Re: [asterisk-users] Connections rejected in DUNDi requests
Chris Bagnall wrote:
Greetings list,
Wondering if anyone's come across this before.
I've configured a couple of our servers with a privatedundi context to
allow calls to still flow between extensions even if they're registered to
different servers . The DUNDi lookups seem to work fine, evidenced by the
following on the originating server:
-- Called private:password@ip/[EMAIL PROTECTED]
shouldn't that be 'private:password@ip/minotaur-201'? I guess you
have a mistake in your dundi mapping
However, on the destination server, I have the following:
May 4 03:50:45 NOTICE[1149]: chan_iax2.c:7354 socket_read: Rejected connect
attempt from 80.68.80.210, request '[EMAIL PROTECTED]' does not exist
I then performed the following:
cronus*CLI show dialplan privatedundi
[ Context 'privatedundi' created by 'pbx_config' ]
'_minotaur-2XX' = 1. NoOp(Connected to ${EXTEN})
[pbx_config]
2. Goto(minotaur|${EXTEN:9}|1)[pbx_config]
Unless I'm missing something, [EMAIL PROTECTED] definitely *does* exist.
I've tried manually specifying minotaur-201 in full rather than as a pattern
match - which works correctly. I'm having exactly the same problem the other
way around (origination and target servers reversed).
What's particularly strange is that other entries in [privatedundi] such as
_clienta-2XX, _clientb-2XX are working fine between the same servers.
So, what's special about _minotaur-2XX vs. _somethingelse-2XX that causes
pattern matching to fail?
If anyone can shed some light on this I'd be most grateful.
Regards,
Chris
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Re: [asterisk-users] SIP RealTime Friends
Forrest Beck wrote: I setup sip realtime. Is it possible to use a type of friend? User and Peer seem to work fine. have you tried? If so, what went wrong? (*hint* ;-) ) -- Met vriendelijke groeten, Remco Post SARA - Reken- en Netwerkdiensten http://www.sara.nl High Performance Computing Tel. +31 20 592 3000Fax. +31 20 668 3167 PGP Key fingerprint = 6367 DFE9 5CBC 0737 7D16 B3F6 048A 02BF DC93 94EC I really didn't foresee the Internet. But then, neither did the computer industry. Not that that tells us very much of course - the computer industry didn't even foresee that the century was going to end. -- Douglas Adams ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Poor man's High Availability solution
On Sun, Apr 29, 2007 at 09:06:53PM +0200, Clayton Milos wrote: Since a PRI is a physical connection as well as a logical one, if you can get the server to shut down when it has a problem you could put a 4-pole relay to change the PRI over to the other box. The ISDN Guard is an excellant product from what I've seen of it,and you would be well served by it.We are in the process of releasing (Product is ready, working on sales channels) a somewhat simpler product, the FSV-4PFS. It will handle two asterisk server redundancy,for significantly less cost.http://www.failsafevoip.com/images/4PFS/FSV-4PFS-Datasheet.pdfA demo of it in action:http://www.failsafevoip.com/images/4PFS/FSV-4PFS_Demo.aviFailSafeVOIP, Inc.www.failsafevoip.comContact for additional info: [EMAIL PROTECTED]___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Poor man's High Availability solution
On Sun, Apr 29, 2007 at 09:06:53PM +0200, Clayton Milos wrote: Since a PRI is a physical connection as well as a logical one, if you can get the server to shut down when it has a problem you could put a 4-pole relay to change the PRI over to the other box. The ISDN Guard is an excellant product from what I've seen of it, and you would be well served by it. We are in the process of releasing (Product is ready, working on sales channels) a somewhat simpler product, the FSV-4PFS. It will handle two asterisk server redundancy for significantly less cost. http://www.failsafevoip.com/images/4PFS/FSV-4PFS-Datasheet.pdf A demo of it in action: http://www.failsafevoip.com/images/4PFS/FSV-4PFS_Demo.avi FailSafeVOIP, Inc. www.failsafevoip.com Contact: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Balancing interrupts.
Steve Edwards wrote: Should I be concerned that cpu1 is servicing only 700,000 interrupts from my te410p while cpu3 is servicing almost 90,000,000? I thought this is what irqbalance was for... Steve, It was my experience that irqbalance used smp affinity to bind the interrupts from each ethernet device to their own CPU. This led to uneven processor utilization on my Asterisk server, so after some research I turned off irqbalance. If you choose to do so, you'll want to confirm that your kernel has been configured to do IRQ balancing. For more details see: Asterisk SMP: Is irqbalance Redundant on 2.6 Kernels? - Resolved http://lists.digium.com/pipermail/asterisk-users/2006-March/146169.html Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoiceXML + Nuance
Voxy - the only way to integrate VoiceXML applications in Asterisk. Configure your dial plan with the URL of your VoiceXML application and it's done. Is something the free and open source Voxy what you are looking for? http://sourceforge.net/projects/voxy On 5/3/07, wendell hamilton [EMAIL PROTECTED] wrote: I've done considerable work with the voxeo Prophecy platform, and it's been successful, albeit challenging at times. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Rousse Sent: Thursday, May 03, 2007 2:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] VoiceXML + Nuance Hello, Is there anyone who has ever done a setup of VoiceXML combined with some licenses from Nuance for the ASR/TTS engine within Asterisk ? I'm currently working with VoiceGenie, for the VoiceXML + ASR/TTS engine, but we are having a couple of issues which I guess are caused by VoiceGenie. If there's an alternative, it would be very interesting for us. Thanks, -- Eric Rousse System Administrator 514.380.2992 450.655.1001 1.888.641.5800 Telmatik inc. 204 Montarville, suite 250 Boucherville, QC, Canada J4B 6S2 www.telmatik.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This message is confidential. It may also be privileged or otherwise protected by work product immunity or other legal rules. If you have received it by mistake, please let us know by e-mail reply and delete it from your system; you may not copy this message or disclose its contents to anyone. Please send us by fax any message containing deadlines as incoming e-mails are not screened for response deadlines. The integrity and security of this message cannot be guaranteed on the Internet. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Codec Translation Table
Hello list, I have always though codec translation table is dircetly connected to system speed, utill i came across this: in my lab, i have 2 boxes, First box is an Intel Celeron 1.7 GHZ with 256M RAM: show translation Translation times between formats (in milliseconds) for one second of data Source Format (Rows) Destination Format (Columns) g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 gsm- -223 21 9 - - 253 - ulaw- 6 -13 21 9 - - 253 - ilbc-106676513 - --7- g726- 73313210 - -26-- Second server is Dual Xeon 2Gh 1G RAM show translation Translation times between formats (in milliseconds) for one second of data Source Format (Rows) Destination Format (Columns) g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g72 gsm- -446 4328 - - 255- ulaw- 7- 14 2126 - - 233- ilbc- 9446 4328 - - - 5- g726-7 221 2126 - - 23 -- Here is the fun part, box1 is faster in converting ulaw to gsm! Is this table accurate? Does it mean asterisk is not handeling multiple cpus very good? both boxes running asterisk 1.4.4 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UK - 2 port ISDN2e cards ...
Gordon Henderson wrote: (aren't you guys getting rid of ISDN anyway? :-) H... Some people would like to think so, but it's going to be here for a long time yet! BT have/are dumping the consumer versions of ISDN2 - home highway which went a while back, but business highway is going soon if it's not already gone, which is a real shame as they had almost all the functionality a small business needed for less price than the full ISDN2e... What is BT offering instead? I ran a company in the UK 6-7 years ago, and we went straight for the business highway. /Per Jessen, Zurich -- ENIDAN Technologies GmbH - managed email security. Starting at SFr1/month/user - http://www.spamchek.ch/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: headsets for linksys/sipura phones?
Per Jessen wrote: Yeah, that's cheap - I've just ordered two M175s at USD40/each. Just in case anyone's interested - I got the M175s this morning and they work just fine with the Sipura/Linksys SPA-921/-941s. /Per Jessen, Zürich -- ENIDAN Technologies GmbH - managed email security. Starting at SFr1/month/user - http://www.spamchek.ch/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - robo dialer
Am Freitag, den 04.05.2007, 00:48 -0400 schrieb Doug Crompton:
Can anyone suggest a source for a free robot dialer or examples? I need to
do some non-commercial auto dialing using Asterisk. Simple phone numbers
in a file, line by line format.
I found one called AstAutoDiaker but I was not able to get it to work and
it appears to not be supported - no email response from author.
Depending on what is required to happen on that call, you might get away
with some simple scripting and .call files. Along the lines of...
#!/usr/bin/perl
while (STDIN) {
$phonenumber = $_;
$callfilecontent = Read the docs.\n.$phonenumber.Text\n;
# Write that stuff to a file on the same physical partition
# as your callfile directory, give it the time stamp of
# the time it is intended to be run, and move it over.
}
... and piping the phonenumbers into the script through stdin.
Of course your dialplan would have to accomodate for the local end,
such that the callee will get some entertainment (announcement,
whatever).
BR
Anselm
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Re: [asterisk-users] UK - 2 port ISDN2e cards ...
On Fri, 4 May 2007, Per Jessen wrote: Gordon Henderson wrote: (aren't you guys getting rid of ISDN anyway? :-) H... Some people would like to think so, but it's going to be here for a long time yet! BT have/are dumping the consumer versions of ISDN2 - home highway which went a while back, but business highway is going soon if it's not already gone, which is a real shame as they had almost all the functionality a small business needed for less price than the full ISDN2e... What is BT offering instead? I ran a company in the UK 6-7 years ago, and we went straight for the business highway. They are offering regular ISDN2e (with the full business features like a DDI number range, etc.) - which puts it out of the price range of even the most enthusiastic home enthusiast... They dropped home business highway and made people pay for a full re-install if they wanted to transition over to ISDN2e. More than a few people are pissed off at this and have transitioned back to a pair of POTS lines, or are now using their ADSL line to carry VoIP. (or are installing a 2nd ADSL line dedicated to their VoIP traffic) Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 1.2.x - 1.4.x upgrade: dialplan block no longer works
Hi,
a block of my extensions.conf no longer works after upgrading from
1.2.17 to 1.4.4. I have:
[macro-dialout]
exten = s,1,Gosub(s-${ARG1},1)
exten = s,n,Congestion
;; default
exten = _s-!,1,Gosub(s-NET,1)
When calling that macro whith no argument ($ARG1 empty):
exten = _0[1-9],1,Macro(dialcapi)
The call is not routed. Apparently _s-! does not match s-:
-- Executing [EMAIL PROTECTED]:1] Macro(SIP/0146472130-0821fe08,
dialcapi) in new stack
-- Executing [EMAIL PROTECTED]:5] Gosub(SIP/0146472130-0821fe08,
s-|1) in new stack
== Auto fallthrough, channel 'SIP/0146472130-0821fe08' status is
'UNKNOWN'
Any idea why?
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Re: [asterisk-users] FXO recommendation
On Thu, 3 May 2007, Kyle Gordon wrote: Hi all, With the gamut of FXO cards out there, I'm looking for a recommendation for home use. I have a nicely working Asterisk 1.4 system that just requires an FXO card to connect my NTL PSTN to it. My previous X101P clone seems to have kicked the bucket. Any suggestions would be greatly appreciated. Another XP100p clone ;-) Or an ATA - I had a recommendation from another group for the Linksys SPA3102... Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Balancing interrupts.
