Re: [asterisk-users] Grandtream Conference issue
Yes I'm aware of g711ulaw and PCMU..but I don't want to use this codec in Phones, as I'm using more then 10 IP phones in my network and going to increase it to connecting other remote offices,, and I don't want to increase the Bandwidth usage due to g711ulaw.. Keshav Jared Smith [EMAIL PROTECTED] wrote: On Thu, 2007-12-27 at 01:32 -0800, Keshav K. wrote: When I'm initiating the conference at that time, IP phone is sending the G711ulaw for the conference call, while in my phone I've set the all codec option to PCMU only. PCMU is another way of saying G711ulaw... they're the same codec. It's your basic 64kbps pulse-code modulated ulaw companded audio codec. (For more information on PCM audio and how it works, see the Digital Telephony section of Asterisk: The Future of Telephony, downloadable at http://www.asteriskdocs.org/ for free.) --- 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 - Looking for last minute shopping deals? Find them fast with Yahoo! Search.___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Grandtream Conference issue
Yes I've testes by this also as only allowing g719 codec, but in that case asterisk is sending 488 Not acceptable here, because INVITE form the phone is having g711ulaw and g711alaw Keshav dave cantera [EMAIL PROTECTED] wrote: keshaw, did you set your sip.conf to only allow g729? disallow=all allow=g729 I don't use g729 so the allow= may not be the correct syntax... here is the config I uise: disallow=all allow=ulaw allow=gsm allow=alaw daveC Keshav K. wrote: Hi, I'm using Grandstream IP phone GXP2000, with Asterisk 1.4.15 I'm using g729 codec and want to use only this codec for the calls. My normal calls are going fine. But issue is coming when I'm using the conference from the Line1 and Line2 Option. When I'm initiating the conference at that time, IP phone is sending the G711ulaw for the conference call, while in my phone I've set the all codec option to PCMU only. Due to this I'm facing issue. Any solution for this problem, please let me know. Regards, Keshav Regards, Kesh Lets change the future...lets change the world. Never miss a thing. Make Yahoo your homepage. ___ --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 No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.516 / Virus Database: 269.17.7/1194 - Release Date: 12/23/2007 05:27 PM -- My wife's sister is in California. I should buy her a Videophone2008! Truly, The Next Best Thing to Being There! -- WorldWideVideoPhones.com 856.380.0894 ___ --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 - Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now.___ --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] application not load
hi, thnks 4 reply, actully i am using asterisk 1.4.15 and that is defined in menuselect file.(xml file) so no need to add entry in module.conf Bhrugu mehta On Dec 27, 2007 7:37 PM, dave cantera [EMAIL PROTECTED] wrote: bhrugu, did you try and load it manually? Modules are compiled in to shared object (.so) files. They are installed to /usr/lib/asterisk/modules and can be turned on and off from loading by editing /etc/asterisk/modules.conf. Modules must include asterisk/modules.h. Modules must also export several functions. The following functions generally return 0 on success and non-zero on failure. Do not define any of these functions as static. http://www.lobstertech.com/doc/ast-12-func/#funcmod daveC Bhrugu Mehta wrote: hi, all I creat new application app_myapp.c for asterisk 1.4.15. I add this in asterisk/apps dir. to load. after compiling asterisk app_myapp.o and app_myapp.so has been created but when i run show applications at cli . my application not displayed. what's wrong??? any suggestion!!! thanks Bhrugu Mehta ___ --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 -- My wife's sister is in California. I should buy her a Videophone2008! Truly, The Next Best Thing to Being There! -- WorldWideVideoPhones.com 856.380.0894 ___ --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] New voicemail app (supports many interfaces, including Audix)
On Dec 28, 2007 12:08 AM, Matt Riddell [EMAIL PROTECTED] wrote: -BEGIN PGP SIGNED MESSAGE- I'm assuming that since you sent it to Asterisk Users (Non-Commercial Discussion) it is free. Is it also Open Source? Classic Matt! And also (to Matt) Classic, Matt! ___ --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] CDR
Grey Man wrote: - Original Message From: Steve Murphy [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, 27 December, 2007 5:44:01 PM Subject: Re: [asterisk-users] CDR Greyman-- No real new functionality in 1.4, except a cdr.conf option that lets you control whether you see one-channel cdrs. I haven't been working on CDR's the last few months in favor of other projects that seem a little more urgent. Plus, I have some folks urging me NOT to proceed until some architectural issues are discussed, which might be wise. I have been working on one bug where I did make some substantive changes to how the CDR's are generated, but it is almost certain that these changes will only show up in trunk. I've reached the limit of what I can do in 1.4; it is simply impossible to do anything with CDR's in 1.4 without tearing the very fabric of time and space, and just plain getting everybody upset... at least, those who were not erased from existence by the tear... on a more serious note, the changes are intrusive enough, the behavior changes big enough, that they really don't qualify to be applied to a current release. It's a huge job! My past work was just in the ZAP channel driver code, and because it's so asynch, and all split up into different code, it's really tough to get the right pieces in the right places at the right time in the right way. What this all says is that I'm most likely NOT doing it the right way. And what worries me most is that there might not be any right way. But I'm still new to this, and will get back around to it hopefully fairly soon. murf Hi Steve, Thanks for the update. I agree it's complicated and looks like it does require a look at the design of Asterisk and where CDR's are generated. As you've already documented and lots of us have discovered generating a single CDR for each bridged call is not suitable when CDR's are used for billing and blind and attended transfers are taking place. For any SIP (can't speak for