[asterisk-users] Call hangup after 60seconds
Hi, I have a client (xlite) connected to my server, on the server I have type=friend and siptimeout=60, canreinvite=yes and dial with tT option, the server is listening on port 5060. However, xlite is connect to a router where the port 5060 is blocked, therefore, I am using 5065 and I have an iptables rule to transfer the incoming packet from 5065 to 5060, I cannot use the port 5065 since some ATA the do not allow the change of the port. When I am calling with xlite the call endup after 60seconds, but in the 60seconds I can talk. Now if I am setting the client (in the sip.conf) in peer everything is working. Someone can explain to me why? What I am doing wrong? Thank you ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk canreinvite option questions
Dear all I have '*' 1.4.11 and 2E1 port hardware installed on it now i have single lan not nat anywhere ( 10.20.1.x ) all phone in single network domain without NAT now i have configured canreinvite=no so that asterisk work in meddle path of RTP so what is suggestable option of canreinvite=yes or no. I have SNOM phone at my users end. - Moody friends. Drama queens. Your life? Nope! - their life, your story. Play Sims Stories at Yahoo! Games. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone use the Linksys phones?
I have had a 941 for a couple of years. It works great for daily use at the office and I'm quite pleased with it. On 9/23/07, Robert Webb [EMAIL PROTECTED] wrote: Is anyone out there using any of the newer linksys phones since Cisco took over? I am more specifically looking at the spa-941 942's. Just curious about call quality, programability, and functionality with asterisk. I have read through the literature, but would like some real world feedback. Thanks ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why does everyone seem to dislike *now?
On 9/21/07, Tim Panton [EMAIL PROTECTED] wrote: I don't think IRC is the natural habitat of people who like NOW, NOW is for people who like web based GUIs. You are much more likely to find them over in the web based digium forums. Since we're talking about this, I have been on the #asterisk channel of Freenode for a few years. I came on as a complete newbie have very rarely used IRC if ever. Before posting on any IRC channel, it's absolutely imperative to lurk there as along as possible to see what kind of people are there, what the sense of humor is like (I'm thinking of say TKDefender or Steve Underwood as examples) and maybe you'll be ready to interface with the group after you've seen a few attacks on people who don't lurk before jumping in. Every extreme exists on tech IRC channels, and the key term here is EXTREME. SOme epople act like robots, immediately calling FLOOD! if someone pastes exactly three lines of a dialplan. Other are saying in explicit terms that if you'd bothered to google for this, blah which is very true 80% of the time and useless for things where an entry-level user whould know what to google for. (DISA? How would you know that term?) I myself am usually patient but I have gotten irritated and even had to resort to the ignore list a few times when after giving a few specific links it becomes obvious that the person just will not go study the stuff but wants to be hand held live. Yes, AsteriskNOW! and Trixbox are NOT the subjects of #asterisk. There is IMO though a need for a less mechanical way to make people understand that without immediate rudeness. For the faint at heart, monitored web forums are probably better. If you think IRC will help, the best way to use it would be literally to luck for days until a question comes up that you have some insight into. At that point, you can actually bring something in, and you'll defacto have become part of the group. By watching the dicsussion for a few days, you'll know all about pastebin, about not asking if you can asj a question, about having a ducks back for the few constantly and systematically rude people (I have no one person in mind here :) ansd you will find yourself laughing at lous to Steve Underwood's poker-face jokes that come out of nowhere. Personally, I will not talk to anyone who does not know what a dialplan is and refuses to go read a link to the explanation. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone use the Linksys phones?
I've installed them in a number of sites. The phones are good and easy to provision. If you need a good speakerphone then choose another phone. If there is something specific you need to know let me know. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Webb Sent: Monday, 24 September 2007 3:13 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Anyone use the Linksys phones? Is anyone out there using any of the newer linksys phones since Cisco took over? I am more specifically looking at the spa-941 942's. Just curious about call quality, programability, and functionality with asterisk. I have read through the literature, but would like some real world feedback. Thanks ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] # to transfer calls
Hi all, I wonder why my call was transferred when I pressed '#' in a conversation. How can I disable this kind of call transfer? Thanks. David ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] # to transfer calls
On Monday 24 September 2007 10:21:44 VoIP Newbie wrote: I wonder why my call was transferred when I pressed '#' in a conversation. How can I disable this kind of call transfer? Thanks. David Take a look at features.conf - probably there is blind transfer enabled on # key. Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sangoma or digium ?
Hi all, We need to get better echo cancellation on an Asterisk gateway. Currently it has two TE410P (1st gen) cards. So would it be possible to just buy two VPM450M cards ? Or do we need to buy two new TE412P cards ? In that case a Sangoma A108d card would be nice as well ? What configuration gives the best audio quality ? Thanks, Leon de Rooij [EMAIL PROTECTED] ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Virtual server Solution
Hello all, I'm looking for a solution to offer Virtual PBX, to my clients. I just saw software with multi-tenant support and I tested it, but no one likes me enought. Finally, I want to offer this service like a kind of hosting. Has you experience with multi-tenant software? Which has you tested? Has anyone experience about vhost, vserver, or something similar to run asterisk on it? Thanks VoIpCrazy ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone use the Linksys phones?
Is anyone out there using any of the newer linksys phones since Cisco took over? I am more specifically looking at the spa-941 942's. Just curious about call quality, programability, and functionality with asterisk. We've used the SPA-942s in most of our recent installs and been very impressed with them. The 941 lacks PoE and most importantly, has no backlight. I've found the bright, clear, backlit screen really makes a difference when it comes to users' first impressions. They also like the more upright nature of it compared to other IP phones - requires substantially less desk space than the Snom 360/370 or Aastra 55i/57i, for example. Firmware is improving fairly rapidly and is well worth updating before deployment. Remote provisioning documentation is somewhat scarce and does seem to involve a fair bit of trial and error, but once you get it working, it seems to work fairly well. I've replaced the Cisco 7960 on my desk with the colour SPA-962. It's about half the buy price of the Cisco, takes less desk space, has more features, and a vastly superior screen. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited For full contact details visit http://www.minotaur.it This email is made from 100% recycled electrons ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Virtual server Solution
Dear Tzafrir, I just try Destar, but one thing I dislike was, that there are no posibilities to login the manager of each virtual PBX. Then customers cannot manage their owns PBX. VoiPCrazy 2007/9/24, Tzafrir Cohen [EMAIL PROTECTED]: On Mon, Sep 24, 2007 at 11:38:38AM +0200, voip crazy wrote: Hello all, I'm looking for a solution to offer Virtual PBX, to my clients. I just saw software with multi-tenant support and I tested it, but no one likes me enought. Finally, I want to offer this service like a kind of hosting. Has you experience with multi-tenant software? Which has you tested? Has anyone experience about vhost, vserver, or something similar to run asterisk on it? Try destar. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma or digium ?
