[asterisk-users] Looking for a Voicemail Lamp device
I'm looking for an external device that can flash when there is new voicemail in a mailbox. I'm using an SPA3000 with a Uniden 5.8 ghz wireless phone system. Problem is, the Uniden system has it's own answering machine, which I don't want to use. But the message lamps are driven solely by the internal answering machine function. Looking for something else to give a visual indication, without being PC based. This is pretty much the one item keeping my wife from getting on board with the new regime.Thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] NAT/firewall/Asterisk/Polycom Phones
Are the Asterisk Implementation below recommended? Scenario One: Asterisk PBX is installed as gateway with two NICs. The internal NIC servers a network LAN of only Polycom IP phones while the other NIC is the public interface(static) I also have a few Polycom IP phones connecting from homes. Specs: Debian Linux, IPtables for Firewall Scenario Two: Asterisk PBX is installed as standalone with one NIC (public). Firewall (IPtables, Debian Linux) with two NICs - The internal NIC serves a network LAN of only Polycom IP phones while the other NIC is the public interface (static) I also have a few Polycom IP phones connecting from homes to the Asterisk PBX. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple TE110P cards in one chassis
Thermal Wetland ha scritto: Does anyone know if you can have multiple TE110P cards in one chassis? One server with two TE110P, shared interrupts, APIC routed irq, all things near ok. Sometime only one of these run HDLC error and some strange error, i think a 0,05% probability of error. :-) signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Psst... Top secret information: Codename Pineapple
* Phones = stations, regardless of where they areAsterisk = SIP Server, Phone = SIP Client Is a Media Server a Phone (ie SIP Client) ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Psst... Top secret information: Codename Pineapple
On Fri, Oct 13, 2006 at 07:00:54PM -0500, Eric ManxPower Wieling wrote: * Phones = stations, regardless of where they are Asterisk = SIP Server, Phone = SIP Client * Trunks = trunks to other SIP servers, bilateral Asterisk and the other server is peer to peer * Services = services you register for, like BroadVoice, Voop or FWD. (where asterisk acts as a phone) Asterisk = SIP Client, Other End = SIP Server Hmm, but I don't see how these ideas map to formal SIP concepts (RFC 3261). Phone = User Agent Client (places outgoing calls) and also User Agent Server (accepts incoming calls) But then Asterisk is both of these too. The term SIP Client does not appear in RFC 3261 at all. The term SIP Server does, in a loose generic way, when they mean SIP Proxy and/or SIP Registrar. Asterisk is never a SIP Proxy, it's a SIP endpoint (UAC/UAS). I think it *is* a registrar though. So what I'm asking is: what's fundamentally different between a phone, and trunk, and a service? How does Asterisk treat them differently? After all, placing a SIP call to a phone (via a dialplan) and routing a SIP call down a trunk (via a dialplan) are the same operation, aren't they? Maybe we need to authenticate to the other side. Maybe we want to require the other side to authenticate to us. But AFAICS that's something you might want to do set (or not) for any SIP endpoint. For instance, you might want to say that all devices with IP address 192.168.1.x can place calls without authentication. Regards, Brian. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call drop and strange CDR records
But, the CDR record looks strange (and this is the only common pointbetween those calls): Both the session timer and the talk timer are the same, but according to the log, the call are all answered after 3 to 5seconds ringing (so those timers should show this difference).Could you elaborate ?Do you mean that :1. For dropped calls and only for them, log files and CDR are not consistant ? 2. If my understanding is correct, from log files you've gotT: Phone is ringingT+3 or 5s: Call is answeredT+X: Call is hangedFrom CDR files :Session duration = Talk duration = X 3. Do you have any example of log files ?Cheers ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoIP+RJ11 Phone existed?
On Sat, Oct 14, 2006 at 09:34:33AM +1000, Paul Hales wrote: The Zulty's 4x5 does (or did) fwiw. Thanks. voipon.co.uk has them for GBP 299 or $557.64 (gulp) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7970 SIP won't update?
I am experiencing the same issue. However, I have not tried the VersionStamp field and will do so tomorrow. If you find an answer please post it to the list. On 10/13/06, Tim Connolly [EMAIL PROTECTED] wrote: Does anyone know what triggers the 7970 to update its config? I was able to get it to update to SIP, but the config I used initially won't go away. I am making small changes to the SEPxxx.cnf.xml file and rebooting the phone, the phone is downloading the (TFTP) new config file, but I don't see any change on the phone itself. I've looked at the VersionStamp and incremented that, but still no go. Any suggestions? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mitchel Constantin Snap - A desktop user interface for Asterisk www.snapanumber.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Centos kernel 34 vs. 42?
