Re: [asterisk-users] Zapata Tormenta 2
On Wed, Mar 26, 2008 at 11:00:21PM -0500, Cody Jarrett wrote: I have an old zapata tormenta 2 quad port pci card. I'd like to get it working and play with it but was curious to see if that was possible. Does anyone know if it will work for 2.6 kernels, or where I can find decent drivers? I tried getting the tor driver from http://www.zapatatelephony.org/linux-zapata-current/ installed, but didn't get very far. Any input is appreciated. Use the tor2 driver included with the main Zaptel dirstrubution. -- 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
[asterisk-users] Asterisk not hanging up after voicemail
Hi, I am having problem with my Asterix server. It does not hand up after play the voicemail. The scenario is this: 1. I make a call to Asterisk's PSTN number; 2. After recording, I hang up and make the same call again. The first call would go through nicely with the voicemail recording, but the second call will hit a message saying the other party is busy. The only way to fix it is to reboot the Asterisk server again. Here is the CLI for the first call: -- Starting simple switch on 'Zap/1-1' -- Executing [EMAIL PROTECTED]:1] Answer(Zap/1-1, ) in new stack -- Executing [EMAIL PROTECTED]:2] NoOp(Zap/1-1, incoming number is 8755048) in new stack -- Executing [EMAIL PROTECTED]:3] Wait(Zap/1-1, 2) in new stack -- Executing [EMAIL PROTECTED]:4] NoOp(Zap/1-1, this is a voice call| not fax) in new stack -- Executing [EMAIL PROTECTED]:5] VoiceMail(Zap/1-1, 2000) in new stack -- Zap/1-1 Playing 'vm-intro' (language 'en') -- Zap/1-1 Playing 'beep' (language 'en') -- Recording the message -- x=0, open writing: /var/spool/asterisk/voicemail/default/2000/tmp/ZyFCyg format: wav, 0x823b2f0 What could be wrong? Thanks, Mark ___ -- 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] ADPCM codec and IAXy device
Hi All; I need to buy one IAXy device, but I discovered that it supports only g711 and ADPCM codec, so I was wonder that it does not support g729 or GSM?! Anyway, what is that ADPCM and how much it consumes bandwitdh? Also, asterisk support such codec? What its name in the configuration? Any advise? Regards Bilal 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
[asterisk-users] Calling users to the external domain using Asterisk
Hi All, I am a newbie to Asterisk. Presently I am working with Asterisk 1.4.17 and using it to make SIP calls. I have a configuration of Asterisk which serves the users in a particular domain, say internal.com I would like to make a SIP call from [EMAIL PROTECTED] to [EMAIL PROTECTED] I have added the following lines in extensions.conf exten = charles,1,Dial(SIP/[EMAIL PROTECTED]) exten = charles,2,Hangup Asterisk does a DNS SRV lookup and resolves the external.com to its proper IP and calls are established. But the problem with the above configuration is that I have manually added users that are in the external domain. Is there any way wherein I can call the users in external.com without adding them in the extensions.conf? Any help would be appreciated. Thanks, Aadil ___ -- 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] SPA400 vs Rhino/Digium card
Hi! I'm a new member in VoIP world. I want to implement a VoIP PBX using asterisk/tribox in the office but I have one doubt. Which is the best way to use actual PSTN lines in the VoIP PBX? Using a box as Linksys SPA400 or installing a PCI card (Rhino/Digium...) into the server? Which benefits/problems will I have with each option? Regards ___ -- 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] IAXy device
Hi All; I have been chocked just when I saw some posts talking about how much the IAXy is bad :) - So I would like to ask, did any one try it later and wether it is good or not? I am asking this because I need to use it as it is NAT Transparent (as I read also, and I did not try it to see how much it is transparent). What about codec? Why it is only support g711 and does not support compressed codec? And what about the IP address and the DNS usage and the DDNS usage? What main porblems contain and any advise? Regards Bilal 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
Re: [asterisk-users] SPA400 vs Rhino/Digium card
On Thu, Mar 27, 2008 at 3:58 AM, Juan Antonio Ibañez Santorum [EMAIL PROTECTED] wrote: Hi! I'm a new member in VoIP world. I want to implement a VoIP PBX using asterisk/tribox in the office but I have one doubt. Which is the best way to use actual PSTN lines in the VoIP PBX? Using a box as Linksys SPA400 or installing a PCI card (Rhino/Digium...) into the server? Which benefits/problems will I have with each option? Older Digium cards had issues with IRQ's where Sangoma did not. If this is just a small installation. Go for onboard EC purchase more ports than you need (provides further upgrade path down the road. I have tested both the Digium and Sangoma four port PCI cards and without looking at the card itself, they both have high quality products. This is coming from a guy that was anti-Digium hardware since day one. By giving Digium a second try on all of their TDM products, I have found that they have done a great job in improving their product line. 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] Asterisk not hanging up after voicemail
Most likely, you don't have any hangup detection available or configured. If these are analogue lines, you will almost certainly need to configure busy detection in order to figure out that the call has been terminated. Do some Googling for asterisk busy detection mark morreny wrote: Hi, I am having problem with my Asterix server. It does not hand up after play the voicemail. The scenario is this: 1. I make a call to Asterisk's PSTN number; 2. After recording, I hang up and make the same call again. The first call would go through nicely with the voicemail recording, but the second call will hit a message saying the other party is busy. The only way to fix it is to reboot the Asterisk server again. Here is the CLI for the first call: -- Starting simple switch on 'Zap/1-1' -- Executing [EMAIL PROTECTED]:1] Answer(Zap/1-1, ) in new stack -- Executing [EMAIL PROTECTED]:2] NoOp(Zap/1-1, incoming number is 8755048) in new stack -- Executing [EMAIL PROTECTED]:3] Wait(Zap/1-1, 2) in new stack -- Executing [EMAIL PROTECTED]:4] NoOp(Zap/1-1, this is a voice call| not fax) in new stack -- Executing [EMAIL PROTECTED]:5] VoiceMail(Zap/1-1, 2000) in new stack -- Zap/1-1 Playing 'vm-intro' (language 'en') -- Zap/1-1 Playing 'beep' (language 'en') -- Recording the message -- x=0, open writing: /var/spool/asterisk/voicemail/default/2000/tmp/ZyFCyg format: wav, 0x823b2f0 What could be wrong? Thanks, Mark ___ -- 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] SPA400 vs Rhino/Digium card
some better faxes handling lie 0 delay assured, important with hylafax for example 2008/3/27, Juan Antonio Ibañez Santorum [EMAIL PROTECTED]: but which would be the best option to include 4 PSTN lines into a VoIP enviroment? As I can see that SPA400 (4 FXO Ports) is a cheaper option than a 4 ports PCI card (Rhino/Digium/Sangoma...). Would I get any benefit on using a PCI card instead of a box as SPA400? Regards ___ -- 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] SPA400 vs Rhino/Digium card
but which would be the best option to include 4 PSTN lines into a VoIP enviroment? As I can see that SPA400 (4 FXO Ports) is a cheaper option than a 4 ports PCI card (Rhino/Digium/Sangoma...). Would I get any benefit on using a PCI card instead of a box as SPA400? Regards ___ -- 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] SPA400 vs Rhino/Digium card
It depends on whether you want to deal with Zaptel or SIP/IAX trunks. My personal choice would be the TDM400P but thats purely because I know it. Barring any problems with IRQ sharing/conflicts, setup is fast and easy, and it just makes sense to me to have all PBX-related hardware in one box But the choice, ultimately, would depend on how you plan expansion, if any. _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Juan Antonio Ibañez Santorum Sent: 27 March 2008 10:58 AM To: Lista Usuarios Asterisk Subject: [asterisk-users] SPA400 vs Rhino/Digium card but which would be the best option to include 4 PSTN lines into a VoIP enviroment? As I can see that SPA400 (4 FXO Ports) is a cheaper option than a 4 ports PCI card (Rhino/Digium/Sangoma...). Would I get any benefit on using a PCI card instead of a box as SPA400? Regards ___ -- 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] Upgraded to 1.4.18 (from 1.2.27) and channels dropping on Zaptel and SIP
--- Darrick Hartman (lists) [EMAIL PROTECTED] wrote: Do yourself a favor and upgrade a Asterisk 1.4 which has a proper implementation of DTMF. It's likely your SIP provider upgraded to something which does not recognize the DTMF tones from Asterisk 1.2. I've upgraded to 1.4.18 (along with zaptel 1.4.9.2) and still experiencing the same problem (not recognizing DTMF on SIP inbound calls) as well as new problems. The new problems are much more severe than the previous problems so I'm starting a new thread with a more descriptive subject. I've changed sip.conf to eliminate warnings for new syntax: insecure=port,invite dtmfmode=rfc2833; Choices are inband, rfc2833, or info Everything else is as-was in sip.conf, extensions.conf, iax.conf, rtp.conf, voicemail.conf, zapata.conf, zaptel.conf (although I looked through the new samples and didn't see anything glaring I needed to change). For the config files I had not changed I took the new sample files. Now in addition to not recognizing DTMF on SIP still, asterisk is now frequently dropping calls when I start to enter DTMF. On console I get lines such as: -- Executing [EMAIL PROTECTED]:1] Goto(SIP/x-081ea720, incoming|s|1) in new stack -- Goto (incoming,s,1) -- Executing [EMAIL PROTECTED]:1] Answer(SIP/x-081ea720, ) in new stack -- Executing [EMAIL PROTECTED]:2] BackGround(SIP/x-081ea720, /home/dnedved/hello) in new stack -- SIP/x-081ea720 Playing '/home/dnedved/hello' (language 'en') == Auto fallthrough, channel 'SIP/x-081ea720' status is 'UNKNOWN' It's also happening on zaptel channels (although not nearly so frequently), so it's not a SIP only problem: [Mar 27 10:42:07] NOTICE[6167]: chan_zap.c:6376 ss_thread: Got event 18 (Ring Begin)... [Mar 27 10:42:08] NOTICE[6167]: chan_zap.c:6376 ss_thread: Got event 2 (Ring/Answered)... [Mar 27 10:42:12] NOTICE[6167]: chan_zap.c:6376 ss_thread: Got event 18 (Ring Begin)... -- Executing [EMAIL PROTECTED]:1] Goto(Zap/4-1, incoming|s|1) in new stack -- Goto (incoming,s,1) -- Executing [EMAIL PROTECTED]:1] Answer(Zap/4-1, ) in new stack -- Executing [EMAIL PROTECTED]:2] BackGround(Zap/4-1, /home/dnedved/hello) in new stack -- Zap/4-1 Playing '/home/dnedved/hello' (language 'en') == Auto fallthrough, channel 'Zap/4-1' status is 'UNKNOWN' -- Hungup 'Zap/4-1' I don't know much about asterisk debugging since it has worked so flawlessly so far, but I would guess that the Auto fallthrough with status UNKNOWN means that the application that was running died, didn't set any return code, so asterisk dropped the call? I'm running in console mode with 5 v's of verbose mode, how do I find more information about why it's dropping these calls? Thanks and best regards, David [EMAIL PROTECTED] 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
[asterisk-users] Problem with socket_process: Call rejected by 127.0.0.1: Busy
Hi I am not sure why this is happening or whether it has anything to do with my iaxmodem setup. When receiving a fax via iaxmodem, I got an error message saying *chan_iax2.c:7542 socket_process: Call rejected by 127.0.0.1: Busy* From faxstat -s, I get: JID Pri S Owner Number Pages Dials TTS Status 58 123 S root 008675533661 0:2 4:12 02:12 No carrier detected Here is the asterisk output: [Mar 28 01:54:00] NOTICE[16753]: chan_iax2.c:6025 update_registry: Restricting registration for peer 'iaxmodem' to 60 seconds (requested 50) -- ast_get_srv: SRV lookup for '_sip._udp.voipuser.org' mapped to host sip.voipuser.org, port 5060 -- Accepting AUTHENTICATED call from 127.0.0.1: requested format = alaw, requested prefs = (), actual format = alaw, host prefs = (alaw), priority = mine -- Executing [EMAIL PROTECTED]:1] NoOp(IAX2/iaxmodem-1, we are at fax-out) in new stack -- Executing [EMAIL PROTECTED]:2] Dial(IAX2/iaxmodem-1, SIP/callwithus/0033661681) in new stack -- Called callwithus/008675533661681 -- Starting simple switch on 'Zap/1-1' -- SIP/callwithus-082370a8 is making progress passing it to IAX2/iaxmodem-1 [Mar 28 01:54:56] NOTICE[16754]: chan_iax2.c:6025 update_registry: Restricting registration for peer 'iaxmodem' to 60 seconds (requested 50) -- Executing [EMAIL PROTECTED]:1] Answer(Zap/1-1, ) in new stack -- Executing [EMAIL PROTECTED]:2] NoOp(Zap/1-1, incoming number is ) in new stack -- Executing [EMAIL PROTECTED]:3] Wait(Zap/1-1, 10) in new stack -- SIP/callwithus-082370a8 answered IAX2/iaxmodem-1 -- Redirecting Zap/1-1 to fax extension == Spawn extension (incoming, fax, 0) exited non-zero on 'Zap/1-1' -- Executing [EMAIL PROTECTED]:1] Answer(Zap/1-1, ) in new stack -- Executing [EMAIL PROTECTED]:2] NoOp(Zap/1-1, getting fax) in new stack -- Executing [EMAIL PROTECTED]:3] Goto(Zap/1-1, fax|s|1) in new stack -- Goto (fax,s,1) -- Executing [EMAIL PROTECTED]:1] Answer(Zap/1-1, ) in new stack -- Executing [EMAIL PROTECTED]:2] NoOp(Zap/1-1, we are at fax) in new stack -- Executing [EMAIL PROTECTED]:3] Dial(Zap/1-1, IAX2/iaxmodem) in new stack -- Called iaxmodem* [Mar 28 01:54:59] WARNING[16753]: chan_iax2.c:7542 socket_process: Call rejected by 127.0.0.1: Busy -- Hungup 'IAX2/iaxmodem-4' == Everyone is busy/congested at this time (1:0/0/1) == Auto fallthrough, channel 'Zap/1-1' status is 'CHANUNAVAIL' -- Hungup 'Zap/1-1' == Spawn extension (fax-out, 0033661681, 2) exited non-zero on 'IAX2/iaxmodem-1' -- Hungup 'IAX2/iaxmodem-1' * Please help me. Really appreciate it. Thanks, Mark ___ -- 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] Had it with Dell Garbage - HP Question