other channels but most likely the same) service providers running Asterisk that are not aware of this problem you will not be getting correct CDR's on blind and attended transfers. Also depending on your dial plan users may be able to send a 302 Redirect response (301 or 302) to an incoming call and get a free outgoing call. This has the potential to cost you money which is very dangerous if any of your users cotton on to it. The easiest way to check your susceptibility is to do call an expensive destination, blind transfer to a free destination and then check the CDRs and pay close attention to the call durations of each CDR. I'll go back to trying to find a way to detect and block dangerous REFER requests at the SIP Proxy before they get to Asterisk. Regards, Aaron Wondering out loud if as the whole CDR defect is looked at, that the issue of blind vs attended transfers should be examined as well. In the rest of the telephony world, there is no difference. It is simply a transfer. An attended ( or announced ) transfer can become a blind transfer simply by the transferrer hanging up. How the original design became what it is is a puzzle, unless the original architect was not too familiar with the time proven practice in PBX systems for many years. Having to differentiate between types of transfer before initiating the transfer is not very user friendly. Once that is corrected will it make fixing the CDR defect easier? In other systems with SMDR/SMDI/CDR the report contains information and timing for the total call, as mostly no one cares who answered the call and then transferred, it is total time that is important for billing. John Novack -- Dog is my co-pilot ___ --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] Definity G3R and MWI
Hey everybody, I've just spent the last two hours Googling and searching the Wiki. I'm trying to find if there are any listings of codes for the Avaya Definity G3R, to allow for an Asterisk system to turn on/off a phones MWI that is attached to a G3. We are looking to use an Asterisk system as a voice mail server. I'm not having any luck, anybody have such information? Thanks, 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 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] Definity G3R and MWI
I've just spent the last two hours Googling and searching the Wiki. I'm trying to find if there are any listings of codes for the Avaya Definity G3R, to allow for an Asterisk system to turn on/off a phones MWI that is attached to a G3. We are looking to use an Asterisk system as a voice mail server. I think you're going to need to integrate via the SMDI feature of Asterisk and figure out what the Definity needs as well to work with an SMDI connection. -- -- Bird's The Word Technologies, Inc. http://www.btwtech.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] Queue members, URI.
Hi all, sorry to rehash this - but I'm having similar issues. I'm on Asterisk 1.0 and have been using Queues without any problems locally. I mean, all the SIP devices on my local server can be added to queues using AddQueueMember. However, I now need to allow agents from other servers to log in to the queue and I thought I could do this with IAX2/calleridnum or something ..but it doesn't work. The only way I was able to get it to work was by defining them as Local/number@context But this has major drawbacks. They are in the queue and can receive calls -- but when the queue directs a call to them, it loses control over it and calls are just transfered to the one agent and don't timeout the caller in the queue isn't really in the queue anymore... The reason it didn't work with IAX2 was that every time an agent logged in ... Add QueueMember would put them in as IAX2/iaxpeer/random port ... because that's where they were connecting over at that very moment. But the queue is unable to locate them at that same port when an actual call comes into the queue! Since they are always moving around ports under the IAX2 protocol. So using Local works cause it uses the dialplan's intelligence in locating an extension on an iaxpeer -- but it's not really a channel like Zap or Sip ... so queue functionality is lost So I'm revisiting this now --- is there any way to use IAX2 peers as queue members? Maybe I'm writing the URI's wrong Or is this something that has been fixed drasically in asterisk 1.2/1.4 anyone know? Ideas/suggestions appreciated ... -- Chris Earle Thomas Kenyon [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Is there an advantage to having a Queue members URI in the form: SIP/User (or indeed IAX2/User) Over Local/number@context ? I know that the latter will allow you to do things like set counting logic etc. through dialplan operations, but the former appears to be a more direct route to calling the party. (and if need be, there is the ability in queues to run a script on connection iirc). TIA for any clarification. ___ --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] CDR
Grey Man wrote: On a separate note does anyone know how to block transfers on a SIP channel? I can block REFER requests from my SIP Proxy but I have to support some transfers so that's not an option. I'd put the SIP devices in a separate context that doesn't include any [twkTWK] in the Dial application. ___ --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 voicemail app (supports many interfaces, including Audix)
Dean, I am saying nothing of the sort. To clarify, I am saying that I do not see the people you mentioned fishing for free ideas or posting commercially to the User's list with the exception of yourself now, when you had affiliation with Mexuar, and a handful of other times. I find it funny when I see the commercials on TV and email spam asking for Inventor's Ideas, all it would take is one sucker to rip off with a great idea to make all those commercials pay off. Some of us have long memories, can put pieces together and will call you out when something is fishy. Thanks, Steve Totaro Dean Collins wrote: So you're saying people like snapanumber, mexuar and other commercially related Asterisk applications cant charge money huh Steve? Maybe this conference call may interest you. http://recordings.talkshoe.com/TC-22622/TS-75263.mp3 Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Thursday, 27 December 2007 7:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] New voicemail app (supports many interfaces, including Audix) Licensing your thoughts, do you have a unique patent or a even a patent on an improvement? Aren't you the guy soliciting the user's list for The Next Geewhiz App idea a while ago? Sharks are everywhere. Anyways, this is the Users, soliciting should be done on the Biz list. Thanks, Steve Totaro Dean Collins wrote: Are you selling/licensing the new voicemail app or just asking if people want to download it? The reason for asking is if you are selling it I have some thoughts on how voicemail on asterisk can be improved and would like to discuss licensing this to you. Not really working for the next few days till after new year though so email replies will be sporadic. Cheers, Dean *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Justin Newman *Sent:* Thursday, 27 December 2007 5:38 PM *To:* asterisk-users@lists.digium.com *Cc:* [EMAIL PROTECTED] *Subject:* [asterisk-users] New voicemail app (supports many interfaces,including Audix) We just completed a new implementation of voicemail for Asterisk. It's much cleaner than Comedian mail and can emulate several voicemail user interfaces, including Audix. It's a great replacement for Audix. All of the sounds/prompts are presently being re-recorded by a professional female voice. If you are interest in the app, let us know at [EMAIL PROTECTED] Justin Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://us.rd.yahoo.com/evt=51734/*http:/tools.search.yahoo.com/newsearc h/ category.php?category=shopping ___ --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] Problems with zaptel and HFC-S PCI card
Hi list, Now that I've got my Asterisk server to recognize my HFC-PCI card, I've run into some serious problems. The first thing I noticed was this message that would show up every five seconds on the CLI: Dec 27 15:46:42 WARNING[12484]: chan_zap.c:2512 pri_find_dchan: No D-channels available! Using Primary channel 3 as D-channel anyway! == Primary D-Channel on span 1 down Second, the syslog and the kern.log were quickly filling up with messages like these: Dec 27 16:52:53 bitis kernel: zaphfc: sync lost, cpu throtteling enabled. Dec 27 16:52:53 bitis kernel: zaphfc: sync lost, pci performance too low. you might have some cpu throtteling enabled. Dec 27 16:52:53 bitis last message repeated 31 times Dec 27 16:52:53 bitis kernel: zaphfc: bchan rx fifo not enough bytes to receive! (z1=4069, z2=4062, wanted 8 got 7), probably a buffer overrun. Asterisk doesn't even have to be running for this to happen, but it can be brought to a halt by unloading the zaphfc module. I'm not aware of any CPU throttling on this system (an AMD Athon running at 1100 MHz). The OS is Debian etch running Linux kernel 2.6.18 (-5-k7). I've installed asterisk and asterisk-bristuff 1.2.13~dfsg-2etch2, as well as zaptel and zaptel-source 1.2.11.dfsg-1 to compile the necessary modules. My current configuration is as follows: cat /proc/zaptel/* Span 1: ZTHFC1 HFC-S PCI A ISDN card 0 [TE] layer 1 ACTIVATED (F7) AMI/CCS 1 ZTHFC1/0/1 Clear (In use) 2 ZTHFC1/0/2 Clear (In use) 3 ZTHFC1/0/3 HDLCFCS (In use) I think TE mode is fine, since I only need it to connect an outside line. Internally, I plan (hope) to use only SIP phones. /etc/asterisk/zapata.conf : [trunkgroups] [channels] language=en context=isdn-in switchtype=euroisdn pridialplan=local prilocaldialplan=unknown nationalprefix = 0 internationalprefix = 00 overlapdial=yes signalling=bri_cpe_ptmp rxwink=300 usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=4.5 txgain=-3 group=1 callgroup=1 pickupgroup=1 immediate=yes #include zapata-channels.conf Incidentally, this needs to work in the Netherlands. /etc/asterisk/zapata-channels.conf switchtype = euroisdn signalling = bri_net channel = 1-2 To connect to an outside line, I think signalling may need to be set to something else, but I'm not sure. The genzaptelconf shell script I used to produce it is buggy, so for all I know these settings may be wrong or even incomplete. /etc/asterisk/modules.conf [modules] autoload=yes noload = pbx_gtkconsole.so noload = pbx_kdeconsole.so noload = app_intercom.so noload = chan_modem.so noload = chan_modem_aopen.so noload = chan_modem_bestdata.so noload = chan_modem_i4l.so noload = chan_capi.so load = res_musiconhold.so noload = chan_alsa.so [global] I've so far made no changes to extensions.conf to use the ISDN card. The linux modules zaptel, xpp and zaphfc get loaded automatically, but I haven't figured out yet from where. I'm thinking the zaphfc module may need to be loaded with a few (extra?) parameters before it starts behaving itself. Any help would be most welcome. Thanks! Jaap ___ --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] Definity G3R and MWI
I have been playing with this some time ago. We used the so called mode code integration. This worked fine. It works simular as described for other Avaya Product. http://www.voip-info.org/wiki/view/Avaya+or+Lucent+Magix+Voicemail+Integration Henk BJ Weschke schreef: I've just spent the last two hours Googling and searching the Wiki. I' trying to find if there are any listings of codes for the Avaya Definity G3R, to allow for an Asterisk system to turn on/off a phones MWI that is attached to a G3. We are looking to use an Asterisk system as a voice mail server. I think you're going to need to integrate via the SMDI feature of Asterisk and figure out what the Definity needs as well to work with an SMDI connection. ___ --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] Performance Issues Degradation After 6 Calls
On 12/27/07, broadband Voice [EMAIL PROTECTED] wrote: I am using Asterisk and A2billing Calling Card Platform and after the 6th call the quality starts to degrade. The way it set up is the user calls into the system then dial out so I have 12 channels being used up but 6 active calls. Here are my specs Asterisk SVN-branch-1.4-r79142 on a i686 running Linux Fedora 6, Pentium 4 Hyper-Threading, 64 bit, 1GB of RAM, 80 GB Sata Drive, bandwidth 4 Mbps (1300GB/Throughput) burstable to 100Mbps. I am planning on upgrading to Intel Core 2 Duo with a clock speed of 1.8GHZ and 2GB Ram. Does anyone have similar situation or advice? 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] call queuing not detecting caller hang up when call originates from voip provider
Dear all
I've got call queuing working when calls originate from my local site.