My experience is that both T1 and TDM cards from Sangoma which come with HWEC (hardware echo cancellation) give you excellent sound quality, better than those without HWEC. They do something which not only removes echo, but improves sound quality as well. Digium I haven't tried because of motherboard conflict issues. Sangoma's reviews are better than Digium's. On 9/24/07, Leon de Rooij [EMAIL PROTECTED] wrote: Hi all, We need to get better echo cancellation on an Asterisk gateway. Currently it has two TE410P (1st gen) cards. So would it be possible to just buy two VPM450M cards ? Or do we need to buy two new TE412P cards ? In that case a Sangoma A108d card would be nice as well ? What configuration gives the best audio quality ? Thanks, Leon de Rooij [EMAIL PROTECTED] ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Zeeshan A Zakaria ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk cli - vi keybindings ?
On 9/24/07, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Mon, Sep 24, 2007 at 02:04:05AM +0100, Ex Vito wrote: Is there any way to setup the asterisk cli to use such keybindings ? ... Set in your environment: AST_EDITOR=vi before starting Asterisk. (See main/asterisk.c) Great ! Thanks a lot. :-) -- exvito ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone use the Linksys phones?
Linksys are great phones. I like them but there only problem is limited line appearances. I prefer Aastra over them because Aastra has more lines appearances. They both are good. If you are not planning to have more than 4 lines, then Linksys is a great phone. On 9/24/07, Chris Bagnall [EMAIL PROTECTED] wrote: Is anyone out there using any of the newer linksys phones since Cisco took over? I am more specifically looking at the spa-941 942's. Just curious about call quality, programability, and functionality with asterisk. We've used the SPA-942s in most of our recent installs and been very impressed with them. The 941 lacks PoE and most importantly, has no backlight. I've found the bright, clear, backlit screen really makes a difference when it comes to users' first impressions. They also like the more upright nature of it compared to other IP phones - requires substantially less desk space than the Snom 360/370 or Aastra 55i/57i, for example. Firmware is improving fairly rapidly and is well worth updating before deployment. Remote provisioning documentation is somewhat scarce and does seem to involve a fair bit of trial and error, but once you get it working, it seems to work fairly well. I've replaced the Cisco 7960 on my desk with the colour SPA-962. It's about half the buy price of the Cisco, takes less desk space, has more features, and a vastly superior screen. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited For full contact details visit http://www.minotaur.it This email is made from 100% recycled electrons ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Zeeshan A Zakaria ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk crash and debug
Hello each 15 days my Asterisk crashes. Every time it happens I try to change something in its configuration to avoid the next crash. I already checked the logs but I don't know what to do. Can someone tell me whats the problem? These are my Asterisk logs: http://vox.fccn.pt/crash Thanks Regards Joao Pereira ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone use the Linksys phones?
On Mon, 2007-09-24 at 16:45 +1000, Klaverstyn, David C wrote: I've installed them in a number of sites. The phones are good and easy to provision. If you need a good speakerphone then choose another phone. SNIP I'd like to echo this. The SPA-942 'looks' the part, it's good value and bulk provision is a breeze. The general sound quality isn't as good as Aastra and Polycom imo, but the price is considerably lower. Kind Regards Dave Walker signature.asc Description: This is a digitally signed message part ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk and OCS integration
Hi List! does anyone played around with the OCS and Asterisk? I want to integrate OCS and Asterisk to enable Office Communicator 7.0 client to make and receive calls from PSTN I know that I need patch Asterisk to support TCP. But I am a bit ( a lot) lost Which more things should I need to keep in mind? Any advise will be wellcome :-) Thank you very much, Marta _ Acepta el reto MSN Premium: Protección para tus hijos en internet. Descárgalo y pruébalo 2 meses gratis. http://join.msn.com?XAPID=1697DI=1055HL=Footer_mailsenviados_proteccioninfantil ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Dropping Calls
Hello,
I am having an issue whereby calls are being dropped randomly. I have an
ISDN 30 E1 line going into a Wildcard TE220 (4th Gen). My Asterisk
install is based on Trixbox 2.0. However, I have updated the source code
to the following. The Asterisk release is asterisk-1.2.20. Zaptel
release is zaptel-1.2.18. And libpri release is libpri-1.2.4.
I have include an extract from the Asterisk log file below that shows
SIP/781 dropping a call when bridged to Zap/3-1. I have also included my
zaptel and zapata conf files.
I have researched the various messages displayed in the log file but
couldn't see anything that would point definitively to why calls are
being dropped.
Has anyone experienced anything similar or can anyone give me a few
ideas on where to start looking for the cause of the drop-outs?
Many thanks.
/var/log/asterisk/full:
Channel 0/3, span 1 got hangup request, cause 16
Sep 18 16:01:03 DEBUG[32377] channel.c: Didn't get a frame from channel:
Zap/3-1
Sep 18 16:01:03 DEBUG[32377] channel.c: Bridge stops bridging channels
SIP/781-b6e1b590 and Zap/3-1
Sep 18 16:01:03 DEBUG[32377] chan_zap.c: Set option AUDIO MODE, value:
ON(1) on Zap/3-1
Sep 18 16:01:03 DEBUG[32377] chan_zap.c: Hangup: channel: 3 index = 0,
normal = 15, callwait = -1, thirdcall = -1
Sep 18 16:01:03 DEBUG[32377] chan_zap.c: Not yet hungup... Calling
hangup once with icause, and clearing call
Sep 18 16:01:03 DEBUG[32377] chan_zap.c: disabled echo cancellation on
channel 3
Sep 18 16:01:03 DEBUG[32377] chan_zap.c: Set option TDD MODE, value:
OFF(0) on Zap/3-1
Sep 18 16:01:03 DEBUG[32377] chan_zap.c: Updated conferencing on 3, with
0 conference users
Sep 18 16:01:03 DEBUG[32377] chan_zap.c: Set option AUDIO MODE, value:
OFF(0) on Zap/3-1
Sep 18 16:01:03 DEBUG[32377] chan_zap.c: disabled echo cancellation on
channel 3
Sep 18 16:01:03 VERBOSE[32377] logger.c: -- Hungup 'Zap/3-1'
Sep 18 16:01:03 DEBUG[32377] app_dial.c: Exiting with DIALSTATUS=ANSWER.
Sep 18 16:01:03 VERBOSE[32377] logger.c: == Spawn extension
(macro-dialout-trunk, s, 20) exited non-zero on 'SIP/781-b6e1b590' in
macro 'dialout-trunk'
Sep 18 16:01:03 VERBOSE[32377] logger.c: == Spawn extension
(macro-dialout-trunk, s, 20) exited non-zero on 'SIP/781-b6e1b590'
Sep 18 16:01:03 VERBOSE[32377] logger.c: -- Executing
Macro(SIP/781-b6e1b590, hangupcall) in new stack
Sep 18 16:01:03 VERBOSE[32377] logger.c: -- Executing
ResetCDR(SIP/781-b6e1b590, w) in new stack
Sep 18 16:01:03 DEBUG[32377] cdr_addon_mysql.c: cdr_mysql: inserting a
CDR record.