Steve Edwards [EMAIL PROTECTED] writes: I see the following on one of my new servers: -ts10::sedwards:~$ cat /proc/interrupts CPU0 CPU1 CPU2 CPU3 0:29790452988620 87780075 87779501IO-APIC-edge timer [...] 225:4611916 681023 84732445 89903138 IO-APIC-level wct4xxp NMI: 0 0 0 0 LOC: 181534588 181534654 181534653 181534652 ERR: 0 MIS: 0 -ts10::sedwards:~$ ps -e | grep bal 2633 ?00:00:00 irqbalance Should I be concerned that cpu1 is servicing only 700,000 interrupts from my te410p while cpu3 is servicing almost 90,000,000? I thought this is what irqbalance was for... Actually, what you *really* want (for performance reasons) is to have one CPU handle *all* the interrupts and all the threads that talk to hardware for that card, if possible. Every time you move the IRQ to a different CPU you lose a bunch of cycles reloading data from main memory into the L2 and L1 cache, cycles that can't be used elsewhere. Binding that interrupt to one specific CPU -- and your NIC to a different CPU -- is generally a good idea. If you can keep the threads that handle those signals and the hardware on that same CPU you increase efficiency a bit more. Moving the IRQ has plenty of cost and isn't a great plan. :) Regards, Daniel -- Digital Infrastructure Solutions -- making IT simple, stable and secure Phone: 0401 155 707email: [EMAIL PROTECTED] http://digital-infrastructure.com.au/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] cpu usuage
Do any one knows the formula to calculate memory and cpu usuage for channel on g729 codec,to know the hardware required for 100 concurrent call. Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. *___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Double DTMF digits
On 5/3/07, Ken Leland III [EMAIL PROTECTED] wrote: When dtmfmode is set to inband for SIP, and i originate a call from sip out to the PSTN, I can hear the DTMF digit twice in the audio stream. Once very briefly and once for normal duration. Our Theory: While Asterisk is parsing the DTMF, for a fraction of a second, while the end user generated DTMF is being detected, the DTMF is passed inband. Once the DTMF is detected Asterisk silences it and regenerates it. Sensitive machines like auto attendants pick up both the brief end user generated tone as well as the full length asterisk generated tone and ultimately perceive each digit twice. Is anyone else experiencing this? I have reproduced this in an environment * with one asterisk server that is both the feature server and the media gateway, and is timing off of network T1s * with two servers, one feature server (timing off of ztdummy) and one media gateway (timing off of network T1s) using IAX as the inter asterisk protocol It is pretty easy to reproduce: -Dial a PSTN number(like your cell) from a sip phone using inband DTMF, and configured in asterisk sip.conf with dtmfmode=inband. -Answer the PSTN end. -Press and hold a digit on the sip phone. On the PSTN phone you will hear a very brief, end user generated, tone. -Let go of the digit on the sip phone. On the PSTN phone you will hear the asterisk generated tone. Can anyone else hear the brief initial tone? Any help is greatly appreciated! Yes, we have a similar issue, but do not normally use inband DTMF because SIP phones very cleanly generate rfc2833 RTP packets directly and remove this issue. On the other hand, asterisk is not alone dealing with this issue in SIP. The Linksys ATAs have exactly the same issue. Strangely, I do not have a problem receiving inband DTMF through Zaptel, which I believe uses the same DSP code for DTMF detection... Or does it? Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXO recommendation
On Thu, May 03, 2007 at 08:17:25PM +0100, Kyle Gordon wrote: Hi all, With the gamut of FXO cards out there, I'm looking for a recommendation for home use. I have a nicely working Asterisk 1.4 system that just requires an FXO card to connect my NTL PSTN to it. My previous X101P clone seems to have kicked the bucket. Any suggestions would be greatly appreciated. Well, I've never connected an NTL, er, Virgin PSTN line to Asterisk, but if I were you I'd consider whether you might want further ports in the future - if so, go for at least a TDM400P with just the one module for now. -- Phil Reynolds o mail: [EMAIL PROTECTED] |L_ \ / Web: http://www.tinsleyviaduct.com/phil/ (_)- \/ Waltham 66, Emley Moor 69, Droitwich 79, Windows 95 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T1/E1 Configuration
On Thu, May 03, 2007 at 11:03:59PM -0400, Nitesh Divecha wrote: Hello All, Can anyone please post their working T1/E1 configuration... Both '/etc/zaptel.conf' and '/etc/asterisk/zapata.conf'. I believe if you run 'genzaptelconf' it created '/etc/asterisk/zapata-channels.conf', so please post that one also. What do you have in /etc/asterisk/zapata.conf ? Here is my configuration which is failing Asterisk to load... I have two cards TE405P and TDM400P: - What error message do you get from asterisk at load time? You'll typically see them in /var/log/asterisk/messages or /var/log/asterisk/full === /etc/zaptel.conf === # T1 Configuration span=1,1,0,esf,b8zs bchan=1-23 dchan=24 span=2,1,0,esf,b8zs bchan=25-47 dchan=48 span=3,1,0,esf,b8zs bchan=49-71 dchan=72 span=4,1,0,esf,b8zs bchan=73-95 dchan=96 fxsks=97 fxsks=98 fxsks=99 fxsks=100 # Global data loadzone= us defaultzone = us /etc/asterisk/zapata-channels.conf group = 1 switchtype = national signalling = pri_cpe context = from-zaptel channel = 1-23 group = 2 switchtype = national signalling = pri_cpe context = from-zaptel channel = 25-47 group = 3 switchtype = national signalling = pri_cpe context = from-zaptel channel = 49-71 group = 4 switchtype = national signalling = pri_cpe context = from-zaptel channel = 73-95 signalling=fxs_ks callerid=asreceived group=0 context=from-zaptel channel = 97 ; context=default signalling=fxs_ks callerid=asreceived group=0 context=from-pstn channel = 98 context=default signalling=fxs_ks callerid=asreceived group=0 context=from-pstn channel = 99 context=default signalling=fxs_ks callerid=asreceived group=0 context=from-pstn channel = 100 context=default Thanking in advance... Cheers, Nitesh ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Users Conference Friday, May 4th at 12:30 PM EDT
AUC is Friday at 12:30 PM EDT. See http://x2z.eu for how to join. http://www.talkshoe.com/talkshoe/web/tscmd/tc/22622 We'll be talking about SIP/IAX providers and I'd like to hear more about asterisk appliances like the Digium and the new D-Link. If the Digium guys are around as they usually are, you can also put any tech questions you may have directly to them, especially release related ones. Hope to see you later today. r ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reinvite after DTMF?
Maybe I missed something here. In my understanding, the only parties in the call at DTMF stage are the originator and Asterisk. The destination is not in the picture yet. Is this correct? What is the purpose of the said DTMF sequence? Do you have a sample dial plan? No, the problem is to receive a call, to dial and send the DTMF to the new dialed number. The dial would normally then bridge the two channels. I'm trying to figure out if there's a way to then remove asterisk from the RTP stream because of the needless distance (crossing the ocean twice is a waste). ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Call interruption
From: Andre Wangler [EMAIL PROTECTED] Date: Fri, 4 May 2007 07:35:38 +0200 Hello all Could someone tell me what happens with running calls when reloading the whole asterisk config files? I think SIP-calls are not Nothing. All calls are maintained according to documentation. Yuan Liu interrupted because of the protocol architecture (signalling vs. media) but what's with other kind of calls like h323 or over analogue interfaces? are they interrupted? I'm quite new with asterisk, so excuse this probably trivial question... Andre ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] need more knowledge about asterisk
Hi all, I am working wit a Mandriva 2007 and have installed the asterisk svn 1.4 with success. I also used the patch for cellphones and it works perfectly. I was that happy that I decided to buy a TDM11B and it works. Now, I want to study a bit the code used by this people. Does anybody know how can I go deeper in this code with funcitonal bloqs in order to understand how is possible to interface that much new technologies? I know there is a book available but I am not in a mood for buying it. Any ideas? Thanks in Advance, Iban _ Un amor, una aventura, compañía para un viaje. Regístrate gratis en MSN Amor Amistad. http://match.msn.es/match/mt.cfm?pg=channeltcid=162349 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR(accountcode) empty in * 1.4.4 (for local chan)
Nops. Its not working. i have restored to original chan_local file. Im also
having another problem now (in asterisk 1.4.4).
The call originates fine, ringing is done, call is accepted, channels
bridged fine. but when either of the channels hangup, asterisk dies and
displays the following msg:
asterisk: symbol lookup error: /usr/lib/asterisk/modules/res_features.so:
undefined symbol: ast_cdr_merge
again, i dont know whats the problem. i'll try n remove the res_features and
then try caling again. Can anybody tell me what other things will be
effected by removing the res_features?
On 5/3/07, Steve Murphy [EMAIL PROTECTED] wrote:
On Thu, 2007-05-03 at 16:47 +0500, Rizwan Hisham wrote:
Hi all,
i just updated to asterisk 1.4.4 from 1.4.2. i was doing this to
forward an unanswered call in 1.4.2
exten= 1,1,Dial(SIP/123,,Ttg)
exten= 1,2,Gotoif($[${DIALSTATUS}=ANSWERED]?:10)
exten= 1,3,Hangup
exten= 1,10,Dial(Local/2,,Ttg)
exten= 1,11,Hangup
exten= 2,1,Dial(SIP/234,,Ttg)
exten= 2,2,Hangup
All the CDR variables for the first channel (SIP/123) are fine. but
when local channel initiates, it does not copy the CDR(accountcode)
variable from the first channel (in asterisk 1.4.4), whereas it did in
1.4.2. so the CDR(accountcode) variable for local channel is empty in
1.4.4. This is a big problem for me as i have to charge the forwarded
calls also and all calls are charged based on account code. If
accountcode is empty, i cant make a decision how to charge the call.
Can anybody fix this for me or do i have to jump back to asterisk
1.4.2?
--
Regards
Rizwan Hisham
Software Engineer
Riswan--
This could easily be my fault. I've attached a fix, that I can commit to
the source, if it works for you.
Here the instructions:
1. save the attachment to a file.
2. cd to your 1.4-source/channels directory
3. patch -p0 localfix
4. cd ..
5. make
6. make install
test
If there's no differences, you still have the same problem, you'd best
restore the source to it's previous condition:
1. cd 1.4-sourcedir/channels
2. mv chan_local.c.orig chan_local.c
3. cd ..
4. make
5. make install
This patch will properly set the accountcode amaflag from the local
channel's owner at channel creation time, and therefore, the local
channels' CDR as well.
--
Steve Murphy
Software Developer
Digium
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Rizwan Hisham
Software Engineer
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Re: [asterisk-users] CDR(accountcode) empty in * 1.4.4 (for local chan)
Nops. removing res_features doesnt work.
On 5/4/07, Rizwan Hisham [EMAIL PROTECTED] wrote:
Nops. Its not working. i have restored to original chan_local file. Im
also having another problem now (in asterisk 1.4.4).
The call originates fine, ringing is done, call is accepted, channels
bridged fine. but when either of the channels hangup, asterisk dies and
displays the following msg:
asterisk: symbol lookup error: /usr/lib/asterisk/modules/res_features.so:
undefined symbol: ast_cdr_merge
again, i dont know whats the problem. i'll try n remove the res_features
and then try caling again. Can anybody tell me what other things will be
effected by removing the res_features?
On 5/3/07, Steve Murphy [EMAIL PROTECTED] wrote:
On Thu, 2007-05-03 at 16:47 +0500, Rizwan Hisham wrote:
Hi all,
i just updated to asterisk 1.4.4 from 1.4.2. i was doing this to
forward an unanswered call in 1.4.2
exten= 1,1,Dial(SIP/123,,Ttg)
exten= 1,2,Gotoif($[${DIALSTATUS}=ANSWERED]?:10)
exten= 1,3,Hangup
exten= 1,10,Dial(Local/2,,Ttg)
exten= 1,11,Hangup
exten= 2,1,Dial(SIP/234,,Ttg)
exten= 2,2,Hangup
All the CDR variables for the first channel (SIP/123) are fine. but
when local channel initiates, it does not copy the CDR(accountcode)
variable from the first channel (in asterisk 1.4.4), whereas it did in
1.4.2. so the CDR(accountcode) variable for local channel is empty in
1.4.4. This is a big problem for me as i have to charge the forwarded
calls also and all calls are charged based on account code. If
accountcode is empty, i cant make a decision how to charge the call.
Can anybody fix this for me or do i have to jump back to asterisk
1.4.2?
--
Regards
Rizwan Hisham
Software Engineer
Riswan--
This could easily be my fault. I've attached a fix, that I can commit to
the source, if it works for you.
Here the instructions:
1. save the attachment to a file.
2. cd to your 1.4-source/channels directory
3. patch -p0 localfix
4. cd ..
5. make
6. make install
test
If there's no differences, you still have the same problem, you'd best
restore the source to it's previous condition:
1. cd 1.4-sourcedir/channels
2. mv chan_local.c.orig chan_local.c
3. cd ..
4. make
5. make install
This patch will properly set the accountcode amaflag from the local
channel's owner at channel creation time, and therefore, the local
channels' CDR as well.