I used to have hundred big HP 9000 boxes running HP-UX 11.0. Having some open th entrailsls of those big boys and due surgery was a damn good feeling. The also came with very very good HW diagnostics and had some place you cold send KERNEL Dumps on troubled system that was often a life saver shadowym wrote: You don't have to build Supermicro stuff yourself if you don't want to. Most Supermicro dealers do it for you if you buy all the parts from them. It's true that what your doing with Dell/HP is paying for emotional support. When it comes to PBX's you not getting any value paying for Dell/HP support. If your getting good stuff you should never need their hardware replacement warranty either. -Original Message- From: Jesse Molina [mailto:[EMAIL PROTECTED] Sent: Tuesday, March 25, 2008 9:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Had it with Dell Garbage - HP Question If you can't troubleshoot a hardware problem, then you should definitely not be thinking about this. Going with a support-yourself plan is not for everyone, especially if you don't have good hands-on hardware ability local to where the systems are. The cost savings can be significant enough that you don't' need to worry about third party support. Just buy some cold-spare systems. Out of eight servers, you can buy three spares and still have money left over -- assuming a three year life span for the systems and related support costs. This is true on 1U and 2U systems -- I'm not sure about larger stuff. After all, what are Dell/IBM/HP-Compaq going to do for you, other than replace the hardware? Nothing. They don't support your custom software and configurations, just hardware. Yank the hard drives, RAID controller, install in a spare system, and you're up and running again. Figure out what went wrong on the old system later. If it's under warranty, get it RMAed at your leisure. If it's not, you've got another two spare systems on the shelf, waiting there 24x7 just for you. Once again, it's not for everyone. If you don't feel comfortable with it, don't do it! It works for some businesses, not for others. It depends on who is supporting your servers. If IBM supports your servers, get IBM support. If you support your servers... then why are you paying them to do nothing??? Don't pay for emotional support. On Tue, Mar 25, 2008 at 06:55:10PM -0400, Al Baker wrote: ok - but, who do you call for HW problem ? HP has all levels of warranties all depending on how much $$ you want to spend. What do you do if you buy and install Supermicro ? HP also has 24x7 support center , again not for free, what do you do with Supermicro ??? I am really interested because I hear a lot of folks putting * on them but I never have worked on them while I have put in bunch of HPs really big boxes.. Thx for sharing your experience Matthew Gibson wrote: I've had good luck with these guys: http://rackmountsetc.com/ supermicro have never failed me yet. On Tue, Mar 25, 2008 at 6:03 PM, Jesse Molina [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: If you want barebones where you add your own processor, RAM, hard drives, and options, try SuperMicro brand servers. They are thousands of dollars less than the big (fat) names like IBM and HP/Compaq, but very good quality. I've built several clusters of computers with SuperMicro systems. They are great if you want to do barebones, clusters, or other special projects. It just takes a little more time to do the assembly work. Try newegg.com http://newegg.com for some sample pricing. ___ -- 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] Question about PCI Slots for DIGIUMs Boards
I had sent this to Digium Sales and cant get a response from them, I don't. know what that means.. So I thought I would ask it hear since i know others have struggled with this. I an considering using *your High Density T*1 cards on a number of servers we are considering purchasing. The vendor lists that his system has: PCI Express*: two x8 slots*, t*wo x8 low profile slots*; *PCI-X: 64-bit/100MHz* Could you please clarify *WHICH* of the above listed *PCI slots* are suitable for use with your *High Density T1 cards*. If there is *more than one combination that is possible*, could you please indicate *WHICH *which high density T1 cards will work in *WHICH* type of PCI slot. Thanks for any guidance and help since DIGIUM sales seems to be too busy to advise a potential customer ___ -- 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] what's a softphone can activer web browser
can anyone help me. I'm finding the softphone which can trigger web browser and use callerid to go web page You don't say on what OS you need it to run. Mine is for Windows and support receiving URL (ex.: Dial(IAX2/7003|20|trw|http://asterisk.org) You can get it here : http://www.marccharbonneau.com/asterisk/mediaxphone.php Let me know what you think of it. hth ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco 7971
Anyone have some up-to-date (within the past 3 months) on Asterisk and the 7971. Searched voip-info, Google, etc., etc., to no avail. Documentation I found was scattered, vague. Thanks in advance. -- J. Oquendo SGFA #579 (FW+VPN v4.1) SGFE #574 (FW+VPN v4.1) wget -qO - www.infiltrated.net/sig|perl http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x3AC173DB smime.p7s Description: S/MIME Cryptographic Signature ___ -- 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] Got SIP response 406 Not Acceptable
try doing a sip debug peer XXX (the problematic extension) and then send a call to him till fail, then see the log, or send a piece here. On Thu, Mar 27, 2008 at 3:10 AM, Al lists [EMAIL PROTECTED] wrote: Nope, Coded is Ulaw on both sides and also this issue happens occasionally with no change. On Wed, Mar 26, 2008 at 6:17 PM, Adrià Vidal [EMAIL PROTECTED] wrote: Seems a codec problem, check the sip.conf from that spa942 On Wed, Mar 26, 2008 at 11:59 PM, Al lists [EMAIL PROTECTED] wrote: I'm getting Got SIP response 406 Not Acceptable back from 10.0.1.2 occasionally when try to dial to SPA942 , anyone has any idea on this before i consider Firmware upgrade? ___ -- 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 -- -- Adrià Vidal [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- 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 -- -- Adrià Vidal [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Had it with Dell Garbage - HP Question
How do you get notifications ? Is this thru one of the add on packages HP sells for the box ? Which One ? Could you be more specific about what you mean by a recovery CD and hod do you get console access below multi used to do recovery ?? What is integrated ILO BIOS Access sounds cool. What O/S you usin and what made you pick it ? What kind and how many RAIDS are you using. The HP site gave like 8 different RAID controllers and like 20 CPUs to chose from. How did you chose ? Thx for sharing !!! Darren Wright wrote: One of the major reasons we use DL320 / DL380's is the ease of swapping drives, and the integrated ILO BIOS level access.We can support remote sites with ease. If a drive dies we get a notification, a new one is sent and a non-techie can replace it with guidance.No onsite visit. That is worth potentially thousands of dollars. We also leave a recovery CD there that can be inserted if we need to rebuild the system remotely. Never had to, but it's worked in the lab. -D This message was sent from D2 Technology, INC. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Astlinux Friday Mar 28 @12 Noon EDT VoIP Users Conference
Friday March 28th;: Astlinux How to be a part of the conference via PSTN or SIP : http://www.VoipUsersConference.org Astlinux is a custom Linux distribution that is centered around the goal of providing Asterisk. Astlinux is highly optimized for Asterisk both in commercial and embedded systems. They will be releasing a new version shortly and Darrick Hartman has graciously offered to be a guest on the conference to answer any questions that the community may have. More information can be found at: http://www.astlinux.org/ http://voipusersconference.org/ning - community site where you can join and post your own blog, videos, images, whatever especially if it's about voip and or asterisk voice over asterisk is a registered trademark of digium. Hi Bill! ___ -- 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] Unable to establish handshaking with fax machine
Hi, I am simulating the sending of fax using sendfax through voip to reach an Asteria server via ZAP/1 ( PSTN phone line ) which then route call to a fax machine at ZAP/2. It seems like I am not able to establish any handshake with the physical fax machine using the sendfax program. Does anyone know why that happens and how to fix it? The scenario will be deployed in remote location in the future, but I am just running a single machine test right now. -- Accepting AUTHENTICATED call from 127.0.0.1: requested format = alaw, requested prefs = (), actual format = alaw, host prefs = (alaw), priority = mine -- Executing [EMAIL PROTECTED]:1] NoOp(IAX2/iaxmodem-1, we are at fax-out) in new stack -- Executing [EMAIL PROTECTED]:2] Dial(IAX2/iaxmodem-1, SIP/voipuser/0033661681) in new stack -- Called voipuser/008675533661681 -- SIP/voipuser-081f99c0 is making progress passing it to IAX2/iaxmodem-1 [Mar 28 04:01:00] NOTICE[16748]: chan_iax2.c:6025 update_registry: Restricting registration for peer 'iaxmodem' to 60 seconds (requested 50) -- Starting simple switch on 'Zap/1-1' -- Executing [EMAIL PROTECTED]:1] Answer(Zap/1-1, ) in new stack -- Executing [EMAIL PROTECTED]:2] NoOp(Zap/1-1, incoming number is ) in new stack -- Executing [EMAIL PROTECTED]:3] Wait(Zap/1-1, 10) in new stack -- SIP/voipuser-081f99c0 answered IAX2/iaxmodem-1 -- Redirecting Zap/1-1 to fax extension == Spawn extension (incoming, fax, 0) exited non-zero on 'Zap/1-1' -- Executing [EMAIL PROTECTED]:1] Answer(Zap/1-1, ) in new stack -- Executing [EMAIL PROTECTED]:2] NoOp(Zap/1-1, getting fax) in new stack -- Executing [EMAIL PROTECTED]:3] Goto(Zap/1-1, fax|s|1) in new stack -- Goto (fax,s,1) -- Executing [EMAIL PROTECTED]:1] Answer(Zap/1-1, ) in new stack -- Executing [EMAIL PROTECTED]:2] NoOp(Zap/1-1, we are at fax) in new stack -- Executing [EMAIL PROTECTED]:3] Dial(Zap/1-1, ZAP/2) in new stack -- Called 2 -- Zap/2-1 is ringing -- Zap/2-1 is ringing -- Zap/2-1 answered Zap/1-1 -- Native bridging Zap/1-1 and Zap/2-1 -- ast_get_srv: SRV lookup for '_sip._udp.voipuser.org' mapped to host sip.voipuser.org, port 5060 Thanks for helping out. I really appreciate it. Thanks, Mark ___ -- 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] Upgraded to 1.4.18 (from 1.2.27) and channels dropping on Zaptel and SIP