Sep 18 16:01:03 DEBUG[32377] cdr_addon_mysql.c: cdr_mysql: SQL command
as follows: INSERT INTO cdr
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,dura
tion,billsec,disposition,amaflags,accountcode) VALUES ('2007-09-18
15:58:30','02072900400','02072900400','08704440730','from-internal',
'SIP/781-b6e1b590','Zap/3-1','ResetCDR','w',153,150,'ANSWERED',3,'')
Sep 18 16:01:03 DEBUG[32377] app_macro.c: Executed application: ResetCDR
Sep 18 16:01:03 VERBOSE[32377] logger.c: -- Executing
NoCDR(SIP/781-b6e1b590, ) in new stack
Sep 18 16:01:03 NOTICE[32377] cdr.c: CDR on channel 'SIP/781-b6e1b590'
not posted
Sep 18 16:01:03 NOTICE[32377] cdr.c: CDR on channel 'SIP/781-b6e1b590'
lacks end
Sep 18 16:01:03 DEBUG[32377] app_macro.c: Executed application: NoCDR
Sep 18 16:01:03 DEBUG[32377] pbx.c: Expression result is '1'
Sep 18 16:01:03 VERBOSE[32377] logger.c: -- Executing
GotoIf(SIP/781-b6e1b590, 1?skiprg) in new stack
Sep 18 16:01:03 VERBOSE[32377] logger.c: -- Goto
(macro-hangupcall,s,6)
Sep 18 16:01:03 DEBUG[32377] app_macro.c: Executed application: GotoIf
Sep 18 16:01:03 DEBUG[32377] pbx.c: Expression result is '1'
Sep 18 16:01:03 VERBOSE[32377] logger.c: -- Executing
GotoIf(SIP/781-b6e1b590, 1?theend) in new stack
Sep 18 16:01:03 VERBOSE[32377] logger.c: -- Goto
(macro-hangupcall,s,9)
Sep 18 16:01:03 DEBUG[32377] app_macro.c: Executed application: GotoIf
Sep 18 16:01:03 VERBOSE[32377] logger.c: -- Executing
Wait(SIP/781-b6e1b590, 5) in new stack
Sep 18 16:01:03 DEBUG[13856] chan_sip.c: Setting NAT on RTP to 524288
Sep 18 16:01:03 DEBUG[13856] chan_sip.c: Stopping retransmission on
'[EMAIL PROTECTED]' of Response 1: Match
Found
Sep 18 16:01:03 DEBUG[13856] chan_sip.c: Setting NAT on RTP to 524288
My zaptel.conf is as follows:
# Autogenerated by /usr/local/sbin/genzaptelconf -- do not hand edit
# Zaptel Configuration File
#
# This file is parsed by the Zaptel Configurator, ztcfg
#
# It must be in the module loading order
# Span 1: WCT1/0 Wildcard TE22xP Card 0
# channel 1, WCT1, unhandled for now
# channel 2, WCT1, unhandled for now
# channel 3, WCT1, unhandled for now
# channel 4, WCT1, unhandled for now
# channel 5, WCT1, unhandled for now
# channel 6, WCT1, unhandled for now
# channel 7, WCT1, unhandled for now
# channel 8, WCT1, unhandled for now
# channel 9, WCT1, unhandled for now
# channel 10, WCT1, unhandled
[asterisk-users] asterisk crash
I am using an asterisk to call another asteisk (i.e
Dial([EMAIL PROTECTED]) in asteriskA). After that, the following error
message displayed and asterisk crashes at once. Anyone has such
experience and can help to fix it?
asterisk version: 1.4.11
zaptel 1.4.5.1
using RealTime
/usr/sbin/safe_asterisk: line 118: 10247 Segmentation fault (core
dumped) nice -n $PRIORITY ${ASTSBINDIR}/asterisk -f ${CLIARGS}
${ASTARGS} /dev/${TTY} /dev/${TTY}
Asterisk ended with exit status 139
Asterisk exited on signal 11.
Automatically restarting Asterisk.
mpg123: no process killed
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Re: [asterisk-users] Asterisk crash and debug
On Monday 24 September 2007 14:06:27 Joao Pereira wrote: Hello each 15 days my Asterisk crashes. Every time it happens I try to change something in its configuration to avoid the next crash. I already checked the logs but I don't know what to do. Can someone tell me whats the problem? These are my Asterisk logs: http://vox.fccn.pt/crash Well, this seems familiar. Notice that the first line of starting asterisk is Sep 24 09:45:56 VERBOSE[7784] logger.c: Asterisk Event Logger Started /var/log/asterisk/event_log And line before is Sep 24 09:45:51 DEBUG[14393] manager.c: Manager received command 'Command' So, you're doing some CLI command trough AMI. I guess, it's show channels ;) I've seen it a lot on 1.2 (am i correct). I get rid of that o stopped only after upgrading to 1.4.10 Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Virtual server Solution
Hai, Is the Asterisk supports PMC-Sierra Analogue Telephone Adapter? Thanks Regards Bincy K Philip ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma or digium ?
Hi all, Thanks everyone for the quick reply. I think I'll go for the Sangoma as I just saw that the VPM450M is no option anyway: http://store.voxilla.com/voip-products/digium-vpm450m.html We have rev1 cards, while the module needs rev3 or greater. Thanks again, Leon de Rooij On Mon, 2007-09-24 at 06:58 -0400, Zeeshan Zakaria wrote: My experience is that both T1 and TDM cards from Sangoma which come with HWEC (hardware echo cancellation) give you excellent sound quality, better than those without HWEC. They do something which not only removes echo, but improves sound quality as well. Digium I haven't tried because of motherboard conflict issues. Sangoma's reviews are better than Digium's. On 9/24/07, Leon de Rooij [EMAIL PROTECTED] wrote: Hi all, We need to get better echo cancellation on an Asterisk gateway. Currently it has two TE410P (1st gen) cards. So would it be possible to just buy two VPM450M cards ? Or do we need to buy two new TE412P cards ? In that case a Sangoma A108d card would be nice as well ? What configuration gives the best audio quality ? Thanks, Leon de Rooij [EMAIL PROTECTED] ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Zeeshan A Zakaria ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX Java Softphone?
All of my testing has shown it be be pretty clean. We have it on our contact us page of our website and we also give that url to overseas (India, Germany, Japan) contacts and some have used it. Some do not want to open up the iax2 port in their firewall, but that is their issue. I wanted to use IAX2 because I knew with NAT and firewalls, that IAX2 was easier for people to use than all of the RTP ports required for SIP. -- -- Steven http://www.glimasoutheast.org Dean Collins [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Steven, how reliable is that freeware? I tried it when it first came out but I couldn't get it to work. It didn't matter at the time as I was working for Mexuar at the time but now I don't have their service anymore I'd like to use it/something like it for my other consultancy services. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Steven Sent: Thursday, 20 September 2007 2:12 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] IAX Java Softphone? I use click2call. http://www.geocities.com/babarnazmi/index2.htm It is an activex control though. -- -- Steven http://www.glimasoutheast.org Matthew Rubenstein [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Does anyone know of an IAX softphone in Java, whether applet or application? Even the most minimum featureset, just voice and dialing, or even embedded in some other app/let. Preferably GPL. Thanks. -- (C) Matthew Rubenstein ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma or digium ?
