[Asterisk-Users] Fwd: [dmuars] Eh up - March 144 results altered
Here you go, Ian..-- Forwarded message --From: G3RIR [EMAIL PROTECTED]Date: 05-Apr-2006 20:54 Subject: [dmuars] Eh up - March 144 results alteredTo: [EMAIL PROTECTED] What's going on here. The results of the MArch 144 UKAC have been re-published and we have lost out considerably. Either I don't understand the rules or we have been robbed We scored 1159 G8VHI 928 G3RIR 154 G0TPH 133 G4OIG 333 G4ARI/P 98 G3CWI/P Totalling 2805 Cray have 2158 G4DBL 238 M3RCV 192 G3SPJ 27 G0KPZ 16 M3CVN/P Totalling 2631 Now we won so have 1000 points Cray should have (2631/2805)*1000 = 938 points They have been given 991 points! Why! Perhaps Peter can point out my error before I raise the issue with the adjudicator. Neil, G3RIR SPONSORED LINKS Craft hobby Hobby and craft supply Ham radio De montfort university YAHOO! GROUPS LINKS Visit your group dmuars on the web. To unsubscribe from this group, send an email to:[EMAIL PROTECTED] Your use of Yahoo! Groups is subject to the Yahoo! Terms of Service. -- Peter BowyerEmail: [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] Re: [dmuars] Eh up - March 144 results altered
Oops! Fat fingers, sorry, all. On 06/04/06, Peter Bowyer [EMAIL PROTECTED] wrote: Here you go, Ian.. ___ --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] VPB cannot call out
Hi DovidActually I dont how to set up my DTMF. Anyway here is the setting :-/etc/vpb/vtcore.conf[general]name = vtcorechannels=12cards=2[card0]type=openpcichannels=4hwplaygain=12hwrecordgain=-12chan = 0/etc/asterisk/vpb.conf[general]type = v4pcicards = 1[interfaces]board = 1echocancel = oncontext = from-pstnUseLoopDrop = 0mode = fxochannel = 0Using this setting, I can get the call. But when I tried to call out, it looks like it didnt set the DTMF. Do I need to configure any bal or txgain and rxgain setting? If so, what should I do? Thanks.Dovid Bender [EMAIL PROTECTED] wrote: Check your DTMF Settings.--- hensem boy <[EMAIL PROTECTED]> wrote: Hi all I have a problem when I want to call out using VPB trunk line, it cannot send the DTMF. Is there anyone has the same problem? Please share with me the solution. Thanks. - New Yahoo! Messenger with Voice. Call regular phones from your PC and save big.___ --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 __Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Blab-away for as little as 1¢/min. Make PC-to-Phone Calls using Yahoo! Messenger with Voice.___ --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] Chan-sccp - Asterisk dies
Hi group. I have install chan sccp drivers following instructions on http://chan-sccp.berlios.de/#build I have setup two Cisco 7970 phones. They register fine. When I call from one sccp phone another it rings, and when I pick up the phone asterisk dies. This is what it shows on CLI: -- SEP0016C87754CE: New call on line 342 -- SEP0016C87754CE: Cisco Digit: 0003 (3) on line 342 -- SEP0016C87754CE: Cisco Digit: 0004 (4) on line 342 -- SEP0016C87754CE: Cisco Digit: 0003 (3) on line 342 -- Executing Dial(SCCP/342-0001, SCCP/343) in new stack -- SEP0016C8528463: Asterisk request to call SCCP/343-0002 -- Called 343 -- SCCP/343-0002 is ringing -- SEP0016C8528463: Taken Offhook -- SEP0016C8528463: Answer the channel 343-2 -- SCCP/343-0002 answered SCCP/342-0001 -- SCCP: Outgoing call has been answered SCCP/342-0001 on [EMAIL PROTECTED] 754CE-1 Illegal instruction This is what I have in my full log file Apr 6 10:55:16 VERBOSE[27237] logger.c: Asterisk Event Logger restarted Apr 6 10:55:16 VERBOSE[27237] logger.c: Asterisk Queue Logger restarted Apr 6 10:55:22 VERBOSE[27242] logger.c: -- SEP0016C87754CE: Taken Offhook Apr 6 10:55:22 VERBOSE[27242] logger.c: -- SEP0016C87754CE: Using line 342 Apr 6 10:55:22 VERBOSE[27502] logger.c: -- SEP0016C87754CE: New call on line 342 Apr 6 10:55:24 VERBOSE[27242] logger.c: -- SEP0016C87754CE: Cisco Digit: 0003 (3) on line 342 Apr 6 10:55:24 VERBOSE[27242] logger.c: -- SEP0016C87754CE: Cisco Digit: 0004 (4) on line 342 Apr 6 10:55:25 VERBOSE[27242] logger.c: -- SEP0016C87754CE: Cisco Digit: 0003 (3) on line 342 Apr 6 10:55:25 VERBOSE[27502] logger.c: -- Executing Dial(SCCP/342-0001, SCCP/343) in new stack Apr 6 10:55:25 VERBOSE[27502] logger.c: -- SEP0016C8528463: Asterisk request to call SCCP/343-0002 Apr 6 10:55:25 VERBOSE[27502] logger.c: -- Called 343 Apr 6 10:55:25 VERBOSE[27502] logger.c: -- SCCP/343-0002 is ringing Apr 6 10:55:28 VERBOSE[27242] logger.c: -- SEP0016C8528463: Taken Offhook Apr 6 10:55:28 VERBOSE[27242] logger.c: -- SEP0016C8528463: Answer the channel 343-2 Apr 6 10:55:28 VERBOSE[27502] logger.c: -- SCCP/343-0002 answered SCCP/342-0001 Apr 6 10:55:28 VERBOSE[27502] logger.c: -- SCCP: Outgoing call has been answered SCCP/342-0001 on [EMAIL PROTECTED] Apr 6 10:55:28 DEBUG[27502] channel.c: Dropping duplicate answer! Anybody knows what could be the problem? -- Tomislav Parcina tparcina#lama.hr ___ --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] CallerID
how do you set two types of caller id one for internal calling and one for external? Basically everyone calling out from asterisk from one context I want to assign a single callerid. On all other contexts I want to assign a caller ID specific to each line for all calls going out to asterisk. Finally for all calls that remain behind the asterisk box (ext to ext) the Caller ID is set to the specific extension of the caller. Thanks Miles ___ --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] How to restrict simultaneous phone registrations
On 17:47, Wed 05 Apr 06, Bryan Mahin wrote: Hello all, I am looking for a way to restrict users from logging in two separate phones with the same authorization name/password at the same time. Meaning, I only want users to be able to place a call from one phone in one location, but have the ability to move from computer to computer. Has anyone found any sort of solution for this type scenario? Use agents. When agent X logs in on location A, the other phone that is logged in as agent X will be logged off. -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Why is it drug addicts and computer afficionados are both called 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] ASTCC: How to reset in-use flag automatically ?
JP Carballo wrote:
Ronald Wiplinger wrote:
I tried now many places to put these lines in. The system still
announces This card number is in use.
Can you give me a place where to put it in?
It's not receiving a card number.
Find the following 3 lines:
#
# At this point we have a valid card number.
#
Insert the whole routine either just before or after these lines.
There I have it
#
# At this point we have a valid card and pin number.
#
if ($phoneno eq RESET_INUSE) {
setinuse($carddata-{number}, 0);
exit(0);
}
checkexpired($carddata-{number});
checkinuse($carddata-{number});
setinuse($carddata-{number}, 1);
I put this into 682 in the extensions.conf
exten = 681,1,DeadAGI(astcc.agi,${CALLERID(num)},BALANCE,1)
exten = 681,2,Hangup
exten = 682,1,DeadAGI(astcc.agi,${CALLERID(num)},RESET_INUSE,2)
exten = 682,2,Hangup
As soon the flag is set, 682 will also tell you: The card number is in
use, try later !
What do I miss?
bye
Ronald Wiplinger
begin:vcard
fn:Ronald Wiplinger
n:Wiplinger;Ronald
org:ELMIT Co., Ltd.
adr:Shilin District;;5F., No.8, Alley 2, Lane 92, Dexing W. Road;Taipei;;11158;Taiwan
email;internet:[EMAIL PROTECTED]
title:CEO
tel;work:+886.2.2835.7765
tel;cell:+886.939.775.516
x-mozilla-html:TRUE
url:http://www.elmit.net
version:2.1
end:vcard
___
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RE: [Asterisk-Users] IAX: Auto-congesting call due to slow response
maybe firewall tends to close iax connection, you can try to decrease qualify check interval (maybe qualify=5000?) PJ Peraphs. 'qualify = 1000' seems to alleviate the problem. Thanks Domenico ___ --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] fax server functionality on Asterisk
how can I put fax server functionality on Asterisk? * as a reliable fax server for 500-1000 fax/day (mostly incoming)? Fax server should be like HylaFax, i.e. stable, low maintenance and functionality like receiving fax as email with PDF attachment, sending faxes per WHFC. Asterisk doesn't natively offer fax support, you can get this only using SpanDsp and managing send/receive by dialplan. To manage this large number of faxes, it's better to use Hylafax and not Asterisk. Obviously, all this IMHO DV ___ --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] legacy Alcatel 4200/4400 and Asterisk (QSIG/PRI)and callerid
Hi, I have same setup: PSTN E1 PRI --- Asterisk --- Crossed E1 cable --- Alcatel 4400 PBX with some IP phones directly connected to Asterisk and a lot of analog/digital phones connected to 4400. When I call from an IP phone to an Alcatel one, I'm able to see full CallerIDName. I set it using: Set(CALLERID(name)=...) but you can use also: SetCIDName(...) even if it is deprecated in 1.2.x I don't know if I'm using Q.Sig or EuroISDN! Bye DV -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Miroslav HOSTINSKY Sent: Wednesday, April 05, 2006 7:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] legacy Alcatel 4200/4400 and Asterisk (QSIG/PRI)and callerid Hello, I have connected asterisk box with legacy PBX Alcatel OmniPCX 4400 (and also another * box connected to A4200). These PBXes have function to assign name to extensions and display it on phone. Asterisk box is connected via PRI with euroISDN signalling (also I have tried QSIG). Is it possible to set callerid with name and display it on alcatel digital phones? With command SetCALLERID I am able only set callerid number (and name) but on phone is always only callerid number... thanks... ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_modem_i4l delay again..
