RE: [Asterisk-Users] Problem with T1 PRI line resetting/droppingcalls.
I'm the one who posted the original question, and yes, I *am* experiencing dropped calls. I haven't yet been able to correlate the dropped calls with anything, and that is why I asked the question about the PRI warnings and notices. Also, our volume of calls is very light...maybe 3-4 calls/hour. But, the dropped calls are very frequent and predictable...predictable in the sense that if you plan to talk for more than 5 minutes, there is a high likelyhood your call will be dropped. We are running on a relatively small system...700mz celeron, 128mb, slow disks, and a bios that tends to like to put as much as possible on the same IRQ. So, I am hoping the messages, and the solution to our problem with dropped calls has to do with a shortage of resources. We plan to upgrade the system to a 3gz P4 in the next few days, and I will post the results of that upgrade when I determine if the problem goes away. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Fri, 4 Jun 2004, Gary Franczyk wrote: I don't think the question you answered is the same as the one I asked. The problem with mine is the dropping of all the lines/calls. It resets all the lines. I get those mystery notices on occasion also, but they don't drop all the lines until I see the Detected Alarm messages for each line. I just signed up for this list, so I might have missed your post. Did the person you were answering see the same problem that I do: Dropping of all the lines? Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Eric Wieling Sent: Friday, June 04, 2004 2:00 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Problem with T1 PRI line resetting/droppingcalls. On Fri, 2004-06-04 at 12:09, Gary Franczyk wrote: I am having a serious problem with my Asterisk system. Every few days, my PRI line resets and drops all calls. I get these errors in the messages log: Jun 3 02:41:11 WARNING[11276]: PRI: Read on 39 failed: Unknown error 500 Jun 3 02:41:11 NOTICE[11276]: PRI got event: 6 on span 1 Jun 3 02:41:11 WARNING[12301]: Detected alarm on channel 1: Red Alarm I answered this same question *today*: From: Eric Wieling [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Mystery PRI NOTICEs WARNINGs Date: Fri, 04 Jun 2004 10:53:42 -0500 On Fri, 2004-06-04 at 09:07, Bruce Komito wrote: Since connecting a PRI to a Digium T100P, I have been seeing the following messages in syslog every few minutes: Jun 4 06:51:54 pbx asterisk[13435]: WARNING[1209214400]: chan_zap.c:6176 in zt_pri_error: PRI: Read on 56 failed: Unknown error 500 Jun 4 06:51:54 pbx asterisk[13435]: NOTICE[1209214400]: chan_zap.c:6913 in pri_dchannel: PRI got event: 8 on span 1 The new Zaptel has the zttest program, which will tell you if interrupts are being locked for too long. This can cause the Error 500 messages you are seeing. Causes of interrupts being locked for too long can be: graphics, frame buffer, IDE DMA. There might be other things that cause this error, I don't know. -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] scandsp, voicetronix and rxfx
[EMAIL PROTECTED] wrote: Hi all, Trying to get fax reception going using a voicetronix openline4 card, however, there are two issues (as far as I can see) (1) Currently the voicetronix world (cards, firmware, drivers, channel prog etc.) does not do fax tone detection. I have spoken to the voicetronix guys , understand what has to happen there and may do some hacking if I can resolve the second issue. Meanwhile, I guess I can do some sort of manual (or pseudo manual) transfer to a spandsp extension. (2) More worringly, when I do force a transfer to rxfax, it does not appear to train with the fax machine that I am using (samsung sf-330). Here is a snippet of log: -- Executing SetVar(vpb/1-1, FAXFILE=/tmp/1086048253.8.tif) in new stack -- Executing RxFAX(vpb/1-1, /tmp/1086048253.8.tif) in new stack Changed from phase 0 to 1 Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Start receiving document Changed from phase 1 to 4 Sending ident CSI: 40 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 DIS: Preferred octets: 256 Can receive fax Supported data signalling rates: V.27ter and V.29 R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85 R8x15.4lines/mm OK Minimum scan line time for higher resolutions: T15.4 = T7.7 DIS: 80 00 ce f0 80 80 01 HDLC underflow in state 9 Changed from phase 4 to 3 T4 timeout in state 9 Changed from phase 3 to 4 Sending ident CSI: 40 08 13 7c 58 f0 7a 15 af 14 c6 ef 10 e7 61 0b 44 1b 20 fb 59 DIS: Preferred octets: 256 Can receive fax Supported data signalling rates: V.27ter and V.29 R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85 R8x15.4lines/mm OK Minimum scan line time for higher resolutions: T15.4 = T7.7 DIS: 80 00 ce f0 80 80 01 T2 timeout Start receiving document ... and this repeats ad nauseum until the fax machine gives up. Any ideas what the issue might be anyone? Could this be voicetronix card-related? Thanks in advance. Greg. Looks like the audio does not reach rxfax Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MYSQL asterisk configuration
Hello We are just working on a billing sysem and config from mysql. so we`ll release it soon and it`ll be distributed under GPL. It`ll take some time (about one more month) so be patient untill then ;) Best Regards Hekuran -- Hekuran Doli Yjet e erenikut 14/2 Gjakova/Kosova [EMAIL PROTECTED] www.ati-kos.com +37744387555 f lamed? I hope not. I have already started reading up on mysql and c and Perl and xml and java and r... So many things I need to get working so little knowledge of coding and so little time. All I can offer anyone right now is good will and future benefits from anything code we eventually develop. I just need help getting started. Oh well, if someone wants to flame me so be it, Ill get over it and they will too. :-) - Terry, not sure if this will help you, since i am not using it but here it is anyway: http://www.codecharge.com/index2.php PSS... and as Scott mentioned... can you send your messages in plain text NOT html . ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DSP Coding