Leon de Rooij wrote: Hi all, We need to get better echo cancellation on an Asterisk gateway. Currently it has two TE410P (1st gen) cards. So would it be possible to just buy two VPM450M cards ? Or do we need to buy two new TE412P cards ? In that case a Sangoma A108d card would be nice as well ? What configuration gives the best audio quality ? Thanks, Leon de Rooij [EMAIL PROTECTED] I have rarely if ever seen echo in non POTS TDM (ISDN) lines that was so bad that it could not be addressed with software EC. Maybe if you describe the gateway and how it functions, plus the phones you are using, you may not have to buy or replace anything. What are your echo can settings? What method of EC are you using? Have you tried working with Digium's support on this issue? Thanks, Steve Totaro ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems Connecting Two Asterisk Installs Via ISDN PRI Cards
At this point I do not think the problem is the wiring. What else should I try? ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone use the Linksys phones?
Is anyone out there using any of the newer linksys phones since Cisco took over? I am more specifically looking at the spa-941 942's. Just curious about call quality, programability, and functionality with asterisk. We like these phones, hundreds deployed to business customers. Mass provisioning is a bit tricky. Make sure you get this worked out in the lab before looking like an idiot in front of the customer. Set option 66 TFTP [IP Address] in the DHCP server, pointed to your config files. You can use a perl script to generate any number of config files with a text file as input of MAC, User Name, Exten Number, Server IP. Users seem to have trouble with the 941's and handling multiple calls, using the soft keys to navigate back and forth between hold calls, for this reason, I don't use the 941's, just the 942's and put first line extension on all line 4 positions, users seem to understand and use this better. Do upgrade to the latest firmware and check for new updates frequently, bug fixes and added and features come relatively quick on this phone, which is a good thing. Paging takes priority over an existing call, so be careful if you plan to do phone-to-phone intercom, it annoys the hell out of users when a page puts a call on hold automatically, you can turn call waiting off on the phone or set the page not to auto answer or in Asterisk you can check the channel state and not page if existing channel exists (this is the best way). I can send you the dial plan to handle this if needed. Set the backlight to 'always on' on the user tab, any SIP messaging or SIP info messages to the phone triggers the backlight, annoys users when the light is on-off-on-off-on-off throughout the day. No way to upload a directory file yet, wish there was. 10 second boot time, up and running, which is really great. Lots of good features, solid mid to low end cost business phone, customers seem to like it and not many support calls once the users get used to using it. JR -- JR Richardson Engineering for the Masses ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone use the Linksys phones?
If you are not planning to have more than 4 lines, then Linksys is a great phone. Out of the hundreds of users I've spoken to, there are only 2 individuals I can think of that routinely juggle more than 2 concurrent calls. The 4 line limitation has never been a problem for the vast majority of people. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited For full contact details visit http://www.minotaur.it This email is made from 100% recycled electrons ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems Connecting Two Asterisk Installs Via ISDN PRI Cards
Brian Alexander wrote: At this point I do not think the problem is the wiring. What else should I try? Have you confirmed that the failing card is working correctly? Maybe the card is at fault. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems Connecting Two Asterisk Installs Via ISDN PRI Cards
On 9/24/07, Doug Lytle [EMAIL PROTECTED] wrote: Have you confirmed that the failing card is working correctly? Maybe the card is at fault. All of the cards have been confirmed to work by themselves. -Brian ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems Connecting Two Asterisk Installs Via ISDN PRI Cards
Brian Alexander wrote: On 9/24/07, *Doug Lytle* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Have you confirmed that the failing card is working correctly? Maybe the card is at fault. All of the cards have been confirmed to work by themselves. The only other suggestion I have would have would be to use IAX instead of PRI for inter-machine communications. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems Connecting Two Asterisk Installs Via ISDN PRI Cards
On 9/24/07, Doug Lytle [EMAIL PROTECTED] wrote: The only other suggestion I have would have would be to use IAX instead of PRI for inter-machine communications. LOL Yeah, normally that is what I would use. Unfortunately it is not an option for this... ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom 501 Phones Rebooting
Finally, press and hold all 4 arrow keys until the phone bleeps, then capture the log files dumped to your provisioning server one last time. If the problem's not obvious from reading the logs, escalate these logs to your Polycom reseller and ask them to open a ticket with Polycom on your behalf. Of course they might recommend upgrading to 2.x ;-) Well, here is what I got. Have no idea how to read these.. 0924095418|so |2|177|soToneStop: Stopping host tone 0x10b892a4 (dial). [0x0] 0924095418|so |2|177|[SoMediaSessC]: Call (0x10b42844) - New State: SoMediaSessStateSetup (Real) 0924095418|so |2|177|[SoMediaSessC]: 0x10b42844: State: SoMediaSessStateSetup, Start Timer: 1000 msecs 0924095418|so |2|177|[SoMediaSessC]: Call (0x10b42844) - New State: SoMediaSessStateOverlap (Real) 0924095418|so |2|177|[SoMediaSessC]: 0x10b42844: State: SoMediaSessStateOverlap, Start Timer: 3 msecs 0924095418|sip |2|177|SipCallMake 8605654321 0924095418|sip |2|177|doDnsNaptrLookup: '192.168.1.1' is an IP, bailing 0924095418|sip |2|177|SipOnEvCallNewState 10edbff0,10b42844 2,Proceeding 0924095418|so |2|177|[SoMediaSessC]: MsgPpsCallStateInd Usr(0x10b42844) Net(0x10edbff0) St(3) 0924095418|so |2|177|[SoMediaSessC]: Call (0x10b42844) - State (SoMediaSessStateOverlap) - Event (SoMediaSessEvLclNetProceeding) 0924095418|so |2|177|[SoMediaSessC]: Call (0x10b42844) - New State: SoMediaSessStateProceeding (Real) 0924095418|so |2|177|[SoMediaSessC]: 0x10b42844: State: SoMediaSessStateProceeding, Start Timer: 6 msecs 0924095418|sip |3|177|407 