Hi, I currently use Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k on a debian sarge with a kernel 2.4.27 on a P4 3Gig with 1Gig of memory When i use i4l on any call, the called party ( on the telco operator side ) ear me with a delay of 1 sec after 1 minutes , 2 sec after 3 minutes and so on... After a quart hour, the delay make the conversation just impossible !!! I use a tdm400P to connect my analogs phones and all is working very well between two zap stations. I have tried different Passive isdn card ( no hfc so I can't use zaphfc driver) Anybody have an idea to fix this problem ? BTW, I have compiled my kernel with the dtmf patch for isdn_tty.c so The cpu usage is 25% during a conversation, 75% idle I have a PCI latency of 32 msec With or without APIC, no changes It seems that the voice is buffered and sended too slowly to the i4l channel and so a delay is present afetr a short time and became bigger minutes after minutes... Alain Degreffe ___ --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] legacy Alcatel 4200/4400 and Asterisk (QSIG/PRI)and callerid
Mimmus napisał(a): Hi, I have same setup: PSTN E1 PRI --- Asterisk --- Crossed E1 cable --- Alcatel 4400 PBX I don't know if I'm using Q.Sig or EuroISDN! 1) it's in config file 2) Should be easy to check when you say what kind of PABX card you use: PRA/PRA2/BRA2 - EuroISDN DLT - qsig -- Krzysztof Drewicz Affordable 2/4 span E1/T1 PCI-cards. 100% Asterisk compatible. See http://4e1.pl ___ --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: Re: H323 problems
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... On 04/04/06 19:20 Tomislav Pareina said the following: Ooh323 channel driver from asterisk-addons-1.2.1 has same problem have you managed to get this working ? I certainly hope so, but I'm not sure. I have applied patch yesterday. Now I'm waiting... :)) Here are instruction that Sam has posted on ooh323 mailing list. You can apply patch, or wait few days when I'll announce does it work :)) P.S. You can install ooh323 channel drivers form Asterisk-addons-1.2.2 they should work also, but I'm unable to install them on Fedora Core 4 (liptoolize/automake problems) -- Tomislav Parcina tparcina#lama.hr *** Subject: ooh323 Deadlocks resolved. From: Sean Lowry [EMAIL PROTECTED] Newsgroups: gmane.comp.telephony.ooh323.c Hello all, I bring you all some great news about ooh323 and deadlocks. I have been running some patched code on a system (full debug) for 24+ hours now without one deadlock, which used to happen quite frequently before ( under and hour ). So thanks to Avin from obj-sys and all his hard work here's how you go about updating to a stable deadlock free channel. Connect to asterisk cvs server and check out latest asterisk-addons cvs co asterisk-addons (the cvs asterisk-addons works perfectly with asterisk 1.2 stable) Goto: asterisk-addons/asterisk-ooh323c/ Download this patch. wget http://www.obj-sys.com/open/changes1.2.2.tar.gz Extract tar zxvf changes1.2.2.tar.gz make clean ./configure vi Makefile change like 72 DEBUG_THREADS = -DDUMP_SCHEDULER -DDEBUG_SCHEDULER -DDEBUG_THREADS -DDETECT_DEADLOCKS #-DDO_CRASH To: DEBUG_THREADS = #-DDUMP_SCHEDULER #-DDEBUG_SCHEDULER #-DDEBUG_THREADS #-DDETECT_DEADLOCKS #-DDO_CRASH Now make make install This will create your new chan_ooh323.so and install it. Hope this helps everyone if you have any trouble don't hesitate to email the list. Regards Sean. --- This SF.net email is sponsored by: Splunk Inc. Do you grep through log files for problems? Stop! Download the new AJAX search engine that makes searching your log files as easy as surfing the web. DOWNLOAD SPLUNK! http://ads.osdn.com/?ad_id=7637alloc_id=16865op=click ___ --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-ooh323, asterisk 1.2.6 and netmeeting
On 04/06/06 05:36 Avi Miller said the following: If I dialled from a SIP phone on Asterisk 1 to the Phone on the Avaya, it worked fine. If I dialled from a phone on the Avaya, the SIP phone would ring, but the call would drop as soon as it was answered because of codec negotiation failure. absolutely the same symptoms. my architecture is as follows: OHPHONE Asterisk SIP Client calls from the SIP client to OHPHONE work fine with audio et al passed both ways. calls from OH PHONE to the SIP client dont. just after the SIP client answers, the call dies. i tried your suggestion of removing all disallow and allow lines in ooh323.conf, but with that, even calls from SIP to H323 (which were working) stop working. it does lend credence to the theory that it's a codec nego issue though. the debug and verbose output of a failed H323 to SIP call is below (6262 is the SIP exten and 6996 is the OHPHONE H.323): Apr 6 13:59:37 VERBOSE[201] logger.c: -- Executing Dial(OOH323/192.168.1.169-0361, SIP/6262|40|owWtT) in new stack Apr 6 13:59:37 DEBUG[201] chan_sip.c: Setting NAT on RTP to 0 Apr 6 13:59:37 DEBUG[201] chan_sip.c: Setting NAT on VRTP to 0 Apr 6 13:59:37 DEBUG[201] acl.c: # Testing 192.168.1.164 with 192.168.1.0 Apr 6 13:59:37 DEBUG[201] chan_sip.c: Outgoing Call for 6262 Apr 6 13:59:37 VERBOSE[201] logger.c: -- Called 6262 Apr 6 13:59:37 DEBUG[201] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 102: Found Apr 6 13:59:37 VERBOSE[201] logger.c: -- SIP/6262-960b is ringing Apr 6 13:59:37 DEBUG[201] channel.c: Driver for channel 'OOH323/192.168.1.169-0361' does not support indication 3, emulating it Apr 6 13:59:37 DEBUG[201] channel.c: Prodding channel 'OOH323/192.168.1.169-0361' Apr 6 13:59:37 DEBUG[201] channel.c: Scheduling timer at 160 sample intervals Apr 6 13:59:37 DEBUG[201] chan_sip.c: Auto destroying call '[EMAIL PROTECTED]' Apr 6 13:59:37 DEBUG[201] acl.c: # Testing 192.168.1.151 with 192.168.1.0 Apr 6 13:59:37 DEBUG[201] chan_sip.c: SIP message could not be handled, bad request: [EMAIL PROTECTED] Apr 6 13:59:38 DEBUG[201] chan_sip.c: Acked pending invite 102 Apr 6 13:59:38 DEBUG[201] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Apr 6 13:59:38 DEBUG[201] chan_sip.c: build_route: Contact hop: sip:[EMAIL PROTECTED]:5060 Apr 6 13:59:38 VERBOSE[201] logger.c: -- SIP/6262-960b answered OOH323/192.168.1.169-0361 Apr 6 13:59:38 WARNING[201] src/chan_h323.c: Don't know how to indicate condition -1 on ooh323c_7 Apr 6 13:59:38 DEBUG[201] channel.c: Scheduling timer at 0 sample intervals Apr 6 13:59:38 VERBOSE[201] logger.c: -- Attempting native bridge of OOH323/192.168.1.169-0361 and SIP/6262-960b Apr 6 13:59:38 DEBUG[201] channel.c: Didn't get a frame from channel: OOH323/192.168.1.169-0361 Apr 6 13:59:38 DEBUG[201] channel.c: Bridge stops bridging channels OOH323/192.168.1.169-0361 and SIP/6262-960b Apr 6 13:59:38 DEBUG[201] chan_sip.c: update_call_counter(6262) - decrement call limit counter Apr 6 13:59:38 DEBUG[201] app_dial.c: Exiting with DIALSTATUS=ANSWER. Apr 6 13:59:38 VERBOSE[201] logger.c: == Spawn extension (macro-stdexten, s-DIAL, 1) exited non-zero on 'OOH323/192.168.1.169-0361' in macro 'stdexten' Apr 6 13:59:38 VERBOSE[201] logger.c: == Spawn extension (macro-stdexten, s-DIAL, 1) exited non-zero on 'OOH323/192.168.1.169-0361' Apr 6 13:59:39 DEBUG[201] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 103: Match Found Apr 6 13:59:40 DEBUG[201] chan_sip.c: Auto destroying call '[EMAIL PROTECTED]' note that channel.c says it didnt get a frame from OHPHONE and that it subsequent stops bridging the channels. now to go figure out why this is so. any pointers would be appreciated. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --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] legacy Alcatel 4200/4400 andAsterisk (QSIG/PRI)and callerid
I don't know if I'm using Q.Sig or EuroISDN! 1) it's in config file 2) Should be easy to check when you say what kind of PABX card you use: PRA/PRA2/BRA2 - EuroISDN DLT - qsig OK, I'm using EuroISDN. Thanks DV ___ --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] Hinting a conference room
Hi there!I was asked to set up a led on a snom phone monitoring a conference room (lit when someone is in conference).I know that there is a patch for hinting parking lots, anyone has made something similiar for conferences ? Tnx for the support!P.S.What about monitoring a global var ?It would be absolutely great variable=0 led off, 1 led on, 2 led blink ... Alessio Focardi ___ --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: Hangupcause is not enough on PRI
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Hi, I'm using Asterisk and a TE110P E1 PRI in Chile. When I call to a disconnected number or any not operational number, the telco sends the Hangupcause disconnection code and an audio message notifying the disconnection cause to the user. Asterisk does not allow the user to hear the audio message form the telco, instead it cuts the call. Any other legacies PRI PBX I've tested allow the user to hear the audio message from the telco. A few months ago I was dealing with this problem (making the user hear the disconnection cause message from the telco) and someone suggested using the Hangupcause code (http://lists.digium.com/pipermail/asterisk-users/2005-December/133374.html) , and it solved the problem momentarily. Now, when I call to a not operational number, depending on the Hangupcause variable, Asterisk plays an internal audio message notifying the user about the disconnection cause, but my client is not satisfied with that, he expect to hear the real audio messager form the telco. I would like to know if somebody solved this issue letting the user hear the real disconnection cause message form the telco. Hi Javier! I have a same problem in Croatia with Optima provider. Please, if you find the solution, mail it to the group. -- Tomislav Parcina tparcina#lama.hr ___ --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] IVR : Can't hear my message
Hello,I've reccorded a voice message for the IVR. (.wav, 16 bits, 8kHz)The file is /var/lib/asterisk/sound/11ivrrecording.wav.When asterisk (1.2.5) starts this file i can't hear it on my phone.Here is the log : Apr 6 17:00:16 VERBOSE[845] logger.c: -- Executing SetCallerID(SIP/11-97b9, Patrice 11) in new stackApr 6 17:00:16 VERBOSE[845] logger.c: -- Executing NoOp(SIP/11-97b9, Using CallerID Patrice 11) in new stack Apr 6 17:00:16 VERBOSE[845] logger.c: -- Executing Playback(SIP/11-97b9, 11ivrrecording) in new stackApr 6 17:00:16 DEBUG[845] channel.c: Scheduling timer at 160 sample intervals Apr 6 17:00:16 VERBOSE[845] logger.c: -- Playing '11ivrrecording' (language 'en')Apr 6 17:00:17 DEBUG[26916] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED] ' of Response 2: Match FoundApr 6 17:00:49 DEBUG[26916] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Apr 6 17:00:50 DEBUG[845] channel.c: Scheduling timer at 0 sample intervalsApr 6 17:00:50 VERBOSE[845] logger.c: == Spawn extension (from-internal, *99, 2) exited non-zero on 'SIP/11-97b9'Anyone has an idea ? Thanks a lot.Antoine ___ --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] Fwd: Hangup Supervision
Hi all, I need help in disconnect supervision. Im running on AAH ver.2.5 at home with TDM400P with 1 FXO and 1 FXS (TDM11B). I have implemented DISA on AAH for origination (PSTN to VOIP bridging). I'm facing problems with disconnection supervision. My calls are not getting disconnected at times and it causes a lot of loss as the provider is charging me. After some serious study i have found that my provider (unusual in other part of the world) is not having a busy tone on disconnection and rather a long tone. (i.e - no ON and OFF). Hence the digium card is not able to identify the disconnection. I have also found out that the tone is having frequency of 425 and its Continous for around 10 seconds after i hangup. Does anybody has a fix for this in the configs so it will help my asterisk identify my hangup. Without this im not able to proceed. Thanks in advance. Dan ___ --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] Incoming call redirected to mobile
Asterisk SVN-trunk-r7353M I have a EuroISDN line. I am sometimes out of the office so I get my extension to ring both my mobile and desk top (7960) phone at the same time. This all works just peachy. However, I have a question regarding callerid. Is there any way of setting the callerid so that I can see the number that is calling me on my mobile (I see it on the desktop) rather than the number assigned to my ISDN line ? Julian. ___ --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] How to restrict simultaneous phone registrations
The only thing registration does is inform Asterisk about what IP the device is at. It has nothing at all to do with Device - Asterisk calls. Registration only affects Asterisk - Device calls. In a Device - Asterisk call, Asterisk does not care what IP the device is at as long as the correct user/password are provided. Bryan Mahin wrote: :) I should rephrase my question. And included a bit more information on what I am trying to accomplish. Solution 1 (preferred) I am working on an asterisk installation where most end users will use softphones. If I am not able to lock down calling to one phone at a time, the end users will share their login information with friends, family, neighbors, and the some girl they meet on myspace. Currently, I am able to register two phones at separate locations with the same account on each phone and make concurrent calls. For example, If I login extension 333 at location A, and 333 at location B, simultaneous calls can be placed from both phones at the exact same time. I only want calls placed from extension 333 to work from either A or B not A and B concurrently. Here is my ideal solution. Location A wants to make a call, but location B has a call in progress. Location B has to either close their phone, or hang up before Location A can make the call. OR.. Solution 2. :) A way I can distinguish in my CDR the IP address or some other recognizable difference between the two locations when they make concurrent calls using the same extension. The complication here is; I can currently the log IP addresses, but as the end phones are on the internet, Nat'd, and I am using a siparator for traversal. As a result, my logs show the IP address of the siparator and I don't have any other data to distinguish the end phones. OR.. Solution 2.5 One thought I've had is to send logs from the siparator to a syslog server, parse them, hunt for simultaneous calls placed by the same accounts from different locations, and bill the end users accordingly. But I really dislike this idea as no one likes to be hit with surcharges. ___ --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] Incoming call redirected to mobile
Julian Lyndon-Smith wrote: Asterisk SVN-trunk-r7353M I have a EuroISDN line. I am sometimes out of the office so I get my extension to ring both my mobile and desk top (7960) phone at the same time. This all works just peachy. However, I have a question regarding callerid. Is there any way of setting the callerid so that I can see the number that is calling me on my mobile (I see it on the desktop) rather than the number assigned to my ISDN line ? The original Caller*ID will be sent. However, your provider may be overriding this information and setting the Caller*ID of your main number. ___ --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] Call transfer to cell phone