Ouch. The 64-port Ai-Logix board goes for (!) $3559 USD. Who is _paying_ for this stuff?!? That's 5x multiplier on what I pay for Asterisk, including the $10 per channel G.729 license. I will say that I'm looking for a good, dense DSP card that can be bodged into Asterisk. I have a line on a DS-3 card that has Linux drivers (channelized to DS-0 level) and I really want to run a DS-3's worth of G.729 or iLBC calls out of a single dual-proc 1u machine, just to say it's been done. However, that is impossible without echo cancellation and offboard DSP's to handle the real number crunching... I've done computations with costs based on RLX blade servers doing transcoding after offload via TDMoE, and it's still pretty pricey to get a DS-3's worth of calls - still in the $25,000-$35,000 range, depending on how you count equipment. That's not to mention the development of the DS-3 interface software... If anyone knows of a good DSP PCI card that could be put to use for G.729/iLBC/GSM/G.726 transcoding, let me know. Never heard about using the GPU to do anything useful. Sounds interesting. Got links? Commetrix: only seems to have a board capable of ~30 high-complexity channels Bittware: maybe. Website not immediately clear. Signalogic: http://www.signalogic.com/index.pl?page=sigc67xx_pci maybe. Aculab: http://www.aculab.com/products/dna/ip_telephony_card.htm only 60 channels. There are about a zillion companies out there selling CompactPCI solutions; someone MUST have a decent generic DSP board on standard PCI that can handle 672 high-complexity calls... JT At 9:42 PM +0800 on 6/3/04, Leo Ann Boon wrote: Heard good things about this card from some of my associates. quite expensive though http://www.ai-logix.com/smartdsp_vr_1.html A wild idea, why not use the graphics card GPU to do the compression? I understand there're some work on using the GPU to do SIMD. Supposed to be faster than the P4 and definitely cheaper than a dedicated dsp board Miroslav Nachev wrote: Hi, I would like to find some way for hardware coding instead software (using the Host CPU). Are there any PCI boards just with codecs (DSP) or other way? Best Regards, Miroslav Nachev [silly .sig snipped] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Transfer with Budgetone
John Fraizer wrote: Oh, and if you like your Grandstream, don't ever handle, let alone USE a Cisco phone. You'll be ruined for life if you do. My grandstream is now a toy phone for my 1 y/o son. It never was much more than a toy to begin with. Oh, and if you like you're money, don't ever order, much less PAY FOR a Cisco phone. You'll be financially ruined for life if you do. My Cisco 7960 is in a museum now where people pay me $1 a look, but if they blink during the look, because it's a Cisco phone, that adds another quarter per blink. It never was much more than a rich man's toy to begin with. In other words, we have Grandstreams all over tarnation; they work just fine for what they cost, and I can seat a lot more people on VoIP with them than I can with Cisco's regally-priced offerings. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ISDN and incoming MSN
I have installed a Billion ISDN card in my Asterisk. Calls between sip and isdn work. The i4l channel has MSN=26. I also put incomingmsn=26,27 in modem.conf. extensions.con: [incoming-isdn] exten = s,1,Dial(SIP/701,20,Ttr) ;will make my extension 701 ring, while exten = 26,1,Dial(SIP/701,20,Ttr) ;will cause an error message in the Cli interface: WARNING[196621]: pbx.c:1814 ast_pbx_run: Channel 'Modem[i4l]/ttyI0' sent into invalid extension 's' in context 'default', but no invalid handler -- Hungup 'Modem[i4l]/ttyI0' Of course, I would like to be able to distinguish between called MSN's so as to be able to experiment with different ways of handling calls. Any suggestions? Thor ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Meetme with moderator
I am having trouble finding links to continue after hangup based on dial.
Can you send me something on that ?
Umar.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Bruce Marler
Sent: 05 June 2004 05:27
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Meetme with moderator
Florian,All,
So I did what was noted below, here what I run into though, how do I set the
DB entry for moderator after the moderator hangs up. I just read the other
posts to the list about continuing after a hangup, but that is based on the
dial command and not meetme.
Here is what I have setup now for my simple testing, basically i dial 100 it
asks for conf # , if they dial 101 it sets moderator code to 1, if they dial
102 it checks it and lets them in if it is 1, unfortuatnely it stays 1 even
after the moderator hangups:
[conferences]
exten = 101,1,Answer
exten = 101,2,Wait(1)
exten = 101,3,DBput(Moderator/5=1)
exten = 101,4,Meetme(5)
exten = 101,6,Hangup
exten = 102,1,Answer
exten = 102,2,Wait(1)
exten = 102,3,DBget(5Admin=Moderator/5)
exten = 102,4,Gotoif($[${5Admin} = 1]?5,1:5550001:1)
exten = 100,1,Answer
exten = 100,2,Wait(1)
exten = 100,3,DigitTimeout,5
exten = 100,4,ResponseTimeout,8
exten = 100,5,BackGround(enter-conf-call-number)
exten = 100,6,Waitexten(20)
exten = 100,7,Goto(100,5)
exten = 5,1,Meetme(5)
exten = 5,2,hangup
-Original Message-
I have been beating my head against a wall trying to figure
out how I would implement a separate moderator code and
participant code for the same conference using meetme, the
deal is I dont want the participants to be able to join until
the moderator is in the conference.
Is it possible to do this using the apps as they are , or is
their a way to use an Agi script, is that the only way?
Simple extension logic is enough to do this:
From a certain extension or with a special pincode or whatever, have
moderator access. Be sure to set a database entry (/MMModerator/Roomnr/ = 1)
before accessing the MeetMe.
For all others, first check this database entry. Only access MeetMe if the
flag is set.
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Re: [Asterisk-Users] DSP Coding
Ouch. The 64-port Ai-Logix board goes for (!) $3559 USD. Who is _paying_ for this stuff?!? Quite a lot of people. I know of at least 1 brand of voice logger that use these boards for real-time compression. It was pretty much necessary considering that some of those loggers were built on 486,586 back-planes. snip Never heard about using the GPU to do anything useful. Sounds interesting. Got links? Take a look at brookgpu, http://graphics.stanford.edu/projects/brookgpu/index.html. It's a compiler for Brook (C with streams) for nvidia,ati GPUs. According to one of the PPT on the site, it's possible to get 2-3x speed up over the CPU. snip -Leo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problem with T1 PRI line resetting/droppingcalls.