I'm now running kernel-2.6.9-42.0.3.EL Not really an answer to your question, but I found out all kernels above 2.6.16 do a better job on asterisk systems then the ones before that. No idea how this is possible as I'm in no way familiar with the inner workings of the linux kernel, but I found out after upgrading one system and because it was better there I upgraded a lot of other boxes and they all became better. Maybe you can get a newer kernel on centos then the 2.6.9 Possibly, but I would have to start worrying about kernel configs, compiling the lot and solving the problem of the box no longer being able to boot the kernel :) I looked for CentOS repo's but cannot find one that will throw a plain vanilla kernel my way. There's only a centos plus kernel but these are basically the same as the original kernels just with some filesystems enabled. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Centos kernel 34 vs. 42?
From: Remco Barendse Possibly, but I would have to start worrying about kernel configs, compiling the lot and solving the problem of the box no longer being able to boot the kernel :) You'd be better off starting with a Fedora kernel. Unfortunately RHEL/CentOS 4 is based on Fedora Core 3 which has been tagged legacy for quite some time now. The last kernel version was around 2.6.13 or so IIRC. And trying to go with a Fedora Core 5, 6 Test or Development (aka Rawhide) might not build because GCC has been upgraded to 4.0/4.1 from 3.4. I looked for CentOS repo's but cannot find one that will throw a plain vanilla kernel my way. And you're not likely to find one. RHEL/CentOS is based on a set kernel version with minimal changes, backporting required fixes/security updates only as necessary. Red Hat's focus with RHEL is 7 years of SLAs with no ABI changes, period - unlike Fedora Core (or Red Hat Linux before it for that matter - which did co-exist with RHEL for 2 years before the trademark change). There's only a centos plus kernel but these are basically the same as the original kernels just with some filesystems enabled. As I hinted above, the changes are just significant enough that Red Hat only backports, to the anal power when it comes to RHEL. And although Fedora Core/Development would be a good start for an updated kernel (far vanilla where countless things would break), there is so much that has changed in the toolchain and user-space of Fedora Core 4-6 that offers a 2.6.16+ release that many people probably haven't bothered. Especially since most people run RHEL/CentOS for its longevity and unchanging ABI/backports approach to an almost anal-level. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] rxfax problem (Trainability test failed)
Dear folks, I couldnt receive faxes and get the following debug traces on the console, I appreciate any help or even hints. Using Spandsp-0.2 app_rxfax.c:76 span_message: FLOW Get at 9600bps, modem 1 app_rxfax.c:76 span_message: FLOW Changed from phase 3 to 5 app_rxfax.c:76 span_message: FLOW Non-ECM carrier up app_rxfax.c:76 span_message: FLOW Non-ECM carrier down app_rxfax.c:76 span_message: FLOW Non-ECM carrier up app_rxfax.c:76 span_message: FLOW Non-ECM carrier trained app_rxfax.c:76 span_message: FLOW Non-ECM carrier down app_rxfax.c:76 span_message: FLOW Trainability test failed - longest run of zeros was 3356 app_rxfax.c:76 span_message: FLOW FTT app_rxfax.c:76 span_message: FLOW Non-ECM carrier up app_rxfax.c:76 span_message: FLOW Non-ECM carrier training failed . . app_rxfax.c:76 span_message: FLOW Trainability test failed - longest run of zeros was 2000 app_rxfax.c:76 span_message: FLOW FTT .. app_rxfax.c:76 span_message: FLOW Non-ECM carrier down app_rxfax.c:76 span_message: FLOW Trainability test failed - longest run of zeros was 1696 . chan_zap.c:4351 __zt_exception: Exception on 23, channel 1 chan_zap.c:3539 zt_handle_event: Got event On hook(1) on channel 1 (index 0) app_rxfax.c:329 rxfax_exec: Got hangup Regards. M. Shokuie Nia _ Don't just search. Find. Check out the new MSN Search! http://search.msn.click-url.com/go/onm00200636ave/direct/01/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP trunk from an Audiocodes mediant 1000