Hi! Is anyone managed to transfer an alredy bridged call, to a cell phone? Some days ago, someone told me to look for the solution in features.conf, but I still haven't found it. I tryied to use de default blindxfer, but it only accept internal extensions. Thanks in advance, Giuseppe ___ --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] Dial out on Zap
Hi, I'm trying to test my dial out function so I did something like this in extensions.conf exten = 999,1,Dial(Zap/g1/02601591) exten = 999,102,Congestion() My Zapata.conf looks something like this [channels] context=from-pstn group=0 switchtype=euroisdn overlapdial=yes faxdetect=no ; PRI port 1 (E1) ; context=1 group=1 signalling=pri_cpe channel=1-15,17-31 I am able to receive the fax just fine with this setting. So I think it's ok. I'm using a Digium card connecting to a PSTN. There're 4 ports on the card, 31 channels each, but we currently use one. When I call extension 999, it was supposed to forward my call to 02601591, right? But it didn't. It just gets silence and it hangs up the call. On my CLI it looks something like this: -- Executing Dial(Zap/1-1, Zap/g1/02601591) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g1/02601591 -- Moving call from channel 1 to channel 2 Apr 6 11:08:08 WARNING[5854]: chan_zap.c:7745 pri_fixup_principle: Can't fix up channel from 1 to 2 because 2 is already in use Apr 6 11:08:08 WARNING[5854]: chan_zap.c:9046 pri_dchannel: Unable to move channel 2! -- Zap/2-1 is proceeding passing it to Zap/1-1 Apr 6 11:08:22 NOTICE[5849]: chan_iax2.c:5691 update_registry: Restricting registration for peer 'hylafax-iaxmodem' to 60 seconds (requested 300) -- Channel 0/2, span 1 got hangup request - I didn't hang up the call. It did by itself. -- Hungup 'Zap/2-1' == Everyone is busy/congested at this time (1:0/0/1) -- Executing Congestion(Zap/1-1, ) in new stack -- Channel 0/1, span 1 got hangup request == Spawn extension (voice, 999, 102) exited non-zero on 'Zap/1-1' -- Executing SetVar(Zap/1-1, HANGUP_TIME=1144314509) in new stack -- Executing NoOp(Zap/1-1, 16) in new stack -- Hungup 'Zap/1-1' I'm a bit confused what I did wrong. Do I need a second line or something?? Pim ___ --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: Can't get Pickup app working
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... I'm trying to set the Pickup feature. I'm setting my extensions.conf as: I'm using pickup from features.conf. I don't need anything better (for now). -- Tomislav Parcina tparcina#lama.hr ___ --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
AW: [Asterisk-Users] Dial out on Zap
Hi, i was able to fix this problem when i added the line pridialplan=local in the zapata.conf but it depends on your telco, i think. marcus -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Gesendet: Donnerstag, 6. April 2006 11:50 An: asterisk-users@lists.digium.com Betreff: [Asterisk-Users] Dial out on Zap Hi, I'm trying to test my dial out function so I did something like this in extensions.conf exten = 999,1,Dial(Zap/g1/02601591) exten = 999,102,Congestion() My Zapata.conf looks something like this [channels] context=from-pstn group=0 switchtype=euroisdn overlapdial=yes faxdetect=no ; PRI port 1 (E1) ; context=1 group=1 signalling=pri_cpe channel=1-15,17-31 I am able to receive the fax just fine with this setting. So I think it's ok. I'm using a Digium card connecting to a PSTN. There're 4 ports on the card, 31 channels each, but we currently use one. When I call extension 999, it was supposed to forward my call to 02601591, right? But it didn't. It just gets silence and it hangs up the call. On my CLI it looks something like this: -- Executing Dial(Zap/1-1, Zap/g1/02601591) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g1/02601591 -- Moving call from channel 1 to channel 2 Apr 6 11:08:08 WARNING[5854]: chan_zap.c:7745 pri_fixup_principle: Can't fix up channel from 1 to 2 because 2 is already in use Apr 6 11:08:08 WARNING[5854]: chan_zap.c:9046 pri_dchannel: Unable to move channel 2! -- Zap/2-1 is proceeding passing it to Zap/1-1 Apr 6 11:08:22 NOTICE[5849]: chan_iax2.c:5691 update_registry: Restricting registration for peer 'hylafax-iaxmodem' to 60 seconds (requested 300) -- Channel 0/2, span 1 got hangup request - I didn't hang up the call. It did by itself. -- Hungup 'Zap/2-1' == Everyone is busy/congested at this time (1:0/0/1) -- Executing Congestion(Zap/1-1, ) in new stack -- Channel 0/1, span 1 got hangup request == Spawn extension (voice, 999, 102) exited non-zero on 'Zap/1-1' -- Executing SetVar(Zap/1-1, HANGUP_TIME=1144314509) in new stack -- Executing NoOp(Zap/1-1, 16) in new stack -- Hungup 'Zap/1-1' I'm a bit confused what I did wrong. Do I need a second line or something?? Pim ___ --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] Call transfer to cell phone [UPDATE]
Hi!
I tried this in features.conf
testfeature = *9,callee,Dial,CAPI/ISDN4/my_phone_number/b,60,T
and it works... but... I would be able to transfer a call to any phone
number,
so I tried to use this line:
testfeature = _*9.,callee,Dial,CAPI/ISDN4/${EXTEN:2}/b,60,T
but... Asterisk crash! (it doesn't want even to reload configuration)
Any idea about how to do so? Thanks a lot!
Giuseppe
--
In my last email I wrote:
Hi!
Is anyone managed to transfer an alredy bridged call, to a cell phone?
Some days ago, someone told me to look for the solution in
features.conf,
but I still haven't found it. I tryied to use de default blindxfer,
but it only
accept internal extensions.
Thanks in advance,
Giuseppe
--
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RE: [Asterisk-Users] can't start chan_capi with asterisk group
Thanks Armin, It works with rw-rw-rw permissions to /dev/capi20. Amaury -Message d'origine- De : Armin Schindler Envoyé : mercredi 5 avril 2006 19:49 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [Asterisk-Users] can't start chan_capi with asterisk group It should work with that permissions. Does it work with other group/user settings? Just for a try, set /dev/capi20 to rw-rw-rw Armin On Wed, 5 Apr 2006, amaury BOSSE wrote: Hello, While upgrading * from 1.0.9 to 1.2.5, I have installed chan-capi-head and I can't start asterisk under asterisk group asterisk -gc -U asterisk and asterisk -gc -U asterisk -G dialout work well but asterisk -gc -U asterisk -G asterisk fail. I am thinking about a group permission configuration but I have exactly the same one than with my old 1.0.9 working config. Log messages when launching asterisk -gc -U asterisk -G asterisk : Apr 5 17:47:21 VERBOSE[5773] logger.c: [chan_capi.so]Apr 5 17:47:21 VERBOSE[5773] logger.c: [chan_capi.so] = (Common ISDN API for Asterisk) Apr 5 17:47:21 VERBOSE[5773] logger.c: == Parsing '/etc/asterisk/capi.conf': Apr 5 17:47:21 VERBOSE[5773] logger.c: == Parsing '/etc/asterisk/capi.conf': Found Apr 5 17:47:21 WARNING[5773] chan_capi.c: CAPI not installed, CAPI disabled! Apr 5 17:47:21 WARNING[5773] loader.c: chan_capi.so: load_module failed, returning -1 Apr 5 17:47:21 WARNING[5773] loader.c: Loading module chan_capi.so failed! Ls -l /dev/capi20 : crw-rw 1 root dialout 68, 0 2006-03-24 14:49 /dev/capi20 id asterisk : uid=105(asterisk) gid=105(asterisk) groupes=105(asterisk),20(dialout),33(www-data) Any idea about why I can't start chan_capi with asterisk group? ___ --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] not get ring tone with chan-capi and avm b1
Hi.
First, pardon my bad English.
I have * configured with one avm b1 and latest chan-capi. I can dial out and receive incoming calls from isdn.
The problem is that i do not know the way to get the ring tone (hear the ringing on the caller phone when i dial with capi)
For example if i dial 0 from an ip phone and any right number i get
outside line and the destination phone is ringing, but i can not hear
anything (busy or ringing tones in the phone that makes the call). If
someone answer the call we both can speak perfectly.
I have tested several capi dial options but i can not find the doc where all the possible parameters or options are specified.
I show you my dial string:
(first there is a menu and user dials 1 (first option) RDSI is the
number of ISDN line, and DESTINATION is a valid target number)
exten =1,n,Dial(CAPI/g1/${RDSI}:${DESTINATION},30)
The ISDN hardware:
capiinfo
Number of Controllers : 1
Controller 1:
Manufacturer: AVM GmbH
CAPI Version: 2.0
Manufacturer Version: 3.11-03 (49.19)
Serial Number: 4007868
BChannels: 2
Global Options: 0x0039
internal controller supported
DTMF supported
Supplementary Services supported
channel allocation supported (leased lines)
B1 protocols support: 0x401f
64 kbit/s with HDLC framing
64 kbit/s bit-transparent operation
V.110 asynconous operation with start/stop byte framing
V.110 synconous operation with HDLC framing
T.30 modem for fax group 3
B2 protocols support: 0x0b1b
ISO 7776 (X.75 SLP)
Transparent
LAPD with Q.921 for D channel X.25 (SAPI 16)
T.30 for fax group 3
ISO 7776 (X.75 SLP) with V.42bis compression
V.120 asyncronous mode
V.120 bit-transparent mode
B3 protocols support: 0x803f
Transparent
T.90NL, T.70NL, T.90
ISO 8208 (X.25 DTE-DTE)
X.25 DCE
T.30 for fax group 3
T.30 for fax group 3 with extensions
0100
0200
3900
1f40
1b0b
3f80
0101 0002
Supplementary services support: 0x03ff
Hold / Retrieve
Terminal Portability
ECT
3PTY
Call Forwarding
Call Deflection
MCID
CCBS
Can someone show me the way to solve this?
I think that i need some parameter or extra option on dial string but i did not find the right one.
Thanks.
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RE: [Asterisk-Users] can't start chan_capi with asterisk group
Okay, so your group settings/permissions are not correct then. Armin On Thu, 6 Apr 2006, amaury BOSSE wrote: Thanks Armin, It works with rw-rw-rw permissions to /dev/capi20. Amaury -Message d'origine- De : Armin Schindler Envoyé : mercredi 5 avril 2006 19:49 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [Asterisk-Users] can't start chan_capi with asterisk group It should work with that permissions. Does it work with other group/user settings? Just for a try, set /dev/capi20 to rw-rw-rw Armin On Wed, 5 Apr 2006, amaury BOSSE wrote: Hello, While upgrading * from 1.0.9 to 1.2.5, I have installed chan-capi-head and I can't start asterisk under asterisk group asterisk -gc -U asterisk and asterisk -gc -U asterisk -G dialout work well but asterisk -gc -U asterisk -G asterisk fail. I am thinking about a group permission configuration but I have exactly the same one than with my old 1.0.9 working config. Log messages when launching asterisk -gc -U asterisk -G asterisk : Apr 5 17:47:21 VERBOSE[5773] logger.c: [chan_capi.so]Apr 5 17:47:21 VERBOSE[5773] logger.c: [chan_capi.so] = (Common ISDN API for Asterisk) Apr 5 17:47:21 VERBOSE[5773] logger.c: == Parsing '/etc/asterisk/capi.conf': Apr 5 17:47:21 VERBOSE[5773] logger.c: == Parsing '/etc/asterisk/capi.conf': Found Apr 5 17:47:21 WARNING[5773] chan_capi.c: CAPI not installed, CAPI disabled! Apr 5 17:47:21 WARNING[5773] loader.c: chan_capi.so: load_module failed, returning -1 Apr 5 17:47:21 WARNING[5773] loader.c: Loading module chan_capi.so failed! Ls -l /dev/capi20 : crw-rw 1 root dialout 68, 0 2006-03-24 14:49 /dev/capi20 id asterisk : uid=105(asterisk) gid=105(asterisk) groupes=105(asterisk),20(dialout),33(www-data) Any idea about why I can't start chan_capi with asterisk group? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: queue issue
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... on a related note, we notice that if we've set atxfer = *1 in features.conf and blindxfer=#1, then attended transfers dont work. somehow, the Queue app captures the '*' and hangs up the call. is this the behaviour others have observed ? obviously, since we've used *2 for auto monitor, that doesnt work as well. Yes, this is well known (problem?). I have solved it by editing features.conf file. -- Tomislav Parcina tparcina#lama.hr ___ --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] qozap errors on junghanns QuadBRI
Is there a fix for these errors for the junghanns card ? Apr 6 13:11:08 asterix qozap: CRC error for HDLC frame on card 1 (cardID 0) S/T port 1 Apr 6 13:11:35 asterix qozap: CRC error for HDLC frame on card 1 (cardID 0) S/T port 3 Apr 6 13:11:39 asterix qozap: CRC error for HDLC frame on card 1 (cardID 0) S/T port 1 Apr 6 13:11:40 asterix qozap: dropped audio card 1 cardid 0 bytes 15 z1 90 z2 59 No interrupts problems on that machine, no errors in zttool. -- Andrzej Wolski e-mail: [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] FXO/FXS and E1 in same system
Hi, can i have a FXO/FXS card and a E1/T1 card in the same system. I have used them seperatly many times before, but not together in one machine. I usually have for the analogue card signalling=fxs_ks channel = 1 and for the e1 card signalling=pri_net group=1 callerid=asreceived channel = 1-15,17-31 how will the channel numbers change with two cards. thanks, yusuf ___ --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] How to restrict simultaneous phone registrations
I apologize if this information is posted elsewhere. Unfortunately I haven't found it yet if it is. I'm not familiar with the channel counting features could you please explain? Also, how are you tagging the phones to account codes? You can limit calls using the set/check group commands. http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+SetGroup Account codes are set either by using the Set function or the accountcode= property in the SIP/IAX conf files. -Jonathan ___ --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] qozap errors on junghanns QuadBRI
Andrzej Wolski napisał(a): Is there a fix for these errors for the junghanns card ? Apr 6 13:11:08 asterix qozap: CRC error for HDLC frame on card 1 Witam, Przepraszam za komercyjny charakter tego maila, ale jeśli byłby Pan zainteresowany to za kilka tygodni otwieramy w pełni sprzedaż kart 4xE1: http://www.4e1.pl/shop/catalog/product_info.php?products_id=28 Są one o wiele tańsze i łatwiejsze w konfiguracji niż BRI. Pozdrawiam, ps. mamy ograniczoną ilość kart które możemy wypożyczać do testów -- Krzysztof Drewicz Affordable 2/4 span E1/T1 PCI-cards. 100% Asterisk compatible. See http://4e1.pl ___ --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 dialing over asterisk to PSTN
hello all soembody can give me an example config how can i let dial a asterisk server via SIP over another asterisk server to a pstn gateway ip?!?! asterisk1: x.x.x.x have to dial over asterisk2: y.y.y.y and then the asterisk2 should forward the call to the PSTN gateway. What i have to set in sip.conf that asterisk1 can dial over asterisk2? regards rene ___ --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] FXS module failed
Hi, I have Wildcard TDM400P with 2 FXS y 2 FXO. After all work fine but now do it: - load driver: wctdm y zaptel (zaptel-1.2.1) Module 0: Installed -- AUTO FXS/DPO Unable to do INITIAL ProSLIC powerup on module 1 Unable to do INITIAL ProSLIC powerup on module 1 Module 1: FAILED FXS (FCC) Module 2: Installed -- AUTO FXO (FCC mode) Module 3: Installed -- AUTO FXO (FCC mode) Found a Wildcard TDM: Wildcard TDM400P REV E/F (3 modules) Registered tone zone 0 (United States / North America) - /etc/zaptel.conf loadzone=us defaultzone=us fxoks=1,2 fxsks=3,4 I don't now why make it. Find in search engines but not look nothing. Thanks for your help. Gabriel ___ --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: What causes deadlock?