That sounds like a timing issue, and that slips are occuring, Ensure all cables are good, reolace if necessary and ensure you are timing off of the network and not providing your own timing. Jason At 23:38 04/06/2004 -0700, you wrote: I'm the one who posted the original question, and yes, I *am* experiencing dropped calls. I haven't yet been able to correlate the dropped calls with anything, and that is why I asked the question about the PRI warnings and notices. Also, our volume of calls is very light...maybe 3-4 calls/hour. But, the dropped calls are very frequent and predictable...predictable in the sense that if you plan to talk for more than 5 minutes, there is a high likelyhood your call will be dropped. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] change cisco ata 186 dial behaviour
I have ata-186 and grandstream connected to asterisk using sip. I have a voip account with ATP, in Australia. In order to ring HK, I need to dial 0011852. Grandstream behaves normally and send the whole series of digits and it connects ok. But ATA-186 somehow only allow only 11 digits. ON the console it was only 0011852. The last 4 digits got truncated. I have tried another trick. This time I prepend 0011 to the 11 digits. Again Grandstream works correctly. But ATA 186 again only sends 0011852. Very strange indeed. On another matter with ATA- 186, I cannot activate line 2 by putting entry in uid1, there is absolute dead. Would it be hardware issue?? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] CODEC and Fax
Kurt [EMAIL PROTECTED] wrote: When reading the feature section of *.ororgt mentions a/ululawwould that imply G711? Also, it said that fax is incomplete. Has there been any more development work on fax? Will * support t.38 anytime soon? Where abouts in *.ororgt did it mention a/ululawwould? -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Immediate partial pattern match
From what I have read in the wiki and in the source code the current pattern '.' will match one or more digits in the dialplan. This allows a match on a part of a dialed extension. E.g. '_0.' matches anything that starts with a 9 and has at least one more digit but not '0' by itself. Can a pattern be written that immediately matches the '0' as well? I want to have a single digit 0 from the incoming channel dial out on the outgoing channel while at the same time allowing the incoming channel to pass '0phone-number' as well. I would like to hand over outgoing calls as fast as possible to the trunk line. It seems easy enough to add a 'match anything, even nothing' to EXTENSION_MATCH_CORE or ast_extension_match. Peter -- Peter Svensson ! Pgp key available by finger, fingerprint: [EMAIL PROTECTED]! 8A E9 20 98 C1 FF 43 E3 07 FD B9 0A 80 72 70 AF Remember, Luke, your source will be with you... always... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco 7960 XML/Configs
Brian D'Arcy [EMAIL PROTECTED] wrote: (Article auto-converted from unnecessary HTML to nice plain text.) I ordered 10 7960s with SIP today (YAY!), I should have them on Monday! So, to be better prepared come Monday morning, I was wondering if anyone knew of any * compatible screen configs for things such as browsing VM, etc, yadda, yadda. I checked out the wiki about ADSI but from what I see, thats not really applicable in a SIP setup? Im guessing its going to be a more XML and static HTML based type of setup. If anyone can point me to some resources or has some scraps of examples laying around, Id love to take a look, as Im really stoked about doing some cool stuff with these phones and *. =) You might find the following page interesting: http://www-106.ibm.com/developerworks/wireless/library/wi-voip/ -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] DTMF and SIP
Title: Message Hi I am using the latest cvs version. I can call other remote systems via PSTN and navigate menu systems with key presses ok ! I am using IP2006 SIP phone. Rgds -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Santiago AguiarSent: 04 June 2004 21:03To: [EMAIL PROTECTED]Subject: Re: [Asterisk-Users] DTMF and SIPhi!I'm having the same problem, I'm connecting through a Planet VIP-450 ITG, and when I send a DTMF code I get a:WARNING: codec_ilbc.c:141 ilbctolin_framein: Huh? An ilbc frame that isn't a multiple of 50 bytes long from RTP (4)?I tried using different dtmf settings in sip.conf, but the message is still there. I don't have problems using a softphone...any ideas???saludos! santiago.Lee Norvall wrote: Hi Just tried that, and still the same with the same error! The spec for the phones includes rfc2833, so I don't think that is it. Rgds -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Justin Carlson Sent: 02 June 2004 19:23 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] DTMF and SIP have you tried commenting out the dtmf lines in your sip.conf we had similar problems with our snom 200's and after commenting out the dtmf lines in sip.conf asterisk reload they worked great :-) On Wed, 2004-06-02 at 11:36, Lee Norvall wrote: Hi I have 2 x SIP hand phones. I have set the DTMF to rfc2833 on the phones and tried both dtmfmode=rfc2833 and sipdtmfmode=rcf2833 (also tried inband) and I get the following error: june 2 17:21:10 WARNING[213006]: codec_ilbc.c:145 ilbctolin_framein: Huh? An ilbc frame that isn't a multiple of 50 bytes long from RTP (4)? This means that I cannot get access to voicemail from the handsets !!! Any clues??? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] BroadVoice usage?
DTMF=inband
Zac
---
Zac Amsler, Technical Team
WNOC.COM http://www.wnoc.com
Phone: (989) 896-3329 X 2000
If you are receiving this e-mail as a respones to a technical issue, Please
respond to me via the support e-mail address provided by your ISP.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Greg Hill
Sent: Friday, June 04, 2004 11:35 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] BroadVoice usage?
On Fri, 4 Jun 2004, Michael Swan wrote:
Yes, we do have a context for incoming calls -- it's used for not only
BroadVoice (which isn't working) and VoiceGlo and iConnectHere (which
are working.) And, yes, we do have a pattern match on our number in our
[general] context in extensions.conf. As of this morning, when we dial
our BroadVoice number, we get a fast busy and our Asterisk server is
never contacted.
similar thing happened when I tried to use voiceglo.. It appeared to be a
problem in the provisioning of the phone number. Their tech support never
responded to fix it, so I dumped 'em and now use Broadvoice instead.
As for outgoing, we did make the change to sip.broadvoice.com
(from proxy.broadvoice.com). This did make a change on our outgoing
calls: the BroadVoice server responded with:
Got SIP response 604 Does not exist anywhere back from
147.135.0.129
This is progress but it's still an error. :-( And, no: our Asterisk
server is not behind a NAT firewall.
I don't see fromuser= and fromdomain= in your config below.. try adding
those. In the SIP debug, you'll probably see that your asterisk server is
putting From : [EMAIL PROTECTED] in the messages to broadvoice. Their
server doesn't like that; it wants to see From:
[EMAIL PROTECTED] in the message. fromuser and fromdomain make
that happen.
PS, a question for Jay. What dtmf mode are you using? I haven't got dtmf
working inbound or out yet.
Greg
At 03:04 AM 6/3/2004 -0500, you wrote:
I have Broadvoice working on three lines. Had the same problems you
have in the beginning.
Trouble-shoot one thing at a time...
Incoming: Do you have a context for incoming calls set up? It belongs
in your [general] section. Once there, do you have a pattern or
extension matching your phone number? If Asterisk can't match the
extension for an incoming call, the call is rejected and BV tells you
it's busy.
Outgoing: Use host=sip.broadvoice.com. Don't use proxy. If your
asterisk sits behind a NAT firewall, it wouldn't hurt to add nat=yes
to your [broadvoice] section.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael Swan
Sent: Tuesday, June 01, 2004 11:51 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] BroadVoice usage?
