var/lib/asterisk/sounds/ on Linux), and go up a directory, and then from there create the licenses/ directory. - Bruce McAlister [EMAIL PROTECTED] wrote: Darryl Dunkin wrote: Make sure there are no other files in the license path other than your valid license for this server. Hi, I have just checked this, and there is only the 1 license file in the /var/lib/asterisk/licenses directory. Is that what you meant? 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 -- Jason Parker Digium ___ --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] Any plans for proper faxing support
I have already tried to contact to persons from Digium and I did not receive a response. I was wondering if there is any plan to support fully faxing in Asterisk, I.E.: A T38 Gateway of sorts. ___ --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] improved SMS?
On 7/12/07, Russ McBride [EMAIL PROTECTED] wrote: Newbie question(s): From what I can determine it sounds like the SMS messaging isn't as robust as it could be (?). I'm wondering if there's active work on that right now or if it's more of an issue about PSTN carrier that one would be using who would be responsible for passing the messages into the PLMN. Background-- I'm looking into the possibility of setting up an emergency messaging system here at the University that would send out voice, SMS, and emails. Any input relevant to that goal would probably be appreciated. Thanks, IMO the support for SMS in Asterisk is very poor. I think you should not consider Asterisk as part of your SMS solution but instead either talk with your preferred mobile carrier and see what sort of arrangements you can work out with them, or look at one of the SMS broker services. Another option could be using a GSM mobile phone along with a serial cable and some software, but I'm sure if you are doing this on a long-term basis it would be better to get a direct connection with the carrier. ___ --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] Any plans for proper faxing support
Andrew Joakimsen wrote: I was wondering if there is any plan to support fully faxing in Asterisk, I.E.: A T38 Gateway of sorts. You can use Asterisk 1.4's T.38 pass-through support in combination with the new OPAL-using t38modem (currently CVS) which now supports SIP (and not just H.323) to terminate T.38 calls. You can also use OPAL and chan_woomera to do essentially the same. Lee. ___ --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] NAT
Hi Stefan - What I want to accomplish: - calls within the LAN are re-invited (RTP goes from endpoint to endpoint) - asterisk detects when a call is going beyond the local LAN (over the NAT), and then stays in the middle. I'm wondering if this is hard to do and how I'm supposed to configure this. I don't really know how hard it would be to do what you describe, but if you're interested in getting the results you want with a minimum of effort, just keep asterisk in the media path all the time. Set canreinvite=no, and your calls should work consistently whether they stay inside the NAT or go outside. - Noah ___ --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] * core file not recognized
Hello, I am trying to debug a machine that segfaults. A core file is produced like /tmp/core.4545 . The command and error: 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 The box was rebooted before I had a change to run gdb, did I miss something? Thank you. JJ Yahoo! oneSearch: Finally, mobile search that gives answers, not web links. http://mobile.yahoo.com/mobileweb/onesearch?refer=1ONXIC ___ --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. Ok, I did the change you specified and now we can receive calls from Nextel phones but get no callerid on any call. How do I apply the patch to libmfcr2? -- 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