After testing I migrated it to calls originating from our voip
provider- it should ring an extension, then queue . All works well
apart from if the caller hangs up when queued: the call hangs around
in the queue as a phantom until one of the extensions answers it and
it is destroyed
Am I doing something wrong? Am using asterisk 1.4.16.2
Relevant part of files:
sip.conf
[voipfone]
type=friend
secret=
username=xx
fromuser=xx
fromdomain=sip.voipfone.co.uk
host=sip.voipfone.co.uk
insecure=very
dtmfmode=rfc2833
context=fromvoipfone
[s450]
type=friend
context=phones
host=dynamic
[xlite]
type=friend
context=phones
host=dynamic
[consult]
type=friend
context=phones
host=dynamic
extensions.conf
[fromvoipfone]
exten= 1234,1,Dial(SIP/consult,3)
exten= 1234,n,Answer
exten= 1234,n,Ringing
exten= 1234,n,Wait(2)
exten= 1234,n,Background(/var/lib/asterisk/sounds/mhqw)
exten= 1234,n,Queue(myqueue|r)
exten= 1234,n,Hangup
[phones]
exten= 1001,1,Dial(SIP/s450)
exten= 1002,1,Dial(SIP/xlite)
exten= 1003,1,Dial(SIP/consult)
exten= _0.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],20,r)
exten= _ZX,1,Dial(SIP/01295${EXTEN:[EMAIL PROTECTED],20,r)
exten= _Z,1,Dial(SIP/01295${EXTEN:[EMAIL PROTECTED],20,r)
queues.conf
[myqueue]
periodic-announce = mhqw
periodic-announce-frequency = 10
music=default
strategy=ringall
timeout=15
retry=5
wrapuptime=0
maxlen=0
announce-frequency=0
announce-holdtime=no
member = SIP/consult,1
context = phones
Any help appreciated!!
John
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Re: [asterisk-users] Definity G3R and MWI
BJ Weschke wrote: I think you're going to need to integrate via the SMDI feature of Asterisk and figure out what the Definity needs as well to work with an SMDI connection. Thanks for the input. 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 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] Definity G3R and MWI
Henk Dick wrote: I have been playing with this some time ago. We used the so called mode code integration. This worked fine. It works simular as described for other Avaya Product. http://www.voip-info.org/wiki/view/Avaya+or+Lucent+Magix+Voicemail+Integration Yes, I saw the page. The Definity wouldn't accept *53 for on, and #*53 for off. For a test, I was using extension 5574, so I did a Dial(ZAP/g1/*535574) from a Asterisk console with no results on the test phone. This system is attached via a PRI. Thanks! 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 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] Performance Issues Degradation After 6 Calls
broadband Voice wrote: On 12/27/07, *broadband Voice* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I am using Asterisk and A2billing Calling Card Platform and after the 6th call the quality starts to degrade. The way it set up is the user calls into the system then dial out so I have 12 channels being used up but 6 active calls. Here are my specs Asterisk SVN-branch-1.4-r79142 on a i686 running Linux Fedora 6, Pentium 4 Hyper-Threading, 64 bit, 1GB of RAM, 80 GB Sata Drive, bandwidth 4 Mbps (1300GB/Throughput) burstable to 100Mbps. I am planning on upgrading to Intel Core 2 Duo with a clock speed of 1.8GHZ and 2GB Ram. Does anyone have similar situation or advice? Thanks. Your system should be able to handle that volume easily. What are you using for PSTN connectivity? I have heard of people having issues with Hyperthreading. That could be a problem, although I have never had any issues myself. What does top look like? When I had a similar issue (voice quality while running monitor on over seventy calls) I found a small Linux CLI app, I cannot remember the name of it but it would give IO stats (I think it may be named IOStat or something similar) and I could see right where the bottleneck was (obviously disc IO but I was able to see exactly where the breaking point was). That may help identify something. Thanks, Steve Totaro ___ --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] Definity G3R and MWI
Doug, Have you checked the feature access code that is defined in the definity. That is the code that needs to be dialed. I always checked the codes from a definity phone to make sure that I was using the right codes. Henk Doug Lytle schreef: Henk Dick wrot I have been playing with this some time ago. We used the so called mode code integration. This worked fine. It works simular as described for other Avaya Product. http://www.voip-info.org/wiki/view/Avaya+or+Lucent+Magix+Voicemail+Integration Yes, I saw the page. The Definity wouldn't accept *53 for on, and #*53 for off. For a test, I was using extension 5574, so I did a Dial(ZAP/g1/*535574) from a Asterisk console with no results on the test phone. This system is attached via a PRI. Thanks! Doug ___ --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] Definity G3R and MWI
Henk Dick wrote: Doug, Have you checked the feature access code that is defined in the definity. That is the code that needs to be dialed. I always checked the codes from a definity phone to make sure that I was using the right I have not been able to find any references to the feature codes available for the Definity G3R. The Definity manager wasn't able to locate any documentation either. 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 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] Definity G3R and MWI
You're looking for Leave Word Calling activation and deactivation. On 12/28/07, Doug Lytle [EMAIL PROTECTED] wrote: Henk Dick wrote: Doug, Have you checked the feature access code that is defined in the definity. That is the code that needs to be dialed. I always checked the codes from a definity phone to make sure that I was using the right I have not been able to find any references to the feature codes available for the Definity G3R. The Definity manager wasn't able to locate any documentation either. 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 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] IVR help, please