Hi all,
I've been trying to use BroadVoice as a SIP service provider. They don't
officially
support * but are helpful when it comes to answering questions for setup
parameters. They claim they have no firewalls or access lists that need
to be set up so I can get access to their servers.
However, something's still not quite right when I use the parameters. It
looks like our Asterisk server is registered with their server and I
believe I have all the other entries set up correctly (we use other
services, too, such as iConnectHere and Voiceglo so I'm pretty sure I've
got my BroadVoice set up correctly.) sip show registry shows my *
server as registered with the BroadVoice IP address.
When I try to make an outbound call, there is no answer from the
BroadVoice server (proxy.boradvoice.com) to Asterisk's INVITE.
When I try to call our assigned BroadVoice number, I immediately get a
BroadVoice message saying the number is busy.
I can provide the sip debug output but it basically shows that
BroadVoice appears to be not communicating with inbound or outbound
requests.
Here are the entries:
sip.conf
register = phonenumber:[EMAIL PROTECTED]:5060/phonenumber
[broadvoice]
type=friend
username=phonenumber
secret=password
host=proxy.broadvoice.com
dtmfmode=inband
disallow=all
allow=ulaw
allow=alaw
extensions.conf
; calls via BroadVoice
exten = _6NXX,1,Dial,SIP/1925${EXTEN:[EMAIL PROTECTED]
exten = _61NXXNXX,1,Dial,SIP/${EXTEN:[EMAIL PROTECTED]
exten = _6011.,1,Dial,SIP/${EXTEN:[EMAIL PROTECTED]
exten = _6.,2,Congestion
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Re: [Asterisk-Users] Silly incoming SIP failure
Am 27.05.2004 um 21:26 schrieb Julian Pawlowski: i upgraded to the actual CVS head from yesterday (27.5.) but can not get incoming SIP calls from my provider (sipgate). If someone calls my number, my asterisk responds with the following error: May 27 21:30:21 NOTICE[1114606512]: chan_sip.c:6351 handle_request: Failed to authenticate user CallerID sip:CallerID@217.10.66.11;tag=as38e9693c Try to put insecure=very into the [sipgate.de] context. Mike ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] QoS in Cisco
I remember seeing a notice about a fix about a month ago, don't remember any specifics. The actual bug was a weird one and required simultaneous use of QoS output service policies, PBR, and multicast PIM-DM to happen. Timothy R. McKee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Friday, June 04, 2004 17:36 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] QoS in Cisco On Fri, 2004-06-04 at 15:02, Timothy R. McKee wrote: Here is what I use on a customer's router. He has a mix of different IP phones which make it a little strange, but it seems to work. Be aware that setting COS on an ethernet had severe bugs up until a service release a month or so ago. I haven't tested the fix yet. What IOS version contains the fix for QoS on Ethernet? -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] illegal instruction
On Saturday 05 June 2004 00:35, dkwok wrote: I have just compiled the latest cvs 040605 and have this illegal instruction error when launched asterisk. It is compiled on Via c5 processor. In the asterisk/Makefile I have set PROC=i586 but it does not help the situation. Disable the PROC setting altogether or patch the Makefile and any source files that might use it; It seems you're well aware of what is causing the problem. :-) -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DSP Tools Technical Support
Email: [EMAIL PROTECTED] FirstName: Miroslav LastName : Nachev Company : COSMOS Software Enterprises, Ltd. Phone: (+359-88) 897-31-95 Fax : Address : P. O. Box 941 Address2 : City : Sofia State: Outside the US, Mexico, or Canada Zip/Postal Code : 1000 Country : BULGARIA SupportType : DSP Tools Issue dsp_using: Other dsp_using_other : I don't know. G.729ab and other voice codecs platform : Other platform_other : PCI Card emulator : None software : None software_other : operating_system : Other operating_system_other: Linux target_os: Other target_os_other : Linux design_stage : Developing Hardware/Software application : Asterisk IP PBX open porject (www.asterisk.org) prob_description: I am looking for some low cost PCI Card running under Linux for DSP Voice processing (coding/transcoding) of G.729, GSM and other Voice Codecs. I would like to use this PCI Card together with the products of the Asterisk IP PBX open porject (www.asterisk.org). The variant of voice channel numbers are 4, 8, 12, 16, 30 and more. Can you give me some suggestions? Thank you in advance. email_copy : yes copy_to_email: [EMAIL PROTECTED] sent_to_email: [EMAIL PROTECTED]; [EMAIL PROTECTED] form_log_path: /support/dsp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with T1 PRI line resetting/droppingcalls.
On Saturday 05 June 2004 02:38, Bruce Komito wrote: I'm the one who posted the original question, and yes, I *am* experiencing dropped calls. I haven't yet been able to correlate the dropped calls with anything, and that is why I asked the question about the PRI warnings and notices. Also, our volume of calls is very light...maybe 3-4 calls/hour. But, the dropped calls are very frequent and predictable...predictable in the sense that if you plan to talk for more than 5 minutes, there is a high likelyhood your call will be dropped. Seems fairly easy to recreate then; can you get q.931 logs from Asterisk to try and help us out? That'll let us know if one side or the other is dropping them on purpose, and if you can so easily recreate the problem it might be possible to beg/borrow/steal a proper T1 test set (your telco should offer this in the first place) to make sure you're not experiencing anything unusual on the physical link. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Feedback needed! FindMe/FollowMe Feature Spec.
On 04/06/2004 at 14:36 James W. Brinkerhoff wrote: On Thursday 03 June 2004 07:05 pm, Andy Powell wrote: chan_btp Hi Brian, You might also like to take a look at chan_btp and the btp daemon which allows the use of bluetooth devices to change routing. Since any old linux box that can handle a bluetooth dongle can report back to a server you can have them all over the place. From what I saw you were looking at timed routing, adding bt to this might make your life a lot easier... Where can one find this chan_btp ... Google didn't turn up anything -jwb you can check it out from CVS - its called btp... :D Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] GS HandyTone Issue
I just got myself a GS HandyTone and it works great, it was a breeze to setup. My only issue is I seem to be hearing a humming noise on the line when I am in calls.. I am using the following: *Product Model: * HT286 *Software Version: * Program--1.0.4.71Bootloader--1.0.0.17 HTML--1.0.0.32VOC--1.0.0.6 This is out of my sip.conf: [201] type=friend secret=** host=dynamic context=intern canreinvite=no dtmfmode=rfc2833 mailbox=201 disallow=all allow=ulaw allow=alaw It seems to be on the device itself so I don't know if I am misconfigured? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk-Users digest, Vol 1 #4041 - 11 msgs
Hi,
You need to set the DialPlan parameter to allow the proper
number of digits to be collected, for all types of numbers
used in your system. I believe that the factory default
value would work for long numbers beginning 0011, but your unit
was probably previously configured for a different environment
or country. Below is an extract from the example in my H.323
firmware; I believe that it's the same for SIP.