--
Steve Murphy
Software Developer
Digium
___
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To UNSUBSCRIBE or update options visit:
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--
Regards
Rizwan Hisham
Software Engineer
--
Regards
Rizwan Hisham
Software Engineer
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Re: [asterisk-users] FXO recommendation
Friday, May 4, 2007, 10:42:13 AM, Phil wrote: On Thu, May 03, 2007 at 08:17:25PM +0100, Kyle Gordon wrote: With the gamut of FXO cards out there, I'm looking for a recommendation for home use. I have a nicely working Asterisk 1.4 system that just requires an FXO card to connect my NTL PSTN to it. My previous X101P clone seems to have kicked the bucket. Any suggestions would be greatly appreciated. Well, I've never connected an NTL, er, Virgin PSTN line to Asterisk, but if I were you I'd consider whether you might want further ports in the future - if so, go for at least a TDM400P with just the one module for now. Well this is a digium list, so here will be digium cards recommendation. But You can use a linksys spa3102, that costs about half price of TDM400P. -- Best regards, Gergomailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Connections rejected in DUNDi requests
-- Called private:password@ip/[EMAIL PROTECTED] shouldn't that be 'private:password@ip/minotaur-201'? I guess you have a mistake in your dundi mapping I've tried both. Sticking @privatedundi on the end was a 4am test because I couldn't think of anything else to try. Normally that wouldn't be there. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited For full contact details visit http://www.minotaur.it/chris.html This email is made from 100% recycled electrons ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] need more knowledge about asterisk
Which book are you talking about. and what are its contents. Is it based on understanding the code used in Asterisk. If it is then plz tell me the name of the book. I'll be happy to buy it. On 5/4/07, Iban Lopetegi Zinkunegi [EMAIL PROTECTED] wrote: Hi all, I am working wit a Mandriva 2007 and have installed the asterisk svn 1.4 with success. I also used the patch for cellphones and it works perfectly. I was that happy that I decided to buy a TDM11B and it works. Now, I want to study a bit the code used by this people. Does anybody know how can I go deeper in this code with funcitonal bloqs in order to understand how is possible to interface that much new technologies? I know there is a book available but I am not in a mood for buying it. Any ideas? Thanks in Advance, Iban _ Un amor, una aventura, compañía para un viaje. Regístrate gratis en MSN Amor Amistad. http://match.msn.es/match/mt.cfm?pg=channeltcid=162349 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Regards Rizwan Hisham Software Engineer ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] need more knowledge about asterisk
Hi Rizwan, You can find the book in the next web page, http://www.oreilly.com/catalog/asterisk/ Iban From: Rizwan Hisham [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [asterisk-users] need more knowledge about asterisk Date: Fri, 4 May 2007 15:44:49 +0500 Which book are you talking about. and what are its contents. Is it based on understanding the code used in Asterisk. If it is then plz tell me the name of the book. I'll be happy to buy it. On 5/4/07, Iban Lopetegi Zinkunegi [EMAIL PROTECTED] wrote: Hi all, I am working wit a Mandriva 2007 and have installed the asterisk svn 1.4 with success. I also used the patch for cellphones and it works perfectly. I was that happy that I decided to buy a TDM11B and it works. Now, I want to study a bit the code used by this people. Does anybody know how can I go deeper in this code with funcitonal bloqs in order to understand how is possible to interface that much new technologies? I know there is a book available but I am not in a mood for buying it. Any ideas? Thanks in Advance, Iban _ Un amor, una aventura, compañía para un viaje. Regístrate gratis en MSN Amor Amistad. http://match.msn.es/match/mt.cfm?pg=channeltcid=162349 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Regards Rizwan Hisham Software Engineer ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Dale rienda suelta a tu tiempo libre. Mil ideas para exprimir tu ocio con MSN Entretenimiento. http://entretenimiento.msn.es/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP RealTime Friends
Hi, Do you know how see the peers statuses like: sip show peers but when sip peers are configured by Relatime method. Thanks 0xception escribió: yes you can use the type friend On 5/3/07, *Forrest Beck* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I setup sip realtime. Is it possible to use a type of friend? User and Peer seem to work fine. -- *** Forrest Beck IAXTEL: 17002871718 [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Con Netfono, puede hablar por telefono, de PC a PC y gratis ! Instale su Netfono desde http://www.netfono.com. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] need more knowledge about asterisk
Oooh. i already have this book (Asterisk The future of Telephony). its not about the code. On 5/4/07, Iban Lopetegi Zinkunegi [EMAIL PROTECTED] wrote: Hi Rizwan, You can find the book in the next web page, http://www.oreilly.com/catalog/asterisk/ Iban From: Rizwan Hisham [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [asterisk-users] need more knowledge about asterisk Date: Fri, 4 May 2007 15:44:49 +0500 Which book are you talking about. and what are its contents. Is it based on understanding the code used in Asterisk. If it is then plz tell me the name of the book. I'll be happy to buy it. On 5/4/07, Iban Lopetegi Zinkunegi [EMAIL PROTECTED] wrote: Hi all, I am working wit a Mandriva 2007 and have installed the asterisk svn 1.4 with success. I also used the patch for cellphones and it works perfectly. I was that happy that I decided to buy a TDM11B and it works. Now, I want to study a bit the code used by this people. Does anybody know how can I go deeper in this code with funcitonal bloqs in order to understand how is possible to interface that much new technologies? I know there is a book available but I am not in a mood for buying it. Any ideas? Thanks in Advance, Iban _ Un amor, una aventura, compañía para un viaje. Regístrate gratis en MSN Amor Amistad. http://match.msn.es/match/mt.cfm?pg=channeltcid=162349 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Regards Rizwan Hisham Software Engineer ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Dale rienda suelta a tu tiempo libre. Mil ideas para exprimir tu ocio con MSN Entretenimiento. http://entretenimiento.msn.es/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Regards Rizwan Hisham Software Engineer ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX Trunk
Hi All, I'd like to thank everyone that answer my question about IAX Trunk. Now I have a working IAX trunking, I just need to tune it. Thank you. Ronaldo. Salvatore Giudice wrote: Yes of course. If you want to limit it, I think you have to set 'incominglimit' and/or 'outgoinglimit'. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ronaldo Sent: Thursday, May 03, 2007 2:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] IAX Trunk OK Steve, Just one more question. Using this configuration can I make more than one call at the same time? Thanks. Steve Kennedy wrote: On Thu, May 03, 2007 at 01:22:07PM -0300, Ronaldo wrote: Can you suggest me any documentation about using IAX trunking? Thank you. There are examples in the iax.conf files I think, but basically just put something like [iax-toremote] type=friend username=whatever secret=somesecret auth=plaintext host=somewhere.com peercontext=some-context qualify=yes trunk=yes then you dial with Dial(iax2/iax-toremote/number) Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Error compiling patched pppd for zapras
hi everybody, i'm tryint to install a asterisk system which acts as a dialin server using a Digium Wildcard 205P. acording to http://www.voip-info.org/wiki/view/Asterisk+cmd+ZapRAS i need a patched version of pppd, but it does not compile on my system. Linux box 2.6.17-gentoo-r8 #1 SMP Tue Sep 26 13:17:23 CEST 2006 x86_64 AMD Athlon(tm) 64 Processor 3200+ GNU/Linux gcc -4.1.1, glibc-2.4 output of make is below. any suggestions ? Alex cd chat; make all make[1]: Entering directory `/usr/src/ppp-2.4.1b2.WORKING/chat' cc -c -O2 -g -pipe -DTERMIOS -DSIGTYPE=void -UNO_SLEEP -DFNDELAY=O_NDELAY -o chat.o chat.c chat.c:215: warning: conflicting types for built-in function 'logf' chat.c:1275:22: warning: trigraph ??) ignored, use -trigraphs to enable cc -o chat chat.o make[1]: Leaving directory `/usr/src/ppp-2.4.1b2.WORKING/chat' cd pppd/plugins; make all make[1]: Entering directory `/usr/src/ppp-2.4.1b2.WORKING/pppd/plugins' gcc -o minconn.so -shared -g -O2 -I.. -I../../include -fPIC minconn.c gcc -o passprompt.so -shared -g -O2 -I.. -I../../include -fPIC passprompt.c make -C pppoe -w pppoe.so make[2]: Entering directory `/usr/src/ppp-2.4.1b2.WORKING/pppd/plugins/pppoe' gcc -g -I.. -I../.. -I../../../include -D_linux_=1 -fPIC -c -o pppoe.o pppoe.c In file included from pppoe.c:21: pppoe.h:109:1: warning: PTT_SRV_NAME redefined In file included from pppoe.h:37, from pppoe.c:21: /usr/include/linux/if_pppox.h:88:1: warning: this is the location of the previous definition In file included from pppoe.c:21: pppoe.h:110:1: warning: PTT_AC_NAME redefined In file included from pppoe.h:37, from pppoe.c:21: /usr/include/linux/if_pppox.h:89:1: warning: this is the location of the previous definition In file included from pppoe.c:21: pppoe.h:111:1: warning: PTT_HOST_UNIQ redefined In file included from pppoe.h:37, from pppoe.c:21: /usr/include/linux/if_pppox.h:90:1: warning: this is the location of the previous definition In file included from pppoe.c:21: pppoe.h:112:1: warning: PTT_AC_COOKIE redefined In file included from pppoe.h:37, from pppoe.c:21: /usr/include/linux/if_pppox.h:91:1: warning: this is the location of the previous definition In file included from pppoe.c:21: pppoe.h:113:1: warning: PTT_VENDOR redefined In file included from pppoe.h:37, from pppoe.c:21: /usr/include/linux/if_pppox.h:92:1: warning: this is the location of the previous definition In file included from pppoe.c:21: pppoe.h:114:1: warning: PTT_RELAY_SID redefined In file included from pppoe.h:37, from pppoe.c:21: /usr/include/linux/if_pppox.h:93:1: warning: this is the location of the previous definition In file included from pppoe.c:21: pppoe.h:115:1: warning: PTT_SRV_ERR redefined In file included from pppoe.h:37, from pppoe.c:21: /usr/include/linux/if_pppox.h:94:1: warning: this is the location of the previous definition In file included from pppoe.c:21: pppoe.h:116:1: warning: PTT_SYS_ERR redefined In file included from pppoe.h:37, from pppoe.c:21: /usr/include/linux/if_pppox.h:95:1: warning: this is the location of the previous definition In file included from pppoe.c:21: pppoe.h:117:1: warning: PTT_GEN_ERR redefined In file included from pppoe.h:37, from pppoe.c:21: /usr/include/linux/if_pppox.h:96:1: warning: this is the location of the previous definition In file included from pppoe.c:21: pppoe.h:118:1: warning: PTT_EOL redefined In file included from pppoe.h:37, from pppoe.c:21: /usr/include/linux/if_pppox.h:87:1: warning: this is the location of the previous definition gcc -g -I.. -I../.. -I../../../include -D_linux_=1 -fPIC -c -o pppoehash.o pppoehash.c In file included from pppoehash.c:11: pppoe.h:109:1: warning: PTT_SRV_NAME redefined In file included from pppoe.h:37, from pppoehash.c:11: /usr/include/linux/if_pppox.h:88:1: warning: this is the location of the previous definition In file included from pppoehash.c:11: pppoe.h:110:1: warning: PTT_AC_NAME redefined In file included from pppoe.h:37, from pppoehash.c:11: /usr/include/linux/if_pppox.h:89:1: warning: this is the location of the previous definition In file included from pppoehash.c:11: pppoe.h:111:1: warning: PTT_HOST_UNIQ redefined In file included from pppoe.h:37, from pppoehash.c:11: /usr/include/linux/if_pppox.h:90:1: warning: this is the location of the previous definition In file included from pppoehash.c:11: pppoe.h:112:1: warning: PTT_AC_COOKIE redefined In file included from pppoe.h:37, from pppoehash.c:11: /usr/include/linux/if_pppox.h:91:1: warning: this is the location of the previous definition In file included from pppoehash.c:11: pppoe.h:113:1: warning: PTT_VENDOR redefined In file included from pppoe.h:37, from
[asterisk-users] Starting Asterisk on Ubuntu 7.04
Hi all, Could someone please tell me how to make Asterisk start at boot on Ubuntu Feisty 7.04? Many thanks, Christian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UK zaptel and zapata.conf for TDM400P
In the UK CLID is sent before the 1st ring. - Original Message - From: Tzafrir Cohen [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, April 24, 2007 11:15 PM Subject: Re: [asterisk-users] UK zaptel and zapata.conf for TDM400P On Tue, Apr 24, 2007 at 09:35:07PM +0100, Ed W wrote: Hi usecallerid=yes cidsignalling=v23 cidstart=polarity Although this is what the wiki recommends, I just couldn't get the cidstart=polarity to play well with immediate=yes, I kept loosing the callerid? Actually: immediate=yes will not work with callerid. The caller ID is passed after the first ring (or even later is other variations) on analog channels. This is what I ended up with and now it avoids the annoying 2 rings before the internal extensions start to ring. However, I still have a problem in that if someone hangs up while still in ringing state then asterisk continues to ring for 2 more rings (roughly). This is annoying because BT appear to do a line test every 30 hours or so and so my lines ring for 2 rings at random times of day or night What do you have on your dialplan for an incoming call? [EMAIL PROTECTED] asterisk]# more 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 usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no echotraining=800 rxgain=0.0 txgain=0.0 group=0 callgroup=1 pickupgroup=1 immediate=no ukcallerid=yes cidsignalling=v23 cidstart=ring ;cidstart=polarity ; Added for UK CLI detection sendcalleridafter=0 immediate=yes ; as we recieve cli info before not after first ring. answeronpolarityswitch=no -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content by ESVA, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk Codec Translation Table