Hi list. I'm new to IVRs and trying to set up one that toggles an auto-forward flag on or off for specific accounts. I'd like to have my users dial an extension and then be prompted to enter the account number. (done) Next I'd like it to jump to the appropriate line in the dial plan that corresponds to the entered account number (if it is valid) and have it play back the current status based on a quick DB query (i.e. - Acct #1234 is currently 'on'). (done) Then I'd like it to prompt the user to Press 1 to turn forwarding on (or 2 for off), but this is where I get stuck. I can't seem to figure out how to do sub menus. I've Googled and checked the wiki, but I can't seem to find exactly what it is I need. Can anyone advise? Thanks in advance, Jay ___ --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] sip.conf realtime
Hi - I'm looking into realtime and I'm having a bit of a problem with the SIP part. My review of the posts seems to indicate that I should use realtime static for the [general] part of my sip.conf including the registration commands: register=did:secret@domain/did context and use realtime realtime (funny name!) for peers and friends: [myprovider] type=peer auth=md5 username=... fromuser=... fromdomain=... secret=... host=... port=5060 nat=yes canreinvite=yes qualify=no disallow=all allow=ulaw dtmfmode=rfc2833 insecure=port,invite context=incoming-sip Is this correct? What's throwing me off is this statment found here:http://www.voip-info.org/wiki/index.php?page=Asterisk%20RealTime%20Static * NOTE:* You can only store a static config OR a RealTime config. You cannot, for example, store sip.conf and use sipfriends via RealTime. This would suggest that I'll have to do a reload when I add a DiD, but a reload won't be necessary if a new SIP client is added. Do I have it right? Also, what's the difference between a peer and a user? I used to think that a user was an agent authorized to call in to my * box, a peer was an agent I could reach and a freind was both. What's throwing me off now is the statement found here:http://www.voip-info.org/wiki/index.php?page=Asterisk%20RealTime%20Static With newer versions of Asterisk the concept of SIP 'users' will be phased out. I can't understand this especially in the context of extconfig.conf that uses both a sipuser and sippeer entry. Could someone clarify for me? Thanks, H ___ --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 voicemail app (supports many interfaces, including Audix)
It's licensed GPL. I'm working on getting the web-site, documentation, and packaging up to par... if you're interested in helping, let me know. Here are some details on it: * Written for Asterisk 1.4.x; not tested with prior versions * Supports both voice and fax mail (including fax detection) * Database support build-in; can use real time as well * Web-based GUI for basic management * Professional non-Allison female prompts (English due mid-Jan 2008) * Consolidated MWI server/client comes with it (for consolidated or distributed voicemail servers) Justin - Original Message From: Matt Riddell [EMAIL PROTECTED] To: Justin Newman [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Cc: [EMAIL PROTECTED] Sent: Thursday, December 27, 2007 3:08:31 PM Subject: Re: [asterisk-users] New voicemail app (supports many interfaces, including Audix) -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Justin Newman wrote: We just completed a new implementation of voicemail for Asterisk. It's much cleaner than Comedian mail and can emulate several voicemail user interfaces, including Audix. It's a great replacement for Audix. All of the sounds/prompts are presently being re-recorded by a professional female voice. If you are interest in the app, let us know at [EMAIL PROTECTED] I'm assuming that since you sent it to Asterisk Users (Non-Commercial Discussion) it is free. Is it also Open Source? What licence? - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHdDBvDQNt8rg0Kp4RArv+AJ43NV5Rtxtx5+nuLf9kOclIOBRuwwCgnuM0 VK4Mg+svmfczGsffotPe24w= =CcGs -END PGP SIGNATURE- Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ ___ --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] Performance Issues Degradation After 6 Calls
On Thu, 27 Dec 2007, broadband Voice wrote: I am using Asterisk and A2billing Calling Card Platform and after the 6th call the quality starts to degrade. The way it set up is the user calls into the system then dial out so I have 12 channels being used up but 6 active calls. Here are my specs Asterisk SVN-branch-1.4-r79142 on a i686 running Linux Fedora 6, Pentium 4 Hyper-Threading, 64 bit, 1GB of RAM, 80 GB Sata Drive, bandwidth 4 Mbps (1300GB/Throughput) burstable to 100Mbps. I am planning on upgrading to Intel Core 2 Duo with a clock speed of 1.8GHZand 2GB Ram. Does anyone have similar situation or advice? Thanks. You don't say how people are calling in+out. If it's via the Internet, then that's where I'd start to look first. If it's via BRI/PRI interface then I'd look at interrupt issues. I'd suggest that the hardware is more than capable if it's set up correctly. But since you mention bandwidth, I'll assume the calls are coming in via that interface - each call will use 80Kb/sec each way (g711). So 5 calls (to make the math easier) is 10 channels of 80Kb/sec each way - so it's only 800Kb/sec each way. Well under your 4Mb limit. However, at 50 packets per second per call (each way), it's 500 packets per second each way - 1000 pps in total... Can your router sustain more than that? Can your ISP deliver more than that? That's where I'd start to look if it's not local hardware issues... 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