# Dial Plan Parameters ===
# -
# Parameter: DialPlan
# Access Code: 926
#Type: Alphanumeric string (199 characters maximum)
#
# Description: Dial plan rules.
#
#Note: No syntax check is performed by the actual implementation.
# It is the responsibility of the provisioner to make sure that
# the dial_plan is syntatically valid.
#
# Programmable strings of dial plan that allow one to specify:
# o special rule -- I{timeout} to control default inter-digit
# timeout - specifying this rule also has the side effect
# of preventing non-matching dial string from being sent out.
# o optional send character to use (e.g. '#' or '*')
# o how many digits before auto send
# o send after timeout at any specified number of digits
# (time out can be changed as digits are entered).
# in the following:
# o . means match any digits
# o - means more digits can be entered, this (if needed) must
# appear at the end of the individual rule
# (i.e. e.g. 1408t5- is legal, but 1408t5-3...
# is illegal).
# o [] Range, means match any digit in the list. '_' indicate
# a range of digits. For example, [135] matches the
# digits 1, 3, and 5, [1_5] matches the digits 1, 2, 3,
# 4 and 5. No # or * is allowed in the range. Range
# doesn't work with repeat, and range can't include
# selection. (feature available after v3.0)
# o (nnn|nnn) Selection, means match any strings in the list.
# The string can be composed by any digit, #, *, ., -
# and range. Selection is not used with any prefix or
# suffix patterns. (feature available after v3.0)
# o # means terminating key to send is #, and termination
# can be applied only after matching hits # (So *
# means terminating char is *, i.e. terminating key
# must follow )
# o rules applied in the order of listed (whichever matched
# completely first will cause trigger the send).
# o tn means timeout is n seconds (note: n is 0-9 and
# a-z -- which ranges 0 to 26).
# o more than one rules are separated by |.
# o ^ Logical not, means match any character except the
# character immediately following the ^ command. ^ can
# be used as a negation before range and selection too.
# o rn means repeat last pattern (except range) n times (note:
# 1. # or tn are modifier, they are not pattern; 2. n is 0-9
# and a-z -- which ranges 0 to 26). Use the repeat modifier
# to specify more rules in less space.
#
# You can also use the modifier 'S' to sieze the rule matching
# (i.e. if a rule matches and the modifier 'S' is seen, all other
# rules after that matching rule will not be used for matching).
#
# Here is the summary of the dial plan rules:
#
# o In: set the default inter-digit time out.
# o Hnxxx: specify the hotline/warmline number. (since v2.14)
# o Pnxxx: specify the prefix. (since v2.14)
# o B: specify the base number. (since v2.16)
# o Rxxx(nnn|nnn): specify the prefix. (since v3.0)
# o C: specify the call blocking numbers. (since v3.0)
# o F: specify the call forwarding blocking numbers. (since
#v3.0)
# o X: specify the call blocking/call forwarding blocking
#numbers. (since v3.0)
# o D: displaying Caller ID. (since v3.0)
#
# More details are available in the eng114487.doc.
#
# Examples 1: The set of dial plan rules:
#
#.t7#..t4-|911|1t7#..t1-|0t4#.t7-
#
# or equivalently
#
#.t7#r6t4-|911|1t7#.r9t1-|0t4#.t7-
#
# consists of the following rules:
#
#
[Asterisk-Users] polycom soundpoint ip500 help
I just received a shipment of ip500's. They came with no documentation and a cd with a bunch of windows stuff on it. I could not find any config or load files on the cd. No problem. I found a pointer to config and load files via the wiki. Fired up the phone, gave it a static ip and watched it asking for tftp files. Copied the files in to place. It successfully download new boot and sip. Now the only thing it will do is just send out CDP packets. No display (other than initial polycom logo) or keyboard response. The folks at polycom explained to me that the reseller should be helping me, but they tried anyway. We were never able to bring it back to life or a factory default. Some how I seem to have turned on the cisco switch and do not know how to get it back. We tried all the magic multi button pushing and hand shakes. Anyone have an suggestions? Can anyone suggest a good polycom reseller that will provide boot and sip load images? I sure like the way these phones look and feel. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Configuring cisco 7940
I've just managed to get hold of a cisco 7940, which looks nice but I'm unable to make it actually do anthing...! All the online manuals say things like see your network administrator which isn't a whole lot of use. First thing I think I need to do is work out how to set the TFTP server IP as it's using the wrong one (it's ignoring the setting in the DHCP server). When you point a browser at the phone it gives you the settings but no opportunity to set them. Also, what is the code of the $8 support option and who sells it (it seems cisco don't sell direct to end users)? The cheapest I've seen is $100 and if it's that kind of price I'll just see how far I can get with the default firmware. Tony ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
FW: [Asterisk-Users] Meetme with moderator
OK, so I am an idiot, with the use if the h extension I can set the
moderator code to 0 on hangup. Cannot believe I missed that one.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Bruce Marler
Sent: Friday, June 04, 2004 11:27 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Meetme with moderator
Florian,All,
So I did what was noted below, here what I run into though, how do I set the
DB entry for moderator after the moderator hangs up. I just read the other
posts to the list about continuing after a hangup, but that is based on the
dial command and not meetme.
Here is what I have setup now for my simple testing, basically i dial 100 it
asks for conf # , if they dial 101 it sets moderator code to 1, if they dial
102 it checks it and lets them in if it is 1, unfortuatnely it stays 1 even
after the moderator hangups:
[conferences]
exten = 101,1,Answer
exten = 101,2,Wait(1)
exten = 101,3,DBput(Moderator/5=1)
exten = 101,4,Meetme(5)
exten = 101,6,Hangup
exten = 102,1,Answer
exten = 102,2,Wait(1)
exten = 102,3,DBget(5Admin=Moderator/5)
exten = 102,4,Gotoif($[${5Admin} = 1]?5,1:5550001:1)
exten = 100,1,Answer
exten = 100,2,Wait(1)
exten = 100,3,DigitTimeout,5
exten = 100,4,ResponseTimeout,8
exten = 100,5,BackGround(enter-conf-call-number)
exten = 100,6,Waitexten(20)
exten = 100,7,Goto(100,5)
exten = 5,1,Meetme(5)
exten = 5,2,hangup
-Original Message-
I have been beating my head against a wall trying to figure
out how I would implement a separate moderator code and
participant code for the same conference using meetme, the
deal is I dont want the participants to be able to join until
the moderator is in the conference.