It's the magical Celeron chip. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Al Sent: Friday, May 04, 2007 3:08 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk Codec Translation Table Hello list, I have always though codec translation table is dircetly connected to system speed, utill i came across this: in my lab, i have 2 boxes, First box is an Intel Celeron 1.7 GHZ with 256M RAM: show translation Translation times between formats (in milliseconds) for one second of data Source Format (Rows) Destination Format (Columns) g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 gsm- -223 21 9 - - 253 - ulaw- 6 -13 21 9 - - 253 - ilbc-106676513 - --7- g726- 73313210 - -26-- Second server is Dual Xeon 2Gh 1G RAM show translation Translation times between formats (in milliseconds) for one second of data Source Format (Rows) Destination Format (Columns) g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g72 gsm- -446 4328 - - 255- ulaw- 7- 14 2126 - - 233- ilbc- 9446 4328 - - - 5- g726-7 221 2126 - - 23 -- Here is the fun part, box1 is faster in converting ulaw to gsm! Is this table accurate? Does it mean asterisk is not handeling multiple cpus very good? both boxes running asterisk 1.4.4 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXO recommendation
Gergo Csibra [EMAIL PROTECTED] Wrote: 5/4/2007 8:12 AM: Friday, May 4, 2007, 1:56:09 PM, Joe wrote: Gergo Csibra [EMAIL PROTECTED] Wrote: 5/4/2007 6:34 AM: Well this is a digium list, so here will be digium cards recommendation. But You can use a linksys spa3102, that costs about half price of TDM400P. I looked up that linksys device. It does not appear that it can replace ad TDM400P. It is not a card at all but a free standing device. More of an ATA, actaully. Yes it is an ATA with an FXS and an FXO port, and you can use as many as you want instead of one TDM400/TDM800/TDM2400. I don't see how that is possible. This device does not connect to the PCI bus, at all. It has two RJ11 ports that can connect to a LAN, or directly to the asterisk box so it may be possible to make it work, somehow, but it cannot replace a TDM card, which is what I thought you were suggesting. I missed the original posting. Since no one else has spoken up, perhaps I am off base. Please help clear up what I am missing. joe a. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] app_txfax, app_rxfax
Hello, I'm trying to compile asterisk from source (1.2.18). Faxing is fairly critical for us, so in the past we've used spandsps app_rxfax and app_txfax to support faxing in asterisk. Unfortunately I can't find these applications on soft-switch.org anymore and even so I have a feeling they wouldn't patch correctly into 1.2.18. Does anyone know how to best handle faxing in 1.2.18? Is it even necessary to compile these two apps into asterisk? what about spandsp? Any advice would be appreciated, Thanks! Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] app_txfax, app_rxfax
mail-lists wrote: Does anyone know how to best handle faxing in 1.2.18? http://iaxmodem.sourceforge.net http://hylafax.sourceforge.net Small foot print, works great with Asterisk and supports error correction. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] app_txfax, app_rxfax
Steve, Am 04.05.2007 um 14:44 schrieb mail-lists: I'm trying to compile asterisk from source (1.2.18). Faxing is fairly critical for us, so in the past we've used spandsps app_rxfax and app_txfax to support faxing in asterisk. Unfortunately I can't find these applications on soft-switch.org anymore Forget them! Use Hylafax and iaxmodem instead. Does anyone know how to best handle faxing in 1.2.18? Is it even necessary to compile these two apps into asterisk? what about spandsp? Any advice would be appreciated, I only have a German howto available. But you should get the idea: http://www.das-asterisk-buch.de/stable/installation-iaxmodem.html http://www.das-asterisk-buch.de/stable/installation-hylafax.html Stefan -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXO recommendation
On Fri, 2007-05-04 at 08:35 -0400, Joe acquisto wrote: Gergo Csibra [EMAIL PROTECTED] Wrote: 5/4/2007 8:12 AM: Friday, May 4, 2007, 1:56:09 PM, Joe wrote: Gergo Csibra [EMAIL PROTECTED] Wrote: 5/4/2007 6:34 AM: Well this is a digium list, so here will be digium cards recommendation. But You can use a linksys spa3102, that costs about half price of TDM400P. I looked up that linksys device. It does not appear that it can replace ad TDM400P. It is not a card at all but a free standing device. More of an ATA, actaully. Yes it is an ATA with an FXS and an FXO port, and you can use as many as you want instead of one TDM400/TDM800/TDM2400. I don't see how that is possible. This device does not connect to the PCI bus, at all. Correct It has two RJ11 ports and an RJ45 that can connect to a LAN, or directly to the asterisk box so it may be possible to make it work, somehow See my post earlier today regarding this device , but it cannot replace a TDM card, which is what I thought you were suggesting. Depends on what you mean by replace. Physically no, but functionally yes. I missed the original posting. Since no one else has spoken up, perhaps I am off base. Please help clear up what I am missing. Should be clearer now. -- Dave Cotton [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Console flooded by WARNING app_meetme messages
Hi there, One of our Asterisk 1.2 machine is experiencing problems with MeetMe. Whenever meetme runs, the console is flooded with warning messages: The messages started as No such file or directory and becomes Resource temporarily unavailable. I couldn't figure out what file MeetMe might be looking for, could anyone help? May 4 08:57:38 WARNING[19032]: app_meetme.c:1563 conf_run: Failed to read frame: No such file or directory May 4 09:01:35 WARNING[19063]: app_meetme.c:1563 conf_run: Failed to read frame: Resource temporarily unavailable I'm currently not subscribed to asterisk-users, if you have an insight on this, please reply to me heison AT chak DOT ca. Thanks -Heison signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP RealTime Friends
Let me check my table Voicemail and CDR in the MySQL database works fine. sip show peers isn't giving me anything. Only the one peer I left setup in sip.conf Here Is what I get from a Dial Command: [May 4 09:14:28] WARNING[7186]: chan_sip.c:1753 __sip_xmit: sip_xmit of 0x9e42d38 (len 705) to 10.110.1.4:0 returned -1: Invalid argument [May 4 09:14:30] WARNING[7186]: chan_sip.c:1753 __sip_xmit: sip_xmit of 0x9de9678 (len 533) to 10.110.1.4:0 returned -1: Invalid argument [May 4 09:14:30] WARNING[7186]: chan_sip.c:1753 __sip_xmit: sip_xmit of 0x9e42d38 (len 705) to 10.110.1.4:0 returned -1: Invalid argument [May 4 09:14:34] WARNING[7186]: chan_sip.c:1753 __sip_xmit: sip_xmit of 0x9de9678 (len 533) to 10.110.1.4:0 returned -1: Invalid argument [May 4 09:14:34] WARNING[7186]: chan_sip.c:1753 __sip_xmit: sip_xmit of 0x9e42d38 (len 705) to 10.110.1.4:0 returned -1: Invalid argumen On 5/4/07, Sergio (Red) [EMAIL PROTECTED] wrote: Hi, Do you know how see the peers statuses like: sip show peers but when sip peers are configured by Relatime method. Thanks 0xception escribió: yes you can use the type friend On 5/3/07, *Forrest Beck* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I setup sip realtime. Is it possible to use a type of friend? User and Peer seem to work fine. -- *** Forrest Beck IAXTEL: 17002871718 [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Con Netfono, puede hablar por telefono, de PC a PC y gratis ! Instale su Netfono desde http://www.netfono.com. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- *** Forrest Beck IAXTEL: 17002871718 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] SIP RealTime Friends
try enabling rtcachefriends -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Forrest Beck Sent: Friday, May 04, 2007 9:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP RealTime Friends Let me check my table Voicemail and CDR in the MySQL database works fine. sip show peers isn't giving me anything. Only the one peer I left setup in sip.conf Here Is what I get from a Dial Command: [May 4 09:14:28] WARNING[7186]: chan_sip.c:1753 __sip_xmit: sip_xmit of 0x9e42d38 (len 705) to 10.110.1.4:0 returned -1: Invalid argument [May 4 09:14:30] WARNING[7186]: chan_sip.c:1753 __sip_xmit: sip_xmit of 0x9de9678 (len 533) to 10.110.1.4:0 returned -1: Invalid argument [May 4 09:14:30] WARNING[7186]: chan_sip.c:1753 __sip_xmit: sip_xmit of 0x9e42d38 (len 705) to 10.110.1.4:0 returned -1: Invalid argument [May 4 09:14:34] WARNING[7186]: chan_sip.c:1753 __sip_xmit: sip_xmit of 0x9de9678 (len 533) to 10.110.1.4:0 returned -1: Invalid argument [May 4 09:14:34] WARNING[7186]: chan_sip.c:1753 __sip_xmit: sip_xmit of 0x9e42d38 (len 705) to 10.110.1.4:0 returned -1: Invalid argumen On 5/4/07, Sergio (Red) [EMAIL PROTECTED] wrote: Hi, Do you know how see the peers statuses like: sip show peers but when sip peers are configured by Relatime method. Thanks 0xception escribió: yes you can use the type friend On 5/3/07, *Forrest Beck* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I setup sip realtime. Is it possible to use a type of friend? User and Peer seem to work fine. -- *** Forrest Beck IAXTEL: 17002871718 [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Con Netfono, puede hablar por telefono, de PC a PC y gratis ! Instale su Netfono desde http://www.netfono.com. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- *** Forrest Beck IAXTEL: 17002871718 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP RealTime Friends
Nevermind. Friday and my mind has gone home! :) I forgot the ipaddr and port setting in the table. On 5/4/07, Forrest Beck [EMAIL PROTECTED] wrote: Let me check my table Voicemail and CDR in the MySQL database works fine. sip show peers isn't giving me anything. Only the one peer I left setup in sip.conf Here Is what I get from a Dial Command: [May 4 09:14:28] WARNING[7186]: chan_sip.c:1753 __sip_xmit: sip_xmit of 0x9e42d38 (len 705) to 10.110.1.4:0 returned -1: Invalid argument [May 4 09:14:30] WARNING[7186]: chan_sip.c:1753 __sip_xmit: sip_xmit of 0x9de9678 (len 533) to 10.110.1.4:0 returned -1: Invalid argument [May 4 09:14:30] WARNING[7186]: chan_sip.c:1753 __sip_xmit: sip_xmit of 0x9e42d38 (len 705) to 10.110.1.4:0 returned -1: Invalid argument [May 4 09:14:34] WARNING[7186]: chan_sip.c:1753 __sip_xmit: sip_xmit of 0x9de9678 (len 533) to 10.110.1.4:0 returned -1: Invalid argument [May 4 09:14:34] WARNING[7186]: chan_sip.c:1753 __sip_xmit: sip_xmit of 0x9e42d38 (len 705) to 10.110.1.4:0 returned -1: Invalid argumen On 5/4/07, Sergio (Red) [EMAIL PROTECTED] wrote: Hi, Do you know how see the peers statuses like: sip show peers but when sip peers are configured by Relatime method. Thanks 0xception escribió: yes you can use the type friend On 5/3/07, *Forrest Beck* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I setup sip realtime. Is it possible to use a type of friend? User and Peer seem to work fine. -- *** Forrest Beck IAXTEL: 17002871718 [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Con Netfono, puede hablar por telefono, de PC a PC y gratis ! Instale su Netfono desde http://www.netfono.com. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- *** Forrest Beck IAXTEL: 17002871718 [EMAIL PROTECTED] -- *** Forrest Beck IAXTEL: 17002871718 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] app_txfax, app_rxfax
mail-lists schrieb: I'm trying to compile asterisk from source (1.2.18). Faxing is fairly critical for us, so in the past we've used spandsps app_rxfax and app_txfax to support faxing in asterisk. Unfortunately I can't find these applications on soft-switch.org anymore and even so I have a feeling they wouldn't patch correctly into 1.2.18. Does anyone know how to best handle faxing in 1.2.18? Is it even necessary to compile these two apps into asterisk? what about spandsp? They do work in 1.2.18 - the archivr just moved to http://www.soft-switch.org/downloads/snapshots/spandsp/test-apps-asterisk-1.2/ IF you don't want to reinvent the wheel and switch to iaxmodem/hylafax, use them instead :-) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Codec Translation Table
On Fri, May 04, 2007 at 01:07:37AM -0600, Al wrote: Here is the fun part, box1 is faster in converting ulaw to gsm! Is this table accurate? Yes. The task of transcoding a single call is done by a single thread and hence a single CPU. Does it mean asterisk is not handeling multiple cpus very good? both boxes running asterisk 1.4.4 If you're only going to have one concurrent transcoding, this will indeed be the case. But in that case a nice little PII will also do :-) When you have multiple calls, each call (or actually: each channel, but that doesn't really matter here) will be handled by a different thread. And threads can be schuled to different CPUs concurrently. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] does Not detected HANGUP and DTMF
Hello all,nbsp;nbsp; I am using HALF DUPLEX modem for TAPI call.the following message is displayed while i am starting the AsteriskNOTICE[1416] chan_tapi.c: Channel format set to ULAW\' ERROR[1416] win32_tapi.c: TAPI Error: 8023 (HCALL 0x0) on lineGetID . If i will receive an Inbound call to modem, i will answer that calland put an wait for infinite time.The caller(Who will originate a call)nbsp;hangup that time TAPI will not receive an EVENT(HangUP).For DTMF issue i am using READ command to receive an DTMF tone..This message is received, but consistently i am not getting the Entered DIgitsnbsp;If i will use an FULL DUPLEX modem can i solve the above problem.please guide me its an urgent issue...Regards,Pandi.P___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Runaway MOH/mp3123 process?