Re: [asterisk-users] Performance Issues Degradation After 6 Calls
On Fri, 28 Dec 2007, Steve Totaro wrote: broadband Voice wrote: On 12/27/07, *broadband Voice* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I am using Asterisk and A2billing Calling Card Platform and after the 6th call the quality starts to degrade. The way it set up is the user calls into the system then dial out so I have 12 channels being used up but 6 active calls. Here are my specs Asterisk SVN-branch-1.4-r79142 on a i686 running Linux Fedora 6, Pentium 4 Hyper-Threading, 64 bit, 1GB of RAM, 80 GB Sata Drive, bandwidth 4 Mbps (1300GB/Throughput) burstable to 100Mbps. I am planning on upgrading to Intel Core 2 Duo with a clock speed of 1.8GHZ and 2GB Ram. Does anyone have similar situation or advice? Thanks. Your system should be able to handle that volume easily. What are you using for PSTN connectivity? I have heard of people having issues with Hyperthreading. That could be a problem, although I have never had any issues myself. What does top look like? When I had a similar issue (voice quality while running monitor on over seventy calls) I found a small Linux CLI app, I cannot remember the name of it but it would give IO stats (I think it may be named IOStat or something similar) and I could see right where the bottleneck was (obviously disc IO but I was able to see exactly where the breaking point was). That may help identify something. Try: vmstat 1 IIRC, iostat is a *BSD type utility, but it's been many years since I touched BSD! It is possible to graph disk IO as well as network packet IO if required using (eg) MRTG. 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
Re: [asterisk-users] New voicemail vs. minivm
This system targets a different market... I like Olle's system. He did a good job. Olle's minivm is a great choice for those wishing to build customized voicemail systems, but as the name suggests, the systems are very basic. Large systems are difficult to maintain in the dial plan and some of the functionality we need would be difficult to implement with that approach. Justin - Original Message From: Tzafrir Cohen [EMAIL PROTECTED] To: Justin Newman [EMAIL PROTECTED] Sent: Friday, December 28, 2007 2:29:28 PM Subject: Re: [asterisk-users] New voicemail app (supports many interfaces, including Audix) Hi On Fri, Dec 28, 2007 at 02:19:35PM -0800, Justin Newman wrote: It's licensed GPL. I'm working on getting the web-site, documentation, and packaging up to par... if you're interested in helping, let me know. Here are some details on it: * Written for Asterisk 1.4.x; not tested with prior versions * Supports both voice and fax mail (including fax detection) * Database support build-in; can use real time as well * Web-based GUI for basic management * Professional non-Allison female prompts (English due mid-Jan 2008) * Consolidated MWI server/client comes with it (for consolidated or distributed voicemail servers) Again: did you get a chance to look at Olle's minivm? He generally broke down the voicemail functionality to separate apps that could be included in the dialplan to make a voicemail menu. IIRC it lacks support for other backends that app_voicemail currently has. But it is probably much less as messy. I wonder if it won't be a better base for extending. Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping___ --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] Building prototype devices?
I know a lot of people on this list are building devices and equipment for Asterisk and communications in general... For those of you building prototype devices, you may want to check out TechShop in the bay area. They are expanding all over the place. http://www.techshop.ws They have lasers, etchers, welders, 3d shaping machines, lathes, and a bunch of other fancy equipment for making equipment prototypes... Justin Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs___ --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] Performance Issues Degradation After 6 Calls
Gordon Henderson wrote: On Fri, 28 Dec 2007, Steve Totaro wrote: broadband Voice wrote: On 12/27/07, *broadband Voice* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I am using Asterisk and A2billing Calling Card Platform and after the 6th call the quality starts to degrade. The way it set up is the user calls into the system then dial out so I have 12 channels being used up but 6 active calls. Here are my specs Asterisk SVN-branch-1.4-r79142 on a i686 running Linux Fedora 6, Pentium 4 Hyper-Threading, 64 bit, 1GB of RAM, 80 GB Sata Drive, bandwidth 4 Mbps (1300GB/Throughput) burstable to 100Mbps. I am planning on upgrading to Intel Core 2 Duo with a clock speed of 1.8GHZ and 2GB Ram. Does anyone have similar situation or advice? Thanks. Your system should be able to handle that volume easily. What are you using for PSTN connectivity? I have heard of people having issues with Hyperthreading. That could be a problem, although I have never had any issues myself. What does top look like? When I had a similar issue (voice quality while running monitor on over seventy calls) I found a small Linux CLI app, I cannot remember the name of it but it would give IO stats (I think it may be named IOStat or something similar) and I could see right where the bottleneck was (obviously disc IO but I was able to see exactly where the breaking point was). That may help identify something. Try: vmstat 1 IIRC, iostat is a *BSD type utility, but it's been many years since I touched BSD! It is possible to graph disk IO as well as network packet IO if required using (eg) MRTG. Gordon http://www.linuxquestions.org/linux/articles/Jeremys_Magazine_Articles/Hunting_I_O_Bottlenecks_with_iostat Thanks, Steve Totaro ___ --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] Voicemail App