David Nedved wrote: --- Darrick Hartman (lists) [EMAIL PROTECTED] wrote: Do yourself a favor and upgrade a Asterisk 1.4 which has a proper implementation of DTMF. It's likely your SIP provider upgraded to something which does not recognize the DTMF tones from Asterisk 1.2. I've upgraded to 1.4.18 (along with zaptel 1.4.9.2) and still experiencing the same problem (not recognizing DTMF on SIP inbound calls) as well as new problems. The new problems are much more severe than the previous problems so I'm starting a new thread with a more descriptive subject. I've changed sip.conf to eliminate warnings for new syntax: insecure=port,invite dtmfmode=rfc2833; Choices are inband, rfc2833, or info Everything else is as-was in sip.conf, extensions.conf, iax.conf, rtp.conf, voicemail.conf, zapata.conf, zaptel.conf (although I looked through the new samples and didn't see anything glaring I needed to change). For the config files I had not changed I took the new sample files. There were several things that changed... Now in addition to not recognizing DTMF on SIP still, asterisk is now frequently dropping calls when I start to enter DTMF. On console I get lines such as: -- Executing [EMAIL PROTECTED]:1] Goto(SIP/x-081ea720, incoming|s|1) in new stack -- Goto (incoming,s,1) -- Executing [EMAIL PROTECTED]:1] Answer(SIP/x-081ea720, ) in new stack -- Executing [EMAIL PROTECTED]:2] BackGround(SIP/x-081ea720, /home/dnedved/hello) in new stack -- SIP/x-081ea720 Playing '/home/dnedved/hello' (language 'en') == Auto fallthrough, channel 'SIP/x-081ea720' status is 'UNKNOWN' Try adding this line in the general section of extensions.conf autofallthrough=no The default behavior in 1.2 was no. In 1.4 it changed to yes. That will be your simplest fix (without seeing your dialplan). Asterisk is moving on to the next step in the dialplan before you enter your digits. You need to have it wait for the digits to be entered. Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com http://www.djhsolutions.com/wiki ___ -- 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] Question about PCI Slots for DIGIUMs Boards
On Thu, Mar 27, 2008 at 6:48 AM, Al Baker [EMAIL PROTECTED] wrote: I had sent this to Digium Sales and cant get a response from them, I don't. know what that means.. So I thought I would ask it hear since i know others have struggled with this. I an considering using *your High Density T*1 cards on a number of servers we are considering purchasing. The vendor lists that his system has: PCI Express*: two x8 slots*, t*wo x8 low profile slots*; *PCI-X: 64-bit/100MHz* Could you please clarify *WHICH* of the above listed *PCI slots* are suitable for use with your *High Density T1 cards*. If there is *more than one combination that is possible*, could you please indicate *WHICH *which high density T1 cards will work in *WHICH* type of PCI slot. Thanks for any guidance and help since DIGIUM sales seems to be too busy to advise a potential customer You never qualify High Density. Maybe Digium thought you might get tired of waiting and find your own answer http://en.wikipedia.org/wiki/Peripheral_Component_Interconnect Besides that, look here http://store.digium.com/products.php?category_id=2 You should be able to figure it out on your own with a little thought. 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] Upgraded to 1.4.18 (from 1.2.27) and channels dropping on Zaptel and SIP
On Thu, Mar 27, 2008 at 8:16 AM, Darrick Hartman (lists) [EMAIL PROTECTED] wrote: David Nedved wrote: --- Darrick Hartman (lists) [EMAIL PROTECTED] wrote: Do yourself a favor and upgrade a Asterisk 1.4 which has a proper implementation of DTMF. It's likely your SIP provider upgraded to something which does not recognize the DTMF tones from Asterisk 1.2. I've upgraded to 1.4.18 (along with zaptel 1.4.9.2) and still experiencing the same problem (not recognizing DTMF on SIP inbound calls) as well as new problems. The new problems are much more severe than the previous problems so I'm starting a new thread with a more descriptive subject. I've changed sip.conf to eliminate warnings for new syntax: insecure=port,invite dtmfmode=rfc2833; Choices are inband, rfc2833, or info Everything else is as-was in sip.conf, extensions.conf, iax.conf, rtp.conf, voicemail.conf, zapata.conf, zaptel.conf (although I looked through the new samples and didn't see anything glaring I needed to change). For the config files I had not changed I took the new sample files. There were several things that changed... Now in addition to not recognizing DTMF on SIP still, asterisk is now frequently dropping calls when I start to enter DTMF. On console I get lines such as: -- Executing [EMAIL PROTECTED]:1] Goto(SIP/x-081ea720, incoming|s|1) in new stack -- Goto (incoming,s,1) -- Executing [EMAIL PROTECTED]:1] Answer(SIP/x-081ea720, ) in new stack -- Executing [EMAIL PROTECTED]:2] BackGround(SIP/x-081ea720, /home/dnedved/hello) in new stack -- SIP/x-081ea720 Playing '/home/dnedved/hello' (language 'en') == Auto fallthrough, channel 'SIP/x-081ea720' status is 'UNKNOWN' Try adding this line in the general section of extensions.conf autofallthrough=no The default behavior in 1.2 was no. In 1.4 it changed to yes. That will be your simplest fix (without seeing your dialplan). Asterisk is moving on to the next step in the dialplan before you enter your digits. You need to have it wait for the digits to be entered. Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com http://www.djhsolutions.com/wiki If the DTMF issue works better in 1.2.X and you do not need the additional features of 1.4.X then you made the right choice going back to 1.2.X. People on the list (mainly dev) want you to test, find bugs, jump through hoops, and post to Mantis (where you bug might just be closed, or a general feeling of You are wrong. All of this testing is free of course due to the Benefit of the Community. In the real world, it would serve you better to do what works best for your business. Don't let the Dev guys push you around, do what makes sense to your business. 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] Had it with Dell Garbage - HP Question
Notifications can be done either thru SNMP traps or SMTP. Insight Manager is free from HP, but any SNMP trapper can work with alerts. The recovery CD is just a build that reloads the majority of the system with a static ip. We backup off site to one of our servers via FTP. ILO access is an integrated IP KVM. So you can see the machine boot, get virtual media access, etc. O/S is CentOS. For smaller systems, RAID 1, and for larger DL380 based systems 0+1 -D -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Al Baker Sent: Thursday, March 27, 2008 8:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Had it with Dell Garbage - HP Question How do you get notifications ? Is this thru one of the add on packages HP sells for the box ? Which One ? Could you be more specific about what you mean by a recovery CD and hod do you get console access below multi used to do recovery ?? What is integrated ILO BIOS Access sounds cool. What O/S you usin and what made you pick it ? What kind and how many RAIDS are you using. The HP site gave like 8 different RAID controllers and like 20 CPUs to chose from. How did you chose ? Thx for sharing !!! Darren Wright wrote: One of the major reasons we use DL320 / DL380's is the ease of swapping drives, and the integrated ILO BIOS level access.We can support remote sites with ease. If a drive dies we get a notification, a new one is sent and a non- techie can replace it with guidance.No onsite visit. That is worth potentially thousands of dollars. We also leave a recovery CD there that can be inserted if we need to rebuild the system remotely. Never had to, but it's worked in the lab. -D This message was sent from D2 Technology, INC. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This message was sent from D2 Technology, INC. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question about PCI Slots for DIGIUMs Boards
On Thu, 2008-03-27 at 06:48 -0400, Al Baker wrote: I an considering using *your High Density T*1 cards on a number of servers we are considering purchasing. Two-port T1 cards? Four-port T1 cards? Could you please clarify *WHICH* of the above listed *PCI slots* are suitable for use with your *High Density T1 cards*. If there is *more than one combination that is possible*, could you please indicate *WHICH *which high density T1 cards will work in *WHICH* type of PCI slot. PCI or PCI-X 3.3 Volt Slots --- TE412P (Quad span T1/E1/J1/PRI with DSP-based Echo Cancellation) TE212P (Dual span T1/E1/J1/PRI with DSP-based Echo Cancellation) PCI 5.0 Volt Slots --- TE407P (Quad span T1/E1/J1/PRI with DSP-based Echo Cancellation) TE207P (Dual span T1/E1/J1/PRI with DSP-based Echo Cancellation) PCI-E (PCI Express) slots - TE420B (Quad Span T1/E1/J1/PRI with DSP-based Echo Cancellation) TE220B (Dual span T1/E1/J1/PRI with DSP-based Echo Cancellation) There are also a variety of models available without the echo cancellation modules onboard, but personally I recommend purchasing the echo cancellation modules in most situations. -- Jared Smith Community Relations Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAXy device
responses inline bilal ghayyad wrote: So I would like to ask, did any one try it later and wether it is good or not? I am asking this because I need to use it as it is NAT Transparent (as I read also, and I did not try it to see how much it is transparent). Thousands and thousands and thousands of people use SIP with NAT. What about codec? Why it is only support g711 and does not support compressed codec? And what about the IP address and the DNS usage and the DDNS usage? The IAXy does not support highly compressed codecs, DNS or DDNS. -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- 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] AGI-python script
I was trying to trap SIGHUP, but could be another signal because it didn't work. I'm doing this class MyScript(): def logsignal(self,signum, frame): self.putCDR() def run(self): signal.signal(signal.SIGHUP, self.logsignal) def putCDR(): put my cdr in my db. I was tryin trap other signals to test this and work well def run(self): signal.signal(signal.SIGALRM, self.logsignal) signal.alarm(3) Thanks a lot! On Wed, Mar 26, 2008 at 4:54 PM, Steve Edwards [EMAIL PROTECTED] wrote: On Wed, 26 Mar 2008, equis software wrote: Hi! I have some IVRs made in python. If the caller hangup before the end of the script I can´t register in my database the cdr. From your description, I'm not sure exactly what you are asking, but 1 of these should solve your problem. 1) Trap SIGHUP. 2) Use the h extension. 3) Use deadagi(). Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- 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] Upgraded to 1.4.18 (from 1.2.27) and channels dropping on Zaptel and SIP
--- Darrick Hartman (lists) [EMAIL PROTECTED] wrote: Try adding this line in the general section of extensions.conf autofallthrough=no The default behavior in 1.2 was no. In 1.4 it changed to yes. That will be your simplest fix (without seeing your dialplan). Asterisk is moving on to the next step in the dialplan before you enter your digits. You need to have it wait for the digits to be entered. Thanks for that. I did see that note in UPGRADE.txt but didn't realize the full importance of it changing the logic of the dialplan. I've got it set back to no and will read the new version of ATFOT to figure out how to restructure my dialplan. So now it seems 1.4.18 is doing the same as 1.2.27 -- working for the most part but completely ignoring DTMF on incoming SIP calls. Best regards, David [EMAIL PROTECTED] 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] Playing sound while talking
On Wed, Mar 26, 2008 at 4:02 PM, Artifex Maximus [EMAIL PROTECTED] wrote: Is it possible play background sounds while talking? I would like to make an outgoing campaign with the possibility playing sounds in background by command. But the extra is I would like to choose which sound to be played. In short operator calls a number, talking to callee and sometimes play different sound (for example different music) in background which should be interruptible and then ask questions from callee. Is it possible? If yes from which version number? I have an old 1.2 system at this time. I am using Asterisk 1.2.10 on Fedora Core 4 now. Looking for plain Asterisk solution and no commercial offers please. Any idea is welcome. Thanks in advance. OK, I had found applicationmap in features.conf. Is there any way for playing sound on both (caller, callee) side? Is there any solution for playing some sound on caller side because caller does not hear anything while sound is playing by applicationmap and have no clue when it will reach the end. bye, a ___ -- 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] IAXy device
bilal ghayyad wrote: Hi All; I have been chocked just when I saw some posts talking about how much the IAXy is bad :) - So I would like to ask, did any one try it later and wether it is good or not? I am asking this because I need to use it as it is NAT Transparent (as I read also, and I did not try it to see how much it is transparent). What about codec? Why it is only support g711 and does not support compressed codec? And what about the IP address and the DNS usage and the DDNS usage? What main porblems contain and any advise? Regards Bilal The device has no echo cancellation and sounds horrible (lots of echo) on about half of the analog phones I tried it on. I wouldn't recommend it unless you absolutely need IAX. It's also very expensive for a 1 port ATA. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zapata Tormenta 2