- Original Message - From: Alex Balashov [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, May 02, 2007 2:35 AM Subject: [asterisk-users] Runaway MOH/mp3123 process? Has anyone noticed a problem with runaway mpg123 processes for music-on-hold eating up ~100% CPU and driving the load on the machine way up? I've seen this problem consistently with multiple Asterisk installs, 1.2.x and 1.4.x, although admittedly it was more common with 1.2.x as far as I can tell. There is no clearly identifiable sequence of events that causes this to occur, although it obviously involves utilisation of the MOH audio blend at some point, which I use both in queues and for hold. But the precise chain of events is never consistent, predictable, nor triggered in any particular temporal relation to when MOH is last used--at least, not one that I can pin down. It does not appear to arise immediately following the activation of a MOH sequence. We had the same problem on our Asterisk ACD. After switching to native mode of MOH, problem goes away. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] app_txfax, app_rxfax
Stefan Wintermeyer wrote: Steve, Am 04.05.2007 um 14:44 schrieb mail-lists: I'm trying to compile asterisk from source (1.2.18). Faxing is fairly critical for us, so in the past we've used spandsps app_rxfax and app_txfax to support faxing in asterisk. Unfortunately I can't find these applications on soft-switch.org anymore Forget them! Use Hylafax and iaxmodem instead. Does anyone know how to best handle faxing in 1.2.18? Is it even necessary to compile these two apps into asterisk? what about spandsp? Any advice would be appreciated, I only have a German howto available. But you should get the idea: http://www.das-asterisk-buch.de/stable/installation-iaxmodem.html http://www.das-asterisk-buch.de/stable/installation-hylafax.html Stefan -- Stefan, My name is spelled Stefan too :) I AM using hylafax/iaxmodem on my production boxes. I guess I'm not entirely clear on WHAT app_rxfax, app_txfax do. Are iaxmodem/hylafax essentially a replacement for these asterisk internal applications? Also, In the past I've installed Trixbox/FreePbx. It uses NvFaxDetect to detect incoming faxes - is this an application I need to build for asterisk? Thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] app_txfax, app_rxfax
Thomas Göttgens wrote: IF you don't want to reinvent the wheel and switch to iaxmodem/hylafax, use them instead :-) I wouldn't consider that re-inventing the wheel. If faxes are 'critical', then I wouldn't use anything but iaxmodem and HylaFAX+ :-) Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] need more knowledge about asterisk
On Fri, May 04, 2007 at 09:46:08AM +, Iban Lopetegi Zinkunegi wrote: Hi all, I am working wit a Mandriva 2007 and have installed the asterisk svn 1.4 with success. I also used the patch for cellphones and it works perfectly. I was that happy that I decided to buy a TDM11B and it works. Now, I want to study a bit the code used by this people. Does anybody know how can I go deeper in this code with funcitonal bloqs in order to understand how is possible to interface that much new technologies? I know there is a book available but I am not in a mood for buying it. How about trying to implement a small feature you need? There is also http://asterisk.org/developers/janitor but I suspect it is not up-to-date. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Starting Asterisk on Ubuntu 7.04
On Fri, May 04, 2007 at 02:04:55PM +0200, Christian wrote: Hi all, Could someone please tell me how to make Asterisk start at boot on Ubuntu Feisty 7.04? Many thanks, Christian apt-get install asterisk Look at the init.d scripts. Note that in Ubuntu, subdirectories under /var/run are deleted at boot, and hence that script generates /var/run/asterisk (with proper ownership) at boot time. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IP Phone Provisioning Tool by voip.com.sg - xml generation
Tom Rymes wrote: On May 3, 2007, at 12:20 PM, Stephen Bosch wrote: Mats Karlsson wrote: Take a look here: http://www.voip.com.sg/voip_products/voip_ip_phone_provisioning_tool.html Ugh. This is a Win32 app, isn't it? Wow, The guy makes a useful application and provides it to the community for free and you have the cojones to bitch and moan b/c it's a windows app? Talk about looking a gift horse in the mouth! A. Yes, I have the cojones. He never mentioned what platform it was for. We need something like this for Linux. I got all excited about it only to be terribly disappointed when I unpacked it. B. It's not a gift horse for me, because it's totally use*less* to me. C. Nobody said you weren't allowed to appreciate it. -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoiceXML + Nuance
Well, basically, I'm looking for something that has the possiblity to use the Nuance licenses, and that can do Text to Speech, as well as Voice Recognition. So far it doesn't seem possible to have a single product that does all this within Asterisk... Rob Townley a écrit : Voxy - the only way to integrate VoiceXML applications in Asterisk. Configure your dial plan with the URL of your VoiceXML application and it's done. Is something the free and open source Voxy what you are looking for? http://sourceforge.net/projects/voxy On 5/3/07, *wendell hamilton* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I've done considerable work with the voxeo Prophecy platform, and it's been successful, albeit challenging at times. -Original Message- From: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]] On Behalf Of Eric Rousse Sent: Thursday, May 03, 2007 2:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] VoiceXML + Nuance Hello, Is there anyone who has ever done a setup of VoiceXML combined with some licenses from Nuance for the ASR/TTS engine within Asterisk ? I'm currently working with VoiceGenie, for the VoiceXML + ASR/TTS engine, but we are having a couple of issues which I guess are caused by VoiceGenie. If there's an alternative, it would be very interesting for us. Thanks, -- Eric Rousse System Administrator 514.380.2992 450.655.1001 1.888.641.5800 Telmatik inc. 204 Montarville, suite 250 Boucherville, QC, Canada J4B 6S2 www.telmatik.com http://www.telmatik.com ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This message is confidential. It may also be privileged or otherwise protected by work product immunity or other legal rules. If you have received it by mistake, please let us know by e-mail reply and delete it from your system; you may not copy this message or disclose its contents to anyone. Please send us by fax any message containing deadlines as incoming e-mails are not screened for response deadlines. The integrity and security of this message cannot be guaranteed on the Internet. ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Eric Rousse System Administrator 514.380.2992 450.655.1001 1.888.641.5800 Telmatik inc. 204 Montarville, suite 250 Boucherville, QC, Canada J4B 6S2 www.telmatik.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T1/E1 Configuration
Thanks Guys... Got the T1/E1 Card working... Digium Engineers helped... According to them TE405P card must load first and then the analog TDM400P. Other thing which I messed up was that I changed the configuration to T1 but forgot to remove the Jumpers from the TE405P card. So that was causing Asterisk to fail... Its working now but can anyone clarify that... Do I have to remove all four jumpers to make T1 card? Anyone can help me with this error? May 4 10:41:10 WARNING[5171]: chan_zap.c:9134 pri_dchannel: PRI Error: We think we're the CPE, but they think they're the CPE too. Regards, Nitesh Tzafrir Cohen wrote: On Thu, May 03, 2007 at 11:03:59PM -0400, Nitesh Divecha wrote: Hello All, Can anyone please post their working T1/E1 configuration... Both '/etc/zaptel.conf' and '/etc/asterisk/zapata.conf'. I believe if you run 'genzaptelconf' it created '/etc/asterisk/zapata-channels.conf', so please post that one also. What do you have in /etc/asterisk/zapata.conf ? Here is my configuration which is failing Asterisk to load... I have two cards TE405P and TDM400P: - What error message do you get from asterisk at load time? You'll typically see them in /var/log/asterisk/messages or /var/log/asterisk/full === /etc/zaptel.conf === # T1 Configuration span=1,1,0,esf,b8zs bchan=1-23 dchan=24 span=2,1,0,esf,b8zs bchan=25-47 dchan=48 span=3,1,0,esf,b8zs bchan=49-71 dchan=72 span=4,1,0,esf,b8zs bchan=73-95 dchan=96 fxsks=97 fxsks=98 fxsks=99 fxsks=100 # Global data loadzone= us defaultzone = us /etc/asterisk/zapata-channels.conf group = 1 switchtype = national signalling = pri_cpe context = from-zaptel channel = 1-23 group = 2 switchtype = national signalling = pri_cpe context = from-zaptel channel = 25-47 group = 3 switchtype = national signalling = pri_cpe context = from-zaptel channel = 49-71 group = 4 switchtype = national signalling = pri_cpe context = from-zaptel channel = 73-95 signalling=fxs_ks callerid=asreceived group=0 context=from-zaptel channel = 97 ; context=default signalling=fxs_ks callerid=asreceived group=0 context=from-pstn channel = 98 context=default signalling=fxs_ks callerid=asreceived group=0 context=from-pstn channel = 99 context=default signalling=fxs_ks callerid=asreceived group=0 context=from-pstn channel = 100 context=default Thanking in advance... Cheers, Nitesh ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IP Phone Provisioning Tool by voip.com.sg - xml generation
Stephen Bosch wrote: Tom Rymes wrote: On May 3, 2007, at 12:20 PM, Stephen Bosch wrote: Mats Karlsson wrote: Take a look here: http://www.voip.com.sg/voip_products/voip_ip_phone_provisioning_tool.html Ugh. This is a Win32 app, isn't it? Wow, The guy makes a useful application and provides it to the community for free and you have the cojones to bitch and moan b/c it's a windows app? Talk about looking a gift horse in the mouth! A. Yes, I have the cojones. He never mentioned what platform it was for. We need something like this for Linux. I got all excited about it only to be terribly disappointed when I unpacked it. Actually the original poster did say this was a Windows program. I believe the exact wording was It runs on any modern flavor of Windows. Note that Mats Karlsson was NOT the original poster. -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Headset for Polycom
Hi, I've been asked for a headset recommandation for Polycom SoundPoint IP phones. Since I believe they use a pretty standard headset jack (correct me if I am wrong) it's really a general recommandation on headset. Regards, Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Headset for Polycom
Yes, they uses a standard headset jack. On Fri, 2007-05-04 at 11:15 -0400, Mike wrote: Hi, I've been asked for a headset recommandation for Polycom SoundPoint IP phones. Since I believe they use a pretty standard headset jack (correct me if I am wrong) it's really a general recommandation on headset. Regards, Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Cargile Explido Software USA Inc. http://www.explido.us ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] T1/E1 Configuration
Anyone can help me with this error? May 4 10:41:10 WARNING[5171]: chan_zap.c:9134 pri_dchannel: PRI Error: We think we're the CPE, but they think they're the CPE too. Both ends of the T1 can't be running in CPE (USER) mode. Typically, the Telco is NETWORK and you are USER (CPE). If you set your end of the T1 to NETWORK mode I'll bet the D/B-channels will come up but **check with your Telco first**. John Treble Ottawa, Ontario, Canada -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Nitesh Divecha Sent: May 4, 2007 10:43 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] T1/E1 Configuration Thanks Guys... Got the T1/E1 Card working... Digium Engineers helped... According to them TE405P card must load first and then the analog TDM400P. Other thing which I messed up was that I changed the configuration to T1 but forgot to remove the Jumpers from the TE405P card. So that was causing Asterisk to fail... Its working now but can anyone clarify that... Do I have to remove all four jumpers to make T1 card? Anyone can help me with this error? May 4 10:41:10 WARNING[5171]: chan_zap.c:9134 pri_dchannel: PRI Error: We think we're the CPE, but they think they're the CPE too. Regards, Nitesh Tzafrir Cohen wrote: On Thu, May 03, 2007 at 11:03:59PM -0400, Nitesh Divecha wrote: Hello All, Can anyone please post their working T1/E1 configuration... Both '/etc/zaptel.conf' and '/etc/asterisk/zapata.conf'. I believe if you run 'genzaptelconf' it created '/etc/asterisk/zapata- channels.conf', so please post that one also. What do you have in /etc/asterisk/zapata.conf ? Here is my configuration which is failing Asterisk to load... I have two cards TE405P and TDM400P: - What error message do you get from asterisk at load time? You'll typically see them in /var/log/asterisk/messages or /var/log/asterisk/full === /etc/zaptel.conf === # T1 Configuration span=1,1,0,esf,b8zs bchan=1-23 dchan=24 span=2,1,0,esf,b8zs bchan=25-47 dchan=48 span=3,1,0,esf,b8zs bchan=49-71 dchan=72 span=4,1,0,esf,b8zs bchan=73-95 dchan=96 fxsks=97 fxsks=98 fxsks=99 fxsks=100 # Global data loadzone= us defaultzone = us /etc/asterisk/zapata-channels.conf group = 1 switchtype = national signalling = pri_cpe context = from-zaptel channel = 1-23 group = 2 switchtype = national signalling = pri_cpe context = from-zaptel channel = 25-47 group = 3 switchtype = national signalling = pri_cpe context = from-zaptel channel = 49-71 group = 4 switchtype = national signalling = pri_cpe context = from-zaptel channel = 73-95 signalling=fxs_ks callerid=asreceived group=0 context=from-zaptel channel = 97 ; context=default signalling=fxs_ks callerid=asreceived group=0 context=from-pstn channel = 98 context=default signalling=fxs_ks callerid=asreceived group=0 context=from-pstn channel = 99 context=default signalling=fxs_ks callerid=asreceived group=0 context=from-pstn channel = 100 context=default Thanking in advance... Cheers, Nitesh ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Headset for Polycom