Is it possible to do this using the apps as they are , or is
their a way to use an Agi script, is that the only way?
Simple extension logic is enough to do this:
From a certain extension or with a special pincode or whatever, have
moderator access. Be sure to set a database entry (/MMModerator/Roomnr/ = 1)
before accessing the MeetMe.
For all others, first check this database entry. Only access MeetMe if the
flag is set.
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RE: [Asterisk-Users] Configuring cisco 7940
Hi Tony, TH First thing I think I need to do is work out how to set the TH TFTP server IP as it's using the wrong one (it's ignoring TH the setting in the DHCP server). http://tinyurl.com/37fe4 HTH Darren -- Comgate TelcoInternetBroadcast -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Tony Hoyle Sent: 05 June 2004 19:22 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Configuring cisco 7940 I've just managed to get hold of a cisco 7940, which looks nice but I'm unable to make it actually do anthing...! All the online manuals say things like see your network administrator which isn't a whole lot of use. First thing I think I need to do is work out how to set the TFTP server IP as it's using the wrong one (it's ignoring the setting in the DHCP server). When you point a browser at the phone it gives you the settings but no opportunity to set them. Also, what is the code of the $8 support option and who sells it (it seems cisco don't sell direct to end users)? The cheapest I've seen is $100 and if it's that kind of price I'll just see how far I can get with the default firmware. Tony ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Configuring cisco 7940
Storer, Darren wrote: http://tinyurl.com/37fe4 Unfortunately those instructions don't seem to relate to my phone (eg. there's no option 6 on the 'Settings' menu). I've found some other documents which seem to help but am unable to change any of the settings even in the unlocked state - it all seems to be hardcoded. I eventually gave up and installed an extra tftp server so I could get an XMLDefault file onto it. Now for some reason it's trying to query the router for something (which isn't going to get very far as it's just a Netgear gateway). I'll try a few packet traces to see if I can fake the responses... presumably as shipped they assume you're running cisco routers etc. Tony ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Configuring cisco 7940
TH Unfortunately those instructions don't seem to relate TH to my phone (eg. there's no option 6 on the 'Settings' TH menu). Sorry Tony, those instructions work well for 12SP and VIP30 phones (although you have to know to use 1 to activate your changes as you exit at the end of the sequence). I'm sure one of the other list readers will be able to help - good luck! Darren -- ComgateInternetBroadcast -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Tony Hoyle Sent: 05 June 2004 21:56 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Configuring cisco 7940 Storer, Darren wrote: http://tinyurl.com/37fe4 Unfortunately those instructions don't seem to relate to my phone (eg. there's no option 6 on the 'Settings' menu). I've found some other documents which seem to help but am unable to change any of the settings even in the unlocked state - it all seems to be hardcoded. I eventually gave up and installed an extra tftp server so I could get an XMLDefault file onto it. Now for some reason it's trying to query the router for something (which isn't going to get very far as it's just a Netgear gateway). I'll try a few packet traces to see if I can fake the responses... presumably as shipped they assume you're running cisco routers etc. Tony ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Configuring cisco 7940
On Sat, 5 Jun 2004, Tony Hoyle wrote: Also, what is the code of the $8 support option and who sells it (it seems cisco don't sell direct to end users)? The cheapest I've seen is $100 and if it's that kind of price I'll just see how far I can get with the default firmware. Search the list. Look for sip 7960 firmware. http://google.com/ sip 7960 firmware site:lists.digium.com About the configuration: http://www.wheely-bin.co.uk/cisco/ It is not quite a configuration howto but you can grabe some information about how to configure your Cisco 7940. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Configuring cisco 7940
Hermann Wecke wrote: Search the list. Look for sip 7960 firmware. http://google.com/ sip 7960 firmware site:lists.digium.com That's no help.. read all of them. The best I can find out is the $8 price on the wiki is bogus and should be removed as it's misleading. The cheapest smartnet is CON-SNT-PKG1 at $75 per year. That's almost ten times what I expected I'd be paying. About the configuration: http://www.wheely-bin.co.uk/cisco/ It is not quite a configuration howto but you can grabe some information about how to configure your Cisco 7940. I got it working, sort of. Out of the box it's just a basic phone (doesn't even support caller ID) but I guess it looks cool sitting on the desk :) Tony ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN and incoming MSN
On Sat, 05 Jun 2004 22:13:00 +0200, [EMAIL PROTECTED] wrote: Hi I have the billion card as well, and I am trying to get it working with bristuff from Klaus. So far it wont even compile the zaphfc driver, so I am very stuck. Hi, I use SuSE Linux. Being from Germany, it has built-in support for ISDN. It is just a matter of defining it in Yast, the configuration program. SuSE recocnized the card and chose the HiSax driver for me. The second step was to make it visible to Asterisk. I modified modem.conf, so now it contains: [interfaces] context=remote stripmsd=0 dialtype=tone mode=immediate msn=26 incomingmsn=26,27,28 device = /dev/ttyI0 Of course, msn should point to one of your msn's. Device goes AFTER msn, I have read. Thor -- Using Opera's revolutionary e-mail client: http://www.opera.com/m2/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Configuring cisco 7940
I have a 7940 and it works fine, caller id etc. What firmware are you running? Are you using the sip firmware, does it say SIP on the top right hand corner of the display? Tony Hoyle wrote: I got it working, sort of. Out of the box it's just a basic phone (doesn't even support caller ID) but I guess it looks cool sitting on the desk :) Tony ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] * to Vonage Connection anyone?