Hi, I am configuring an audiocodes Medant1000 to talk to my asterisk box. So far I have successfull in landing a single call from mediant to my *box. my sip conf is as follows: [general] context=sip bindport=5060 bindaddr=0.0.0.0 srvlookup=yes [3911700] type=friend host=dynamic dtmfmode=info secret=blah context=sip where 3911700 is my E1 telephone no. in my extensions.conf I have exten = 3911700,1,Dial(SIP/100) When I dial from outside to my E1 number calls are coming like the following: INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.9.230;branch=z9hG4bKac806223297 Max-Forwards: 70 From: sip:[EMAIL PROTECTED];tag=1c806218385 To: sip:[EMAIL PROTECTED];user=phone Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE Contact: sip:[EMAIL PROTECTED] Supported: em,100rel,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Remote-Party-ID: sip:[EMAIL PROTECTED];party=called;npi=1;ton=4 Remote-Party-ID: sip:[EMAIL PROTECTED];party=calling;privacy=off;screen=yes;screen-ind=1;npi=1;ton=0 User-Agent: Audiocodes-Sip-Gateway-Mediant 1000/v.4.60A.016.003 Content-Type: application/sdp Content-Length: 348 and the call get's connected to SIP/100 via the line in extensions.conf But what I am expecting is that the calls to come to the context's 's' extension. I am not sure if the changes are to be done in Asterisk or to Mediant. Any help in this will be much appreciated. raj ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 12 port FXx PCI card
Hi, http://www.openvox.com.cn/products_detail.php?genre_id=17id=45 The A1200P is a 12 port card, that used the same modules as a TDM400P. I have been looking at this card, and I want to know if anybody has used this card and what their experiences were? thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looking for a Voicemail Lamp device
Tom, The uniden TRU446 and the CLX465 both are supposed to detect stutter dial tone (SDT) from the phone company and light the MWI. When used with asterisk the SPA3000 can generate SDT. I'm not sure it can do so on its own. I gave up on the SPA 3000 due to echo problems. http://www.amazon.com/Uniden-CLX465-Expandable-Cordless-Handset/dp/B0007TIYCU http://www0.epinions.com/content_70491344516 Hope that helps, a little. Bob... Tom Lynn wrote: I'm looking for an external device that can flash when there is new voicemail in a mailbox. I'm using an SPA3000 with a Uniden 5.8 ghz wireless phone system. Problem is, the Uniden system has it's own answering machine, which I don't want to use. But the message lamps are driven solely by the internal answering machine function. Looking for something else to give a visual indication, without being PC based. This is pretty much the one item keeping my wife from getting on board with the new regime. Thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoipSupply? [Semi-Urgent]
Contact them again... they have always been very good... I'm chocking this up to the snow storm. On 10/13/06, Shaw Terwilliger [EMAIL PROTECTED] wrote: Matt wrote: Hi, Does anyone know what is going on with voipsupply? My sales guy hasn't been online in several days, their 800 number is fasy busy, as are their direct lines. And the canadian store website is down. What the heck is going on? If you search the archives from a few months ago you'll find a few unhappy voipsupply customers (including me). They never shipped what I ordered, didn't respond to any e-mail or calls. The president saw the list traffic and sent me a long apology (stating his commitment to service) and offered to send me an extra component that I had cancelled the order for--free of charge--as a show of good will. It's been two or three months since that promise, and I never received the part. He hasn't responded to my follow-up did you really mean it? e-mail either. -- Shaw Terwilliger [EMAIL PROTECTED] SourceGear LLC ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looking for a Voicemail Lamp device
My uniden phone is the TRU8885-3HS.On 10/14/06, Bob Chiodini [EMAIL PROTECTED] wrote: Tom,The uniden TRU446 and the CLX465 both are supposed to detect stutterdial tone (SDT) from the phone company and light the MWI.When used with asterisk the SPA3000 can generate SDT.I'm not sure it can do soon its own.I gave up on the SPA 3000 due to echo problems. http://www.amazon.com/Uniden-CLX465-Expandable-Cordless-Handset/dp/B0007TIYCUhttp://www0.epinions.com/content_70491344516Hope that helps, a little. Bob...Tom Lynn wrote: I'm looking for an external device that can flash when there is new voicemail in a mailbox.I'm using an SPA3000 with a Uniden 5.8 ghz wireless phone system.Problem is, the Uniden system has it's own answering machine, which I don't want to use.But the message lamps are driven solely by the internal answering machine function.Looking for something else to give a visual indication, without being PC based. This is pretty much the one item keeping my wife from getting on board with the new regime. Thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Centos kernel 34 vs. 42?