Hi Mike, Yes, they use a standard headset jack. In our implementations so far we've just had the customers continue to use their existing headsets. We take one of them from the customer ahead of time and test it out... So long as it works well, we replace their phones and keep the headsets. I can't say that we've found ones that work better than others. I'm sure that there are some really cheap ones that wouldn't work as well, but I've found that the customer has already invested a bit into the headsets since their employees will be wearing them all day long. The headsets are of good quality, and all seem to work about the same. Cheers, Alex Robar On 5/4/07, Mike [EMAIL PROTECTED] wrote: Hi, I've been asked for a headset recommandation for Polycom SoundPoint IP phones. Since I believe they use a pretty standard headset jack (correct me if I am wrong) it's really a general recommandation on headset. Regards, Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T1/E1 Configuration
Thanks John, How can I change my conf to NETWORK? Where can I find this information? Regards, Nitesh John Treble wrote: Anyone can help me with this error? May 4 10:41:10 WARNING[5171]: chan_zap.c:9134 pri_dchannel: PRI Error: We think we're the CPE, but they think they're the CPE too. Both ends of the T1 can't be running in CPE (USER) mode. Typically, the Telco is NETWORK and you are USER (CPE). If you set your end of the T1 to NETWORK mode I'll bet the D/B-channels will come up but **check with your Telco first**. John Treble Ottawa, Ontario, Canada -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Nitesh Divecha Sent: May 4, 2007 10:43 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] T1/E1 Configuration Thanks Guys... Got the T1/E1 Card working... Digium Engineers helped... According to them TE405P card must load first and then the analog TDM400P. Other thing which I messed up was that I changed the configuration to T1 but forgot to remove the Jumpers from the TE405P card. So that was causing Asterisk to fail... Its working now but can anyone clarify that... Do I have to remove all four jumpers to make T1 card? Anyone can help me with this error? May 4 10:41:10 WARNING[5171]: chan_zap.c:9134 pri_dchannel: PRI Error: We think we're the CPE, but they think they're the CPE too. Regards, Nitesh Tzafrir Cohen wrote: On Thu, May 03, 2007 at 11:03:59PM -0400, Nitesh Divecha wrote: Hello All, Can anyone please post their working T1/E1 configuration... Both '/etc/zaptel.conf' and '/etc/asterisk/zapata.conf'. I believe if you run 'genzaptelconf' it created '/etc/asterisk/zapata- channels.conf', so please post that one also. What do you have in /etc/asterisk/zapata.conf ? Here is my configuration which is failing Asterisk to load... I have two cards TE405P and TDM400P: - What error message do you get from asterisk at load time? You'll typically see them in /var/log/asterisk/messages or /var/log/asterisk/full === /etc/zaptel.conf === # T1 Configuration span=1,1,0,esf,b8zs bchan=1-23 dchan=24 span=2,1,0,esf,b8zs bchan=25-47 dchan=48 span=3,1,0,esf,b8zs bchan=49-71 dchan=72 span=4,1,0,esf,b8zs bchan=73-95 dchan=96 fxsks=97 fxsks=98 fxsks=99 fxsks=100 # Global data loadzone= us defaultzone = us /etc/asterisk/zapata-channels.conf group = 1 switchtype = national signalling = pri_cpe context = from-zaptel channel = 1-23 group = 2 switchtype = national signalling = pri_cpe context = from-zaptel channel = 25-47 group = 3 switchtype = national signalling = pri_cpe context = from-zaptel channel = 49-71 group = 4 switchtype = national signalling = pri_cpe context = from-zaptel channel = 73-95 signalling=fxs_ks callerid=asreceived group=0 context=from-zaptel channel = 97 ; context=default signalling=fxs_ks callerid=asreceived group=0 context=from-pstn channel = 98 context=default signalling=fxs_ks callerid=asreceived group=0 context=from-pstn channel = 99 context=default signalling=fxs_ks callerid=asreceived group=0 context=from-pstn channel = 100 context=default Thanking in advance... Cheers, Nitesh ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Headset for Polycom
On Fri, 2007-05-04 at 11:15 -0400, Mike wrote: Hi, I've been asked for a headset recommandation for Polycom SoundPoint IP phones. Since I believe they use a pretty standard headset jack (correct me if I am wrong) it's really a general recommandation on headset. Regards, Mike We use the Plantronics SupraPlus SL #P361N-U10P ($102 USD) Binaural, noise canceling, soft leather, etc. (Does not require a separate amplifier.) A bit on the spendy side, but for those who live on the phone, it makes being in a call center, sales, or service role a little nicer. Jim ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IP Phone Provisioning Tool by voip.com.sg - xml generation
A. Yes, I have the cojones. He never mentioned what platform it was for. We need something like this for Linux. I got all excited about it only to be terribly disappointed when I unpacked it. From the original announcement : It runs on any modern flavor of Windows. It is not like if he said runs on windows or better, then Linux would seem appropriate ;) You have at least two option beside having a windows machine : 1 - try it in WINE 2 - install Windows in a vmware machine (only way to run windows without ever rebooting your machine) hth ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reinvite after DTMF?
From: Wilson Pickett [EMAIL PROTECTED] Date: Fri, 4 May 2007 11:37:41 +0200 Maybe I missed something here. In my understanding, the only parties in the call at DTMF stage are the originator and Asterisk. The destination is not in the picture yet. Is this correct? What is the purpose of the said DTMF sequence? Do you have a sample dial plan? No, the problem is to receive a call, to dial and send the DTMF to the new dialed number. The dial would normally then bridge the two While it is not possible to reinvite in the middle of a call (based on whatever event), I'm thinking more in the way of a workaround. Does this DTMF sequence absolutely have to be sent in the MIDDLE of the call or can it be sent at the beginning, i.e., before any conversation starts? Yuan Liu channels. I'm trying to figure out if there's a way to then remove asterisk from the RTP stream because of the needless distance (crossing the ocean twice is a waste). ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ASA-2007-013: IAX2 users can cause unauthorized data disclosure
Asterisk Project Security Advisory - ASA-2007-013 +--+ | Product| Asterisk | |--+---| | Summary| IAX2 users can cause unauthorized data disclosure | |--+---| | Nature of Advisory | Unauthorized information disclosure | |--+---| |Susceptibility| Remote authenticated sessions | |--+---| | Severity | Low | |--+---| |Exploits Known| No | |--+---| | Reported On | April 27, 2007 | |--+---| | Reported By | Tim Panton, Mexuar, [EMAIL PROTECTED] | | | | | | Birgit Arkesteijn, Westhawk, [EMAIL PROTECTED] | |--+---| | Posted On | May 4, 2007 | |--+---| | Last Updated On| May 4, 2007 | |--+---| | Advisory Contact | [EMAIL PROTECTED] | |--+---| | CVE Name | CVE-2007-2488 | +--+ +--+ | Description | From: Tim Panton [EMAIL PROTECTED] | | | | | | Date: 27 April 2007 08:02:36 BDT | | | | | | To: Kevin P. Fleming [EMAIL PROTECTED] | | | | | | Subject: Possible IAX2 vulnerability (Minor) | | | | | | | | | | | | We've stumbled on a bug in the way Asterisk's IAX2 handles text | | | | | | frames. | | | | | | I'm emailing you because it is a borderline security | | | vulnerability, | | | | | | and my | | | | | | friends in the security world tell me that I should notify the | | | | | | vendor privately | | | | | | first. If you feel it isn't a security issue, let me know and| | | I'll | | | | | | put it in
Re: [asterisk-users] app_txfax, app_rxfax
On Fri, 4 May 2007, mail-lists wrote: Stefan Wintermeyer wrote: Steve, Am 04.05.2007 um 14:44 schrieb mail-lists: I'm trying to compile asterisk from source (1.2.18). Faxing is fairly critical for us, so in the past we've used spandsps app_rxfax and app_txfax to support faxing in asterisk. Unfortunately I can't find these applications on soft-switch.org anymore Forget them! Use Hylafax and iaxmodem instead. Does anyone know how to best handle faxing in 1.2.18? Is it even necessary to compile these two apps into asterisk? what about spandsp? Any advice would be appreciated, I only have a German howto available. But you should get the idea: http://www.das-asterisk-buch.de/stable/installation-iaxmodem.html http://www.das-asterisk-buch.de/stable/installation-hylafax.html Stefan -- Stefan, My name is spelled Stefan too :) I AM using hylafax/iaxmodem on my production boxes. I guess I'm not entirely clear on WHAT app_rxfax, app_txfax do. Are iaxmodem/hylafax essentially a replacement for these asterisk internal applications? I've used rx_fax (never tx_fax though, but I don't imagine an issue with it) So, as I see it: You answer an incoming channel, and plumb it directly into rx_fax. rx_fax is a combined software modem and fax receptor. It spits out a TIF file. I've used this on Zap channels with a good degree of success. iaxmodem is a softare modem. It's a program which takes an IAX channel and gives you a serial-line like interface. You can send AT commands to it and get/send digital data directly. This is what people connect HylaFax to. HylaFax is a suit pf programs that have been about for donkeys years - originally designed to talk to real modems (I used hylafax with a USR modem many years ago and remember it being a PITA to setup which is why I've never bothered to look at it - however that might have changed recently...?) So if you have a lot of experience with HylaFax, then the suggestion may be to get iaxmodem and carry on with HylaFax, else consider rx_fax/tx_fax. The downside of rx_fax is that you need to compile it into asterisk. The downside of iaxmodem is that (to my knowledge) you can't easilly implement an auto-answer/detect fax/voice/ auto attendant/voicemail system. The channel must be dedicated to faxing, and that's that. This may or may not be an issue for you though. The last fax setup I did was for a small 2-person office where they had an existing fax machine that answered, listened for the remote fax squawk, if it didn't get it, then it rung the phones daisy-chained to it, and if they didn't answer it went to answering machine. I implemented this in asterisk fairly easilly with rx_fax. I'm not sure if you can do that with iaxmodem. An upside of iaxmodem is that you can run it on a seprate server. You then rely on an IP connection between the asterisk box and the fax box... And we know how fussy modem signals can be over IP links.. On a LAN it ought to be OK though. If I'm missing something obvious (or my knowledge is out of date!) someone can correct me please. Cheers, Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [asterisk-users] Starting Asterisk on Ubuntu 7.04
Hi, I have already done: apt-get build-dep asterisk and then installed libpri, zaptel and asterisk from the latest sources. So what should i do then? New to Ubuntu. many thanks, Christian On 2007-05-04 at 17:00 Tzafrir Cohen wrote: On Fri, May 04, 2007 at 02:04:55PM +0200, Christian wrote: Hi all, Could someone please tell me how to make Asterisk start at boot on Ubuntu Feisty 7.04? Many thanks, Christian apt-get install asterisk Look at the init.d scripts. Note that in Ubuntu, subdirectories under /var/run are deleted at boot, and hence that script generates /var/run/asterisk (with proper ownership) at boot time. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Headset for Polycom
As far as a call center is concerned it really depends on the type of employees you have working for you. Most call center reps tend to be... umm... well less than trust worth to put it nicely. I have seen reps pour coffee into computers, sit on their headsets, get up and walk away with their headset still attached (those safety catches only last so long), and do other such nasty things. At such call centers I would go with the cheapest headset you can get. The difference in quality for the more expensive ones does not make up for distructive behavior of reps. You will probably be replacing headsets every two months with the cheap ones verses every four for the expensive ones. And the cheap ones are more than half the cost. However if your call center is full of trustworthy people I would go with a the nice Plantronics. If properly taken care of they will last for a few years. On Fri, 2007-05-04 at 09:11 -0700, Jim Rice wrote: On Fri, 2007-05-04 at 11:15 -0400, Mike wrote: Hi, I've been asked for a headset recommandation for Polycom SoundPoint IP phones. Since I believe they use a pretty standard headset jack (correct me if I am wrong) it's really a general recommandation on headset. Regards, Mike We use the Plantronics SupraPlus SL #P361N-U10P ($102 USD) Binaural, noise canceling, soft leather, etc. (Does not require a separate amplifier.) A bit on the spendy side, but for those who live on the phone, it makes being in a call center, sales, or service role a little nicer. Jim ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Cargile Explido Software USA Inc. http://www.explido.us ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXO recommendation
Friday, May 4, 2007, 3:06:02 PM, Dave wrote: On Fri, 2007-05-04 at 08:35 -0400, Joe acquisto wrote: Gergo Csibra [EMAIL PROTECTED] Wrote: 5/4/2007 8:12 AM: Yes it is an ATA with an FXS and an FXO port, and you can use as many as you want instead of one TDM400/TDM800/TDM2400. It has two RJ11 ports Yes, one for FXO, one for FXS and an RJ45 Two RJ45, one for local network, one for Internet (if you use this box for voip subscription). , but it cannot replace a TDM card, which is what I thought you were suggesting. Depends on what you mean by replace. Physically no, but functionally yes. You can reach the FXS and FXO port as a simple SIP client, so it works without zaptel. I missed the original posting. Since no one else has spoken up, perhaps I am off base. Please help clear up what I am missing. Original poster wants to have an FXO port. There's many sollution for that, TDM400P is only one (and maybe the most expensive) of them. -- Best regards, Gergomailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] app_txfax, app_rxfax