Thanks, got it working. Also, does the zapata tormenta 2 card have only T1/E1 ports, or are they also FXS/FX0 ports? Cody Jarrett IT Freedom direct 512.351.4965 [EMAIL PROTECTED] office 512.351.7990 : fax 512.351.7991 On Mar 27, 2008, at 1:42 AM, Tzafrir Cohen wrote: On Wed, Mar 26, 2008 at 11:00:21PM -0500, Cody Jarrett wrote: I have an old zapata tormenta 2 quad port pci card. I'd like to get it working and play with it but was curious to see if that was possible. Does anyone know if it will work for 2.6 kernels, or where I can find decent drivers? I tried getting the tor driver from http://www.zapatatelephony.org/linux-zapata-current/ installed, but didn't get very far. Any input is appreciated. Use the tor2 driver included with the main Zaptel dirstrubution. -- 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 ___ -- 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] Zapata Tormenta 2
On Thu, Mar 27, 2008 at 09:57:39AM -0500, Cody Jarrett wrote: Thanks, got it working. Also, does the zapata tormenta 2 card have only T1/E1 ports, or are they also FXS/FX0 ports? No. Just E1/T1 ports. Analogs ports are different beasts. -- 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] DTMF suddenly stopped working on SIP channel
--- Eric Wieling [EMAIL PROTECTED] wrote: Inband only works with the ulaw and alaw codecs. I think you might be onto something here. I don't have any explicit allow or disallow lines, just taking the defaults. I've got plenty of bandwidth and CPU, I'm much more concerned about calls going through. Without knowing what codecs my provider uses and not seeing anything specific in the logs, is there a setting that would be better than default for reliability? I had originally set to inband for outgoing calls because the default wasn't working for dialing into voicemail systems, etc. Switching to inband fixed the outgoing DTMF issue and incoming worked fine for months until earlier this week. Thanks for any suggestions. Best regards, David [EMAIL PROTECTED] 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
Re: [asterisk-users] Question about PCI Slots for DIGIUMs Boards
On Thu, 27 Mar 2008 06:48:58 -0400, Al Baker [EMAIL PROTECTED] wrote: I an considering using *your High Density T*1 cards on a number of servers we are considering purchasing. The vendor lists that his system has: PCI Express*: two x8 slots*, t*wo x8 low profile slots*; *PCI-X: 64-bit/100MHz* Could you please clarify *WHICH* of the above listed *PCI slots* are suitable for use with your *High Density T1 cards*. None of the above listed are PCI slots. PCI != PCI Express -- Godwin Stewart - Horwich IT services ___ -- 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] Upgraded to 1.4.18 (from 1.2.27) and channels dropping on Zaptel and SIP
On 27/03/2008, David Nedved [EMAIL PROTECTED] wrote: So now it seems 1.4.18 is doing the same as 1.2.27 -- working for the most part but completely ignoring DTMF on incoming SIP calls. Perhaps you now need to delve deeper. Capture a UDP trace between your VoIP provider and Asterisk, and another of the same call between Asterisk and a handset. Do this for an ordinary voice call, no IVR menus etc etc. 1) Can you hear the DTMF being sent by the far end by the way? 2) If you use Wireshark to do a VoIP call analysis of the traces, do you receive any DTMF signalling in the RTP stream, or in INFO packets from your VoIP provider? I'm sure there is more... Steve ___ -- 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] Had it with Dell Garbage - HP Question
I've got two DL385s and a DL320, and they all rock. iLO especially rocks, but to leverage the full functionality, you'll need to get the Advanced License, which opens up full blown remote console capabilities (via Java). It's a separate piece of hardware that, as long as the server PSUs have power flowing into them, lets you do things like remotely power on/off/reset the machine (referred to as virtual power), monitor OS crashes (picks up Windows BSoDs and NetWare ABENDShaven't Oopsed Linux on one yet). iLO's allowed me to do everything from BIOS upgrades to fixing NetWare boot issues all from the comfort of my home at 3am in the morning. The build quality is superb...more metal than plastic, so they can weight a bit more, but I expect that of my servers versus desktop boxes. I myself use RAID5 in my DL385 G1's (AMD Opteron), which hold up to six Ultra 320 SCSI drives, on a HP SmartArray controller (64MB of cache thoughneed to upgrade that). The DL320 is a RAID1 on 2x 10k rpm SAS drives, on a...P400 I think w/ 256MB of cache. All battery backed. OS Support is great so far. Just check HP's site for the Proliant Support Packs specific to an OS, as they sometimes provide better drivers than what ships stock w/ the OS (this is especially true for NetWare). --J -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Al Baker Sent: Thursday, March 27, 2008 8:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Had it with Dell Garbage - HP Question How do you get notifications ? Is this thru one of the add on packages HP sells for the box ? Which One ? Could you be more specific about what you mean by a recovery CD and hod do you get console access below multi used to do recovery ?? What is integrated ILO BIOS Access sounds cool. What O/S you usin and what made you pick it ? What kind and how many RAIDS are you using. The HP site gave like 8 different RAID controllers and like 20 CPUs to chose from. How did you chose ? Thx for sharing !!! Darren Wright wrote: One of the major reasons we use DL320 / DL380's is the ease of swapping drives, and the integrated ILO BIOS level access.We can support remote sites with ease. If a drive dies we get a notification, a new one is sent and a non-techie can replace it with guidance.No onsite visit. That is worth potentially thousands of dollars. We also leave a recovery CD there that can be inserted if we need to rebuild the system remotely. Never had to, but it's worked in the lab. -D This message was sent from D2 Technology, INC. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- 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] Question about PCI Slots for DIGIUMs Boards
This will be 4 T1s to a card.So, just so I am not confused. In my original e-mail I said I an considering using *your High Density T*1 cards on a number of servers we are considering purchasing. The VENDOR lists that his SYSTEM has: PCI Express*: two x8 slots*, twoo x8 low profile slots*; *PCI-X: 64-bit/100MHz* And you list 3 possible PCI slots for your DIGIUM Quad T1 Cards PCI or PCI-X 3.3 Volt Slots PCI 5.0 Volt Slots PCI-E (PCI Express) slots == Here is the Confusion - 1) Can ANY of your QUAD cards go in the PCI Express - two x8 slots - Yes or NO ? 2 Can ANY of your QUAD cards go in the PCI Express - two x8 LOW Profile Slots - Yes or No ? 3) Can your Quad cards you list as fitting a PCI or PCI-X 3.3 Volt Slots fit in the PCI-X: 64-bit/100MHz slot OR would I have to ask the vendor Is this PCI-X: 64-bit/100MHz definitely a 3.3 Volt Slot ? Yes - It would absolutely definitely fit OR NO - It will NOT fit OR To answer that I need more info ? THX for your help Jared Smith wrote: On Thu, 2008-03-27 at 06:48 -0400, Al Baker wrote: I an considering using *your High Density T*1 cards on a number of servers we are considering purchasing. Two-port T1 cards? Four-port T1 cards? Could you please clarify *WHICH* of the above listed *PCI slots* are suitable for use with your *High Density T1 cards*. If there is *more than one combination that is possible*, could you please indicate *WHICH *which high density T1 cards will work in *WHICH* type of PCI slot. PCI or PCI-X 3.3 Volt Slots --- TE412P (Quad span T1/E1/J1/PRI with DSP-based Echo Cancellation) TE212P (Dual span T1/E1/J1/PRI with DSP-based Echo Cancellation) PCI 5.0 Volt Slots --- TE407P (Quad span T1/E1/J1/PRI with DSP-based Echo Cancellation) TE207P (Dual span T1/E1/J1/PRI with DSP-based Echo Cancellation) PCI-E (PCI Express) slots - TE420B (Quad Span T1/E1/J1/PRI with DSP-based Echo Cancellation) TE220B (Dual span T1/E1/J1/PRI with DSP-based Echo Cancellation) There are also a variety of models available without the echo cancellation modules onboard, but personally I recommend purchasing the echo cancellation modules in most situations. ___ -- 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] Had it with Dell Garbage - HP Question
On 08:02, Thu 27 Mar 08, Al Baker wrote: How do you get notifications ? Is this thru one of the add on packages HP sells for the box ? Which One ? Could you be more specific about what you mean by a recovery CD and hod do you get console access below multi used to do recovery ?? What is integrated ILO BIOS Access sounds cool. What O/S you usin and what made you pick it ? What kind and how many RAIDS are you using. The HP site gave like 8 different RAID controllers and like 20 CPUs to chose from. How did you chose ? Thx for sharing !!! I'm not the op, but sending a reply anyways. The notifications come from the HP tools you can download for free from their website. The recovery cd is probably a selfmade installer for their setup. At least that's what we have. the ILO stuff is to give you access to the box like you were sitting right in front of it with a physical keyboard and monitor, but over IP. You can boot the machine, access the cd in your local machine etc, even if the box is on the other side of the moon. We use Debian. HP even supports it on their DL380 boxen. We use the P400 raid controller. Setup RAID5 with 3 disks. CPU we use right now is the Intel E5405 Hope this helps a bit. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question about PCI Slots for DIGIUMs Boards
No actually I gave it a LOT of thought and I and even asked two different techs who repair PCs and both said what the to vendors are saying is not sufficiently clear that I would make a purchasing decision based on what you have in hand from them But, nice try at the cheap shot. Steve Totaro wrote: On Thu, Mar 27, 2008 at 6:48 AM, Al Baker [EMAIL PROTECTED] wrote: I had sent this to Digium Sales and cant get a response from them, I don't. know what that means.. So I thought I would ask it hear since i know others have struggled with this. I an considering using *your High Density T*1 cards on a number of servers we are considering purchasing. The vendor lists that his system has: PCI Express*: two x8 slots*, t*wo x8 low profile slots*; *PCI-X: 64-bit/100MHz* Could you please clarify *WHICH* of the above listed *PCI slots* are suitable for use with your *High Density T1 cards*. If there is *more than one combination that is possible*, could you please indicate *WHICH *which high density T1 cards will work in *WHICH* type of PCI slot. Thanks for any guidance and help since DIGIUM sales seems to be too busy to advise a potential customer You never qualify High Density. Maybe Digium thought you might get tired of waiting and find your own answer http://en.wikipedia.org/wiki/Peripheral_Component_Interconnect Besides that, look here http://store.digium.com/products.php?category_id=2 You should be able to figure it out on your own with a little thought. 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 ___ -- 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] Question about PCI Slots for DIGIUMs Boards
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Al Baker Sent: Thursday, March 27, 2008 12:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Question about PCI Slots for DIGIUMs Boards No actually I gave it a LOT of thought and I and even asked two different techs who repair PCs and both said what the to vendors are saying is not sufficiently clear that I would make a purchasing decision based on what you have in hand from them But, nice try at the cheap shot. The two techs you spoke with are obviously not familiar enough with the technology at hand to be able to make that determination, because all of the necessary information is contained within the original e-mail and the specs available on Digium's website. To simplify, however, here is the answer: The TE420 cards will fit in the regular x8 (not low-profile) PCI-E slots, and the TE412P will fit into the PCI-X slots. Regards, - Brad ___ -- 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] callers in queue passed to agents who accept only one call at a time