Does anyone have any configuration info for the Vonage sip client? -Original Message- From: Greg Blakely [mailto:[EMAIL PROTECTED] Sent: Friday, June 04, 2004 11:59 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] * to Vonage Connection anyone? If you use their soft phone, it will work with Asterisk if you use port 5061 rather than port 5060. Incoming works well all the time; outgoing is somewhat problematic, especially if you are using Asterisk to proxy for one of your internal SIP phones. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Roy Sent: Friday, June 04, 2004 10:09 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] * to Vonage Connection anyone? Listonians, Anyone get * to work together with Vonage? Thanks, Jerry ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems with PPP internet T1
Patrick/All, I have updated http://www.voip-info.org/tiki-index.php?page=Asterisk+Data+Configuration page. Regards, Vasyl Patrick J. Conroy wrote: I've tried building the 2.4.21 and the 2.4.20 kernels with the appropriate hdlc patch and I continue to have the same results. I'm thinking this is a problem with the routing table rather than getting hdlc compiled correctly, but I'm pretty much at a loss at this point. I have tried the one route statement that I have seen on posts of other people using these boards with a data T: route add -net 64.80.211.0 netmask 255.255.255.252 gw 64.80.211.41 If anyone has any suggestions, I would love to see them. Thanks, Patrick -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Patrick J. Conroy Sent: Friday, May 28, 2004 6:28 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Problems with PPP internet T1 I have tried with and without CONFIG_OLD_HDLC_API. If I try to compile with just CONFIG_OLD_HDLC_API uncommented I get function not implemented when I run ztcfg. I tried with both options uncommented, just to see what would happen, and I got compile errors. If I leave both commented out, I get function not implemented when I run ztcfg. The only way I can get the data T1 to come up is if I compile with CONFIG_ZAPATA_NET uncommented and CONFIG_OLD_HDLC_API commented out. Also, I don't get any errors when I run sethdlc hdlc0 ppp to set up the device, but when I run sethdlc hdlc0 with no options, I get: Interface unknown: 0x1 It seems that people have had luck with the 2.4.21 kernel, so I am going to try to build that and see if I have any luck, but if anyone can help explain what is going wrong, I would greatly appreciate it. Thanks, Patrick -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Vasyl Rublyov Sent: Friday, May 28, 2004 4:45 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Problems with PPP internet T1 Patrick, I seen the problem with HDLC on kernels 2.4.20, but this is explainable. Did you try compile zaptel with CONFIG_OLD_HDLC_API option? Patrick J. Conroy wrote: We are using redhat 8 with kernel 2.4.18-14. We recompiled the kernel with the hdlc-2.4.20-1.14a.patch from http://hq.pm.waw.pl/hdlc/. That site stated that this was the patch to use for 2.4.20 and earlier kernels. The kernel seemed to compile and sethdlc seemed to compile fine and the hdlc module loads and we see the hdlc0 network device. Patrick -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Vasyl Rublyov Sent: Friday, May 28, 2004 12:11 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Problems with PPP internet T1 What is your kernel version? Patrick J. Conroy wrote: Hello all, We have a TE405P set up with span 1 running to a channel bank, a PRI running into span 2, and a PPP internet T1 running into span 3. We have the first 2 spans up and running without a problem. We have hdlc compiled into the kernel and after making the appropriate changes to zaptel.conf and loading the zaptel, wct4xxp, and hldc modules we can bring up the third span with the internet T1, but we can't seem to communicate with the ISP. We ran the following commands: sethdlc hdlc0 ppp ifconfig hdlc0 our serial ip pointopoint isp gateway ip netmask isp subnet mask -arp Now we can ping our serial ip, but can't ping the isp gateway ip. ifconfig shows us transmitting packets, but we don't receive any. Any help would be greatly appreciated. Thanks, Patrick -- Thanks and regards, Vasyl Rublyov IonIdea, Inc. 3913, Old Lee Highway, Suite 33B Fairfax, VA 22030 Tel: (703) 691-0400 Mob: (703) 395-0238 Fax: (703) 691-0401 www.ionidea.com A CMM Level III and ISO 9001 Company - This e-mail (including any attachments) is confidential and may be legally privileged. If you are not an intended recipient or an authorized representative of an intended recipient, you are prohibited from using, copying or distributing the information in this e-mail or its attachments. If you have received this e-mail in error, please notify the sender immediately byreturn e-mail and delete all copies of this message and any attachments. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content, and is believed to be clean. -- Thanks and regards, Vasyl Rublyov IonIdea, Inc. 3913, Old Lee Highway, Suite 33B Fairfax, VA 22030 Tel: (703) 691-0400 Mob: (703) 395-0238 Fax: (703) 691-0401 www.ionidea.com A CMM Level III and ISO 9001 Company - This e-mail (including any attachments) is confidential and may be legally privileged. If you are not an intended recipient or an authorized representative of an intended recipient, you are prohibited
Re: [Asterisk-Users] DSP Coding
..And I'd like a time machine, and a supercomputer, and a submarine too... I think you are way-over ambitious for a 1RU space. Heat, noise, all the bad things computers do get worse in confined spaces. Perhaps Cray, or someone will come out with a 1RU supercomputer soon. But I'd not hold my breath. Even the telco's breakout of a DS-3 takes more space than you think. How would you troubleshoot one DS0? (Very carefully I'd imagine) I guess we can wish however... (I can all ready imagine the Inermod/crosstalk, RFI of all those DSP's crammed into a confined space..brrr) I know you're wishing, maybe you'll hit the lottery too, and start designing boards to do exactly that...(grin) WE can wish too;) Marc I will say that I'm looking for a good, dense DSP card that can be bodged into Asterisk. I have a line on a DS-3 card that has Linux drivers (channelized to DS-0 level) and I really want to run a DS-3's worth of G.729 or iLBC calls out of a single dual-proc 1u machine, just to say it's been done. However, that is impossible without echo cancellation and offboard DSP's to handle the real number crunching... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FWD network from Asterisk through NAT
Hi there, I'm trying to dial into the FWD network using Asterisk, though a NAT. The sources I've read say that it's unconfirmed to work through a NAT, but I'm wondering if anyone's done it anyway. So, anyone got a clue how to do this? Hank ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FWD network from Asterisk through NAT
I use the new IAX service at FWD. Much easier than trying to sort out the whole proxy thing with SIP. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of hank smith Sent: Saturday, June 05, 2004 8:18 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] FWD network from Asterisk through NAT Hi there, I'm trying to dial into the FWD network using Asterisk, though a NAT. The sources I've read say that it's unconfirmed to work through a NAT, but I'm wondering if anyone's done it anyway. So, anyone got a clue how to do this? Hank ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Configuring cisco 7940
Tony Hoyle wrote: That's no help.. read all of them. The best I can find out is the $8 price on the wiki is bogus and should be removed as it's misleading. The cheapest smartnet is CON-SNT-PKG1 at $75 per year. That's almost ten times what I expected I'd be paying. Not true. I just bought the 7960 SPECIFIC support contracts, and they were $8.30 each. Here is what you need to be looking for: http://www.ams.net/public/products/product_info.cfm?Product_ID=7993 This was from a google search, but it's a little high. Search for con-snt-7940 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] * to Vonage Connection anyone?