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Hi What causes deadlock? Apr 5 14:02:43 WARNING[2413] channel.c: Avoided initial deadlock for '0x82acb10', 10 retries! Apr 5 14:02:43 WARNING[2413] channel.c: Avoided initial deadlock for '0x8298160', 10 retries! Does this happen with ooh323 channel driver? -- Tomislav Parcina tparcina#lama.hr ___ --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: What causes deadlock?
Tomislav Parčina wrote: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Hi What causes deadlock? Apr 5 14:02:43 WARNING[2413] channel.c: Avoided initial deadlock for '0x82acb10', 10 retries! Apr 5 14:02:43 WARNING[2413] channel.c: Avoided initial deadlock for '0x8298160', 10 retries! Does this happen with ooh323 channel driver? -- Tomislav Parcina tparcina#lama.hr ___ --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 sip to sip channels as well ___ --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] TDM2400P problems
I am having issues with a TDM2400P. It appears when the ZAP channel dials out, it randomly chops the first digit off of the number. I have tried relaxdtmf=yes, turning up and down the txgain, turned off and on the echo cancellation, generated new zaptel (with updated spinlock.h)... I am at a loss. Can someone please offer some help? Thanks. TJ ___ --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: CallerID
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
how do you set two types of caller id one for internal calling and one
for external? Basically everyone calling out from asterisk from one
context I want to assign a single callerid. On all other contexts I
want to assign a caller ID specific to each line for all calls going out
to asterisk.
Finally for all calls that remain behind the asterisk box (ext to ext)
the Caller ID is set to the specific extension of the caller.
That's easy.
In sip.conf define caller id for every telephone that you wont them to have in
internal calls.
In every context put something like this.
exten = _0.,1,Set(CALLERID(name)=Lama.hr)
exten = _0.,n,Set(CALLERID(number)=00038521495148)
exten = _0.,n,Dial(OOH323/${EXTEN:[EMAIL PROTECTED],60,TW)
exten = _0.,n,Hangup
Hope it helps.
--
Tomislav Parcina
tparcina#lama.hr
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Re: [Asterisk-Users] TDM2400P problems
We have had this problem with the TDM400 and just about every thing we
have ever had... it isn't the card that is chopping off the first
digit. It is the fact that it picks up too quickly and starts to dial.
Change your dial to be Zap/g0/w${EXTEN} and see if that takes care of
the problem
Tim Jackson wrote:
I am having issues with a TDM2400P. It appears when the ZAP channel dials
out, it randomly chops the first digit off of the number. I have tried
relaxdtmf=yes, turning up and down the txgain, turned off and on the echo
cancellation, generated new zaptel (with updated spinlock.h)...
I am at a loss. Can someone please offer some help?
Thanks.
TJ
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[Asterisk-Users] Voicemaster
HI all, Any of you having experience with voice master? I tried using the openh323 channel it doesn't give me voice at all. THere's no packet coming in. There's no problem with any other equipment but voicemaster doesn't send voice at all. Funny thing, i have an old version of OpenPhone, it's working. So please if any of you knows this problem, please share. THx a bunch ___ --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] CallerID
AFAIK, you can use database lookups to fetch the internal caller id
and external caller id depending on the channel that is placing the
call. Then, simply set the corresponding caller id before placing the
call. Alternatively, which is what I currently do, since I don't use
account codes, I set the accountcode parameter in my sip peer
definitions to the external caller id I want to show, and then I
force the caller id to the ${CDR(accountcode)} variable before
placing external calls.
I don't know if there are any other more efficient methods.
- Waldo
On Apr 6, 2006, at 3:02 AM, Miles Scruggs wrote:
how do you set two types of caller id one for internal calling and
one for external? Basically everyone calling out from asterisk
from one context I want to assign a single callerid. On all other
contexts I want to assign a caller ID specific to each line for all
calls going out to asterisk.
Finally for all calls that remain behind the asterisk box (ext to
ext) the Caller ID is set to the specific extension of the caller.
Thanks
Miles
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Re: [Asterisk-Users] TDM2400P problems
I am having issues with a TDM2400P. It appears when the ZAP channel dials
out, it randomly chops the first digit off of the number. I have tried
relaxdtmf=yes, turning up and down the txgain, turned off and on the echo
cancellation, generated new zaptel (with updated spinlock.h)...
I am at a loss. Can someone please offer some help?
* is probably starting to dial too fast. Try to add a w in your dial
string to make it wait.
Like : Dial(ZAP/g0,w${EXTEN})
If I'm not mistaken, w adds half a second pause. You can put more w to
make it wait longer.
hth
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Re: [Asterisk-Users] Re: queue issue
On Thu, 06 Apr 2006 13:17:29 +0200, Tomislav Parčina [EMAIL PROTECTED] wrote: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... on a related note, we notice that if we've set atxfer = *1 in features.conf and blindxfer=#1, then attended transfers dont work. somehow, the Queue app captures the '*' and hangs up the call. is this the behaviour others have observed ? obviously, since we've used *2 for auto monitor, that doesnt work as well. Yes, this is well known (problem?). I have solved it by editing features.conf file. How did you modify it? and will the ATXFR be perceived as a discharge from the queue system as a blind transfer using #? Yours l. -- Loway Research - Home of QueueMetrics http://queuemetrics.loway.it ___ --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] Questions on call recording and conference.
On 03/31/06 08:24 Wai Wu said the following: In Asterisk, what happens to the files when both legs of the call hangs up? Is there a way to create a conference room on the flight? i.e. without pre-defining the conference ID in meetme.conf. look at the 'd' option to MeetMe. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --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] Fedora Core 4 - problem with kernel 2.6.16-1.2069_FC4
I've had a similar problem with CentOS and yummed kernels. The problem seems to be that the zaptel doesn't quite know where to put the modules. If you check the directory for your current kernel version, you'll see they're not there. I have fixed this in two different ways: 1) Per the wiki - - As root: # ln -s /lib/modules/`uname -r`/build /usr/src/linux-2.6 (I don't know if a link to 'linux' is needed) # ln -s /lib/modules/`uname -r`/build /usr/src/linux - 2) 'rpm -e' the old kernel Thanks, Bob McDowell -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of William M Conlon Sent: Wednesday, April 05, 2006 6:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Fedora Core 4 - problem with kernel 2.6.16-1.2069_FC4 I was just getting to work on fax for my * system, so I thought I would bring everything up to date since there would be some new compilations involved. yum update gave me kernel-2.6.16-1.2069_FC4 but after recompiling zaptel, I kept getting FATAL module zaptel not found Chased this for an hour with multiple recompiles and reboots. Finally dropped back to 2.6.15-1.1833_FC4, which worked before, and still works now. Bill ___ --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 *** PRIVILEGED AND CONFIDENTIAL CLIENT COMMUNICATION *** This e-mail message and all attachments, if any, may contain confidential and privileged material and are intended only for the intended recipient. Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient, please contact the sender by reply e-mail or by calling (417) 869-9192 and destroy the original and any copies of this e-mail. ___ --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] Using Call Progress
I'm attempting to use callprogress in my system, and I'm having trouble. Callprogress always can tell if the line is busy or ringing, but when the line is answered, the call does not get bridged. Messages showing that "line is ringing" stop in the console and if the called party hangs up, asterisk reports the line is busy. Are there any settings that I could use to help with this issue? I am using asterisk 1.2.4 with TDM04B (FXO) cards on a RHEL3 system. Something in indications.conf or zonedata.c/dsp.c in the sourcethat can be tweaked? Any help would be appreciated! Thanks! - Eric Buruschkin ___ --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] Increase volume on trunk
Hello All I am wondering whether you can increase the volume on the trunk port when it is running on pure VoIP with no channels involved. Sam ___ --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] (no subject)
Hi, I'm using IPSwitchboard v 1.8.10, a sort of Operator Panel, to monitor my Asterisk's extensions. Recently I noticed that on the official site (http://ipswitchboard.thorben.dk/), where I downloaded the software some weeks ago, this project is no longer supported. Is there anyone that can say me where I can find the Italian version of IPswitchboard or if there is a way to translate the its messages? Thanks in advance, Marco. ___ --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] Call Parking and multiple contexts
Is there any way to define call parking parameters for different contexts? For example, if I have a client in context 100 and another client in context 200, can they both define parking positions, say, from 701-710, where 701 in context 100 is different from 701 in context 200? Or even better, can context 100 define parking positions 701-710 and context 200 define parking positions 801-810? Thanks, Waldo ___ --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: queue issue
On 04/06/06 19:17 Tomislav Parèina said the following: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... on a related note, we notice that if we've set atxfer = *1 in features.conf and blindxfer=#1, then attended transfers dont work. somehow, the Queue app captures the '*' and hangs up the call. is this the behaviour others have observed ? obviously, since we've used *2 for auto monitor, that doesnt work as well. Yes, this is well known (problem?). I have solved it by editing features.conf file. i've opened a bug and provided a fix for this at http://bugs.digium.com/view.php?id=6897 on investigation into the source, it wasnt the queue app but rather chan_agent which was doing this. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --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] Open channels
First, I'm not sure is this Asterisk or ooh323 channel problem. It seams that I have solved (I do hope so!) deadlock problem with ooh323 (thanks to Sean and his patch). Now I have another one. It seams that some channels stay open even they should not. This is what I see from CLI: pbx*CLI show channels Channel Location State Application(Data) SIP/302-924a [EMAIL PROTECTED]:3 RingDial(OOH323/[EMAIL PROTECTED] SIP/302-ce2d [EMAIL PROTECTED]:3 RingDial(OOH323/[EMAIL PROTECTED] SIP/302-58f3 [EMAIL PROTECTED]:3 RingDial(OOH323/[EMAIL PROTECTED] SIP/302-2933 [EMAIL PROTECTED]:3RingDial(OOH323/[EMAIL PROTECTED] SIP/301-3dfd [EMAIL PROTECTED]:3 RingDial(OOH323/[EMAIL PROTECTED] 5 active channels 5 active calls pbx*CLI How to solve this one? How can I check why are channels open? One is for sure, headphones are down on the hook :)) -- Tomislav Parcina tparcina#lama.hr ___ --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] Hinting a conference room
Look at hints for Local Channel. That may be what you are looking for. Alex From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alessio FocardiSent: Thursday, April 06, 2006 4:34 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Hinting a conference room Hi there!I was asked to set up a led on a snom phone monitoring a conference room (lit when someone is in conference).I know that there is a patch for hinting parking lots, anyone has made something similiar for conferences ? Tnx for the support!P.S.What about monitoring a global var ?It would be absolutely great variable=0 led off, 1 led on, 2 led blink ... Alessio Focardi ___ --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] Using Call Progress
Welcome to the painful world of analog phone lines. Unless you are using a digital line, there really is no true call progress detection available. In many situations this is not a problem, where we see this the most is when you are trying to ring a zip device and a zap channel at the same time, the zap call progress indicates an answered line the moment the zap channel goes active, NOT when the far side answers. If you have a ring group with sip and zap channels, what typically happens is that the sip phone will ring once, but as soon as the TDM card places the outbound call, it is considered "answered" and the sip phone stops ringing. Yes, you can enable callprogress and several other tweaks but the end result is often the far side answering and Asterisk still playing ring tones because there is no signal on the PSTN to indicate a far side answer. So, what to do when you find yourself in this situation and adding a PRI is not a solution, the only way we have worked around this is to make those outbound calls over a SIP or IAX service provider (and no, using a SIP gateway like a Mediatrix 1204 does not solve the problem as it is a PSTN issue) I know some people will argue this, but this was the result of almost 12 hours of work with us and Digium to figure out this issue. After MUCH debate and many hours of testing, this became the official word. Don't shoot the messenger. Kerry GarrisonDirector of Technical ServicesTech Data Pros - Orange County's Mobile IT Service Provider(949)502-7819 x200- [EMAIL PROTECTED]http://www.techdatapros.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric BuruschkinSent: Thursday, April 06, 2006 6:19 AMTo: Asterisk-UsersSubject: [Asterisk-Users] Using Call Progress I'm attempting to use callprogress in my system, and I'm having trouble. Callprogress always can tell if the line is busy or ringing, but when the line is answered, the call does not get bridged. Messages showing that "line is ringing" stop in the console and if the called party hangs up, asterisk reports the line is busy. Are there any settings that I could use to help with this issue? I am using asterisk 1.2.4 with TDM04B (FXO) cards on a RHEL3 system. Something in indications.conf or zonedata.c/dsp.c in the sourcethat can be tweaked? Any help would be appreciated! Thanks! - Eric Buruschkin ___ --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] How to restrict simultaneous phone registrations
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Hash: SHA1
Jonathan k. Creasy wrote:
I apologize if this information is posted elsewhere. Unfortunately I
haven't found it yet if it is. I'm not familiar with the channel
counting features could you please explain? Also, how are you tagging
the phones to account codes?