Hi, I like Centos as a basic platform, but I always then upgrade the kernel to the latest stable release from kernel.org The latest ones are using Centos 4.4 x86_64 with kernel 2.6.18 For simplicity, I always start with the .config from the original Centos kernel. My install sequence is: Untar the kernel source in /usr/src/kernels ln -s /usr/src/kernels/linux-2.6.18 /usr/src/linux (changing 2.6.18 to the version in use...) cd to /usr/src/linux do 'make mrproper' to clean out any leftover garbage. Older kernel versions often would not build if you missed this step, so I do it by habit now. Copy the .config file from the last installed Centos kernel directory (under /usr/src/kernels/2.6.) to /usr/src/linux Still in /usr/src/linux, do 'make oldconfig' and just press enter all the way through. This should give you a new kernel config matching the Centos distribution. (If you need anything other than the standard Centos kernel config, make changes here) Then do:- make make modules make modules_install make install If everything builds OK, reboot and get ready to select the new kernel as the Grub menu appears. If it all boots OK, you can edit /etc/grub.conf and change the default line to 0 so the new kernel is booted in future. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bryan J. Smith Sent: 14 October 2006 10:48 To: asterisk-users@lists.digium.com; asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Centos kernel 34 vs. 42? From: Remco Barendse Possibly, but I would have to start worrying about kernel configs, compiling the lot and solving the problem of the box no longer being able to boot the kernel :) You'd be better off starting with a Fedora kernel. Unfortunately RHEL/CentOS 4 is based on Fedora Core 3 which has been tagged legacy for quite some time now. The last kernel version was around 2.6.13 or so IIRC. And trying to go with a Fedora Core 5, 6 Test or Development (aka Rawhide) might not build because GCC has been upgraded to 4.0/4.1 from 3.4. I looked for CentOS repo's but cannot find one that will throw a plain vanilla kernel my way. And you're not likely to find one. RHEL/CentOS is based on a set kernel version with minimal changes, backporting required fixes/security updates only as necessary. Red Hat's focus with RHEL is 7 years of SLAs with no ABI changes, period - unlike Fedora Core (or Red Hat Linux before it for that matter - which did co-exist with RHEL for 2 years before the trademark change). There's only a centos plus kernel but these are basically the same as the original kernels just with some filesystems enabled. As I hinted above, the changes are just significant enough that Red Hat only backports, to the anal power when it comes to RHEL. And although Fedora Core/Development would be a good start for an updated kernel (far vanilla where countless things would break), there is so much that has changed in the toolchain and user-space of Fedora Core 4-6 that offers a 2.6.16+ release that many people probably haven't bothered. Especially since most people run RHEL/CentOS for its longevity and unchanging ABI/backports approach to an almost anal-level. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looking for a Voicemail Lamp device
I can get stutter dialtone using my spa3000, but the uniden doesn't respond to it by lighting the lamp. All it sees is an incoming call from the spa.It looks to me that I'll either need an external MWI device or I'm going to have to replace the Uniden phones. On 10/14/06, Tom Lynn [EMAIL PROTECTED] wrote: My uniden phone is the TRU8885-3HS.On 10/14/06, Bob Chiodini [EMAIL PROTECTED] wrote: Tom,The uniden TRU446 and the CLX465 both are supposed to detect stutterdial tone (SDT) from the phone company and light the MWI.When used with asterisk the SPA3000 can generate SDT.I'm not sure it can do soon its own.I gave up on the SPA 3000 due to echo problems. http://www.amazon.com/Uniden-CLX465-Expandable-Cordless-Handset/dp/B0007TIYCUhttp://www0.epinions.com/content_70491344516 Hope that helps, a little. Bob...Tom Lynn wrote: I'm looking for an external device that can flash when there is new voicemail in a mailbox.I'm using an SPA3000 with a Uniden 5.8 ghz wireless phone system.Problem is, the Uniden system has it's own answering machine, which I don't want to use.But the message lamps are driven solely by the internal answering machine function.Looking for something else to give a visual indication, without being PC based. This is pretty much the one item keeping my wife from getting on board with the new regime. Thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Generate Random Numbers in dialplan
Steve, Is RAND available in the latest trunk or do I need the 1.4 beta? If I do show function RAND it says its not available. Thanks, Jon - Original Message - From: Steve Murphy [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Saturday, October 14, 2006 12:30 AM Subject: [asterisk-users] Re: Generate Random Numbers in dialplan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anybody using inphonex service?