challenge received 0924095418|sip |2|177|doDnsNaptrLookup: '192.168.1.1' is an IP, bailing 0924095418|sip |2|177|SipOnEvCallNewState 10edbff0,10b42844 2,Proceeding 0924095418|so |2|177|[SoMediaSessC]: MsgPpsCallStateInd Usr(0x10b42844) Net(0x10edbff0) St(3) 0924095418|so |2|177|[SoMediaSessC]: Call (0x10b42844) - State (SoMediaSessStateProceeding) - Event (SoMediaSessEvLclNetProceeding) 0924095418|so |2|177|[SoNcasC]: Receiving MsgType 0x848 0924095418|sip |2|177|SipOnEvCallNewState 10edbff0,10b42844 3,NULL 0924095418|so |2|177|[SoMediaSessC]: MsgPpsCallStateInd Usr(0x10b42844) Net(0x10edbff0) St(4) 0924095418|so |2|177|[SoMediaSessC]: Call (0x10b42844) - State (SoMediaSessStateProceeding) - Event (SoMediaSessEvLclNetRingback) 0924095418|so |2|177|soToneStop: Stopping host tone 0x10b892a4 (dial). [0x0] 0924095418|so |2|177|soToneStop: Stopping host tone 0x10b892a4 (dial). [0x0] 0924095418|so |2|177|[SoMediaSessC]: Call (0x10b42844) - New State: SoMediaSessStateRingBack (Real) 0924095418|so |2|177|[SoMediaSessC]: 0x10b42844: State: SoMediaSessStateRingBack, Start Timer: 6 msecs 0924095418|sip |2|177|SipOnEvNewCodec 101a8c0,0 PCMU/8000 18536,2232 ptime=0,dir 2 index 0 0924095418|so |2|177|[SoMediaSessC]: [0x10b42844] pps[0x10edbff0] i[0x0] d[2] p[0] pn[0] lp[2232] rip[192.168.1.1] rp[18536] dp[30333] 0924095418|so |2|177|[SoMediaSessC]: Call (0x10b42844) - State (SoMediaSessStateRingBack) - Event (SoMediaSessEvLclNetMediaInfoInd) 0924095418|sip |2|177|SipOnEvNewCodec 101a8c0,101 telephone-event/8000 18536,2232 ptime=0,dir 2 index 0 0924095418|so |2|177|soStreamAddrSet DestIP: local RTP port=2232 dest IP=192.168.1.1 dest port=18536 (10B359C0) 0924095418|so |2|177|soStreamNetConn attempt to re-connect RTP stream to network 0924095418|so |2|177|soStreamNetConn attempt to re-open RTCP port 0924095418|so |2|177|soStreamLclTermConn: receive-only (10B359C0) 0924095418|so |2|177|soStreamNetConn attempt to re-connect RTP stream to network 0924095418|so |2|177|[SoStreamC]: 1st rtp pkt rx now. 0924095418|so |2|177|soStreamNetConn attempt to re-open RTCP port 0924095418|so |2|177|soStreamLclTermConn: send-and-receive (10B359C0) 0924095418|so |2|177|[SoMediaSessC]: [0x10b42844] pps[0x10edbff0] i[0x0] d[2] p[120] pn[6] lp[2232] rip[192.168.1.1] rp[18536] dp[101] 0924095418|so |2|177|[SoMediaSessC]: Call (0x10b42844) - State (SoMediaSessStateRingBack) - Event (SoMediaSessEvLclNetMediaInfoInd) 0924095418|so |2|177|soStreamNetConn attempt to re-connect RTP stream to network 0924095418|so |2|177|soStreamNetConn attempt to re-open RTCP port 0924095418|so |2|177|soStreamLclTermConn: receive-only (10B359C0) 0924095418|so |2|177|soStreamNetConn attempt to re-connect RTP stream to network 0924095418|so |2|177|soStreamNetConn attempt to re-open RTCP port 0924095418|so |2|177|soStreamLclTermConn: send-and-receive (10B359C0) 0924095418|so |2|177|[SoStreamC]: 1st rtp pkt tx now. 0924095423|sip |2|177|SipOnEvNewCodec 101a8c0,0 PCMU/8000 18536,2232 ptime=0,dir 2 index 0 0924095423|so |2|177|[SoMediaSessC]: [0x10b42844] pps[0x10edbff0] i[0x0] d[2] p[0] pn[0] lp[2232] rip[192.168.1.1] rp[18536] dp[30333] 0924095423|so |2|177|[SoMediaSessC]: Ignoring Media Info - No Critical Change Detected 0924095423|sip |2|177|SipOnEvNewCodec 101a8c0,101 telephone-event/8000 18536,2232 ptime=0,dir 2 index 0 0924095423|so
Re: [asterisk-users] Asterisk and OCS integration
I would use SER or OpenSER as a middle man. Set it up to receive via TCP and send it on to the asterisk server using UDP. Kind Regards Jon Leren Schøpzinsky Solution Engineer Dansk Erhvervs-Telefon A/S tlf: +45 88200336 mob: +45 31206709 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of dadsadsadf dsadasdsa Sent: 24. september 2007 13:29 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk and OCS integration Hi List! does anyone played around with the OCS and Asterisk? I want to integrate OCS and Asterisk to enable Office Communicator 7.0 client to make and receive calls from PSTN I know that I need patch Asterisk to support TCP. But I am a bit ( a lot) lost Which more things should I need to keep in mind? Any advise will be wellcome :-) Thank you very much, Marta _ Acepta el reto MSN Premium: Protección para tus hijos en internet. Descárgalo y pruébalo 2 meses gratis. http://join.msn.com?XAPID=1697DI=1055HL=Footer_mailsenviados_proteccioninfantil ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4.12 Release?
Hi All, I read rumors of a potential Asterisk 1.4.12 release for last week. I would like to start testing Asterisk 1.4 on Solaris, but, the fix for the segfault in res_features is only in the current development trunk. I would much rather test a release version, and as such, need to wait for the release of 1.4.12, my question is, do we have a guestimate on when it will be released, 1 week, 2 weeks, a month? Thanks Bruce begin:vcard fn:Bruce McAlister n:McAlister;Bruce org:Blueface Ltd adr:;;8 Clanwilliam Terrace;Dublin;Dublin;Dublin 2;Ireland email;internet:[EMAIL PROTECTED] tel;work:+353 1 524 2009 x-mozilla-html:FALSE url:http://www.blueface.ie version:2.1 end:vcard ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems Connecting Two Asterisk Installs Via ISDN PRI Cards
Brian Alexander wrote: On 9/24/07, *Doug Lytle* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: The only other suggestion I have would have would be to use IAX instead of PRI for inter-machine communications. LOL Yeah, normally that is what I would use. Unfortunately it is not an option for this... You mentioned that one of the machines have two PRI cards. If it's in the one machine that's failing, maybe try it with just one PRI to see if it's the causing issues. -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone use the Linksys phones? (Zeeshan Zakaria)
Note that the newish SPA962 has 6 appearances and a color screen. I've noticed that the bright color screen does impress people when they first see it. PoE is also very nice and web provisioning was quite easy. I've yet to try a more automated provisioning method on it. I know that getting the polycom's to auto provision wasn't very straight forward. I do provision some the linksys PAP2Ts via HTTP and that works quite well, so I suspect the SPA's to be relatively similar. Norman Franke ASD, Inc. www.myasd.com On Sep 24, 2007, at 7:06 AM, [EMAIL PROTECTED] wrote: Linksys are great phones. I like them but there only problem is limited line appearances. I prefer Aastra over them because Aastra has more lines appearances. They both are good. If you are not planning to have more than 4 lines, then Linksys is a great phone. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk cli - vi keybindings ?