You can limit calls using the set/check group commands.
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+SetGroup
Account codes are set either by using the Set function or the
accountcode= property in the SIP/IAX conf files.
-Jonathan
Exactly, I'll post a sample dialplan. This dialplan is for ASTPP but
should give you the idea.
# exten = _1XX,1,Set(GROUP()=${ACCOUNTCODE})
# exten = _1XX,2,AGI(astpp-authorize.agi,${ACCOUNTCODE},${EXTEN})
# exten = _1XX,3,GotoIf($[${CALLSTATUS} = 0]?60) ; Checks
if account has sufficient funds
# exten = _1XX,4,GotoIf($[${CALLSTATUS} = 1]?70) ; Checks
if the phone number exists
# exten = _1XX,5,GotoIf($[${CALLSTATUS} = 2]?80) ; Check
if account exists
# exten = _1XX,6,GotoIf($[${GROUP_COUNT()}
${MAXCHANNELS}]?90) ; Verify number of outgoing channels
#
; assigned to account.
# exten = _1XX,7,Set(GROUP(${TRUNK1}-OUTBOUND)=OUTBOUND)
# exten =
_1XX,8,GotoIf($[${GROUP_COUNT([EMAIL PROTECTED])}
${TRUNK1_MAXCHANNELS}]?10)
# exten = _1XX,9,Dial(${LCRSTRING1}||${TIMELIMIT}|${OPTIONS})
# exten = _1XX,110,Busy
# exten = _1XX,10,Set(GROUP(${TRUNK2}-OUTBOUND)=OUTBOUND)
# exten =
_1XX,11,GotoIf($[${GROUP_COUNT([EMAIL PROTECTED])}
${TRUNK2_MAXCHANNELS}]?13)
# exten = _1XX,12,Dial(${LCRSTRING2}||${TIMELIMIT}|${OPTIONS})
# exten = _1XX,113,Busy
# exten = _1XX,13,Set(GROUP(${TRUNK2}-OUTBOUND)=OUTBOUND)
# exten =
_1XX,14,GotoIf($[${GROUP_COUNT([EMAIL PROTECTED])}
${TRUNK3_MAXCHANNELS}]?16)
# exten = _1XX,15,Dial(${LCRSTRING3}||${TIMELIMIT}|${OPTIONS})
# exten = _1XX,116,Busy
# exten = _1XX,16,Set(GROUP(${TRUNK4}-OUTBOUND)=OUTBOUND)
# exten =
_1XX,17,GotoIf($[${GROUP_COUNT([EMAIL PROTECTED])}
${TRUNK4_MAXCHANNELS}]?19)
# exten = _1XX,18,Dial(${LCRSTRING4}||${TIMELIMIT}|${OPTIONS})
# exten = _1XX,119,Busy
# exten = _1XX,19,Set(GROUP(${TRUNK5}-OUTBOUND)=OUTBOUND)
# exten =
_1XX,20,GotoIf($[${GROUP_COUNT([EMAIL PROTECTED])-OUTBOUND}
${TRUNK5_MAXCHANNELS}]?22)
# exten = _1XX,21,Dial(${LCRSTRING5}||${TIMELIMIT}|${OPTIONS})
# exten = _1XX,122,Busy
# exten = _1XX,22,Goto(100)
# exten = _1XX,60,Congestion ; '0' Tells them they do not have
enough money
# exten = _1XX,61,Hangup
# exten = _1XX,70,Congestion '1' Bad Phone Number
# exten = _1XX,71,Hangup
# exten = _1XX,80,Congestion
# exten = _1XX,81,Hangup
# exten = _1XX,90,Congestion; Their outgoing channel limit
is full already
# exten = _1XX,91,Hangup
# exten = _1XX,100,Congestion; No Route Available
# exten = _1XX,101,Hangup
Some of the group counts are for outgoing trunks. It's just the first
one that you need.
- --
Darren Wiebe
[EMAIL PROTECTED]
Aleph Communications
ASTPP - Open Source Voip Billing Calling Cards
www.aleph-com.net/astpp
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Re: [Asterisk-Users] FXO/FXS and E1 in same system
Hi, It will work - it is just a matter of the order in which the zaptel driver for the particular card is loaded. Just insert your card, load necessary driver and see /proc/zaptel/* - it is self explanatory. Ondrej yusuf wrote: Hi, can i have a FXO/FXS card and a E1/T1 card in the same system. I have used them seperatly many times before, but not together in one machine. I usually have for the analogue card signalling=fxs_ks channel = 1 and for the e1 card signalling=pri_net group=1 callerid=asreceived channel = 1-15,17-31 how will the channel numbers change with two cards. thanks, yusuf ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: What causes deadlock?
i am also getting this warning since upgrading to 1.2 when running asterisk with -p param (realtime priority)On 4/6/06, Raymond Chen [EMAIL PROTECTED] wrote:Tomislav Parčina wrote: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Hi What causes deadlock? Apr5 14:02:43 WARNING[2413] channel.c: Avoided initial deadlock for '0x82acb10', 10 retries! Apr5 14:02:43 WARNING[2413] channel.c: Avoided initial deadlock for '0x8298160', 10 retries! Does this happen with ooh323 channel driver? -- Tomislav Parcina tparcina#lama.hr ___ --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 sip to sip channels as well___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Call Parking and multiple contexts
Once upon a time there was an app called app_valetparking, and its big brother SUPERvaletparking. They both addressed that very senario. However, the brothers proved to be a expensive load on the PBX as searching within and moving throughout the Multiple parking lots required much time and processing power, even in broad daylight. Alas with the new zoning changes that have happened since 1.2.0, the parking lots are no longer welcome in the neighborhood. But don't give up hope!! Olle (OEJ) with his sultry Swedish voice and his ability to ruin a perfectly good weekend! Has proposed a new and inproved parking system that fits in with the new zoning guidelines set by the Developers. He has even set up a Magic Kingdom of sorts to let those play before he opens it up to the world. http://svn.digium.com/view/asterisk/team/oej/test-this-branch/. Then along came Rizzo with his new way of organizing and finding spaces, and Olle asked Rizzo, to please merge and reorganinze the parking lots in the Kingdom of Olle. So we wait for the lots to be repainted and repaved, so that we can tell the cars to park either in the Red, Blue, or Green Parking Lots. Oh, Did I mention that some lots need Valet and the others are Self-serve? Olle's MultiParking: http://bugs.digium.com/view.php?id=6113 Rizzo's New Search Routine: http://bugs.digium.com/view.php?id=6144 Valet Parking and Examples (does not compile on current release w/o patches, I don't have the patches) Description: http://www.loligo.com/asterisk/misc/apps/app_valetparking.README Software: http://www.loligo.com/asterisk/misc/apps/app_valetparking.c (This version will not work with asterisk-1.0.0) SuperValetParking - Latest from BKW 26/11/2004: http://www.asterlink.com/svp/ -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Waldo Rubinstein Sent: Thursday, April 06, 2006 9:41 AM To: Non-Commercial Discussion Asterisk Subject: [Asterisk-Users] Call Parking and multiple contexts Is there any way to define call parking parameters for different contexts? For example, if I have a client in context 100 and another client in context 200, can they both define parking positions, say, from 701-710, where 701 in context 100 is different from 701 in context 200? Or even better, can context 100 define parking positions 701-710 and context 200 define parking positions 801-810? Thanks, Waldo ___ --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] audiocodes with asterisk:- newbie
Hello friends, I am using SIP on Asterisk 1.2.4. All my configurations are working perfectly on a Welltech fxo box. But today I changed to an audiocodes MP104 fxo box. All the sip signalling works fine but the noise is something like an alien invasion, I mean, its completely outrageous. I dont know what to do. Has anyone got an audiocodes with asterisk working. Please help me with some configurations in audiocodes With warm regards. Vivek J. Joshi. [EMAIL PROTECTED] Trikon electronics Pvt. Ltd. All science is either physics or stamp collecting. -- Ernest Rutherford ___ --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] chan_sccp and hinting
Ok, so multiple people have said that hinting is possible with chan_sccp on the 7940/7960's and such, has anyone got this working? How do you go about getting this to work? I'd use the wiki, but it's link to the mailing list topic on that doesn't work anymore :( -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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 Parking and multiple contexts
You sound very poetic. Thanks for the info. - Waldo On Apr 6, 2006, at 10:27 AM, Alexander Lopez wrote: Once upon a time there was an app called app_valetparking, and its big brother SUPERvaletparking. They both addressed that very senario. However, the brothers proved to be a expensive load on the PBX as searching within and moving throughout the Multiple parking lots required much time and processing power, even in broad daylight. Alas with the new zoning changes that have happened since 1.2.0, the parking lots are no longer welcome in the neighborhood. But don't give up hope!! Olle (OEJ) with his sultry Swedish voice and his ability to ruin a perfectly good weekend! Has proposed a new and inproved parking system that fits in with the new zoning guidelines set by the Developers. He has even set up a Magic Kingdom of sorts to let those play before he opens it up to the world. http://svn.digium.com/view/asterisk/team/oej/test-this-branch/. Then along came Rizzo with his new way of organizing and finding spaces, and Olle asked Rizzo, to please merge and reorganinze the parking lots in the Kingdom of Olle. So we wait for the lots to be repainted and repaved, so that we can tell the cars to park either in the Red, Blue, or Green Parking Lots. Oh, Did I mention that some lots need Valet and the others are Self-serve? Olle's MultiParking: http://bugs.digium.com/view.php?id=6113 Rizzo's New Search Routine: http://bugs.digium.com/view.php?id=6144 Valet Parking and Examples (does not compile on current release w/o patches, I don't have the patches) Description: http://www.loligo.com/asterisk/misc/apps/app_valetparking.README Software: http://www.loligo.com/asterisk/misc/apps/app_valetparking.c (This version will not work with asterisk-1.0.0) SuperValetParking - Latest from BKW 26/11/2004: http://www.asterlink.com/svp/ -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Waldo Rubinstein Sent: Thursday, April 06, 2006 9:41 AM To: Non-Commercial Discussion Asterisk Subject: [Asterisk-Users] Call Parking and multiple contexts Is there any way to define call parking parameters for different contexts? For example, if I have a client in context 100 and another client in context 200, can they both define parking positions, say, from 701-710, where 701 in context 100 is different from 701 in context 200? Or even better, can context 100 define parking positions 701-710 and context 200 define parking positions 801-810? Thanks, Waldo ___ --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] Routing SIP calls via URI
But is there a way of doing this without a prefix?
because people should dial without prefixes: [EMAIL PROTECTED] , not like:
[EMAIL PROTECTED]
How can we make this without a prefix? something like:
if( !uri=~@mydomain.pt ){
forward the all to the Internet
}
:)
Thanks
Joao Pereira
Shad Mortazavi wrote:
Dear Group,
I was able to fix this problem;
The solution was to use a prefix to dial out.