I have a queue I configured as strict and a cron script I use to QueueAdd and QueueRemove agents according to my company's requirements. Usually I have 2 or 3 agents at a time and the ring strategy is ringall. These agents use non-open-source Windows softphones that do not let you configure it so that if they're on the phone, a second call will be rejected (agent busy). Instead, it's as if they had call waiting and incoming calls keep popping up while they're conversating with the first caller and they would like to avoid this. I guess the easiest solution would be to find an open-source or free softphone that can be configured to accept only one call at a time (currently using SJphone). Another solution would be if I could tell the Queue() application that if an agent is InUse then don't pass the call. Still another yet more delicate solution would be to have a custom script receive manager events related to the queue which in turn replies with an agi command. For example, whenever an agent answers a call I think that an event such as QueueMemberStatus can be triggered (although I don't know how). If the custom script could receive this event in realtime then it would run an agi command such as QueueRemove(busyagent...). When the agent is free again I suppose the same event is triggered and the custom script can QueueAdd(freeagent...). Could anyone please give me some pointers on this? Thanks! Vieri 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] Asterisk not picking up (some) calls due to zaptel detecting and clearing alarms
Hi All, For the most part, the PBX works as it should. Occasionally people complain that they call and the PBX doesn't pick up. Other times it looks like the call is answered by Asterisk but I still hear ringing and I start listening to the IVR menu a few seconds into it. As for Asterisk not picking up, I see the following in the logs: [Mar 27 13:32:29] NOTICE[13197]: chan_zap.c:7071 ss_thread: Got event 18 (Ring Begin)... [Mar 27 13:32:30] NOTICE[13197]: chan_zap.c:7071 ss_thread: Got event 2 (Ring/Answered)... -- Executing [EMAIL PROTECTED]:1] Answer(Zap/2-1, ) in new stack -- Executing [EMAIL PROTECTED]:2] Wait(Zap/2-1, 1) in new stack [Mar 27 13:32:30] WARNING[13197]: chan_zap.c:4203 get_alarms: Unable to determine alarm on channel 2 [Mar 27 13:32:30] WARNING[13197]: chan_zap.c:4424 zt_handle_event: Detected alarm on channel 2: No Alarm == Spawn extension (incoming, s, 2) exited non-zero on 'Zap/2-1' -- Hungup 'Zap/2-1' [Mar 27 13:32:31] NOTICE[13197]: chan_zap.c:7514 handle_init_event: Alarm cleared on channel 2 The above messages repeat themselves a number of times as the ringing continues and causes Asterisk to try and pick up the call again (and fails with the alarm thrown which then gets cleared). Any suggestions?? I'm using Asterisk 1.6.0-beta4 and Zaptel 1.4.9.2. Thanks in advance! Gonzalo ___ -- 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 not picking up (some) calls due to zaptel detecting and clearing alarms
On Thu, Mar 27, 2008 at 01:40:48PM -0300, Gonzalo Servat wrote: Hi All, For the most part, the PBX works as it should. Occasionally people complain that they call and the PBX doesn't pick up. Other times it looks like the call is answered by Asterisk but I still hear ringing and I start listening to the IVR menu a few seconds into it. As for Asterisk not picking up, I see the following in the logs: [Mar 27 13:32:29] NOTICE[13197]: chan_zap.c:7071 ss_thread: Got event 18 (Ring Begin)... [Mar 27 13:32:30] NOTICE[13197]: chan_zap.c:7071 ss_thread: Got event 2 (Ring/Answered)... -- Executing [EMAIL PROTECTED]:1] Answer(Zap/2-1, ) in new stack -- Executing [EMAIL PROTECTED]:2] Wait(Zap/2-1, 1) in new stack [Mar 27 13:32:30] WARNING[13197]: chan_zap.c:4203 get_alarms: Unable to determine alarm on channel 2 [Mar 27 13:32:30] WARNING[13197]: chan_zap.c:4424 zt_handle_event: Detected alarm on channel 2: No Alarm == Spawn extension (incoming, s, 2) exited non-zero on 'Zap/2-1' -- Hungup 'Zap/2-1' [Mar 27 13:32:31] NOTICE[13197]: chan_zap.c:7514 handle_init_event: Alarm cleared on channel 2 The above messages repeat themselves a number of times as the ringing continues and causes Asterisk to try and pick up the call again (and fails with the alarm thrown which then gets cleared). Any suggestions?? I'm using Asterisk 1.6.0-beta4 and Zaptel 1.4.9.2. A freshly-built Asterisk? Built vs. zaptel 1.4.9.2 ? Shouldn't it have picked up the alarm as a red alarm on the channel? (Besides the problem. Is 1.4 SVN recommended for that at the moment?) -- 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] IAXy device
Sean Dennis wrote: bilal ghayyad wrote: Hi All; I have been chocked just when I saw some posts talking about how much the IAXy is bad :) - So I would like to ask, did any one try it later and wether it is good or not? I am asking this because I need to use it as it is NAT Transparent (as I read also, and I did not try it to see how much it is transparent). What about codec? Why it is only support g711 and does not support compressed codec? And what about the IP address and the DNS usage and the DDNS usage? What main porblems contain and any advise? Regards Bilal The device has no echo cancellation and sounds horrible (lots of echo) on about half of the analog phones I tried it on. I wouldn't recommend it unless you absolutely need IAX. It's also very expensive for a 1 port ATA. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Echo may be the result of latency on the network. I've not had any echo problems that I remember with my IAXy and I make ten calls a day, five days a week, for the last few years, to all sorts of numbers/areas. I know that this isn't representative of typical business use, but residential use, but I've been using in my business and have never been disappointed :) I will agree that's is fairly expensive, but I WOULD recommend it to people who are on the go often. After setup, it really is plug-n-play IMO. Moj ___ -- 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] ADPCM codec and IAXy device
bilal ghayyad wrote: Hi All; I need to buy one IAXy device, but I discovered that it supports only g711 and ADPCM codec, so I was wonder that it does not support g729 or GSM?! Anyway, what is that ADPCM and how much it consumes bandwitdh? Also, asterisk support such codec? What its name in the configuration? Any advise? Regards Bilal 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 I've heard, and I think Eric Wieling just confirmed it on the mailing list today, The IAXy does not support highly compressed codecs... I seem to recall that there's a space for ADPCM in the IAXy provisioning file, but I also seem to remember that this codec was not implemented in the IAXy's firmware. I've never tested it, so I don't know for sure. And if this was true, of course it could have changed by 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] IAXy device
You will never get latency on a network low enough for echo to be perceived as sidetone (like on analog). If you want to get rid of echo you must cancel echo. Mojo with Horan Company, LLC wrote: Sean Dennis wrote: bilal ghayyad wrote: Hi All; I have been chocked just when I saw some posts talking about how much the IAXy is bad :) - So I would like to ask, did any one try it later and wether it is good or not? I am asking this because I need to use it as it is NAT Transparent (as I read also, and I did not try it to see how much it is transparent). What about codec? Why it is only support g711 and does not support compressed codec? And what about the IP address and the DNS usage and the DDNS usage? What main porblems contain and any advise? Regards Bilal The device has no echo cancellation and sounds horrible (lots of echo) on about half of the analog phones I tried it on. I wouldn't recommend it unless you absolutely need IAX. It's also very expensive for a 1 port ATA. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Echo may be the result of latency on the network. I've not had any echo problems that I remember with my IAXy and I make ten calls a day, five days a week, for the last few years, to all sorts of numbers/areas. I know that this isn't representative of typical business use, but residential use, but I've been using in my business and have never been disappointed :) I will agree that's is fairly expensive, but I WOULD recommend it to people who are on the go often. After setup, it really is plug-n-play IMO. Moj ___ -- 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 -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- 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 not picking up (some) calls due to zaptel detecting and clearing alarms
On Thu, Mar 27, 2008 at 1:56 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote: Any suggestions?? I'm using Asterisk 1.6.0-beta4 and Zaptel 1.4.9.2. A freshly-built Asterisk? Built vs. zaptel 1.4.9.2 ? Yes, I built 1.6.0-beta4 just recently with zaptel 1.4.9.2. As per your suggestion on IRC, I've checked out, compiled and installed Zaptel from SVN (1.4 branch). I reloaded the zaptel modules but ... no go. Do I need to recompile Asterisk too? Shouldn't it have picked up the alarm as a red alarm on the channel? I've no idea to be honest. (Besides the problem. Is 1.4 SVN recommended for that at the moment?) Also no idea. I was told to use 1.4 if I'm using Asterisk 1.6 so I went with that. - Gonzalo ___ -- 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] IAXy device
I guess I've never run asterisk without ANY echo cans :) It's just that the echo was minor enough that MG2 et. al did a fine job. Thanks! Moj Eric Wieling wrote: You will never get latency on a network low enough for echo to be perceived as sidetone (like on analog). If you want to get rid of echo you must cancel echo. Mojo with Horan Company, LLC wrote: Sean Dennis wrote: bilal ghayyad wrote: Hi All; I have been chocked just when I saw some posts talking about how much the IAXy is bad :) - So I would like to ask, did any one try it later and wether it is good or not? I am asking this because I need to use it as it is NAT Transparent (as I read also, and I did not try it to see how much it is transparent). What about codec? Why it is only support g711 and does not support compressed codec? And what about the IP address and the DNS usage and the DDNS usage? What main porblems contain and any advise? Regards Bilal The device has no echo cancellation and sounds horrible (lots of echo) on about half of the analog phones I tried it on. I wouldn't recommend it unless you absolutely need IAX. It's also very expensive for a 1 port ATA. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Echo may be the result of latency on the network. I've not had any echo problems that I remember with my IAXy and I make ten calls a day, five days a week, for the last few years, to all sorts of numbers/areas. I know that this isn't representative of typical business use, but residential use, but I've been using in my business and have never been disappointed :) I will agree that's is fairly expensive, but I WOULD recommend it to people who are on the go often. After setup, it really is plug-n-play IMO. Moj ___ -- 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] Question about PCI Slots for DIGIUMs Boards
Ok - that is even more confusing , since one of them looks like it MIGHT be a PCI slot in the list PCI Express*: two x8 slots*, t*wo x8 low profile slots*; *PCI-X: 64-bit/100MHz* | /\ | | | | | | So now I am really confused, because Jared Smith -Community Relations Manager,Digium, Inc. said in His Post Quote *PCI or PCI-*X 3.3 Volt Slots --- TE412P (Quad span T1/E1/J1/PRI with DSP-based Echo Cancellation) TE212P (Dual span T1/E1/J1/PRI with DSP-based Echo Cancellation) *So, what is the *_Real Answer_* here ? Any and all help greatly appreciatted* Horwich IT Services (Godwin Stewart) wrote: On Thu, 27 Mar 2008 06:48:58 -0400, Al Baker [EMAIL PROTECTED] wrote: I an considering using *your High Density T*1 cards on a number of servers we are considering purchasing. The vendor lists that his system has: PCI Express*: two x8 slots*, t*wo x8 low profile slots*; *PCI-X: 64-bit/100MHz* Could you please clarify *WHICH* of the above listed *PCI slots* are suitable for use with your *High Density T1 cards*. None of the above listed are PCI slots. PCI != PCI Express ___ -- 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] Upgraded to 1.4.18 (from 1.2.27) and channels dropping on Zaptel and SIP