For incoming, it's a simple entry in sip.conf: [general] register = 16126051544:[EMAIL PROTECTED]:5061/200 ; This will register username 16126051544 with password of QjrT56svW to server atlas3.atlas.vonage.net on port 5061. Incoming calls will ring to extension 200, as defined in extensions.conf ; Outgoing is a little trickier. I've had better luck with SIP using iConnectHere. And IAX providers make the easiest of all outgoing connections. (I use voicepulse). -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin Sent: Saturday, June 05, 2004 7:56 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] * to Vonage Connection anyone? Does anyone have any configuration info for the Vonage sip client? -Original Message- From: Greg Blakely [mailto:[EMAIL PROTECTED] Sent: Friday, June 04, 2004 11:59 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] * to Vonage Connection anyone? If you use their soft phone, it will work with Asterisk if you use port 5061 rather than port 5060. Incoming works well all the time; outgoing is somewhat problematic, especially if you are using Asterisk to proxy for one of your internal SIP phones. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Roy Sent: Friday, June 04, 2004 10:09 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] * to Vonage Connection anyone? Listonians, Anyone get * to work together with Vonage? Thanks, Jerry ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DSP Coding
Even the telco's breakout of a DS-3 takes more space than you think. How would you troubleshoot one DS0? (Very carefully I'd imagine) In software, naturally. A physical DS0 needn't exist. (I can all ready imagine the Inermod/crosstalk, RFI of all those DSP's crammed into a confined space..brrr) Why would you do something that crazy? You could put 8 high-end DSPs on a half-height PCI card and have each one handle a DS2's worth of channels (up to 96) and then have the 8th do general housekeeping of the entire DS3 and PCI interface. Why would you use one DSP per channel? -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DSP Coding
But even at that rate, that box is going to be mighty busy... Why would you do something that crazy? You could put 8 high-end DSPs on a half-height PCI card and have each one handle a DS2's worth of channels (up to 96) and then have the 8th do general housekeeping of the entire DS3 and PCI interface. Why would you use one DSP per channel? -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DSP Coding
At 10:19 PM -0400 on 6/5/04, Andrew Kohlsmith wrote:
Even the telco's breakout of a DS-3 takes more space than you think.
How would you troubleshoot one DS0? (Very carefully I'd imagine)
In software, naturally. A physical DS0 needn't exist.
(I can all ready imagine the Inermod/crosstalk, RFI of all those DSP's
crammed into a confined space..brrr)
Why would you do something that crazy? You could put 8 high-end DSPs on a
half-height PCI card and have each one handle a DS2's worth of channels (up
to 96) and then have the 8th do general housekeeping of the entire DS3 and
PCI interface. Why would you use one DSP per channel?
-A.
Andrew's points are correct.
There exist already cards that will do this, that are even PCI (CPCI,
though.) They tend to be crazy expensive despite relatively
inexpensive parts, as has already been noted. They also tend to be
surrounded in marketing gobbledy-gook that makes it impossible to
determine the true capabilities of the equipment without getting a
'sales engineer' to cut through the BS and tell you what the card
actually does. And as a last nail in the coffin, typically these
boards are part of larger architectures which are impossible to
purchase in individually useful or programmable components. (OH!
You want the SOFTWARE LICENSE, then, as well! That's a separate
contract and price sheet!)
We here in the Asterisk community sometimes fail to see the larger
possibilities that surround us, and focus only on what the hobbyist
or single IT person working alone can afford and understand. The
telephony hardware market is huge, and has an impressive array of
vendors producing some really nice cards. Alas, most of them are
overpriced because of the niche nature of some of this gear - if you
spend $300,000 developing the hardware, software, and certifications
for a card then you can't charge $750 for it, even though that might
be the cost of the chips and manufacturing.
We (the * community) have this single-minded focus because of the
items I mention in paragraph 1. If it's too difficult to understand,
purchase, or if it's too much money to afford experimentation, we
won't use it. That's a shame, since I think there could be some
really cool parallel-CPU stuff done with third party cards
(encryption, transcoding, echo cancellation, faxing) if they became
more available and approachable by the open-source community. Look
at the neat stuff that OpenBSD does with the PCI-based encryption
cards.
I expect a DS-3's worth of physical and transcoding traffic can be
pushed through a PCI bus machine and into Asterisk, if the
appropriate amount of 'real' development was put towards the effort.
('real' in this context equals a team of developers working full
time, for money.) I have some doubts if it could be marketed and
sold in a cost-effective manner by anyone other than Digium at this
point, though, so it's a moot point.
There have been discussions here on this list already on the
availability of boards like SBEI's channelized DS-3 card (they've
been a reasonably approachable vendor.) All that we need is what
Andrew describes (a few high-end DSP's on a card) and the software
extensions to glue all of that into Asterisk. Markets exist for such
a combination(I know - I've been in three firms now that would have
bought such a system) but the real revenues are out there in the land
of slick salespeople and big trade show booths, which jack up the
prices out of the range where anyone running Asterisk would be
interested. I think if that could be delivered for $5000 (not
including the PC) then there would be some buyers. Compare against
buying a used (not new) Cisco DS-3 card for a 58xx or a Quintum or a
Nuera with the same capacity. I will say that the big problem with
this whole discussion is that when you reach DS-3 levels, running PRI
just isn't elegant (but certain it's possible.) Implementing SS7 on
Asterisk is a much larger issue, and more fraught with danger. That
being said, I can also get M-13 DS3-to-T1 muxes pretty cheap these
days, so just the space savings of a DS3 into a single Asterisk box
still makes it look appealing versus a slew of PRI's and associated
card madness.
I don't expect any real comments to come out of this post, and I'm
uncertain why I even made it. The people reading this list (you know
who you are - Hi, guys!) who have an interest in high-density
Asterisk installations have not and will not ever post to this list
directly. There are dozens of companies in this situation (ssh!
It's a secret that they run Asterisk! What embarrassment that an
RBOC was using gasp OPEN SOURCE!) and it's a shame that this type
of platform will not be developed due to everyone's reluctance to
practice what they preach with open source information.
Anyone want to fund an egg or a chicken?
JT
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