Gordon Henderson wrote: The downside of iaxmodem is that (to my knowledge) you can't easilly implement an auto-answer/detect fax/voice/ auto attendant/voicemail system. The channel must be dedicated to faxing, and that's that. This may or may not be an issue for you though. If you needed this you would handle the fax/voice detection in Asterisk, and only route the call to the iaxmodem if it detected fax. Lee. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Starting Asterisk on Ubuntu 7.04
Christian, You can follow this procedure http://www.aussievoip.com/wiki/freePBX-Ubuntu Regards, Nitesh Christian wrote: Hi, I have already done: apt-get build-dep asterisk and then installed libpri, zaptel and asterisk from the latest sources. So what should i do then? New to Ubuntu. many thanks, Christian On 2007-05-04 at 17:00 Tzafrir Cohen wrote: On Fri, May 04, 2007 at 02:04:55PM +0200, Christian wrote: Hi all, Could someone please tell me how to make Asterisk start at boot on Ubuntu Feisty 7.04? Many thanks, Christian apt-get install asterisk Look at the init.d scripts. Note that in Ubuntu, subdirectories under /var/run are deleted at boot, and hence that script generates /var/run/asterisk (with proper ownership) at boot time. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re[2]: [asterisk-users] Starting Asterisk on Ubuntu 7.04
If you do make config when compiling zaptel and asterisk, it should put the script in /etc/init.d, and add the relevant entries to the various start levels. Thanks, James Texter On Fri, 2007-05-04 at 18:44 +0200, Christian wrote: Hi, I have already done: apt-get build-dep asterisk and then installed libpri, zaptel and asterisk from the latest sources. So what should i do then? New to Ubuntu. many thanks, Christian On 2007-05-04 at 17:00 Tzafrir Cohen wrote: On Fri, May 04, 2007 at 02:04:55PM +0200, Christian wrote: Hi all, Could someone please tell me how to make Asterisk start at boot on Ubuntu Feisty 7.04? Many thanks, Christian apt-get install asterisk Look at the init.d scripts. Note that in Ubuntu, subdirectories under /var/run are deleted at boot, and hence that script generates /var/run/asterisk (with proper ownership) at boot time. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] app_txfax, app_rxfax
I've deployed Iaxmodem as part of a Unified messaging platform for a Fortune 100 company and it works great. * detects fax tones and vectors to fax extension which iaxmodem terminates. Kevin Collins -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lee Howard Sent: Friday, May 04, 2007 1:17 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] app_txfax, app_rxfax Gordon Henderson wrote: The downside of iaxmodem is that (to my knowledge) you can't easilly implement an auto-answer/detect fax/voice/ auto attendant/voicemail system. The channel must be dedicated to faxing, and that's that. This may or may not be an issue for you though. If you needed this you would handle the fax/voice detection in Asterisk, and only route the call to the iaxmodem if it detected fax. Lee. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] app_txfax, app_rxfax
Gordon Henderson wrote: iaxmodem is a softare modem. It's a program which takes an IAX channel and gives you a serial-line like interface. You can send AT commands to it and get/send digital data directly. This is what people connect HylaFax to. HylaFax is a suit pf programs that have been about for donkeys years - originally designed to talk to real modems (I used hylafax with a USR modem many years ago and remember it being a PITA to setup which is why I've never bothered to look at it - however that might have changed recently...?) So if you have a lot of experience with HylaFax, then the suggestion may be to get iaxmodem and carry on with HylaFax, else consider rx_fax/tx_fax. My personal experience - as part of a recent migration to Asterisk - Hylafax was very easily set up with a normal analog modem, which I opted for initially as I (somehow) got the impression that faxing wasn't all that straight forward with Asterisk/VoIP. Just today I've switched to using iaxmodem, which was equally easy. Anyone in the market for a plain 56K analog fax modem? The downside of iaxmodem is that (to my knowledge) you can't easilly implement an auto-answer/detect fax/voice/ auto attendant/voicemail system. The channel must be dedicated to faxing, and that's that. This may or may not be an issue for you though. In my case, we have a dedicated fax number. /Per Jessen, Zürich -- ENIDAN Technologies GmbH - managed email security. Starting at SFr1/month/user - http://www.spamchek.ch/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] zaptel compile error
I get the following error when trying to compile zaptel on CentOS 5 kernel 2.6.18-8.1.3.el5 CC [M] /root/asterisk-src/zaptel-1.2.17.1/xpp/xbus-core.o /root/asterisk-src/zaptel-1.2.17.1/xpp/xbus-core.c: In function â /root/asterisk-src/zaptel-1.2.17.1/xpp/xbus-core.c:171: error: â has no member named â make[3]: *** [/root/asterisk-src/zaptel-1.2.17.1/xpp/xbus-core.o] Error 1 make[2]: *** [/root/asterisk-src/zaptel-1.2.17.1/xpp] Error 2 make[1]: *** [_module_/root/asterisk-src/zaptel-1.2.17.1] Error 2 make[1]: Leaving directory `/usr/src/kernels/2.6.18-8.1.3.el5-i686' make: *** [all] Error 2 I'm kind of at my wits end with this - been trying for several hours.. Thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AsteriskNow!
Does anyone know how to gain access directly to the configuration files in AsteriskNow? I have dual NICs and need to change the binding in the config file. I believe they blocked ssh2 access by default. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] app_txfax, app_rxfax
On Fri, 4 May 2007, Lee Howard wrote: Gordon Henderson wrote: The downside of iaxmodem is that (to my knowledge) you can't easilly implement an auto-answer/detect fax/voice/ auto attendant/voicemail system. The channel must be dedicated to faxing, and that's that. This may or may not be an issue for you though. If you needed this you would handle the fax/voice detection in Asterisk, and only route the call to the iaxmodem if it detected fax. Does that work though? Asterisk has answered the call and listened for the initial tone, then it dils the IAX channel - which has to detect RING, then answer it and start modem initiation... Hm. thinking about it now, I can see that it would work. Asterisk dials an IAX channel which does ring at the other end... I think :) Cheers, Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T1/E1 Configuration
On 5/4/07, Nitesh Divecha [EMAIL PROTECTED] wrote: Thanks John, How can I change my conf to NETWORK? Where can I find this information? #signalling = pri_cpe signalling = pri_net ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Starting Asterisk on Ubuntu 7.04
On Fri, May 04, 2007 at 01:31:19PM -0400, Nitesh Divecha wrote: Christian, You can follow this procedure http://www.aussievoip.com/wiki/freePBX-Ubuntu If you like hard work, that is. I wonder how our frePBX debs fare on Ubuntu (deb http://updates.xorcom.com etch main ). Theretically they should work. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] allowing call to my pabx every 15 minutes
Gotoiftime() core show application gotoiftime Thats the best bet it sounds like, but your question was kind of hard to understand exactly, or why you would want to do this. -bkruse Goke Aruna wrote: Hello all, I have a set up that answer my customer. and its working well, however, the number of call to technical dept is what i want to reduce. I want all call to get to voice prompt except that that enter when minutes is 15, 30, 45, 60(in multiples of 15 minutes). how can i achieve this and what application can i use to get this done. I will be glad, if someone can give me a hint on this. Goksie ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] question about more than one drop file
hello there all, if i have a script that writes drop files into /var/spool/asterisk/outgoing asterisk picks up the file and initiates the call just fine. however, what is supposed to happen if more than one gets dropped in there within like a second. Will it wait till the first is complete to initiate the second ? Do they dissapear ? thanks shawn ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RE: Autologoff
I am having an issue with the autologoff fuction in agents.conf I am running Asterisk BE and I am testing with agent 1656. I log in, and then make a call into the queue. The agent's phone rings, and if I answer it, all's fine/ I am trying to have this agent automatically be logged off if he does not answer the queue callback within 5 seconds, however the agents extension keeps ringing until the call eventually goes to the extension's voice mail, which I am also trying to avoid. Here is my agents.conf [general] persistentagents=yes [agents] autologoff=5 multiplelogin=no recordagencalls=yes monitor-join=yes createlink=yes updatecdr=yes musiconhold=default recordformat=wav49 urlprefix=http://xxx.xxx.xxx.xxx/calls/ savecallsin=/var/www/html/calls agent = 1650,1650, agent = 1656,1656,Ed Here is my queues.conf [general] persistentmembers=yes [noi-cust-serv-spanish] strategy = leastrecent announce-frequency = 90 announce-holdtime = yes announce-round-seconds = 10 timeout=180 monitor-format=wav49 monitor-join=yes joinwhenempty = strict leavewhenempty = yes musiconhold = default eventwhencalled = yes queue-youarenext = queue-youarenext; (You are now first in line.) queue-thereare = queue-thereare; (There are) queue-callswaiting = queue-callswaiting; (calls waiting.) queue-holdtime = queue-holdtime; (The current est. holdtime is) queue-minutes = queue-minutes ; (minutes.) queue-seconds = queue-seconds ; (seconds.) queue-thankyou = queue-thankyou; (Thank you for your patience.) queue-lessthan = queue-less-than ; (less than) member = Agent/1656 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IP Phone Provisioning Tool by voip.com.sg - xml generation
On May 4, 2007, at 10:08 AM, Stephen Bosch wrote: Tom Rymes wrote: On May 3, 2007, at 12:20 PM, Stephen Bosch wrote: Mats Karlsson wrote: Take a look here: http://www.voip.com.sg/voip_products/ voip_ip_phone_provisioning_tool.html Ugh. This is a Win32 app, isn't it? Wow, The guy makes a useful application and provides it to the community for free and you have the cojones to bitch and moan b/c it's a windows app? Talk about looking a gift horse in the mouth! [snip] B. It's not a gift horse for me, because it's totally use*less* to me. [snip] It *is* a gift horse; he gave it to you, after all. The expression To look a gift horse in the mouth implies that someone gave you a horse and you looked into its mouth to see how old it is and whether it is of value to *you*. A modern comparison might be pulling up froogle to see how much a gift cost as someone gives it to you. In other words, taking a gift from someone and questioning its value to you in front of that person's face is rude, inconsiderate, and bad form. I would understand a post along the lines of: Wow, that looks like a great program from the description, but it's a windows only app, and I don't run Windows. Does anyone know of something similar for Linux? I dunno, I guess I'm not your mother, but then again, it seemed pretty rude considering the guy offered the program for free and you were criticizing the fact that he didn't develop a free linux app for you, too. Tom ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Unable to Execute System Command From DialPlan
Doh! That was it. It was a permissions issue. Thanks for your help! Victor ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T1/E1 Configuration
On Fri, May 04, 2007 at 02:28:32PM -0400, Andreas van dem Helge wrote: On 5/4/07, Nitesh Divecha [EMAIL PROTECTED] wrote: Thanks John, How can I change my conf to NETWORK? Where can I find this information? #signalling = pri_cpe signalling = pri_net nitpicking: ;signalling = pri_cpe signalling = pri_net (The comment character is ';' . '#' is reserved for special directives of the sort of #include) -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] E1 config for chile
Hello, I try to configure a Patton smart node with a E1 for the Chile. I need to know wich parameters I must set for Chile. If someone have informations it's welcoms. Laurent ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR(accountcode) empty in * 1.4.4 (for local chan)
On Fri, 2007-05-04 at 15:25 +0500, Rizwan Hisham wrote:
Nops. removing res_features doesnt work.
Rizwan--
This is strange; It would seem your main/cdr.c and res/res_features.c
are out of sync!
The code chunk I sent you does not contain any references to
ast_cdr_merge, and
does not have anything to do with res_features... so... you should have
seen this
problem with or without my patch! Can you investigate and make sure
something hasn't been mixed into your release?
murf
On 5/4/07, Rizwan Hisham [EMAIL PROTECTED] wrote:
Nops. Its not working. i have restored to original chan_local
file. Im also having another problem now (in asterisk 1.4.4).
The call originates fine, ringing is done, call is accepted,
channels bridged fine. but when either of the channels hangup,
asterisk dies and displays the following msg:
asterisk: symbol lookup
error: /usr/lib/asterisk/modules/res_features.so: undefined
symbol: ast_cdr_merge
again, i dont know whats the problem. i'll try n remove the
res_features and then try caling again. Can anybody tell me
what other things will be effected by removing the
res_features?
On 5/3/07, Steve Murphy [EMAIL PROTECTED] wrote:
On Thu, 2007-05-03 at 16:47 +0500, Rizwan Hisham
wrote:
Hi all,
i just updated to asterisk 1.4.4 from 1.4.2. i was
doing this to
forward an unanswered call in 1.4.2
exten= 1,1,Dial(SIP/123,,Ttg)
exten= 1,2,Gotoif($[${DIALSTATUS}=ANSWERED]?:10)
exten= 1,3,Hangup
exten= 1,10,Dial(Local/2,,Ttg)
exten= 1,11,Hangup
exten= 2,1,Dial(SIP/234,,Ttg)
exten= 2,2,Hangup
All the CDR variables for the first channel
(SIP/123) are fine. but
when local channel initiates, it does not copy the
CDR(accountcode)
variable from the first channel (in asterisk 1.4.4),
whereas it did in
1.4.2. so the CDR(accountcode) variable for local
channel is empty in
1.4.4. This is a big problem for me as i have to
charge the forwarded
calls also and all calls are charged based on
account code. If
accountcode is empty, i cant make a decision how to
charge the call.