On Thu, Mar 27, 2008 at 08:32:28AM -0400, Steve Totaro wrote: People on the list (mainly dev) want you to test, find bugs, jump through hoops, and post to Mantis (where you bug might just be closed, or a general feeling of You are wrong. All of this testing is free of course due to the Benefit of the Community. In the real world, it would serve you better to do what works best for your business. Don't let the Dev guys push you around, do what makes sense to your business. Ok, Steve... I understand both sides of this issue, and the one you're handwaving is how much did you pay for Asterisk? Nothing in life is free, and people who prefer to use the no-Cost Asterisk as a PBX base instead of paying Nortel mumble-thousand for an Option 11 still ought to be prepared to invest *something* in their outcome. Being a participating member of the open source community; feeding bugs back to the developers and the like; that's how you 'pay your bill' when the software doesn't cost anything. Sure, *everyone's* not *required* to do it. But people inclined to use Asterisk ought to be figuring some of this into their value equation. If it's too troublesome...well, buy a box from someone. No? Cheers, -- jr 'I am not now, nor have I ever been a dev' a -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- 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] Question about PCI Slots for DIGIUMs Boards
On Thu, 2008-03-27 at 12:16 -0400, Al Baker wrote: == Here is the Confusion - 1) Can ANY of your QUAD cards go in the PCI Express - two x8 slots - Yes or NO ? Yes, the TE420B will fit. It's a 4-port T1/E1/J1/PRI card with echo cancellation on-board. It only fits in PCI Express slots. 2 Can ANY of your QUAD cards go in the PCI Express - two x8 LOW Profile Slots - Yes or No ? No, none of our PCI Express cards are low-profile (as far as I know) 3) Can your Quad cards you list as fitting a PCI or PCI-X 3.3 Volt Slots fit in the PCI-X: 64-bit/100MHz slot OR would I have to ask the vendor Is this PCI-X: 64-bit/100MHz definitely a 3.3 Volt Slot ? Yes - It would absolutely definitely fit OR NO - It will NOT fit OR To answer that I need more info ? You would need to ask your vendor whether or not the PCI-X slot is a 3.3 volt or a 5.0 volt slot. I'm guessing it's 3.3 volts, but that's just a guess. Here's a link to a decent article explaining the difference between PCI and PCI Express slots, if you're interested. http://www.geeks.com/techtips/2006/techtips-24sept06.htm -- Jared Smith Community Relations Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calling users to the external domain using Asterisk
Aadilkhan Maniyar wrote: Hi All, I am a newbie to Asterisk. Presently I am working with Asterisk 1.4.17 and using it to make SIP calls. I have a configuration of Asterisk which serves the users in a particular domain, say internal.com I would like to make a SIP call from [EMAIL PROTECTED] to [EMAIL PROTECTED] I have added the following lines in extensions.conf exten = charles,1,Dial(SIP/[EMAIL PROTECTED]) exten = charles,2,Hangup Asterisk does a DNS SRV lookup and resolves the external.com to its proper IP and calls are established. But the problem with the above configuration is that I have manually added users that are in the external domain. Is there any way wherein I can call the users in external.com without adding them in the extensions.conf? Any help would be appreciated. Thanks, Aadil ___ -- 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 I could be wrong about this, but isn't that what a switch statement is for? So you might check to see if the dialed number is local to internal.com, then you might do a switch statement to external.com's dialplan if it wasn't local? moj ___ -- 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] Upgraded to 1.4.18 (from 1.2.27) and channels dropping on Zaptel and SIP
On Thu, Mar 27, 2008 at 2:38 PM, Jay R. Ashworth [EMAIL PROTECTED] wrote: On Thu, Mar 27, 2008 at 08:32:28AM -0400, Steve Totaro wrote: People on the list (mainly dev) want you to test, find bugs, jump through hoops, and post to Mantis (where you bug might just be closed, or a general feeling of You are wrong. All of this testing is free of course due to the Benefit of the Community. In the real world, it would serve you better to do what works best for your business. Don't let the Dev guys push you around, do what makes sense to your business. Ok, Steve... I understand both sides of this issue, and the one you're handwaving is how much did you pay for Asterisk? Nothing in life is free, and people who prefer to use the no-Cost Asterisk as a PBX base instead of paying Nortel mumble-thousand for an Option 11 still ought to be prepared to invest *something* in their outcome. Being a participating member of the open source community; feeding bugs back to the developers and the like; that's how you 'pay your bill' when the software doesn't cost anything. Sure, *everyone's* not *required* to do it. But people inclined to use Asterisk ought to be figuring some of this into their value equation. If it's too troublesome...well, buy a box from someone. No? Cheers, I am a user and a high level integrator, none of what you mention applies to me. Maybe in a lab if I had time... I run multi million dollar call centers and very demanding PBXs, it is not in customer's best interest to run buggy code, therefore it is also not in my best interest. It is a similar relationship to corporations and their stockholders, the corp must do what is in the best interest of the shareholder. I like to call it good business, none of this rebooting daily, weekly, monthly crap. Maybe if you lost $26k/hr due to outages, you might feel differently Asterisk is a loss leader for the hardware (cards, appliances, support, ABE) that is why it is free. Otherwise Asterisk would be vaporware. Anyways, Asterisk has many costs but I guess you never took Econ 101 or above in college. I have brought Asterisk to the attention of CSC, The US State Dept, large corporations, and foreign governments, is that some form of contribution to the community? I think promotion is a full time job in some outfits. By the way, I use the best components to build my systems and my consulting fee is pretty nice, so you are right, nothing is free. 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] Question about PCI Slots for DIGIUMs Boards
Are you looking to purchase a server or just looking for a card for an existing server? Either way, post the specs from the manufacturer (and also make sure those slots are open) You will get your answer pretty quickly, and much less painfully this way. Thanks, Steve Totaro On Thu, Mar 27, 2008 at 2:30 PM, Jared Smith [EMAIL PROTECTED] wrote: On Thu, 2008-03-27 at 12:16 -0400, Al Baker wrote: == Here is the Confusion - 1) Can ANY of your QUAD cards go in the PCI Express - two x8 slots - Yes or NO ? Yes, the TE420B will fit. It's a 4-port T1/E1/J1/PRI card with echo cancellation on-board. It only fits in PCI Express slots. 2 Can ANY of your QUAD cards go in the PCI Express - two x8 LOW Profile Slots - Yes or No ? No, none of our PCI Express cards are low-profile (as far as I know) 3) Can your Quad cards you list as fitting a PCI or PCI-X 3.3 Volt Slots fit in the PCI-X: 64-bit/100MHz slot OR would I have to ask the vendor Is this PCI-X: 64-bit/100MHz definitely a 3.3 Volt Slot ? Yes - It would absolutely definitely fit OR NO - It will NOT fit OR To answer that I need more info ? You would need to ask your vendor whether or not the PCI-X slot is a 3.3 volt or a 5.0 volt slot. I'm guessing it's 3.3 volts, but that's just a guess. Here's a link to a decent article explaining the difference between PCI and PCI Express slots, if you're interested. http://www.geeks.com/techtips/2006/techtips-24sept06.htm -- 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 ___ -- 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] IAXy device
I had a customer using an IAXY (old gen) for an FXO fax machine and it worked almost all the time so it cannot be that bad. Maybe because the fax was very old and did not have high transmit rates. Thanks, Steve Totaro On Thu, Mar 27, 2008 at 2:11 PM, Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote: I guess I've never run asterisk without ANY echo cans :) It's just that the echo was minor enough that MG2 et. al did a fine job. Thanks! Moj Eric Wieling wrote: You will never get latency on a network low enough for echo to be perceived as sidetone (like on analog). If you want to get rid of echo you must cancel echo. Mojo with Horan Company, LLC wrote: Sean Dennis wrote: bilal ghayyad wrote: Hi All; I have been chocked just when I saw some posts talking about how much the IAXy is bad :) - So I would like to ask, did any one try it later and wether it is good or not? I am asking this because I need to use it as it is NAT Transparent (as I read also, and I did not try it to see how much it is transparent). What about codec? Why it is only support g711 and does not support compressed codec? And what about the IP address and the DNS usage and the DDNS usage? What main porblems contain and any advise? Regards Bilal The device has no echo cancellation and sounds horrible (lots of echo) on about half of the analog phones I tried it on. I wouldn't recommend it unless you absolutely need IAX. It's also very expensive for a 1 port ATA. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Echo may be the result of latency on the network. I've not had any echo problems that I remember with my IAXy and I make ten calls a day, five days a week, for the last few years, to all sorts of numbers/areas. I know that this isn't representative of typical business use, but residential use, but I've been using in my business and have never been disappointed :) I will agree that's is fairly expensive, but I WOULD recommend it to people who are on the go often. After setup, it really is plug-n-play IMO. Moj ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Upgraded to 1.4.18 (from 1.2.27) and channels dropping on Zaptel and SIP
On Thu, Mar 27, 2008 at 02:58:31PM -0400, Steve Totaro wrote: I am a user and a high level integrator, none of what you mention applies to me. Maybe in a lab if I had time... If you are a high-level integrator, then it seems to me you make direct profit off the backs of the developers you decline to support. I run multi million dollar call centers and very demanding PBXs, it is not in customer's best interest to run buggy code, therefore it is also not in my best interest. Rockwell Galaxy's are great stuff. It is a similar relationship to corporations and their stockholders, the corp must do what is in the best interest of the shareholder. I like to call it good business, none of this rebooting daily, weekly, monthly crap. Maybe if you lost $26k/hr due to outages, you might feel differently Yup. And if I had lots of outages and that was an issue, I might run a Galaxy and pay the price. But in fact, not such a problem. Asterisk is a loss leader for the hardware (cards, appliances, support, ABE) that is why it is free. Otherwise Asterisk would be vaporware. Well, most of our cards are Sangomas, actually. Anyways, Asterisk has many costs but I guess you never took Econ 101 or above in college. Clearly, *you* failed reading comprehension. :-) My entire point was that there are many different costs -- and that you were shirking the most important one I could see. I have brought Asterisk to the attention of CSC, The US State Dept, large corporations, and foreign governments, is that some form of contribution to the community? I think promotion is a full time job in some outfits. Sure. Everyone contributes something different. And thanks. :-) By the way, I use the best components to build my systems and my consulting fee is pretty nice, so you are right, nothing is free. See? We're in violent agreement. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- 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] Help with cisco 7960 phone
I have a cisco 7960 phone. Worked fine in the office. I took it home. At home I have a linksys router that the phone is plugged into. The linksys router has DHCP enabled. I am getting the following error on the console from the 7960. I have tried it with nat=yes and nat=no in the sip.conf file. --- Transmitting (NAT) to 192.168.1.69:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.69:5060;branch=z9hG4bK1415360297;received=192.168.1.69;rport=5060 From: Display Name sip:[EMAIL PROTECTED];tag=1683635072 To: Display Name sip:[EMAIL PROTECTED];tag=as4c59a734 Call-ID: [EMAIL PROTECTED] CSeq: 3091 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=4a1c350c Content-Length: 0 -- The username and secret are the same as they were in the office when it worked. I figure it has to be something easy but I have not found it yet. the sip.conf entry for this phone is: [570] type=friend dtmfmode=rfc2833 username=570 secret=XXX disallow=all allow=ulaw allow=alaw host=dynamic context=local-sip callerid=Home 570 570 nat=no What might I try to get the phone working from home? Thanks, Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Star Wars Echo Sound