Tzafrir Cohen wrote: On Mon, Sep 24, 2007 at 02:04:05AM +0100, Ex Vito wrote: This might sound lika a small issu, but here it goes: I'm a long time unix user and my shell history usage and editing is configured to use vi keybindings; it's something that's already built into my fingers and using different bindings, like the arrow keys to fetch previous lines, really blows me !... :-( Is there any way to setup the asterisk cli to use such keybindings ? I took a quick glance at 1.4.11 source and found readline.[ch] files, but asterisk is not behaving to my inputrc configuration... Googled for a while to no effect. Set in your environment: AST_EDITOR=vi AWESOME! ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Adding allowed codecs to Asterisk
Hi, In what direction should I start looking so that I can let Asterisk forward (pass through) sip calls containing strange codecs without transcoding? (Not just the G.729, which already seems to have some sort of support in Asterisk.) Are there some config files to fix, or should I reprogram/recompile? What files should I start looking at? Thanks in advance, Matti Zemack, BBC RI, Kingswood, UK http://www.bbc.co.uk/ This e-mail (and any attachments) is confidential and may contain personal views which are not the views of the BBC unless specifically stated. If you have received it in error, please delete it from your system. Do not use, copy or disclose the information in any way nor act in reliance on it and notify the sender immediately. Please note that the BBC monitors e-mails sent or received. Further communication will signify your consent to this. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone use the Linksys phones?
Interesting comment on the speakerphone. Have you found a reasonably priced desk set with a good speakerphone? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Klaverstyn, David C Sent: Monday, September 24, 2007 2:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Anyone use the Linksys phones? I've installed them in a number of sites. The phones are good and easy to provision. If you need a good speakerphone then choose another phone. If there is something specific you need to know let me know. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom 501 Phones Rebooting
Douglas Garstang wrote: Wow. Polycom phones are STILL doing that? I haven't been involved with Polycom phones since before January, and it was a problem back then too. Jeez... Doug -- he's using 1.6.7 firmware. -Stephen- ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom 501 Phones Rebooting
Hi, Greg: I really can't recommend upgrading to a 2.x firmware highly enough. Many people had the spontaneous reboot problems and I think they were all solved by going to current 2.x firmware. -Stephen- Gregory Boehnlein wrote: Finally, press and hold all 4 arrow keys until the phone bleeps, then capture the log files dumped to your provisioning server one last time. If the problem's not obvious from reading the logs, escalate these logs to your Polycom reseller and ask them to open a ticket with Polycom on your behalf. Of course they might recommend upgrading to 2.x ;-) Well, here is what I got. Have no idea how to read these.. 0924095418|so |2|177|soToneStop: Stopping host tone 0x10b892a4 (dial). [0x0] 0924095418|so |2|177|[SoMediaSessC]: Call (0x10b42844) - New State: SoMediaSessStateSetup (Real) 0924095418|so |2|177|[SoMediaSessC]: 0x10b42844: State: SoMediaSessStateSetup, Start Timer: 1000 msecs 0924095418|so |2|177|[SoMediaSessC]: Call (0x10b42844) - New State: SoMediaSessStateOverlap (Real) 0924095418|so |2|177|[SoMediaSessC]: 0x10b42844: State: SoMediaSessStateOverlap, Start Timer: 3 msecs 0924095418|sip |2|177|SipCallMake 8605654321 0924095418|sip |2|177|doDnsNaptrLookup: '192.168.1.1' is an IP, bailing 0924095418|sip |2|177|SipOnEvCallNewState 10edbff0,10b42844 2,Proceeding 0924095418|so |2|177|[SoMediaSessC]: MsgPpsCallStateInd Usr(0x10b42844) Net(0x10edbff0) St(3) 0924095418|so |2|177|[SoMediaSessC]: Call (0x10b42844) - State (SoMediaSessStateOverlap) - Event (SoMediaSessEvLclNetProceeding) 0924095418|so |2|177|[SoMediaSessC]: Call (0x10b42844) - New State: SoMediaSessStateProceeding (Real) 0924095418|so |2|177|[SoMediaSessC]: 0x10b42844: State: SoMediaSessStateProceeding, Start Timer: 6 msecs 0924095418|sip |3|177|407 challenge received 0924095418|sip |2|177|doDnsNaptrLookup: '192.168.1.1' is an IP, bailing 0924095418|sip |2|177|SipOnEvCallNewState 10edbff0,10b42844 2,Proceeding 0924095418|so |2|177|[SoMediaSessC]: MsgPpsCallStateInd Usr(0x10b42844) Net(0x10edbff0) St(3) 0924095418|so |2|177|[SoMediaSessC]: Call (0x10b42844) - State (SoMediaSessStateProceeding) - Event (SoMediaSessEvLclNetProceeding) 0924095418|so |2|177|[SoNcasC]: Receiving MsgType 0x848 0924095418|sip |2|177|SipOnEvCallNewState 10edbff0,10b42844 3,NULL 0924095418|so |2|177|[SoMediaSessC]: MsgPpsCallStateInd Usr(0x10b42844) Net(0x10edbff0) St(4) 0924095418|so |2|177|[SoMediaSessC]: Call (0x10b42844) - State (SoMediaSessStateProceeding) - Event (SoMediaSessEvLclNetRingback) 0924095418|so |2|177|soToneStop: Stopping host tone 0x10b892a4 (dial). [0x0] 0924095418|so |2|177|soToneStop: Stopping host tone 0x10b892a4 (dial). [0x0] 0924095418|so |2|177|[SoMediaSessC]: Call (0x10b42844) - New State: SoMediaSessStateRingBack (Real) 0924095418|so |2|177|[SoMediaSessC]: 0x10b42844: State: SoMediaSessStateRingBack, Start Timer: 6 msecs 0924095418|sip |2|177|SipOnEvNewCodec 101a8c0,0 PCMU/8000 18536,2232 ptime=0,dir 2 index 0 0924095418|so |2|177|[SoMediaSessC]: [0x10b42844] pps[0x10edbff0] i[0x0] d[2] p[0] pn[0] lp[2232] rip[192.168.1.1] rp[18536] dp[30333] 0924095418|so |2|177|[SoMediaSessC]: Call (0x10b42844) - State (SoMediaSessStateRingBack) - Event (SoMediaSessEvLclNetMediaInfoInd) 0924095418|sip |2|177|SipOnEvNewCodec 101a8c0,101 telephone-event/8000 18536,2232 ptime=0,dir 2 index 0 0924095418|so |2|177|soStreamAddrSet DestIP: local RTP port=2232 dest IP=192.168.1.1 dest port=18536 (10B359C0) 0924095418|so |2|177|soStreamNetConn attempt to re-connect RTP stream to network 0924095418|so |2|177|soStreamNetConn attempt to re-open RTCP port 0924095418|so |2|177|soStreamLclTermConn: receive-only (10B359C0) 0924095418|so |2|177|soStreamNetConn attempt to re-connect RTP stream to network 0924095418|so |2|177|[SoStreamC]: 1st rtp pkt rx now. 0924095418|so |2|177|soStreamNetConn attempt to re-open RTCP port 0924095418|so |2|177|soStreamLclTermConn: send-and-receive (10B359C0) 0924095418|so |2|177|[SoMediaSessC]: [0x10b42844] pps[0x10edbff0] i[0x0] d[2] p[120] pn[6] lp[2232] rip[192.168.1.1] rp[18536] dp[101] 0924095418|so |2|177|[SoMediaSessC]: Call (0x10b42844) - State (SoMediaSessStateRingBack) - Event (SoMediaSessEvLclNetMediaInfoInd) 0924095418|so |2|177|soStreamNetConn attempt to re-connect RTP stream to network 0924095418|so |2|177|soStreamNetConn attempt to re-open RTCP port 0924095418|so |2|177|soStreamLclTermConn: receive-only (10B359C0) 0924095418|so |2|177|soStreamNetConn attempt to re-connect RTP stream to network 0924095418|so |2|177|soStreamNetConn attempt to re-open RTCP port 0924095418|so |2|177|soStreamLclTermConn: send-and-receive (10B359C0) 0924095418|so |2|177|[SoStreamC]: 1st rtp pkt tx now. 0924095423|sip |2|177|SipOnEvNewCodec 101a8c0,0 PCMU/8000 18536,2232 ptime=0,dir 2 index 0 0924095423|so |2|177|[SoMediaSessC]:
Re: [asterisk-users] Anyone use the Linksys phones?