The next challenge was to send the SIP Domain over IAX2!. I found that
if I included @SIPDOMAIN it would break the IAX2 communications.
exten = _6.,1,Dial(IAX2/bxx:[EMAIL PROTECTED]/[EMAIL PROTECTED]),
breakes because @SIPDOMAIN is treated as the target context. You also
can not include @Context after the @SIPDOMAIN.
I created a new variable DS which was a concatenation of EXTEN and
SIPDOMAIN separated by % and not @ and I was now able to pass this over
IAX2;
DS = EXTEN%SIPDOMAIN.
exten = _6.,1,Dial(IAX2/bxx:[EMAIL PROTECTED]/${DS}).
At the other end I used the CUT command and substring facilities in
Asterisk to split DS by the % eliminator; I re-formed a new variable
which was
DS = [EMAIL PROTECTED]
I can now pass calls from my internal Asterisk server to my external
Asterisk server using IAX2 and then call any external VoIP number.
Warm Regards
Shad Mortazavi
--
Nexus Group Technical Manager
n|m Nexus Management Inc
-Original Message-
From: Shad Mortazavi
Sent: Thursday, March 30, 2006 10:30 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Routing SIP calls via URI
Dear Group;
I can confirm that I have read through the three examples in
www.voip-info.org.
These examples are excellent and address a couple of the questions. I
have IAX2 working between several asterisk servers on our VPN and
between the DMZ and our LAN.
Also
exten = shad,1,Dial(IAX2/bxx:[EMAIL PROTECTED]/${EXTEN})
This answers part of the question;
However what I want to do is to send any outbound sip calls via our
external SIP server.
i.e;
VPN LANIAX2DMZ Internet
Internal UA --- Internal (*) -- External (*)--
ExternalUA
We have an extensive internal dial plan, X dial the UK, Y dial USA, 1XXX
for Voicemail, 2xxx for Meetme, etc.
Do I need to setup a prefix to dial the internet? And then route all
calls to the External(*) based on this prefix?
Thanks
Shad Mortazavi
--
Nexus Group Technical Manager
n|m Nexus Management Inc
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[Asterisk-Users] pause / unpausequeuemember
Hi,
I wanted to use the same extensions for Pausing and
UnPausing queue members.
Is that a variable that is set up with the agent
status (on call, available, not logged, paused) so that I could use it to make
some logic in this extension?
exten =
111,1,Set(AGENTEPARADESLOGAR=${$[AGENTBYCALLERID_${CALLERIDNUM}]})exten
= 111,2,PauseQueueMember(|Agent/${AGENTEPARADESLOGAR})exten =
111,3,Hangup
Or the only way out is to have different extensions
for pausing and unpausing?
Thank you
Dov
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Re: [Asterisk-Users] Using Call Progress
Eric Buruschkin wrote:
I'm attempting to use callprogress in my system, and I'm having
trouble. Callprogress always can tell if the line is busy or ringing,
but when the line is answered, the call does not get bridged.
If the call is not bridged as soon as * is done dialing, then you have a
configuration problem and its likely to be in extensions.conf. Please
post the section that includes the dial statement for the zap interface.
If your dial statement includes an r option, take it out and test
again. You should be using something like:
exten = _9XXX,1,Dial(Zap/4/${EXTEN})
Messages
showing that line is ringing stop in the console and if the called
party hangs up, asterisk reports the line is busy.
The call progress function in asterisk is known to not be all that
accurate or useful. If you are using busydetect, then do something like
this:
busydetect=yes
busycount=6
callprogress=no
where the busycount represents the number of tone cycles to listen to
before judging whether its a busy signal or not. (Values less then six
will oftentimes result in inaccurate detection.)
Are there any settings that I could use to help with this issue? I am
using asterisk 1.2.4 with TDM04B (FXO) cards on a RHEL3 system.
Something in indications.conf or zonedata.c/dsp.c in the source that can
be tweaked?
I'm assuming you are located in the US. If not, there are significant
variations from one country to another in terms of tones used, answer
supervision, etc.
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Re: [Asterisk-Users] Increase volume on trunk
Sam Tam wrote: Hello All I am wondering whether you can increase the volume on the trunk port when it is running on pure VoIP with no channels involved. No, there isn't any such settings. ___ --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] Cisco 7960 - hints
Is the Cisco 7960 capable of monitoring other extensions (hint status) with a SIP implementation? Seems like it could, just can't find any info on it... Sean ___ --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 behind NAT
Hello to all Can we put Asterisk in a company that has an ADSL connection with just one public IP address? Because with just one public IP, Asterisk must have a private (NATed) IP... but the idea is to make him dial other SIP domains. Can Asterisk work behing NAT, and still route calls to the Internet? And he can still receive calls from the Internet? Thanks Joao Pereira ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 - hints
Sadly, no. The SIP firmware on the Cisco phones doesn't support subscribing to other lines. I heard chan_sccp does though.. now to figure out how. Aaron On Thu, 6 Apr 2006, Sean Cook wrote: Is the Cisco 7960 capable of monitoring other extensions (hint status) with a SIP implementation? Seems like it could, just can't find any info on it... Sean ___ --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 -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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] Planet VIP-320 DECT gateway with Asterisk?
Hello, I just received what seems to be a nice SIP-DECT gateway but can't make it work with asterisk. The manual is very unclear (written in chinese english) and the web configurator is ambiguous as well. Has anyone succeeded in making one of these babies work with * ? info: http://www.planet.com.tw/product/product_dm.php?product_id=367menu_id=3 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 - hints
Are you using chan_sccp for you cisco implementation? Aaron Daniel wrote: Sadly, no. The SIP firmware on the Cisco phones doesn't support subscribing to other lines. I heard chan_sccp does though.. now to figure out how. Aaron On Thu, 6 Apr 2006, Sean Cook wrote: Is the Cisco 7960 capable of monitoring other extensions (hint status) with a SIP implementation? Seems like it could, just can't find any info on it... Sean ___ --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] Got SIP response 302 Moved temporarely
Hi all Hmm, often when my Asterisk tryes to register, it get's the answer back: Got SIP response 302 Moved temporarely (and an IP). But it looks like it's not respecting this redirection and tryes again and again to register to the server configured in sip.conf instead of the one the SIP provider tryes to redirect to. Any known issues? Mit freundlichen Grüssen Benoit Panizzon -- I m p r o W a r e A G-System Services __ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 PrattelnFax +41 61 826 93 01 Schweiz Web http://www.imp.ch __ pgpnFN06MP2sl.pgp Description: PGP 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] Asterisk in production as a fax server, anyone?
Julio Arruda escreveu: Paulo, He is mentioning E1/PRI, so I assume the well known collect call on E1/R2 thingie doesn't apply to him. Julio, I have 1 E1 from telefonica and 1 from Embratel. Telefonica has a better deal for incoming calls (gave us more DIDs) but Embratel has better rates. I've had a real hard time trying to make E1/PRI signaling work with Embratel, with no success. In the end, I had to use MFC/5C. Telefonica and Embratel will not block collect calls for you, they dont care, its easy money. May be he is linking to a smaller and more flexible telco, or may be he will put the * box behind another PBX that has better support for MFC/5C than libmfcr2. I'm just curious anyway. The automated collect call system in Brazil is really dumb and unfair, and is abused so many ways... I want to beat the crap out of the genius who invented this system where the callee does not have to explicitly accept a collect call. Anatel (the telco government agency in Brazil) dont even acknowledge this as problem, because they will not accept complaints against Anatel regulations, just against the telcos, and the telcos are following this dumb rules to the letter. Its because regulatory agencies in Brazil are here not to protect the citizens, just to extort money from private companies to burn in our corrupt political engine. Sorry for the rant, but I would like to hear from other people running * in Brazil, how they address this trouble. -- Paulo ___ --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 connection refused between 2 asterisks 1.2.5
Can you post your iax.conf? On 4/4/06, Marco Mouta [EMAIL PROTECTED] wrote: Password and username are ok. On 4/4/06, Joshua Colp [EMAIL PROTECTED] wrote: Marco Mouta wrote: Hi all, I've 2 * tryning to connect each other Server A is already registred on server B But server B never registers in server A I always get this: Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGREJ Timestamp: 00018ms SCall: 4 DCall: 3 [XXX.XXX.XXX.XX:4569] CAUSE : Registration Refused CAUSE CODE : 29 Any tip? Best regards, Marco Mouta ___ --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 Check everything you can: username, passwords, etc. -- Joshua Colp Software Developer Digium P - 256-428-6066 C - 506-878-0147 [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 ___ --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] Loading module chan_zap.so failed! PLZ help me!
ali asma wrote: I have recompiled my zaptel drivers but I still get the same error --- Derek Whitten [EMAIL PROTECTED] a écrit : ali asma wrote: I modified the configuration but I still have the same error. Please tell me in whach directory should I execute: modprobe zaptel modprobe wcfxo becose it seems that my card not had been detected Thanks, --- Lee Archer [EMAIL PROTECTED] a écrit : I run suse 10 and have an X100p. But I use fxsks=1 in the /etc/zaptel.conf not /etc/asterisk/zaptel.conf. Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of ali asma Sent: 04 April 2006 10:13 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Loading module chan_zap.so failed! PLZ help me! Hi, Sorry my card is X101P. My config is : /etc/asterisk/zaptel.conf : loadzone=us defaultzone=us fxoks=1 and /etc/asterisk/zapata.conf : [trunkgroups] [channels] context=mainmenu signalling=fxo_ks faxdetect=incoming usecallerid=yes echocancel=yes echocancelwhenbridged=no echotraining=800 language=en channel=1 please help me --- ali asma [EMAIL PROTECTED] a écrit : Hi, I' ve just connected a carte X100M to my asterisk server running zaptel-1.2.5, libpri-1.2.2 and asterisk-1.2.6 on SUSE 10.0. When I make modprobe wcfxo and modprobe zaptel I haven't any error, I have also chan_zap.so module existing in /usr/lib/asterisk/modules. But, when i run ztcfg, it shows me this: Zaptel Configuration == Channel map: 0 channels configured. and when I run asterisk it shows me this: Asterisk Dynamic Loader Starting: == Parsing '/etc/asterisk/modules.conf': Found [chan_zap.so]Apr 4 09:45:58 WARNING[9975]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/chan_zap.so: undefined symbol: ast_pickup_call Apr 4 09:45:58 WARNING[9975]: loader.c:499 load_modules: Loading module chan_zap.so failed! Where do i look, how can i debug? Thanks in advance, ___ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.yahoo.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 ___ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.yahoo.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 has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.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 ___ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. === message truncated === ___ --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 ___ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.yahoo.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 You need to load res_features.so before loading chan_zap.so, that will make ast_pickup_call resolve. This can be accomplished by explicitly putting it in your modules.conf to be loaded -- Joshua Colp Software Developer Digium P - 256-428-6066 C - 506-878-0147 [EMAIL PROTECTED] ___ --Bandwidth and
Re: [Asterisk-Users] IAX connection refused between 2 asterisks 1.2.5
Password and username are ok. On 4/4/06, Joshua Colp [EMAIL PROTECTED] wrote: Marco Mouta wrote: Hi all, I've 2 * tryning to connect each other Server A is already registred on server B But server B never registers in server A I always get this: Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGREJ Timestamp: 00018ms SCall: 4 DCall: 3 [XXX.XXX.XXX.XX:4569] CAUSE : Registration Refused CAUSE CODE : 29 Any tip? Best regards, Marco Mouta ___ --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 Check everything you can: username, passwords, etc. -- Joshua Colp Software Developer Digium P - 256-428-6066 C - 506-878-0147 [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] Frustrated with echo...