We have a location that is having a really odd issue. We have a sangoma POTs card. We are running software echo cancellation with the card (through asterisk) to try to eliminate some major echoing problems. I've turned on both EC and echotrain, which seemed to have gotten rid of the echo for the most part. However, we are now running into an issue where the outside caller hears a star wars type of sound. I expierenced this myself when talking to them. By this, I mean you hear a few words from them, then a few seconds lagging behind, you'll hear a muffled (darth vader) version of the same thing. Has anyone seen this? Thanks, Rob ___ -- 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] Star Wars Echo Sound
That's probably just someone at the NSA snooping your lines and playing tricks on you... g --J -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Rob Schall Sent: Thursday, March 27, 2008 4:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Star Wars Echo Sound We have a location that is having a really odd issue. We have a sangoma POTs card. We are running software echo cancellation with the card (through asterisk) to try to eliminate some major echoing problems. I've turned on both EC and echotrain, which seemed to have gotten rid of the echo for the most part. However, we are now running into an issue where the outside caller hears a star wars type of sound. I expierenced this myself when talking to them. By this, I mean you hear a few words from them, then a few seconds lagging behind, you'll hear a muffled (darth vader) version of the same thing. Has anyone seen this? Thanks, Rob ___ -- 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] Star Wars Echo Sound
I wanna say that's the echotraining taking effect. What it does is try to cause some echo so it can dynamically reconfigure the levels on the fly -- right at the start of the call. I know this happens with digium cards -- not sure if the Sangoma cards behave the exact same way. It's only at the start of the call right? once that occurs, the EC is kicked in and everything is fine? -- Chris Earle System Solutions Specialist, Network Technologies Division CBL Data Recovery w: http://www.cbltech.com Rob Schall [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] We have a location that is having a really odd issue. We have a sangoma POTs card. We are running software echo cancellation with the card (through asterisk) to try to eliminate some major echoing problems. I've turned on both EC and echotrain, which seemed to have gotten rid of the echo for the most part. However, we are now running into an issue where the outside caller hears a star wars type of sound. I expierenced this myself when talking to them. By this, I mean you hear a few words from them, then a few seconds lagging behind, you'll hear a muffled (darth vader) version of the same thing. Has anyone seen this? Thanks, Rob ___ -- 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] Problem when leaving voicemail
Hi, I am investigating an issue with voicemail and realtime. What we are seeing is the following: 1. Caller calls in and goes to an IVR 2. Presses 101 to go to voicemail 3. app_voicemail start and tries to connect to the database trhough res_config_mysql. However, it takes too long to be able to connect (~15 minutes) It seems like it first attemots to connect to the database on 16:25:03 and manage to connect at 16:40:24. [Mar 26 16:25:03] VERBOSE[19786] logger.c: -- AGI Script agi://127.0.0.1/enswitch?stype=external completed, returning 0 [Mar 26 16:25:03] DEBUG[19786] pbx.c: Launching 'Answer' [Mar 26 16:25:03] VERBOSE[19786] logger.c: -- Executing [ [EMAIL PROTECTED]:1] Answer(SIP/5060-ac017e30, ) in new stack [Mar 26 16:25:03] DEBUG[19786] pbx.c: Expression result is '0' [Mar 26 16:25:03] DEBUG[19786] pbx.c: Launching 'GotoIf' [Mar 26 16:25:03] VERBOSE[19786] logger.c: -- Executing [ [EMAIL PROTECTED]:2] GotoIf(SIP/5060-ac017e30, 0?5) in new stack [Mar 26 16:25:03] DEBUG[19786] pbx.c: Not taking any branch [Mar 26 16:25:03] DEBUG[19786] pbx.c: Launching 'VoiceMail' [Mar 26 16:25:03] VERBOSE[19786] logger.c: -- Executing [ [EMAIL PROTECTED]:3] VoiceMail(SIP/5060-ac017e30, [EMAIL PROTECTED]|us[EMAIL PROTECTED]) in new stack [Mar 26 16:25:03] DEBUG[19786] app_voicemail.c: Before find_user [Mar 26 16:25:03] DEBUG[19786] app_voicemail.c: In find_user_realtime for mailbox 101 context 708 [Mar 26 16:40:24] VERBOSE[14269] logger.c: Really destroying SIP dialog '[EMAIL PROTECTED]' Method: OPTIONS [Mar 26 16:40:28] ERROR[19786] res_config_mysql.c: MySQL RealTime: Ping failed (2006). Trying an explicit reconnect. [Mar 26 16:40:28] DEBUG[19786] res_config_mysql.c: MySQL RealTime: Server Error (2006): MySQL server has gone away [Mar 26 16:40:28] DEBUG[19786] res_config_mysql.c: MySQL RealTime: Successfully connected to database. [Mar 26 16:40:28] DEBUG[19786] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM mailboxes WHERE mailbox = '101' AND context = '708' The database and the connection seems to be ok. It is only this query that's taking long. We are experiencing this on Asterisk 1.4.17 and 1.4.18 but difficult to reproduce. Anybody have any idea what might be the cause and how to procceed and figure out what's wrong? Thanks in advance. R ___ -- 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] Help with cisco 7960 phone
Enable NAT on the phone itself and leave it enabled in *. Jerry Geis wrote: I have a cisco 7960 phone. Worked fine in the office. I took it home. At home I have a linksys router that the phone is plugged into. The linksys router has DHCP enabled. I am getting the following error on the console from the 7960. I have tried it with nat=yes and nat=no in the sip.conf file. --- Transmitting (NAT) to 192.168.1.69:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.69:5060;branch=z9hG4bK1415360297;received=192.168.1.69;rport=5060 From: Display Name sip:[EMAIL PROTECTED];tag=1683635072 To: Display Name sip:[EMAIL PROTECTED];tag=as4c59a734 Call-ID: [EMAIL PROTECTED] CSeq: 3091 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=4a1c350c Content-Length: 0 -- The username and secret are the same as they were in the office when it worked. I figure it has to be something easy but I have not found it yet. the sip.conf entry for this phone is: [570] type=friend dtmfmode=rfc2833 username=570 secret=XXX disallow=all allow=ulaw allow=alaw host=dynamic context=local-sip callerid=Home 570 570 nat=no What might I try to get the phone working from home? Thanks, Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- 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] Star Wars Echo Sound
them, then a few seconds lagging behind, you'll hear a muffled (darth vader) version of the same thing. I had a similar experience where people claimed it sounded like a 'Dalek' (yes, in UK). The sound is somewhat similar to Darth Vader, I suppose. This was down to a buggy Echocancellation/Silence Detection implementation in the softphone (iaxcomm). Conrad ___ -- 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] Cisco 7971
What are you trying to do? I run a 7970 here with SIP. Thanks, Matt On Thu, Mar 27, 2008 at 7:02 AM, J. Oquendo [EMAIL PROTECTED] wrote: Anyone have some up-to-date (within the past 3 months) on Asterisk and the 7971. Searched voip-info, Google, etc., etc., to no avail. Documentation I found was scattered, vague. Thanks in advance. -- J. Oquendo SGFA #579 (FW+VPN v4.1) SGFE #574 (FW+VPN v4.1) wget -qO - www.infiltrated.net/sig|perlhttp://www.infiltrated.net/sig%7Cperl http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x3AC173DB ___ -- 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] Developer Conference, Aug 5-7, Chicago
Question: is anyone planning on going to the Cluecon convention this year? (www.cluecon.com http://www.cluecon.com/ ) I'm hoping to go this year and I'm hoping to meet other OSS telephony users and developers. BTW, Anthony Minessale said that there is a need for Asterisk speakers, so if you're an Asterisk user (or expert) and you're in the Chicago area in early August then perhaps you could check out the conference and possibly even be a guest speaker... I'm sure that the attendees would like to hear from developers and contributors about their experiences the past year with 1.4 as well as what's happening with 1.6 beta. -MC ___ -- 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] IAXy device
It's not bad in the sense of stability (well the original ones are claimed to have overheating issues..). But its that it lacks ANY features. The IAXy has no features at all. Also no security, it MUST be placed behind a firewall, as the configuration doesn't have any sort of security whatsoever. Did I mention it has no features besides DHCP? Not even DNS. Also it's very expensive. I could understand if it was a full-featured device with a webinterface, DNS support 2 Ethernet phone ports I wouldn't complain of the price. But it was released at approx USD $100 at a time when most full-featured adapters sold for a little less, and still sells for $90 today. If they sold them for $40 I wouldn't bash them either.. because honestly thats what they really should be worth. I'd rather use a Grandstream HT than an IAXY honestly. On Thu, Mar 27, 2008 at 3:08 PM, Steve Totaro [EMAIL PROTECTED] wrote: I had a customer using an IAXY (old gen) for an FXO fax machine and it worked almost all the time so it cannot be that bad. Maybe because the fax was very old and did not have high transmit rates. Thanks, Steve Totaro On Thu, Mar 27, 2008 at 2:11 PM, Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote: I guess I've never run asterisk without ANY echo cans :) It's just that the echo was minor enough that MG2 et. al did a fine job. Thanks! Moj Eric Wieling wrote: You will never get latency on a network low enough for echo to be perceived as sidetone (like on analog). If you want to get rid of echo you must cancel echo. Mojo with Horan Company, LLC wrote: Sean Dennis wrote: bilal ghayyad wrote: Hi All; I have been chocked just when I saw some posts talking about how much the IAXy is bad :) - So I would like to ask, did any one try it later and wether it is good or not? I am asking this because I need to use it as it is NAT Transparent (as I read also, and I did not try it to see how much it is transparent). What about codec? Why it is only support g711 and does not support compressed codec? And what about the IP address and the DNS usage and the DDNS usage? What main porblems contain and any advise? Regards Bilal The device has no echo cancellation and sounds horrible (lots of echo) on about half of the analog phones I tried it on. I wouldn't recommend it unless you absolutely need IAX. It's also very expensive for a 1 port ATA. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Echo may be the result of latency on the network. I've not had any echo problems that I remember with my IAXy and I make ten calls a day, five days a week, for the last few years, to all sorts of numbers/areas. I know that this isn't representative of typical business use, but residential use, but I've been using in my business and have never been disappointed :) I will agree that's is fairly expensive, but I WOULD recommend it to people who are on the go often. After setup, it really is plug-n-play IMO. Moj ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- 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] callers in queue passed to agents who accept only one call at a time
On Thu, Mar 27, 2008 at 6:32 PM, Vieri [EMAIL PROTECTED] wrote: I have a queue I configured as strict and a cron script I use to QueueAdd and QueueRemove agents according to my company's requirements. Usually I have 2 or 3 agents at a time and the ring strategy is ringall. These agents use non-open-source Windows softphones that do not let you configure it so that if they're on the phone, a second call will be rejected (agent busy). Instead, it's as if they had call waiting and incoming calls keep popping up while they're conversating with the first caller and they would like to avoid this. I guess the easiest solution would be to find an open-source or free softphone that can be configured to accept only one call at a time (currently using SJphone). Another solution would be if I could tell the Queue() application that if an agent is InUse then don't pass the call. Still another yet more delicate solution would be to have a custom script receive manager events related to the queue which in turn replies with an agi command. For example, whenever an agent answers a call I think that an event such as QueueMemberStatus can be triggered (although I don't know how). If the custom script could receive this event in realtime then it would run an agi command such as QueueRemove(busyagent...). When the agent is free again I suppose the same event is triggered and the custom script can QueueAdd(freeagent...). Could anyone please give me some pointers on this? In queues.conf set ringinuse=no Also make sure that you don't use realtime sip peers (or use rtcachefriends with that). Probably you also need call-limit set to any value in sip.conf For more info see http://www.voip-info.org/wiki-Asterisk+config+sip.conf Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- 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] callers in queue passed to agents who accept only one call at a time