Tried them for all three - a tad pricey but good service imho.On 10/12/06, Crazy Boy [EMAIL PROTECTED] wrote:Hi,I want to register with http://www.inphonex.com VoIP provider. I want to configure my Trixbox and Asterisk servers with inphonex. Anybody using this service? Mainly, I want to do three tasks. They are1) Able to make calls to USA 2) Able to make international dialing3) Able to receive incoming calls through my DID. (Are they offering DID numbers?)If anybody using this inphonex service, please tell me your feedback. Looking forward to your response. Thank you. Regards,Chandra. Yahoo! Messenger with Voice. Make PC-to-Phone Calls to the US (and 30+ countries) for 2¢/min or less. ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: Generate Random Numbers in dialplan
Use the AGI I sent. It looks like the email did not put a CR correctly. Run it from the commandline and see if you get output. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Jon Weisman Sent: Saturday, October 14, 2006 12:45 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Re: Generate Random Numbers in dialplan Steve, Is RAND available in the latest trunk or do I need the 1.4 beta? If I do show function RAND it says its not available. Thanks, Jon - Original Message - From: Steve Murphy [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Saturday, October 14, 2006 12:30 AM Subject: [asterisk-users] Re: Generate Random Numbers in dialplan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: SIP stuck channel soft hangup?
On 2006-10-10 23:14:45 -0700, Martin Joseph [EMAIL PROTECTED] said: On 2006-10-10 20:25:44 -0700, Nic Bellamy [EMAIL PROTECTED] said : I am seeing occasional stuck SIP channels that seem to occur when the fricking Nokia E60 drifts out of WIFI range in the midst of a call. snipI wonder if there is some way to automatically soft hangup these channels when the qualify fails? Take a look at rtptimeout in sip.conf - that might do what you need. Ok, short version. I did a little more studying and found that rtptimeout only works in the sip.conf general section or in a peer definition. Since my e60 was a friend, I guess this was my issue. I added the rtptimeout=60 to my general section in sip.conf, and now when the e60 goes out of wifi range, 61 seconds later, my channels are clear! Sweet. Thanks again Nic. marty ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] A Call centre module on Asterisk
try http://astguiclient.sourceforge.netOn 10/7/06, Marnus van Niekerk [EMAIL PROTECTED] wrote: Yes, you can easily use asterisk for a call center, start looking here http://www.voip-info.org/wiki/view/Asterisk+call+queues M Imed Imed wrote: Hi, I'm a novice in asterisk. I'm just want to know if we can develop a Call centre application on an asterisk ? ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie question about meetme
yes to ztdummy: but you may have trouble when you try and run multiple simultaneous meetme sessions.On 10/5/06, Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote:omar parihuana wrote: Is possible use meetme feature without Zaptel card? (ztdummy will be the solution? )Yup. :P Thanks in advanced..--Mojo [EMAIL PROTECTED]Office Manager, Horan Company, LLC(907) 747- x112___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: SIP fails when internet connection lost.
On 2006-10-11 03:22:00 -0700, Thomas Kenyon [EMAIL PROTECTED] said: I have been seeing this problem for a long time and it occurs in 1.4.0b2 (as well as 1.2.0-1.2.12.1). If the internet connection is lost and I have SIP services that require me to register, any SIP devices attached to the system stop working. I have an IAX phone connected to one of my servers that I've been having this problem with which will work fine (and filover to the PSTN) the problem is that SIP handsets and softphones can no longer register or make calls. Is this normal behaviour or have I got something wrong with each server? I think this was recently discussed and a likely issue is the lack of a fully qualified domain name for you servers. Try searching the list archives for hosts. Basically if the server box tries to resolve itself via the network you have a problem. HTH, Marty ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Asterisk 'Hosting'
On 2006-08-21 02:44:55 -0700, Benny Amorsen [EMAIL PROTECTED] said: MR == Matt Riddell (NZ) [EMAIL PROTECTED] writes: MR And so you're thinking it would be better to run several hundred MR Asterisk instances?! Why not? As long as you stay away from the things that need zap timing, asterisk is really not much of a load. Don't forget translations or recording! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: VoipSupply? [Semi-Urgent]
On 2006-10-14 07:36:51 -0700, Matt [EMAIL PROTECTED] said: Contact them again... they have always been very good... I'm chocking this up to the snow storm. Yes, might still be too early, I see over 200K still without power in there neck of the woods (Buffalo, NY). Massive tree damage beyond belief, very fast down pour of heavy wet stuff. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Test to list
Sorry, just checking if my mail is working. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: SIP stuck channel soft hangup?