On Wednesday 05 April 2006 07:26, Eric ManxPower Wieling wrote: We reboot all our Asterisk servers once per week if they have a TDM400P in them. If we don't do that, then the TDM400P modules stop working. I have *never* rebooted an Asterisk system because of the TDM400. Granted, the driver did have a signed/unsigned variable issue but it's been fixed quite literally for months. When that *was* an issue, I would of course stop asterisk and unload/reload the wctdm module, but as I said that has not been a problem for six months, if not longer. *CLI zap show status Description Alarms IRQbpviol CRC4 Wildcard TDM400P REV E/F Board 1 OK 0 0 0 *CLI show uptime System uptime: 8 weeks, 1 day, 11 hours, 26 seconds *CLI show version Asterisk SVN-trunk-r8643M built by root @ asterisk on a i686 running Linux on 2006-01-25 12:57:55 UTC # w 08:48:24 up 57 days, 10:57, 1 user, load average: 0.16, 0.03, 0.01 As a community we *really* need to stop pushing these old issues as if they were current. There *were* problems, but they *have* been fixed. -A. ... hell, I'm even sharing interrupts on this TDM400P: # cat /proc/interrupts CPU0 0: 496444645 XT-PIC timer 1: 2 XT-PIC keyboard 2: 0 XT-PIC cascade 8: 1 XT-PIC rtc 10:4947152 XT-PIC eth0 11: 669495590 XT-PIC wctdm, usb-uhci 14: 168829 XT-PIC ide0 NMI: 0 ERR: 0 Not as wow as when I was using a diferent system and sharing TDM400P interrupts with the NIC (the box was also an NFS server), but seriously... the old rumours and bugs that DID exist have been quite squashed, in my opinion. We need to move on and start complaining about the current bugs! :-) -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] Re: E1 te110p problem
Hi, What kind of problem happens? Show your dialplan. Daniel On 4/4/06, Toke [EMAIL PROTECTED] wrote: Hi Antonio, What problems are you having with it? Which operator give you E1 connectivity?? If you want mail me directly and we will try to have it working if it is possible. Regards On 4/4/06 10:38, Antonio Almodóvar [EMAIL PROTECTED] wrote: Hi all. I'm using a te110p in spain. ;zaptel.conf span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 I'm getting problems dialing out through this span. ¿How can I debug its behaviour? Thank you in advance. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] voicemail context issue
If you have a temporary message set up, it always uses the temporary greeting. If you want it to use the regular busy/unavailable messages, you have to remove the temporary greeting. Aaron On Tue, 4 Apr 2006, Dov Bigio wrote: Hi, I know this has already been discussed here, but I still have the problem even with 1.2.6: When I call a mailbox in a context company is doesn't play my busy message... It goes directly to the temp message... Am I doing something wrong? == Everyone is busy/congested at this time (1:0/1/0) -- Executing NoOp(SIP/200.234.208.250-0840f548, Voicemail de [EMAIL PROTECTED]) in new stack -- Executing VoiceMail(SIP/200.234.208.250-0840f548, [EMAIL PROTECTED]) in new stack -- Playing '/var/spool/asterisk/voicemail/bawm/87/temp' (language 'pt') -- Playing 'vm-intro' (language 'pt') == Spawn extension (macro-ramais_sip, s, 224) exited non-zero on 'SIP/200.234.208.250-0840f548' in macro Here are the show voicemail users for company results ContextMbox User Zone NewMsg company 87Dov Bigio 0 Thank you Dov -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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] Pickup() h323
Pavel Jezek wrote: Hello Jeremy, do you think, that adding features to original h323 channel is perspective? is still maintained or will be replaced eg. with ooh323 (from asterisk add-ons)? anyway I'm currently using original h323, it working prety fine for me (with ooh323/oh323 I had problem with callerid between h323-asterisk)... chan_h323 is very much supported, just nobody has bothered to give me any valid information on what needs to be fixed. I have totally removed H.323 from my operation, so I no longer utilize chan_h323 for anything. Thus it is now up to the community to report issues they find. Digium paid for ooh323, for whatever reasons that is beyond me, but it has proven to be no better than any H.323 channel driver, so I hope they got their money back. Jeremy McNamara ___ --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] AST eating CPU 100%-Resource temporarily unavailable
Ing. Oscar Andrés Carriles
I got a CPU hog of 100% running asterisk 1.0.9
The problem is caused by a single process capturing all available CPU in
one call. When this call hang up seldom others continue in normal
service.
I have all 30 SIP softPhones eyebean, 1E1 AFT101 Sangoma card signalling
MFCR2
When the problem arrives in the call center, people from outside hears
so good, but from inside the voice becomes choppy.
I did a little trace in the related process as attached-
-Not related to heavy load
-May occur with 2 calls or 20 up
--
No virus found in this outgoing message.
Checked by AVG Free Edition.
Version: 7.1.385 / Virus Database: 268.3.5/300 - Release Date:
03/04/2006
read(28, \324\325\325\325\325\325U\325\325U\325\325\325\324\325..., 1024) =
160
ioctl(28, 0xc0044a09, 0xbd7f5160) = 0
gettimeofday({1144187314, 639148}, NULL) = 0
gettimeofday({1144187314, 639191}, NULL) = 0
time([1144187314]) = 1144187314
stat64(/etc/localtime, {st_mode=S_IFREG|0644, st_size=377, ...}) = 0
stat64(/etc/localtime, {st_mode=S_IFREG|0644, st_size=377, ...}) = 0
stat64(/etc/localtime, {st_mode=S_IFREG|0644, st_size=377, ...}) = 0
sendto(247, \200\2107J\0\23\6\20QD\244\3\324\325\325\325\325\325U\325...,
172, 0, {sa_family=AF_INET, sin_port=htons(8828),
sin_addr=inet_addr(192.168.250.58)}, 16) = 172
poll([{fd=28, events=POLLIN|POLLPRI}, {fd=247, events=POLLIN|POLLPRI,
revents=POLLIN}, {fd=249, events=POLLIN|POLLPRI}, {fd=296,
events=POLLIN|POLLPRI}], 4, -1) = 1
read(296, 0xbd7f5fc8, 4)= -1 EAGAIN (Resource temporarily
unavailable)
recvfrom(247, \200\10([EMAIL PROTECTED]..., 8192, 0, {sa_family=AF_INET,
sin_port=htons(8828), sin_addr=inet_addr(192.168.250.58)}, [16]) = 172
time([1144187314]) = 1144187314
write(28, \325\324\325\325\325\325\324\325\325\325\325\325..., 160) =
160
poll([{fd=247, events=POLLIN|POLLPRI, revents=POLLIN}, {fd=249,
events=POLLIN|POLLPRI}, {fd=296, events=POLLIN|POLLPRI}, {fd=28,
events=POLLIN|POLLPRI}], 4, -1) = 1
read(296, 0xbd7f5fc8, 4)= -1 EAGAIN (Resource temporarily
unavailable)
recvfrom(247, \200\10)\0\31\v@@\371\30\261\325\325U\325UU\325\325\325...,
8192, 0, {sa_family=AF_INET, sin_port=htons(8828),
sin_addr=inet_addr(192.168.250.58)}, [16]) = 172
time([1144187314]) = 1144187314
write(28, \325\325U\325UU\325\325\325\325\325UUU\325\325\325UU\325..., 160) =
160
poll([{fd=28, events=POLLIN|POLLPRI}, {fd=247, events=POLLIN|POLLPRI,
revents=POLLIN}, {fd=249, events=POLLIN|POLLPRI}, {fd=296,
events=POLLIN|POLLPRI}], 4, -1) = 1
read(296, 0xbd7f5fc8, 4)= -1 EAGAIN (Resource temporarily
unavailable)
recvfrom(247, \200\10[EMAIL PROTECTED]..., 8192, 0, {sa_family=AF_INET,
sin_port=htons(8828), sin_addr=inet_addr(192.168.250.58)}, [16]) = 172
time([1144187314]) = 1144187314
write(28, \325UUT\325\325U\325\325UU\325\325UU\325\325U\325\324..., 160)
= 160
poll([{fd=247, events=POLLIN|POLLPRI, revents=POLLIN}, {fd=249,
events=POLLIN|POLLPRI}, {fd=296, events=POLLIN|POLLPRI}, {fd=28,
events=POLLIN|POLLPRI}], 4, -1) = 1
read(296, 0xbd7f5fc8, 4)= -1 EAGAIN (Resource temporarily
unavailable)
recvfrom(247, \200\10/[EMAIL PROTECTED]..., 8192, 0, {sa_family=AF_INET,
sin_port=htons(8828), sin_addr=inet_addr(192.168.250.58)}, [16]) = 172
time([1144187314]) = 1144187314
write(28, UUU\325UTUUTU\325UU\325\325\325UU\325\325\325\325U\325..., 160) =
-1 EAGAIN (Resource temporarily unavailable)
write(28, UUU\325UTUUTU\325UU\325\325\325UU\325\325\325\325U\325..., 160) =
-1 EAGAIN (Resource temporarily unavailable)
write(28, UUU\325UTUUTU\325UU\325\325\325UU\325\325\325\325U\325..., 160) =
-1 EAGAIN (Resource temporarily unavailable)
write(28, UUU\325UTUUTU\325UU\325\325\325UU\325\325\325\325U\325..., 160) =
-1 EAGAIN (Resource temporarily unavailable)
write(28, UUU\325UTUUTU\325UU\325\325\325UU\325\325\325\325U\325..., 160) =
-1 EAGAIN (Resource temporarily unavailable)
write(28, UUU\325UTUUTU\325UU\325\325\325UU\325\325\325\325U\325..., 160) =
-1 EAGAIN (Resource temporarily unavailable)
write(28, UUU\325UTUUTU\325UU\325\325\325UU\325\325\325\325U\325..., 160) =
-1 EAGAIN (Resource temporarily unavailable)
write(28, UUU\325UTUUTU\325UU\325\325\325UU\325\325\325\325U\325..., 160) =
-1 EAGAIN (Resource temporarily unavailable)
write(28, UUU\325UTUUTU\325UU\325\325\325UU\325\325\325\325U\325..., 160) =
-1 EAGAIN (Resource temporarily unavailable)
write(28, UUU\325UTUUTU\325UU\325\325\325UU\325\325\325\325U\325..., 160) =
-1 EAGAIN (Resource temporarily unavailable)
write(28, UUU\325UTUUTU\325UU\325\325\325UU\325\325\325\325U\325..., 160) =
-1 EAGAIN (Resource temporarily unavailable)
write(28, UUU\325UTUUTU\325UU\325\325\325UU\325\325\325\325U\325..., 160) =
-1 EAGAIN (Resource temporarily unavailable)
write(28, UUU\325UTUUTU\325UU\325\325\325UU\325\325\325\325U\325..., 160) =
-1 EAGAIN
Re: [Asterisk-Users] voicemail context issue
Dov Bigio wrote: When I call a mailbox in a context company is doesn't play my busy message... It goes directly to the temp message... Am I doing something wrong? If you have a temp message, it is supposed to override your other messages. ___ --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] voipstunt: Forbidden - wrong password ...
voipstunt: Forbidden - wrong password on authentication for INVITE to
I have paid, the password was not changed, ... I have no idea why.
Is there anything what I can do to get this failed call over to
another provider, so that the user can complete the call?
(Dialstatus was an idea, but the line does not show up in CLI)
[Apr 5 09:22:36] -- Executing SetCIDNum(SIP/601-5039, 601|a) in
new stack
[Apr 5 09:22:36] -- Executing EnumLookup(SIP/601-5039,
+12124615222) in new stack
[Apr 5 09:22:36] -- Executing Dial(SIP/601-5039,
SIP/[EMAIL PROTECTED]) in new stack
[Apr 5 09:22:36] -- Called [EMAIL PROTECTED]
[Apr 5 09:22:37] WARNING[4274]: chan_sip.c:9613 handle_response_invite:
Forbidden - wrong password on authentication for INVITE to 'Ronald
Hotline sip:[EMAIL PROTECTED];tag=as36296964'
[Apr 5 09:22:37] -- SIP/voipstunt-b7e6 is circuit-busy
[Apr 5 09:22:37] == Everyone is busy/congested at this time (1:0/1/0)
[Apr 5 09:22:37] -- Executing Hangup(SIP/601-5039, ) in new stack
[Apr 5 09:22:37] == Spawn extension (default, 912124615222, 204)
exited non-zero on 'SIP/601-5039'
[Apr 5 09:22:37] -- Executing Hangup(SIP/601-5039, ) in new stack
[Apr 5 09:22:37] == Spawn extension (default, h, 1) exited non-zero
on 'SIP/601-5039'
exten = _91Z.,103,Dial(SIP/00${EXTEN:[EMAIL PROTECTED])
exten = _91Z.,104,NoOp(Line 104 ${DIALSTATUS})
bye
Ronald Wiplinger
___
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[Asterisk-Users] Querying number of people in a call queue from dialplan
Is there any way to query the number of people in a call queue from the dialplan? Our freephone provider has a feature where if we busy a call they record the voicemail and email it to us. This enables us to divert calls to them if our incoming lines start to get full. In order to do this we need to decide whether to busy the call before passing it into the queue. Thanks Gareth ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk svn starting problem
On Wed, 2006-04-05 at 08:52 +0200, René Enskat [Teamware GmbH] wrote: hi i updated asterisk today via svn no i can'T start asterisk i get core dumps. i have to comment some modules then i can start: noload = format_au.so noload = format_mp3.so noload = format_pcm_alaw.so.so noload = format_pcm_alaw.so compiling was fine just some warnings somebody has any idea? And make install didn't mention anything about /usr/lib/asterisk/modules? -- Dave Cotton [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ASTCC: How to reset in-use flag automatically ?
JP Carballo wrote:
Ronald Wiplinger wrote:
Insert this in astcc.agi; anywhere after the calls for it to load
and connect to the db.
if ($phoneno eq RESET_INUSE) {
setinuse($carddata-{number}, 0);
exit(0);
}
Thanks!