Red Tiger is Java based, so it will run on any Java VM (i.e., Windows, MacOS, Linux, Unix, etc.) There are some JNI-based additions for Linux which give it more capabilities, but Red Tiger itself runs cross platform. You could run Asterisk on Linux, but have Red Tiger and all of your applications running on Windows, MacOS, Linux, etc. Only the base libraries must be on the Asterisk machine... the rest can run anywhere. - Original Message From: Tammy A. Wisdom [EMAIL PROTECTED] To: Justin Newman [EMAIL PROTECTED] Sent: Friday, December 28, 2007 3:51:34 PM Subject: Re: Voicemail App What platform does red tiger run on? Thanks --Tammy Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs___ --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] Definity G3R and MWI
Tom Lynn wrote: You're looking for Leave Word Calling activation and deactivation. Thank you, I'll pass that on to him. 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 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] [OT] black dogs
Philipp Kempgen wrote: Drew Gibson wrote: A well-written application should attempt to minimize the amount of 'conversion' the user/programmer has to do. Therefore the command structure SHOULD be in a form that is natural for the user/programmer, NOT to the machine. Personally, I would vote for show dogs colour black but maybe I've spent too much time with Cisco's IOS! :-) show me all the colored dogs now. hurry up! don't spent any time in those other threads. and while you're at it would you please fix that stupid mistake i made in the config file Is this in the schedule for 1.6? regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] Not Able To tar zxvf zaptel-*.tar.gz
I figured it out. The ftp site was not named well and corrected. The other problem I have it after the extraction and make; it was suppose to go under /etc but that did not happen. I am trying to figure out why. On 12/28/07, broadband Voice [EMAIL PROTECTED] wrote: I successfully downloaded the Asterisk package from Digium but not able tar zxvf zaptel-*.tar.gz. See log below. Thanks. [EMAIL PROTECTED] src]# wget --passive-ftp http://www.digium.com/elqNow/elqRedir.htm?ref=http://downloads.digium.com/pub/asterisk/releases/asterisk-1.4.16.2.tar.gz --10:15:59-- http://www.digium.com/elqNow/elqRedir.htm?ref=http://downloads.digium.com/pub/asterisk/releases/asterisk-1.4.16.2.tar.gz = `elqRedir.htm?ref=http:%2F%2Fdownloads.digium.com%2Fpub%2Fasterisk%2Freleases%2Fasterisk- 1.4.16.2.tar.gz' Resolving www.digium.com... 216.207.245.16 Connecting to www.digium.com|216.207.245.16|:80... connected. HTTP request sent, awaiting response... 200 OKk/releases/asterisk- 1.4.16.2.tar.g Length: 2,403 (2.3K) [text/html] 100%[==] 2,403 --.--K/s 10:15:59 (278.29 MB/s) - `elqRedir.htm?ref=http:%2F%2Fdownloads.digium.com%2Fpub%2Fasterisk%2Freleases%2Fasterisk- 1.4.16.2.tar.gz' saved [2403/2403] [EMAIL PROTECTED] src]# ls -all total 20 drwxr-xr-x 2 root root 4096 2007-12-28 10:15 . drwxr-xr-x 14 root root 4096 2007-10-27 13:24 .. -rw-r--r-- 1 root root 2403 2007-10-30 00:01 elqRedir.htm?ref=http:%2F%2Fdownloads.digium.com%2Fpub%2Fasterisk%2Freleases%2Fasterisk- 1.4.16.2.tar.gz [EMAIL PROTECTED] src]# tar zxvf zaptel-*.tar.gz tar: zaptel-*.tar.gz: Cannot open: No such file or directory tar: Error is not recoverable: exiting now tar: Child returned status 2 tar: Error exit delayed from previous errors ___ --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 1.4 Fax
what method is preferred: haylafax and Iaxmodem or spnadsp for faxing. ___ --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 Fax
Al lists wrote: what method is preferred: haylafax and Iaxmodem or spnadsp for faxing. I think that you mean to say HylaFAX and IAXmodem or txfax/rxfax ... because spandsp is but a DSP/DCE library, and it cannot work alone, and iaxmodem uses spandsp. Thanks, 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] Asterisk 1.4 Fax
Al lists wrote: what method is preferred: haylafax and Iaxmodem or spnadsp for faxing. HylaFAX+ and iaxmodem (That includes SpanDSP). 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 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] Problems with zaptel and HFC-S PCI card
Hi list, Just thought I'd let you know that the problems outlined in my previous post apparently had to do with a bad card. After swapping it out for another one the messages went away. Of course, I still have some problems. For instance, there's this error that keeps appearing in my syslog and kern.log: zaphfc: empty HDLC frame or bad CRC received Any idea how to get rid of it? Thanks, Jaap == Quoting Jaap Winius [EMAIL PROTECTED]: Hi list, Now that I've got my Asterisk server to recognize my HFC-PCI card, I've run into some serious problems. The first thing I noticed was this message that would show up every five seconds on the CLI: Dec 27 15:46:42 WARNING[12484]: chan_zap.c:2512 pri_find_dchan: No D-channels available! Using Primary channel 3 as D-channel anyway! == Primary D-Channel on span 1 down Second, the syslog and the kern.log were quickly filling up with messages like these: Dec 27 16:52:53 bitis kernel: zaphfc: sync lost, cpu throtteling enabled. Dec 27 16:52:53 bitis kernel: zaphfc: sync lost, pci performance too low. you might have some cpu throtteling enabled. Dec 27 16:52:53 bitis last message repeated 31 times Dec 27 16:52:53 bitis kernel: zaphfc: bchan rx fifo not enough bytes to receive! (z1=4069, z2=4062, wanted 8 got 7), probably a buffer overrun. ___ --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 Fax