MJ == Martin Joseph [EMAIL PROTECTED] writes: MJ I added the rtptimeout=60 to my general section in sip.conf, and MJ now when the e60 goes out of wifi range, 61 seconds later, my MJ channels are clear! Sweet. Does this work with canreinvite=yes? (I can't see how it could, but I'd like to be surprised) /Benny ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looking for a Voicemail Lamp device
Tom, There are a couple of SIP based cordless phones out there. A little pricey, however. Such as: http://www.voipsupply.com/product_info.php?manufacturers_id=35products_id=923 It might be compatible with your existing cordless hand sets. Uniden seems pretty good about that. Or: http://www.voipsupply.com/product_info.php?manufacturers_id=40products_id=1007 Bob... Tom Lynn wrote: I can get stutter dialtone using my spa3000, but the uniden doesn't respond to it by lighting the lamp. All it sees is an incoming call from the spa. It looks to me that I'll either need an external MWI device or I'm going to have to replace the Uniden phones. On 10/14/06, *Tom Lynn* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: My uniden phone is the TRU8885-3HS. On 10/14/06, *Bob Chiodini* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Tom, The uniden TRU446 and the CLX465 both are supposed to detect stutter dial tone (SDT) from the phone company and light the MWI. When used with asterisk the SPA3000 can generate SDT. I'm not sure it can do so on its own. I gave up on the SPA 3000 due to echo problems. http://www.amazon.com/Uniden-CLX465-Expandable-Cordless-Handset/dp/B0007TIYCU http://www0.epinions.com/content_70491344516 http://www0.epinions.com/content_70491344516 Hope that helps, a little. Bob... Tom Lynn wrote: I'm looking for an external device that can flash when there is new voicemail in a mailbox. I'm using an SPA3000 with a Uniden 5.8 ghz wireless phone system. Problem is, the Uniden system has it's own answering machine, which I don't want to use. But the message lamps are driven solely by the internal answering machine function. Looking for something else to give a visual indication, without being PC based. This is pretty much the one item keeping my wife from getting on board with the new regime. Thanks! ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Switchtype,Signalling,rxwink warnings
What's in zapata.conf? On 10/13/06, Remi Quezada [EMAIL PROTECTED] wrote: When I reload the asterisk I get the following warnings: Oct 13 08:29:17 WARNING[10170]: chan_zap.c:10874 setup_zap: Ignoring switchtype Oct 13 08:29:17 WARNING[10170]: chan_zap.c:10874 setup_zap: Ignoring signalling Oct 13 08:29:17 WARNING[10170]: chan_zap.c:10874 setup_zap: Ignoring rxwink Everything works fine as far as I know, I can dial and complete calls. So why am I getting this warning. Is there anyway to fix this? Thanks, Remi ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Tellabs and a PRI
Doug Lytle wrote: Another question, Is anybody using the Tellabs 2572 EC with a PRI. When we moved from our analog lines to a PRI, I thought it would be simple stuff moving the EC to the PRI. Changed signaling, made sure that channel 24 wasn't being ECd and everything came up. But, I was getting complaints of random echo on the PRI. Local echo. Also, we weren’t able to do any kind of modem dial-outs (Adit 600 supplying the dialtone). Funny thing is that inbound/outbound faxing was working fine. Figured this one out. The Tellabs cards that I've been purchasing have the send side echo canceller (Option 38). This works great on analog lines, but apparently won't work on a PRI. Turned it off and now I can use modems. Since this was put into place over the weekend, it remains to be seen if it takes care of the local echo. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: Generate Random Numbers in dialplan
-- Forwarded message --From:Alexander Lopez [EMAIL PROTECTED]To:Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.comDate:Sat, 14 Oct 2006 13:04:08 -0400 Subject:RE: [asterisk-users] Re: Generate Random Numbers in dialplanUse the AGI I sent. It looks like the email did not put a CRcorrectly.Run it from the commandline and see if you get output. The AGI works ok for me. You have to insert a carriage return before the second echo. You also have to remove the single quote inserted after the 5. from -5'` to -5` See corrected script below.#!/bin/shRANDNUM=`echo $RANDOM$RANDOM | cut -c1-5`echo SET VARIABLE asteriskrandom $RANDNUM \\\n ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Strange FXS disconnection problem.