Can anybody fix this for me or do i have to jump
back to asterisk
1.4.2?
--
Regards
Rizwan Hisham
Software Engineer
Riswan--
This could easily be my fault. I've attached a fix,
that I can commit to
the source, if it works for you.
Here the instructions:
1. save the attachment to a file.
2. cd to your 1.4-source/channels directory
3. patch -p0 localfix
4. cd ..
5. make
6. make install
test
If there's no differences, you still have the same
problem, you'd best
restore the source to it's previous condition:
1. cd 1.4-sourcedir/channels
2. mv chan_local.c.orig chan_local.c
3. cd ..
4. make
5. make install
This patch will properly set the accountcode amaflag
from the local
channel's owner at channel creation time, and
therefore, the local
channels' CDR as well.
--
Steve Murphy
Software Developer
Digium
___
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--
Regards
Rizwan Hisham
Software Engineer
--
Regards
Rizwan Hisham
Software Engineer
Re: [asterisk-users] app_txfax, app_rxfax
Forget them! Use Hylafax and iaxmodem instead. I wondering, how do you guys handle multiple calls? We frequently get many concurrent faxes, sometimes even to the same number. As far as I know, one instance of iaxmodem can only support one fax session at a time. So essentially you need a pool of iaxmodems running on different ports, and then Dial() them until you find one that accepts your call. Or did I get that wrong? That seems really like a drawback to me, that's why we're sticking to app_rxfax, which in the newer versions also supports error correction. With app_rxfax you are always guaranteed that that there is someone to answer the fax, given sufficient resources (CPU and memory). The biggest drawback with app_rxfax is that if it crashes for whatever reason (happens sometimes), it will take down the entire PBX and all sessions with it. --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] question about more than one drop file
Good question shawn, The callfile does get deleted once the call has been finished (I believe its FINISHED, not processed) No, they are not executed sequentially..exactly. Well, from your point of view, you can drop tons of them in there and all of the calls will fire up. I have dropped over 200 in at the same time (well, as fast as mv can write them.) And the asterisk box went through all of them. So if its a time thing, you should not have to worry about it. -bkruse shawn bright wrote: hello there all, if i have a script that writes drop files into /var/spool/asterisk/outgoing asterisk picks up the file and initiates the call just fine. however, what is supposed to happen if more than one gets dropped in there within like a second. Will it wait till the first is complete to initiate the second ? Do they dissapear ? thanks shawn ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk registration SIP confusion. Can someone explain this?
We have an Asterisk v1.2.16 box registering with an ITSP using SIP. The registration succeeds, and is confirmed with SIP SHOW REGISTER. However, we frequently (every few minutes) see this on our console: REGISTER attempt 1 to [EMAIL PROTECTED] REGISTER attempt 2 to [EMAIL PROTECTED] Any ideas what is going on? In particular 1. What causes the two register attempt messages above? 2. Why is our asterisk box being associated with the entryunauthorized context, not the entryinternal context? (See below) 3. Why is the contact sip:[EMAIL PROTECTED]:5060 in our SIP messages, why s@ anything? Thanks MD -- Contents of sip.conf at ITSP: [999] context=entryinternal ; I know this context exists! This is the right context. type=friend username=999 secret= callerid=Test 999 host=dynamic nat=no canreinvite=no allow=ulaw allow=alaw dtmfmode=rfc2833 --- Console log from ITSP show strange SIP traffic: --- Scheduling destruction of call mailto:'[EMAIL PROTECTED]' '[EMAIL PROTECTED]' in 15000 ms pbx*CLI pbx*CLI -- SIP read from 123.183.86.231:5060: REGISTER sip:pbx.itsp.com SIP/2.0 Via: SIP/2.0/UDP 123.183.86.231:5060;branch=z9hG4bK1a8cee82;rport From: sip:[EMAIL PROTECTED];tag=as3218ff14 To: sip:[EMAIL PROTECTED] Call-ID: mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] CSeq: 103 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Authorization: Digest username=999, realm=pbx.itsp.com, algorithm=MD5, uri=sip:pbx.itsp.com, nonce=5cec66c0, response=6451967016fc38f896efeb7247523fe1, opaque= Expires: 120 Contact: sip:[EMAIL PROTECTED]:5060 Event: registration Content-Length: 0 --- (13 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 123.183.86.231 : 5060 (NAT) Transmitting (no NAT) to 123.183.86.231:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 123.183.86.231:5060;branch=z9hG4bK1a8cee82;received=123.183.86.231;rport=506 0 From: sip:[EMAIL PROTECTED];tag=as3218ff14 To: sip:[EMAIL PROTECTED] Call-ID: mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] CSeq: 103 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- Transmitting (no NAT) to 123.183.86.231:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 123.183.86.231:5060;branch=z9hG4bK1a8cee82;received=123.183.86.231;rport=506 0 From: sip:[EMAIL PROTECTED];tag=as3218ff14 To: sip:[EMAIL PROTECTED];tag=as7d680d48 Call-ID: mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] CSeq: 103 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Expires: 120 Contact: sip:[EMAIL PROTECTED]:5060;expires=120 Date: Fri, 04 May 2007 19:27:58 GMT ontent-Length: 0 -- SIP read from 123.183.86.231:5060: OPTIONS sip:pbx.itsp.com SIP/2.0 Via: SIP/2.0/UDP 123.183.86.231:5060;branch=z9hG4bK36c1df86;rport From: asterisk sip:[EMAIL PROTECTED];tag=as6e5334cf To: sip:pbx.itsp.com Contact: sip:[EMAIL PROTECTED]:5060 Call-ID: mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 04 May 2007 19:38:36 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- (12 headers 0 lines) --- Looking for s in entryunauthorized (domain pbx.itsp.com) Transmitting (no NAT) to 123.183.86.231:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 123.183.86.231:5060;branch=z9hG4bK36c1df86;received=123.183.86.231;rport=506 0 From: asterisk sip:[EMAIL PROTECTED];tag=as6e5334cf To: sip:pbx.itsp.com;tag=as51d476cd Call-ID: mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:74.110.57.25 Accept: application/sdp Content-Length: 0 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T1/E1 Configuration
Thanks Everyone for the help... Got the T1 UP and insvc with Cisco AS5350, but I am failing to send the call. On the Cisco side I do not see any incoming call and on Asterisk side I get message saying Channels unavailable, while all channels are available. Can anyone post a working configuration for Asterisk T1 and Cisco conf? Please... Thank you. Regards, Nitesh Tzafrir Cohen wrote: On Fri, May 04, 2007 at 02:28:32PM -0400, Andreas van dem Helge wrote: On 5/4/07, Nitesh Divecha [EMAIL PROTECTED] wrote: Thanks John, How can I change my conf to NETWORK? Where can I find this information? #signalling = pri_cpe signalling = pri_net nitpicking: ;signalling = pri_cpe signalling = pri_net (The comment character is ';' . '#' is reserved for special directives of the sort of #include) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] app_txfax, app_rxfax
Luki wrote: Forget them! Use Hylafax and iaxmodem instead. I wondering, how do you guys handle multiple calls? We frequently get I have 23 iaxmodems running on my each of my Asterisk/HylaFAX+ servers. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [asterisk-users] Starting Asterisk on Ubuntu 7.04
Hi, Many thanks got it working now. All the best, Christian On 2007-05-04 at 13:31 Nitesh Divecha wrote: Christian, You can follow this procedure http://www.aussievoip.com/wiki/freePBX-Ubuntu Regards, Nitesh Christian wrote: Hi, I have already done: apt-get build-dep asterisk and then installed libpri, zaptel and asterisk from the latest sources. So what should i do then? New to Ubuntu. many thanks, Christian On 2007-05-04 at 17:00 Tzafrir Cohen wrote: On Fri, May 04, 2007 at 02:04:55PM +0200, Christian wrote: Hi all, Could someone please tell me how to make Asterisk start at boot on Ubuntu Feisty 7.04? Many thanks, Christian apt-get install asterisk Look at the init.d scripts. Note that in Ubuntu, subdirectories under /var/run are deleted at boot, and hence that script generates /var/run/asterisk (with proper ownership) at boot time. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.2 on CentOS 5?
Just wondering if anyone has tried using Asterisk 1.2 on CentOS 5. Is it worth considering for a Production install yet? Did they fix that spinlock.h Kernel problem? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] app_txfax, app_rxfax
Luki wrote: So essentially you need a pool of iaxmodems running on different ports, and then Dial() them until you find one that accepts your call. Or did I get that wrong? That seems really like a drawback to me The biggest drawback with app_rxfax is that if it crashes for whatever reason (happens sometimes), it will take down the entire PBX and all sessions with it. So you'd rather have the entire PBX crash in order to avoid creating sufficient iaxmodem instances to handle your fax call load? Lee. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Semi-OT: useful things to do with XML browsers inphones
Requests 1. Directories linked to their databases 2 weather broadcasts 3 local traffic info. 4 local news headlines 5 sms send / receive 6 alarm on the phone of calendar events - not a call back, simply a beep and notice without pulling out a pda or opening outlook Couple of other one of's too, some very esoteric like evening's TV lineup, Dave Bour Desktop Solution Center 905.381.0077 [EMAIL PROTECTED] For those who just want it to work... Giving you complete IT peace of mind. (Sent via Blackberry - hence message may be shorter than my usual verbose responses) PIN 4cc364db (as of March 24, 2007) - Original Message - From: [EMAIL PROTECTED] [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Thu May 03 10:18:06 2007 Subject: [asterisk-users] Semi-OT: useful things to do with XML browsers inphones Greetings list, It seems that more and more phones these days are coming with XML mini-browsers. I'd like to have a go at developing something useful to use on them, but in all honesty, most of our customers use their phones to make and take calls and very little else. So I'm open to suggestions. What useful applications are you developing for these mini-browsers? What sort of things do your customers want to use on them? Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited For full contact details visit http://www.minotaur.it/chris.html This email is made from 100% recycled electrons ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] app_txfax, app_rxfax
So you'd rather have the entire PBX crash in order to avoid creating sufficient iaxmodem instances to handle your fax call load? No, but so far this occurred only once in about a year of service. Not ideal, but acceptable considering Asterisk itself segfaults or deadlocks every now for no apparent reason. I had more trouble when trying to use T.38 with the newest app_rxfax so I abandoned it for now. And iaxmodem cannot do T.38 anyway... So you are saying a pool if iaxmodems and a loop through Dial() to find an open one is the way to go? --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: USB T1/E1 Interface?
[EMAIL PROTECTED] wrote: Why? There used to be a saying 'usb is for mice, firewire is for men', though USB has grown a bit in bandwidth since then, it is still not very well suited for a high sustained bandwidth. NOw T1/E1 is not that big, I suspect a lack of demand. Havng a E1 termintae in your laptop is quite useless, and a server usually has plenty of slots (if not, buy a bigger server ;-). Imagestream's low cost (about US$500) Envoy T1/E1 router actually uses a USB T1/E1 WAN 'Card'. I wonder how difficult is it to repurpose that card for voice :). Leo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re[2]: [asterisk-users] Starting Asterisk on Ubuntu 7.04
From: James Texter [EMAIL PROTECTED] Date: Fri, 04 May 2007 12:28:39 -0500 If you do make config when compiling zaptel and asterisk, it should put the script in /etc/init.d, and add the relevant entries to the various start levels. Not with 1.4 at least. makefile is not looking in the right place and not the right script. Yuan Liu Thanks, James Texter On Fri, 2007-05-04 at 18:44 +0200, Christian wrote: Hi, I have already done: apt-get build-dep asterisk and then installed libpri, zaptel and asterisk from the latest sources. So what should i do then? New to Ubuntu. many thanks, Christian On 2007-05-04 at 17:00 Tzafrir Cohen wrote: On Fri, May 04, 2007 at 02:04:55PM +0200, Christian wrote: Hi all, Could someone please tell me how to make Asterisk start at boot on Ubuntu Feisty 7.04? Many thanks, Christian apt-get install asterisk Look at the init.d scripts. Note that in Ubuntu, subdirectories under /var/run are deleted at boot, and hence that script generates /var/run/asterisk (with proper ownership) at boot time. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Starting Asterisk on Ubuntu 7.04
What I do is add an entry in the crontab file as such: * * * * * if [ ! `/bin/pidof -s asterisk` ] ; then /usr/sbin/asterisk; fi Its simple and it works. Additionally if asterisk crashes then cron restarts the server in about a minute. Just be careful with your configs. Mark Coccimiglio IS Manager Payroll Services Hawaii, Inc. http://www.psh-inc.com Christian wrote: Hi all, Could someone please tell me how to make Asterisk start at boot on Ubuntu Feisty 7.04? Many thanks, Christian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