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Bagnall Sent: Monday, 24 September 2007 6:15 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Anyone use the Linksys phones? Is anyone out there using any of the newer linksys phones since Cisco took over? I am more specifically looking at the spa-941 942's. Just curious about call quality, programability, and functionality with asterisk. I've replaced the Cisco 7960 on my desk with the colour SPA-962. It's about half the buy price of the Cisco, takes less desk space, has more features, and a vastly superior screen. -- I'd like to second the SPA962 - I've deployed a couple of them now and they're great, clients get a kick out of sticking the company logo in colour on the screen and as of fw 5.1.15 the SPA932 sidecar supports asterisk for BLF and speed dial. They're also supposed to to support RSS for stock ticker type scrollies but haven't played with this yet. The only nasty thing I've found is that whenever the handsets resync they reboot even if no settings have changed. When this occurs anything connected to the phones second Ethernet port will drop connection for a few seconds. Craig ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone use the Linksys phones?
On 9/24/07, Craig Guy [EMAIL PROTECTED] wrote: The only nasty thing I've found is that whenever the handsets resync they reboot even if no settings have changed. When this occurs anything connected to the phones second Ethernet port will drop connection for a few seconds. The phones can send a parameter to the provisioning server to indicate that they want an Update if they do this, and you send no network or other major config parameters, the phone does not reboot. Look at the Linksys provisioning PDF for more details of the parameter. :) Simple. Steve ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Samsung Sprint CDMAoIP
If you pay for the free calling via wifi, TMobile bases cost of call where it initiates. So you call your long lost buddy from the house, jump in the car, drive for an hour, the entire call is free. If your buddy calls you as you're pulling in the driveway, you have the same hour long call, you'd better have plenty of minutes. If you don't pay for the free calling via wifi, it doesn't matter, the only benefit of using the wifi is coverage in dead spots. Paul wrote: Does it switch back to wifi from gsm tower? If so, I would hope they count total gsm seconds for the call to determine how many minutes get deducted from the wireless plan. Otherwise, somebody could get clipped a full minute for every time he leaned out the window to yell at the kids. [EMAIL PROTECTED] wrote: Actually TMobile offers best of both worlds: If you only concern is poor coverage, you pay no additional money, but your phone calls are transported via wifi. If you want free calling via wifi, you pay $10 for a single line, $20 for a family plan. Eric Chamberlain wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Jason Parker Sent: Thursday, September 20, 2007 11:49 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] OT: Samsung Sprint CDMAoIP If you're referring to T-Mobile's [EMAIL PROTECTED] service, it's actually $20 per month, per line on the account (unless it's changed very recently). As far as how it works on T-Mobile, I recently had some questions answered by them about that.. They use UMA over wifi, and it does automatic switching between the wifi and the gsm towers (ie; your call stays up). Quote from the tech I talked to: [EMAIL PROTECTED] does not use a VoIP protocol, as the voice data is transferred from the Internet directly to our UMA Gateway and then through our regular Mobile Switching Centers. Pretty interesting stuff. Interesting from a marketing and sales perspective that one can get people to buy a box, pay for the bandwidth used by the box, and then pay an extra $20/month per phone, all for coverage problems the carrier should address. But then again these carriers have managed to convince people to pay close to a thousand dollars per megabyte for SMS messages. -- Eric Chamberlain, CISSP Chief Technical Officer Voxilla - http://voxilla.com/ ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone use the Linksys phones?
On 9/24/07, Steve Davies [EMAIL PROTECTED] wrote: The phones can send a parameter to the provisioning server to indicate that they want an Update if they do this, and you send no network or other major config parameters, the phone does not reboot. Look at the Linksys provisioning PDF for more details of the parameter. Really? I've been through this document several times looking for something like this and haven't found a single reference to it. Could you provide more details or at least a page number in the Linksys SPA provisioning doc? Thanks! ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4.12 Release?
On Mon, Sep 24, 2007 at 03:18:53PM +0100, Bruce McAlister wrote:
Hi All,
I read rumors of a potential Asterisk 1.4.12 release for last week. I
would like to start testing Asterisk 1.4 on Solaris, but, the fix for
the segfault in res_features is only in the current development trunk. I
would much rather test a release version, and as such, need to wait for
the release of 1.4.12, my question is, do we have a guestimate on when
it will be released, 1 week, 2 weeks, a month?
Have you tried the svn version of branches/1.4 ?
To help you start your testing, here's a nice little script to package a
tarball from current SVN. It's for Zaptel, and adjustments to Asterisk
are left as an exercise to the reader.
It's not that well tested, but then again, that's what this list is
for...
--
Tzafrir Cohen
icq#16849755 jabber:[EMAIL PROTECTED]
+972-50-7952406 mailto:[EMAIL PROTECTED]
http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir
#!/bin/sh
# svn_tarball: Generate a tarball from svn.digium.com irelease branch.
# Tzafrir Cohen [EMAIL PROTECTED]
set -e
BRANCH_NAME=1.4
REV=HEAD
PROJECT=zaptel
TARBALLS_DIR_DEBIAN=/home/svn/tarballs
VENDORS_SUBDIR=vendors
TARBALLS_DIR=$TARBALLS_DIR_DEBIAN/$VENDORS_SUBDIR
me=`basename $0`
say() {
echo $me: $@
}
usage() {
echo 2 $0: packages uploading script
echo 2
echo 2 $0 [-r REV] [-2] [-p PROJECT]
echo 2 $0 -h | --help: This message
echo 2
echo 2 Options:
echo 2-2 --zap12: Use 1.2.
echo 2-p --project PROJECT: Use PROJECT instead of $PROJECT.
echo 2-r --rev REV: extract tarball from this revision ($REV).
echo 2-s --show:Only show the versions you will create.