On Fri, Oct 13, 2006 at 09:05:54PM +0100, David Bath wrote: Hey all, Just as an update, incoming calls are fine. I have had several long calls today inbound on the PSTN with no drops. From the log it does sort of look like a hangup is being detected.. but its certainly not correct! Could anyone help me debug this a little further? As you probably recall, I have asked you to post debug log snippets just above the ones you posted. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Polycom HDVoice
According to docs I've seen this is the same codec that they use in the "Communicator" product that they target at Skype users. I have one of these and it does sound great. It's integration with the Skype API is primarily for the buttons that access the Skype app to start/stop calls. Aotherwise it's just a USB audio device with great hardware based echo cancellation. Michael --Original Message Text--- From: Peter Johnson Date: Sat, 14 Oct 2006 05:47:37 +1000 If your phones reinvite then it doesn't matter if Asterisk supports G722, only that both endpoints support the codec. Peter From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jessee J Holmes Sent: Saturday, 14 October 2006 4:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom HDVoice Actually, come to think of it, I don't know who will support it. Does Asterisk support G.722? From what I know it doesn't, is it included in the 1.4 beta? Will they support it? If Asterisk doesn't support it, then the phone won't do "HD" anyways. So then the questions comes to, what other PBX system or service provider will support this new "HD" standard? Jessee Holmes Atacomm / Ataractic Corporation www.atacomm.com V: 1-877-700-VOIP [EMAIL PROTECTED] Looking for voice over IP products? Visit our VoIP store at http://voipstore.atacomm.com/ On Oct 13, 2006, at 11:14 AM, Forrest Beck wrote: Has anyone used the Polycom HDvoice phone yet? I am curious if it uses a different codec. Does it actually sound any better? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: How do you like TrixBox?
I first learned asterisk via [EMAIL PROTECTED] Then I went to straight asterisk. This seems to be a theme. Getting your feet wet with [EMAIL PROTECTED]/Trixbox is not a bad way to go, especially if you want to get a functioning system up and running quickly. After tinkering with Trixbox then go back and do a plain Asterisk install. You will learn a lot, both about Asterisk and Trixbox. I've modified the Trixbox install scripts a bit to tailor them to my needs and ended up with the best of both worlds: a Trixbox installation that is more than plain vanilla but less than the somewhat cluttered Trixbox stock install. -MC ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Strange FXS disconnection problem.
Hi there, Thanks for the reply. I pasted everything from the console, with logger set to all, and verbose set to around 15. (as asked in original email) I have the whole session, from placing the call to the call hanging up, but it's pretty long, so I wasn't sure if you wanted to full thing. I'm sorry if I've misunderstood what you asked for - would you like it all? If not, would you mind please clarifying what I should post? Many thanks for your patience, Cheers, Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: 15 October 2006 01:35 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Strange FXS disconnection problem. On Fri, Oct 13, 2006 at 09:05:54PM +0100, David Bath wrote: Hey all, Just as an update, incoming calls are fine. I have had several long calls today inbound on the PSTN with no drops. From the log it does sort of look like a hangup is being detected.. but its certainly not correct! Could anyone help me debug this a little further? As you probably recall, I have asked you to post debug log snippets just above the ones you posted. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Centos kernel 34 vs. 42? [was: asterisk-users Digest, Vol 27, Issue 72]
Remco Barendse [EMAIL PROTECTED] wrote: I'm not running trixbox but normal Centos 4 with asterisk installed. I tried to find some further info on this but couldn't find any. Do audio problems occur with normal Centos and the latest kernel version too? (In other words, should every centos user downgrade??) I can't give you a definitive answer, but can provide a data point. I started out with an AAH installation early this year, then enabled yum and watched it upgrade various packages. Before long it upgraded the kernel and I found myself having to rebuild zaptel or be careful to always reboot the original kernel that zaptel was compiled against. A little later, zaptel wouldn't recompile against the newer kernel sources. So, a few weeks ago, I bit the bullet and cleaned out the stable, so to speak - got the Centos 2.6.9-42.EL kernel sources, figured out how CentOS wanted them installed, and also downloaded the latest Asterisk, zaptel, freepbx, etc. and rebuilt the whole lot. My config is now kernel 2.6.9-42.EL-smp, with Asterisk 1.2.12.1, zaptel 1.2.9.1, freepbx 2.1.2, etc. Cutting to the chase: I'm not aware of any audio problems, but our system doesn't get heavy use (only two lines and eight phones). Best, --- Les Bell, RHCE, CISSP [http://www.lesbell.com.au] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Generate Random Numbers in dialplan