calllimit in sip.conf and you are done Vieri escribió: I have a queue I configured as strict and a cron script I use to QueueAdd and QueueRemove agents according to my company's requirements. Usually I have 2 or 3 agents at a time and the ring strategy is ringall. These agents use non-open-source Windows softphones that do not let you configure it so that if they're on the phone, a second call will be rejected (agent busy). Instead, it's as if they had call waiting and incoming calls keep popping up while they're conversating with the first caller and they would like to avoid this. I guess the easiest solution would be to find an open-source or free softphone that can be configured to accept only one call at a time (currently using SJphone). Another solution would be if I could tell the Queue() application that if an agent is InUse then don't pass the call. Still another yet more delicate solution would be to have a custom script receive manager events related to the queue which in turn replies with an agi command. For example, whenever an agent answers a call I think that an event such as QueueMemberStatus can be triggered (although I don't know how). If the custom script could receive this event in realtime then it would run an agi command such as QueueRemove(busyagent...). When the agent is free again I suppose the same event is triggered and the custom script can QueueAdd(freeagent...). Could anyone please give me some pointers on this? Thanks! Vieri 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 smime.p7s Description: S/MIME Cryptographic Signature ___ -- 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] Unable to establish handshaking with fax machine
mark morreny wrote: I am simulating the sending of fax using sendfax through voip Ooops. Please see: http://hylafax.sourceforge.net/docs/fax-over-voip.pdf 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
[asterisk-users] SPA-962+ SPA-932- blf function
Asterisk Users, I am running Asterisk 1.4.11 on Debian Etch system with the TDM03B wildcard. I recently purchased a SPA-962 and SPA-932- the sidecar for our receptionist. After reading many forum postings on how to configure the side car, I uprgraded the SPA-962 software to 5.1.18(SC) version. I got the sidecar to subscribed to an extension on the Asterisk server, but the LED state on the SPA-932 never changes even when I am a call with that extension on another VOIP phone- SPA-941. I got the speed dial function to work, but the blf function does not appear to work. Did anybody get the blf function to work? What I am doing wrong? Any input would be greatly appreciated. Thanks in advance. Regards, John _ How well do you know your celebrity gossip? http://originals.msn.com/thebigdebate?ocid=T002MSN03N0707A___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problem about voice when using TDM2400p with VPMADT032 echo canceller module
link with the same problem: http://www.asteriskguru.com/archives/asterisk-users-tdm2400-hardware-echo-cancel-vt96394.html?highlight=tdm2400 nobody can solve the problem ? Vu AnhTuan [EMAIL PROTECTED] wrote: hi you, I'm having problem with voice quality on my trixbox using TDM2400B.The trixbox is connected via 20 FXO ports on a TDM2400 with the hardware echo cancel module. Echo cancel almost works, but the users hear what they describe as a 'background crackle/buzz' coming back when they talk. anyone have the same problem? pls help me. thanks a lot. my trixbox and config file: trixbox version 2.4 (Linux kernel 2.6.18, Zaptel 1.4.7) zaptel.conf # Autogenerated by /usr/sbin/genzaptelconf -- do not hand edit # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # # It must be in the module loading order # Span 1: WCTDM/0 Wildcard TDM2400P Board 1 fxsks=1 fxsks=2 fxsks=3 fxsks=4 fxsks=5 fxsks=6 fxsks=7 fxsks=8 fxsks=9 fxsks=10 fxsks=11 fxsks=12 fxsks=13 fxsks=14 fxsks=15 fxsks=16 fxsks=17 fxsks=18 fxsks=19 fxsks=20 # channel 21, WCTDM, no module. # channel 22, WCTDM, no module. # channel 23, WCTDM, no module. # channel 24, WCTDM, no module. # Global data loadzone = us defaultzone = us zapata.conf -- ; Zapata telephony interface ; ; Configuration file [trunkgroups] [channels] language=en context=from-zaptel signalling=fxs_ks rxwink=300 ; Atlas seems to use long (250ms) winks ; ; Whether or not to do distinctive ring detection on FXO lines ; ;usedistinctiveringdetection=yes usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no ;default ;echotraining=800 ;default rxgain=0.0 txgain=0.0 group=0 callgroup=1 pickupgroup=1 immediate=no busydetect=yes busycount=0 relaxdtmf=yes ;faxdetect=both faxdetect=incoming ;faxdetect=outgoing ;faxdetect=no ;Include genzaptelconf configs #include zapata-channels.conf group=1 ;Include AMP configs #include zapata_additional.conf zapata_additional.conf --- ; Autogenerated by /usr/sbin/genzaptelconf -- do not hand edit ; Zaptel Channels Configurations (zapata.conf) ; ; This is not intended to be a complete zapata.conf. Rather, it is intended ; to be #include-d by /etc/zapata.conf that will include the global settings ; ; Span 1: WCTDM/0 Wildcard TDM2400P Board 1 ;;; line=1 WCTDM/0/0 signalling=fxs_ks callerid=asreceived group=0 context=from-zaptel channel = 1 context=default ;;; line=2 WCTDM/0/1 signalling=fxs_ks callerid=asreceived group=0 context=from-zaptel channel = 2 context=default ;;; line=3 WCTDM/0/2 signalling=fxs_ks callerid=asreceived group=0 context=from-zaptel channel = 3 context=default ;;; line=4 WCTDM/0/3 signalling=fxs_ks callerid=asreceived group=0 context=from-zaptel channel = 4 context=default ;;; line=5 WCTDM/0/4 signalling=fxs_ks callerid=asreceived group=0 context=from-zaptel channel = 5 context=default ;;; line=6 WCTDM/0/5 signalling=fxs_ks callerid=asreceived group=0 context=from-zaptel channel = 6 context=default ;;; line=7 WCTDM/0/6 signalling=fxs_ks callerid=asreceived group=0 context=from-zaptel channel = 7 context=default ;;; line=8 WCTDM/0/7 signalling=fxs_ks callerid=asreceived group=0 context=from-zaptel channel = 8 context=default ;;; line=9 WCTDM/0/8 signalling=fxs_ks callerid=asreceived group=0 context=from-zaptel channel = 9 context=default ;;; line=10 WCTDM/0/9 signalling=fxs_ks callerid=asreceived group=0 context=from-zaptel channel = 10 context=default ...more... [IP-PBX ~]# ztcfg -vv -- Zaptel Version: 1.4.7-3259 Echo Canceller: OSLEC Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) Channel 03: FXS Kewlstart (Default) (Slaves: 03) Channel 04: FXS Kewlstart (Default) (Slaves: 04) Channel 05: FXS Kewlstart (Default) (Slaves: 05) Channel 06: FXS Kewlstart (Default) (Slaves: 06) Channel 07: FXS Kewlstart (Default) (Slaves: 07) Channel 08: FXS Kewlstart (Default) (Slaves: 08) Channel 09: FXS Kewlstart (Default) (Slaves: 09) Channel 10: FXS Kewlstart (Default) (Slaves: 10) Channel 11: FXS Kewlstart (Default) (Slaves: 11) Channel 12: FXS Kewlstart (Default) (Slaves: 12) Channel 13: FXS Kewlstart (Default) (Slaves: 13) Channel 14: FXS Kewlstart (Default) (Slaves: 14) Channel 15: FXS Kewlstart (Default) (Slaves: 15) Channel 16: FXS Kewlstart (Default) (Slaves: 16) Channel 17: FXS Kewlstart (Default) (Slaves: 17) Channel 18: FXS Kewlstart (Default) (Slaves: 18) Channel 19: FXS Kewlstart (Default) (Slaves: 19) Channel 20: FXS Kewlstart (Default) (Slaves: 20) 20 channels to configure. cat
[asterisk-users] Newbie Polycom: DHCP/boot server supporting 2 models of phones
I have a question about DHCP and boot server supporting more than 1 model of Polycom phones. According to Polycom standards, Polycom phone boots up to get a DHCP address and at the same time getting a boot server string (with username and password) to logon to boot server to download SIP, bootROM and etc. That is okay if there is only one type of phone (that requires a specific SIP and bootROM release). What about if the environment has to support two or more models of Polycom phones? On the boot server side, I can define another home directory like /home/polycom1 and /home/polycom2 to store different SIP and bootROM releases. However, the issue is how different polycom phone model can get a different user account and password to log on to different home directories. I understand the issue here is DHCP and not the boot server but I am a bit of a newbie here. Can anyone help please? ___ -- 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] Newbie Polycom: DHCP/boot server supporting 2 models of phones
For this, I would recommend using a smart DHCP device, which supports the passing of 'option 66' - for example, the edgemarc series of routers. With that, you could pass ftp://user1:[EMAIL PROTECTED] via dhcp in order to provision the phone, and different credentials if you are concerned about mixing up bootroms, and application loads. As far as I am aware ( I think I tried this once) it is not possible to pass ftp://username:[EMAIL PROTECTED]/directory1 via option 66. I am not too sure how much of an issue the versions is is with polycoms, in their reseller training, they give a list of upgrade paths, but I am not sure if this affects new versions, which may just be upgradable out of the box without any clashes Robert On Thu, Mar 27, 2008 at 9:38 PM, Lee, John (Sydney) [EMAIL PROTECTED] wrote: I have a question about DHCP and boot server supporting more than 1 model of Polycom phones. According to Polycom standards, Polycom phone boots up to get a DHCP address and at the same time getting a boot server string (with username and password) to logon to boot server to download SIP, bootROM and etc. That is okay if there is only one type of phone (that requires a specific SIP and bootROM release). What about if the environment has to support two or more models of Polycom phones? On the boot server side, I can define another home directory like /home/polycom1 and /home/polycom2 to store different SIP and bootROM releases. However, the issue is how different polycom phone model can get a different user account and password to log on to different home directories. I understand the issue here is DHCP and not the boot server but I am a bit of a newbie here. Can anyone help please? ___ -- 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] SPA-962+ SPA-932- blf function
We have BLF buttons working fine on the SPA932 side-car. What does show hints tell you under Asterisk, and what syntax did you use when configuring the side-car buttons? John Meksavan wrote: Asterisk Users, I am running Asterisk 1.4.11 on Debian Etch system with the TDM03B wildcard. I recently purchased a SPA-962 and SPA-932- the sidecar for our receptionist. After reading many forum postings on how to configure the side car, I uprgraded the SPA-962 software to 5.1.18(SC) version. I got the sidecar to subscribed to an extension on the Asterisk server, but the LED state on the SPA-932 never changes even when I am a call with that extension on another VOIP phone- SPA-941. I got the speed dial function to work, but the blf function does not appear to work. Did anybody get the blf function to work? What I am doing wrong? Any input would be greatly appreciated. Thanks in advance. Regards, John How well do you know your celebrity gossip? Talk celebrity smackdowns here. http://originals.msn.com/thebigdebate?ocid=T002MSN03N0707A ___ -- 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] Newbie Polycom: DHCP/boot server supporting 2 models of phones
All Polycom phones use the same firmware and bootroms - one reason why the sip.ld is so damn large for them. Lee, John (Sydney) wrote: I have a question about DHCP and boot server supporting more than 1 model of Polycom phones. According to Polycom standards, Polycom phone boots up to get a DHCP address and at the same time getting a boot server string (with username and password) to logon to boot server to download SIP, bootROM and etc. That is okay if there is only one type of phone (that requires a specific SIP and bootROM release). What about if the environment has to support two or more models of Polycom phones? On the boot server side, I can define another home directory like /home/polycom1 and /home/polycom2 to store different SIP and bootROM releases. However, the issue is how different polycom phone model can get a different user account and password to log on to different home directories. I understand the issue here is DHCP and not the boot server but I am a bit of a newbie here. Can anyone help please? ___ -- 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