I use it here:
elsif ($phoneno eq BALANCE) {
setinuse($carddata-{number}, 0);
exit(0);
}
elseif ($phoneno eq RESET_INUSE) {
setinuse($carddata-{number}, 0);
exit(0);
}
bye
Ronald Wiplinger
And this in extensions.conf:
exten = s,n,DeadAGI(astcc.agi,${CARDNO},RESET_INUSE,2)
I leave it to you to capture ${CARDNO} :)
I don't enable this in the IVR unless the person has entered a valid
account number, for obvious reasons.
Wouldn't that totally disable inuse??? It would be possible that a
user uses two or more soft phones and make phone calls on multiple
places!
Nope. I don't want that to happen either.
Because the 2nd argument is normally the phone number to call, the
test will be false and the routine will be skipped if the customer
intends to call.
Besides, if the routine does evaluate to true, it will exit the agi
and not process any calls anyway.
Set this up as a separate extension that you can call if an account is
locked in use.
I've only ever used this when testing new trunks because an account
with the inuse flag set means the previous call ended prematurely.
In my case, I want customers to make one and only one call at a time
so I left the inuse handling mostly intact.
I want it to be anal.
If a customer complains about it, I'm more worried about a trunk
failing than a cheating caller.
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Re: [Asterisk-Users] WOW! Sphinx is awesome... but.... (asterisk+sphinx+menus)
Matt, it is the first time I hear positive about Sphinx. Do you have a menu for the installation you did? He's just exceptionally easy to please. :-) Is there a problem with Sphinx that I have missed? So far it really seems to be hitting the words right on. ___ --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] Ideal Setup for T1/PRI and TE110P - second try
Hi all, I'm sure something similar has been discussed, but one can only wade through the archives for so long. I'm setting up a T1 and my telco has a bunch of questions it wants me to answer. I know much of the TE110p is configurable to do any of this, but I wanted to know if there is an optimal or preferred setup. Any help would be appreciated. Here is the quiz my telco is giving me: Wiring: 4-Wire | Coax | Fiber (I'm assuming 4-wire is the correct interface) Jack type: RJ45 | 48? (I'm pretty sure the TE110p is RJ45 - correct?) Dial Tone: None | Yes-Precise | Yes-SCC Framing: SF | ESF (I'm assuming ESF) Line Coding: AMI | B8ZS Signaling Start: Ground Start | EM | Loop Start w/ring | Loop Start w/o ring (which of these does kewlstart deal with?) Pulse Mode: DTMF | MF (I assume DTMF) Outpulse Mode: Wink | Immediate | Seizure (If Seizure, then Origination or Digit Collection) Will ANI delivery be required for Toll-Free service? (I'm assuming Yes if we want to pass our caller id?) Thanks a ton for your time, JT ___ --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] queueue recording and what to do next
On 14:36, Tue 04 Apr 06, Anton Krall wrote: Guys, if you define recording on queues.conf and also define a monitor_filename var on your dialplna, you can record a queue call but, isthere a way to do something with the file after the call ends? I need to move the file to some other place but I cant find where to define a command to run after a queue call finishes. Any hints? You can use the exten = h,1,deadagi() to process it. At least that's how we do it with faxes. exten = h,1,deadagi(processfax.php) ;put the fax in db and generate pdf on filesys Good luck -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.info GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Why is it drug addicts and computer afficionados are both called 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] Opensource solutions to SPIT
Hi, I have been listening to Blue Box: The VoIP Security Podcast - http://www.blueboxpodcast.com, and thought that SPIT could pose a problem if not already one. Like to know if there are any OSS solutions, within Asterisk or can integrates well with it, that focus in this area? Regards Andy Tan -- Andy Tan [EMAIL PROTECTED] -- http://www.fastmail.fm - And now for something completely different ___ --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] Anyone have a definitive list of Managereventsper category?
Title: [Asterisk-Users] Anyone have a definitive list of Manager eventsper category? hm, I have to try that. I am using for third party control so the need to know all the events. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Josh McAllisterSent: Tuesday, April 04, 2006 4:19 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] Anyone have a definitive list of Managereventsper category? My understanding is that is exactly what these categories do for you. IE. If I were to create a user with read=call, that user would only get events in the call category. Am I wrong? If my assumption is correct, it would be of great benefit to know exactly which events are in which category. Josh From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wai WuSent: Tuesday, April 04, 2006 1:35 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] Anyone have a definitive list of Manager eventsper category? I don't think you can selectively receive events. I am also write an app using heavy manager actions, and I put the filters on my app. So far, I have not seen traffic from these events do a dent to my application/network performance. From: [EMAIL PROTECTED] on behalf of Josh McAllisterSent: Tue 4/4/2006 2:59 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] Anyone have a definitive list of Manager eventsper category? Can anyone provide a complete list of events and to which category theyare in? (ie. system,call,log,verbose,command,agent,user).I'm using * Manager in various ways with heavy call volume and wouldlike to limit the events per connection as much as possible.Any help would be appreciated.Thanks,Josh McAllister___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Applying patch.
Title: [Asterisk-Users] Anyone have a definitive list of Manager eventsper category? Hi, After apply patch and make clean; make install. Do I have to do a make sample to have new asterisk running? ___ --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: Asterisk in production as a fax server, anyone?
Don Pobanz escreveu: Frame slips are NOT motherboard related! I had problems with some combinations of motherboards, memory sizes and linux kernel versions. There are timing problems that also causes frame slips, like buffer overruns or underruns, but these are software related. -- Paulo ___ --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 in production as a fax server, anyone?
I have a worst issue for you... If your fax solution is ever going to receive fax in Brazil, how would you block collect calls? I have made a fax server solution with cheap Digium hardware that works in Brazil (2 E1s). -- Paulo Adolfo R. Brandes escreveu: Greetings, All-Knowing Asterisk Users List, My company needs to build a reliable fax server that can handle at least 30 simultaneous incoming faxes from the PSTN, using PRI. We realize that this can be solved in any number of ways using a Linux box, but since IVR is also a must, Asterisk popped up as the most promising solution. After combing these lists for clues, we began experimenting extensively with Asterisk and its software DSP and fax capabilities in most of their incarnations, such as Rxfax or Iaxmodem/Hylafax, together with Digium's E1 cards in server-grade Intel motherboards, all in a dedicated test environment. Unfortunately, though, we have yet to achieve reliable and satisfactory results, even with only 1 fax call at a time. I won't go into the details because we don't need technical support, given that this is, as of yet, a very loosely defined test. What we want is is merely a pointer in the right direction. So here it comes: Has anybody ever achieved, or know of someone who has, reliable 30 simultaneous PRI fax calls using Asterisk and Asterisk-compatible hardware and software? We are hardware agnostic, so if you say Sangoma's cards do it better than Digium's, or that Eicon Diva cards' hardware DSP and chan_capi are the only solution, we have no problem going there. I would be most thankful, however, for detailed explanations of successful scenarios, including such things as motherboard make and model, processor speed, Linux distribution and version, and anything else you decide to be even marginally pertinent. Thank you very much, Adolfo R. Brandes ___ --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] Frustrated with echo...
Lorentz Hinrichsen wrote: I've had very poor results with the Digium cards, I am using a couple of the new Sangoma ones now (they are cheaper and have hardware echo cancellation). Which boards are cheaper _and_ have hardware echo cancellation? ___ --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] Phones are all auto answering
Kind of like DND, but some phones seem ok. They all give the message even if it rings through that the person is on the phone even if they are not. Normally it says that when they are on the phone, and it says unavailable if they are not on the phone but never answer... Almost like astericks thinks all the phones are busy, at least by the recording it gives. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Tuesday, April 04, 2006 10:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Phones are all auto answering What phones you using? On 4/4/06, Christian Buchter [EMAIL PROTECTED] wrote: Strange, but all the phones when called immediately return a user is on the phone and the phone never rings. Anyone else ever experience this before? TIA _ This email has been scanned by MessageLabs on behalf of E-INS ___ --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 _ This email has been scanned by MessageLabs on behalf of E-INS _ This email has been scanned by MessageLabs on behalf of E-INS ___ --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] Monitor or mixmonitor
I'm using MixMonitor. Be aware that some people encounter a bug where MixMonitor stops recording at random (see http://bugs.digium.com/view.php?id=6457). There are a couple of working patches for it. Thanks.On 4/3/06, Wai Wu [EMAIL PROTECTED] wrote: Hi all,I am setting up a script to record all the call. There are two app for recording. Monitor and Mixmonitor, one mixing the audio on the fly and one mixing it at the end but also allow a option not to mixing the audio at all. If mixing the audio on the fly is not that taxing on the CPU, I would like to use 'mixmonitor' app. My question is, what is penalty on the CPU when mixing the audio on the fly? I know this is the better option, but I don't really need the 'in' and 'out' audio mixed until it's played back, and which happens less than 5% of the time. What are your thoughts? ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX connection refused between 2 asterisks 1.2.5
Marco Mouta wrote: Password and username are ok. On 4/4/06, Joshua Colp [EMAIL PROTECTED] wrote: Marco Mouta wrote: Hi all, I've 2 * tryning to connect each other Server A is already registred on server B But server B never registers in server A I always get this: Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGREJ Timestamp: 00018ms SCall: 4 DCall: 3 [XXX.XXX.XXX.XX:4569] CAUSE : Registration Refused CAUSE CODE : 29 Any tip? Best regards, Marco Mouta ___ --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 Check everything you can: username, passwords, etc. -- Joshua Colp Software Developer Digium P - 256-428-6066 C - 506-878-0147 [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 Well can you post the entries in question? -- Joshua Colp Software Developer Digium P - 256-428-6066 C - 506-878-0147 [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] R2 protocol error
a mirror to soft-switch can be found at: http://zarzamora.com.mx/mirror/www.soft-switch.org/ regards On 4/3/06, Steve Underwood [EMAIL PROTECTED] wrote: Hi Dennis, Update to libmfcr2-0.0.3 pre9. I made a slip in pre8. Sorry. Steve Dennis Nacino wrote: Hi, I have three R2 installation on different carriers, all shows the same inconsistency at varying degree. But, on most test calls we made, it reaches T3. The worst part of these, the carrier claims that it's my R2 box that is not responding in time. Please, check the attached file and take note of the timestamp, you'll find that in some call, it already contradict what the carrier claims but they too have logs to counter my claim. So, I hope people, please give me a good insight and direction to resolve this problem. I have the following for my R2 box: unicall-0.0.3pre8 libmfcr2-0.0.3 libsupertone-0.0.2 libunicall-0.0.3 spandsp-0.0.2pre25 asterisk-1.2.6 zaptel-1.2.5 wanpipe-2.3.3-2 2.6.11-1.1369_FC4smp sangoma A101 in my zaptel.conf I got the following: span=1,0,0,cas,hdb3 loadzone = us defaultzone=us cas=1-15:1101 cas=17-31:1101 in my unicall.conf I got these lines: [channels] context=default usecallerid=yes hidecallerid=no callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no supertones=ph loglevel=255 protocolclass=mfcr2 protocolvariant=ph,10,3,12 protocolend=co group = 1 channel = 1-15 channel = 17-31 Thanks a lot. Dennis __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com Apr 3 11:34:54 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 0001 [1/ 1/Idle /Idle ] Apr 3 11:34:54 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Detected Apr 3 11:34:54 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Making a new call with CRN 32769 Apr 3 11:34:54 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 1101 - [2/ 2/Idle /Idle ] Apr 3 11:34:54 WARNING[17334]: chan_unicall.c:2644 handle_uc_event: Unicall/1 event Detected Apr 3 11:34:55 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 3 on [2/ 2/Seize ack /Seize ack] Apr 3 11:34:55 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 1 on - [2/ 2/Seize ack /Seize ack] Apr 3 11:34:55 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 3 off [2/ 2/Group A /DNIS request ] Apr 3 11:34:55 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 1 off - [2/ 2/Group A /DNIS request ] Apr 3 11:34:55 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 3 on [2/ 2/Group A /DNIS request ] Apr 3 11:34:55 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 1 on - [2/ 2/Group A /DNIS request ] Apr 3 11:34:55 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 3 off [2/ 2/Group A /DNIS request ] Apr 3 11:34:55 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 1 off - [2/ 2/Group A /DNIS request ] Apr 3 11:34:55 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 3 on [2/ 2/Group A /DNIS request ] Apr 3 11:34:55 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 5 on - [2/ 2/Group A /DNIS request ] Apr 3 11:34:55 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 1 on [2/ 2/Group A /Category req ] Apr 3 11:35:20 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 1 off [2/ 2/Group A /ANI request ] Apr 3 11:35:20 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 5 off - [2/ 2/Group A /ANI request ] Apr 3 11:35:20 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 R2 prot. err. [2/ 2/Group A /ANI request ] cause 32771 - T3 timed out Apr 3 11:35:20 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 1001 - [1/ 1/Idle /Idle ] Apr 3 11:35:20 WARNING[17334]: chan_unicall.c:2644 handle_uc_event: Unicall/1 event Protocol failure -- Unicall/1 protocol error. Cause 32771 Apr 3 11:35:20 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Channel echo cancel Apr 3 11:35:20 WARNING[17334]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 -