On Sat, 2006-10-14 at 12:00 -0700, [EMAIL PROTECTED] wrote: Steve, Is RAND available in the latest trunk or do I need the 1.4 beta? If I do show function RAND it says its not available. Thanks, Jon Jon-- Forgive me, you didn't say which version you were using in your original post -duh-- I should have guessed. Nope, RAND is not in 1.2. But, if you are desparate, you could steal the 1.4 RAND function, and redo the module stuff in it to work under 1.2; there are plenty of patterns to follow. murf - Original Message - From: Steve Murphy [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Saturday, October 14, 2006 12:30 AM Subject: [asterisk-users] Re: Generate Random Numbers in dialplan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users smime.p7s Description: S/MIME cryptographic signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Student Research - Asterisk H323 Video
I am currently doing my thesis on an implementation of Video into Asterisk using H323 So I know that they are various mailing lists that demonstrate that SIP is the way forward, but sometimes It helps to use old equipment that one already owns so I am just looking for some simple ideas as to if possible Provide a quick and simple method of passing video through asterisk between 2 softphones I currently have the following Channels installed on the various systems Fedora Core 2 Asterisk-0h323 on Asterisk 1.1.00 Fedora Core 5 Asterisk h323 on Asterisk 1.2.12 Fedora Core 5 Asterisk h323 on Asterisk 1.2.12 Fedora Core 5 Asterisk h323 on Asterisk 1.4.Beta 2 Just looking for any pointers and decent directions Thanks in advance ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] New and Improved
Well, this is mostly just new. I thought this would be a good place to announce the opening of the forums on The Asterisk Blog. Check it out and leave a few messages to help get it's feet wet. Thanks guys! http://www.AsteriskBlog.com/forum-- www.AsteriskBlog.comYour home for easy to learn Asterisk stuff. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Codec swap (reinvite)
Hi, I've finally given up on trying to fax over my Digium TDM400 card. I've found that fax over VoIP is quite more reliable (at least I can receive the faxes). My ITSP supports G729 and alaw/ulaw. As I won't be receiving faxes everyday (just ocasionally), i pretend on using g729, unless a fax is detected. Is there any way to force asterisk to make a reinvite, and swap the codec on the fly? Something like this would be great: exten = fax,1,CodecChange(ulaw) exten = fax,2,rxfax(blablabla) Thanks, Julián J. M. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: How do you like TrixBox?
FWIW, I too started with AAH, but got really upset when tempted with an upgrade and learning the path was a total re-install. I hear things have gotten better since.In response, I went completely minimalist and turned to AstLinux. My primary reason was my only hardware resource was a PC without a hard drive. It cost me 1/2 as much to buy an IDE to Compact Flash adaptor and a 512MB CF card than a new hard drive. With my dusty old driveless PC now converted to a brand new, yet still dusty * system, I turned to the O'Reilly book, freely downloadable from the net. Working through the book, I learned everything I needed to configure a beginning system. I now run my home on AstLinux, and regardless of how much I hear about TrixBox being upgradeable, I still see it's a love/hate relationship every time a new release is put forth. I also read a lot about audio quality and AstLinux gets great marks in this respect. I'd recommend it to anybody new to * because it's minimalist, embedded systems approach teaches you so much about the tradeoffs you have to make to maintain a stable system that pleases it's users. TomOn 10/14/06, Michael Collins [EMAIL PROTECTED] wrote: I first learned asterisk via [EMAIL PROTECTED] Then I went to straight asterisk.This seems to be a theme.Getting your feet wet with [EMAIL PROTECTED]/Trixbox is nota bad way to go, especially if you want to get a functioning system up and running quickly.After tinkering with Trixbox then go back and do aplain Asterisk install.You will learn a lot, both about Asterisk andTrixbox.I've modified the Trixbox install scripts a bit to tailor them to my needs and ended up with the best of both worlds: a Trixboxinstallation that is more than plain vanilla but less than the somewhatcluttered Trixbox stock install.-MC___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] two SIP phones as one line
Hi, I am looking to replace a quirk of our old PBX system functionality with asterisk but after searching, archives, wiki, etc.. I cannot figure out how. Here is what I would like to do: PhoneA is a SIP hard phone, phoneB is an analogue phone connected to a SIP ATA. When an incoming call comes in, I would like to ring both phones, but if phoneA is answered first, I would like phoneB to be answered as well and left in a off hook state so that when someone picks up the receiver of phoneB, they can hear and participate in the conversation between the calling party and phoneA. I believe I would have to put both phones in a MeetMe conference, but how to I auto-answer phoneB when phoneA has answered the call? I suspect that this may not be possible with asterisk, but would like confirmation of that. Thanks in advance. -m signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