So HylaFax and IaxModem is more preferred than using rxfax/txfax ? any reason? On Dec 28, 2007 6:40 PM, Lee Howard [EMAIL PROTECTED] wrote: Al lists wrote: what method is preferred: haylafax and Iaxmodem or spnadsp for faxing. I think that you mean to say HylaFAX and IAXmodem or txfax/rxfax ... because spandsp is but a DSP/DCE library, and it cannot work alone, and iaxmodem uses spandsp. Thanks, 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 ___ --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] Problems with zaptel and HFC-S PCI card
On Sat, Dec 29, 2007 at 02:46:18AM +0100, Jaap Winius wrote: Hi list, Just thought I'd let you know that the problems outlined in my previous post apparently had to do with a bad card. After swapping it out for another one the messages went away. Of course, I still have some problems. For instance, there's this error that keeps appearing in my syslog and kern.log: zaphfc: empty HDLC frame or bad CRC received Any idea how to get rid of it? Hmm... I think that those problems should have been solved by florz's patch. Try using the zaptel packages from: deb http://updates.xorcom.com/rapid etch main (those include built zaptel-modules for default Etch kernel) -- Tzafrir Cohen icq#16849755 jabber:[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] Not Able To tar zxvf zaptel-*.tar.gz
On Fri, Dec 28, 2007 at 07:56:39PM -0500, broadband Voice wrote: I figured it out. The ftp site was not named well and corrected. The other problem I have it after the extraction and make; it was suppose to go under /etc but that did not happen. I am trying to figure out why. On 12/28/07, broadband Voice [EMAIL PROTECTED] wrote: I successfully downloaded the Asterisk package from Digium but not able tar zxvf zaptel-*.tar.gz. See log below. Thanks. [EMAIL PROTECTED] src]# wget --passive-ftp http://www.digium.com/elqNow/elqRedir.htm?ref=http://downloads.digium.com/pub/asterisk/releases/asterisk-1.4.16.2.tar.gz wget http://downloads.digium.com/pub/asterisk/releases/asterisk-1.4.16.2.tar.gz What you got is a short redirection page that uses javascript. Sadly wget is not that capable yet. -- Tzafrir Cohen icq#16849755 jabber:[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 voicemail app (supports many interfaces, including Audix)
Hi Justin, On Thu, 2007-12-27 at 15:38 -0800, Justin Newman wrote: Yes, I wrote nvfaxdetect and a number of other modules. I don't have any nvfaxdetect updates planned for public release unless someone would like to integrate some of my changes in the GPL version...we could do this though. Perhaps you could send the diff to Antonio Gallo who started the agx-ast-addons project which includes faxdetect and backgrounddetect ported to 1.4. He seems open to enhancements/additions. His email is agx at users.sourceforge.net The project can be found at: http://sourceforge.net/projects/agx-ast-addons http://agx-ast-addons.svn.sourceforge.net/viewvc/agx-ast-addons/trunk/ Regards, Patrick - Original Message From: Matt Riddell [EMAIL PROTECTED] Justin Newman wrote: We just completed a new implementation of voicemail for Asterisk. It's much cleaner than Comedian mail and can emulate several voicemail user interfaces, including Audix. It's a great replacement for Audix. All of the sounds/prompts are presently being re-recorded by a professional female voice. Also, are you the guy who wrote nvfaxdetect et al? Any chance of an update for 1.4 etc? __ Looking for last minute shopping deals? Find them fast with Yahoo! Search. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] Performance Issues Degradation After 6 Calls
On Fri, 28 Dec 2007, Steve Totaro wrote: Gordon Henderson wrote: On Fri, 28 Dec 2007, Steve Totaro wrote: broadband Voice wrote: On 12/27/07, *broadband Voice* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I am using Asterisk and A2billing Calling Card Platform and after the 6th call the quality starts to degrade. The way it set up is the user calls into the system then dial out so I have 12 channels being used up but 6 active calls. Here are my specs Asterisk SVN-branch-1.4-r79142 on a i686 running Linux Fedora 6, Pentium 4 Hyper-Threading, 64 bit, 1GB of RAM, 80 GB Sata Drive, bandwidth 4 Mbps (1300GB/Throughput) burstable to 100Mbps. I am planning on upgrading to Intel Core 2 Duo with a clock speed of 1.8GHZ and 2GB Ram. Does anyone have similar situation or advice? Thanks. Your system should be able to handle that volume easily. What are you using for PSTN connectivity? I have heard of people having issues with Hyperthreading. That could be a problem, although I have never had any issues myself. What does top look like? When I had a similar issue (voice quality while running monitor on over seventy calls) I found a small Linux CLI app, I cannot remember the name of it but it would give IO stats (I think it may be named IOStat or something similar) and I could see right where the bottleneck was (obviously disc IO but I was able to see exactly where the breaking point was). That may help identify something. Try: vmstat 1 IIRC, iostat is a *BSD type utility, but it's been many years since I touched BSD! It is possible to graph disk IO as well as network packet IO if required using (eg) MRTG. Gordon http://www.linuxquestions.org/linux/articles/Jeremys_Magazine_Articles/Hunting_I_O_Bottlenecks_with_iostat Ah, intersting, so I was about to suggest it might be a distro thing, but digging deeper, I find there is an iostat for Debian - under the generic package sysstat which is why I've never found it in the past. The iostat I remember for BSD had a screen/curses interface, but scrolling might help you see trends. Cheers, 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] Digium Asterisk Appliance voicemail logs
Does anyone know how much space the appliance has for voicemail and/or logs? Doesn't have an embedded disk from what I can see, and only a 1G flash card? -- Barry D. Hassler President, HCST http://www.hcst.net/ 937-427-9000 ___ --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