}
opt_showonly=no
options=`getopt -o 2hp:r:s --long zap12,help,project:,rev:,revision:,show --
$@`
if [ $? != 0 ] ; then echo 2 Terminating... ; exit 1 ; fi
# Note the quotes around `$TEMP': they are essential!
eval set -- $options
echo $@
while true ; do
case $1 in
-2|--zap12) BRANCH_NAME=1.2;;
-p|--project) PROJECT=$2; shift ;;
-r|--rev|--revision) REV=$2; shift ;;
-s|--show) opt_showonly=yes ;;
-h|--help) usage; exit 0;;
--) shift ; break ;;
esac
shift;
done
BRANCH=branches/$BRANCH_NAME
PROJECT_BASE=http://svn.digium.com/svn/$PROJECT
PROJECT_URL=$PROJECT_BASE/$BRANCH
set -e
# Get the name of the previous version for this release.
# The idea is to look at the latest tag for that branhch. Tags are
# global, and hence we filter tag names by branch name.
#
# Note: this strips any minor version number.
# e.g: if last releast was 1.4.5.1, this will still return 1.4.5 . Here
# we rely on the fact that the revision number will be added.
zap_ver=`svn ls -r $REV $PROJECT_BASE/tags | grep ^$BRANCH_NAME \
| sed -e s/\($BRANCH_NAME\.[0-9]\+\)[/.-].*/\1/ \
| sort -nu -t . -k 3 | tail -n 1`
real_rev=`svn info -r $REV $PROJECT_URL \
| awk '/^Last Changed Rev: /{print $4}'`
ver_full=$zap_ver.9.svn.$real_rev
tar_name=zaptel-$ver_full
tar_ball_full=$TARBALLS_DIR/$tar_name.tar.gz
say Version: $ver_full (ver: $zap_ver, rev: $real_rev)
say Tarball: $tar_ball_full
if [ $opt_showonly = 'yes' ]; then
exit 0;
fi
CHECKOUT_DIR=`mktemp -d zaptel_checkout_dir_XX`
# Package a tarball from the subversion, using 'make dist':
svn export -q -r $REV $PROJECT_URL $CHECKOUT_DIR/$tar_name
echo $ver_full $CHECKOUT_DIR/$tar_name/.version
tar cz -C $CHECKOUT_DIR -f $tar_ball_full $tar_name
rm -rf $CHECKOUT_DIR
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[asterisk-users] CallerID problem Asterisk 1.4.2
When receiving inbound calls from a Vonage Softphone extension, I'm unable
to view/maniupulate calledid data. but it shows up in the CDR records and on
called handsets.. any ideas?
exten = asda,n,NoOp(callerID is ${CALLERID})
exten = asda,n,NoOp(CallerID is ${CALLERIDNAME})
exten = asda,n,NoOp(CallerID is ${CALLERIDNUM})
-- Executing [EMAIL PROTECTED]:2] Wait(SIP/asd1-086775b8, 1) in new
stack
-- Executing [EMAIL PROTECTED]:3] NoOp(SIP/asd-086775b8, callerID is )
in
new stack
-- Executing [EMAIL PROTECTED]:4] NoOp(SIP/asd-086775b8, CallerID is )
in
new stack
-- Executing [EMAIL PROTECTED]:5] NoOp(SIP/asd-086775b8, CallerID is )
in
new stack
But CDR data is there.
WIRELESS CALLER
asd,asd,100,pstn-in,SIP/asd-086775b8,SIP/peter5-08689458,Dial
,SIP/peterS
IP/peter2SIP/peter3SIP/peter4SIP/peter5|25|m,2007-09-24
11:19:27,2007-09-24 11:19:27,2007-09-24 11:19:44,17,17,ANSWER
ED,DOCUMENTATION,,1190657967.4165,
Peter Kranz
Founder/CEO - Unwired Ltd
www.UnwiredLtd.com
Desk: 510-868-1614 x100
Mobile: 510-207-
[EMAIL PROTECTED]
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[asterisk-users] Spur error with Siemens Hi Path
Hi, I have an IBM server running latest asterisk 1.4.x connected to a Siemens hi-path user a TE120P single-span. Approx every 8 hours (although not every 8 hours and sometimes 2 in a row) at exactly the same time I see the following errors Does anyone have any suggestions / ideas ? Thanks Ruairi Sep 24 11:56:30 asterisk01 -- MARK -- Sep 24 11:59:02 asterisk01 ntpd[3615]: synchronized to 172.16.22.60, stratum 4 Sep 24 11:59:28 asterisk01 kernel: wcte12xp: NMF workaround on! Sep 24 11:59:28 asterisk01 kernel: wcte12xp: Setting yellow alarm Sep 24 11:59:29 asterisk01 kernel: wcte12xp: NMF workaround off! Sep 24 11:59:34 asterisk01 kernel: wcte12xp: Clearing yellow alarm Sep 24 12:16:30 asterisk01 -- MARK -- Sep 24 12:17:01 asterisk01 /USR/SBIN/CRON[8325]: (root) CMD ( cd / run-parts --report /etc/cron.hourly) Sep 24 12:36:31 asterisk01 -- MARK -- Sep 24 12:56:31 asterisk01 -- MARK -- Sep 24 12:59:28 asterisk01 kernel: wcte12xp: NMF workaround on! Sep 24 12:59:28 asterisk01 kernel: wcte12xp: Setting yellow alarm Sep 24 12:59:28 asterisk01 kernel: wcte12xp: NMF workaround off! Sep 24 12:59:33 asterisk01 kernel: wcte12xp: Clearing yellow alarm Sep 24 13:16:31 asterisk01 -- MARK -- The siemens see a F5413 Spur error... A more detailed asterisk log shows [Sep 24 12:59:28] WARNING[6374] chan_zap.c: Detected alarm on channel 2: Red Alarm [Sep 24 12:59:28] WARNING[6374] chan_zap.c: Unable to disable echo cancellation on channel 2 [Sep 24 12:59:28] WARNING[6374] chan_zap.c: Detected alarm on channel 3: Red Alarm [Sep 24 12:59:28] WARNING[6374] chan_zap.c: Unable to disable echo cancellation on channel 3 [Sep 24 12:59:28] WARNING[6374] chan_zap.c: Detected alarm on channel 4: Red Alarm [Sep 24 12:59:28] WARNING[6374] chan_zap.c: Unable to disable echo cancellation on channel 4 [Sep 24 12:59:28] WARNING[6374] chan_zap.c: Detected alarm on channel 5: Red Alarm [Sep 24 12:59:28] WARNING[6374] chan_zap.c: Unable to disable echo cancellation on channel 5 [Sep 24 12:59:28] WARNING[6374] chan_zap.c: Detected alarm on channel 6: Red Alarm [Sep 24 12:59:28] WARNING[6374] chan_zap.c: Unable to disable echo cancellation on channel 6 [Sep 24 12:59:28] WARNING[6374] chan_zap.c: Detected alarm on channel 7: Red Alarm [Sep 24 12:59:28] WARNING[6374] chan_zap.c: Unable to disable echo cancellation on channel 7 [Sep 24 12:59:28] WARNING[6374] chan_zap.c: Detected alarm on channel 8: Red Alarm [Sep 24 12:59:28] WARNING[6374] chan_zap.c: Unable to disable echo cancellation on channel 8 [Sep 24 12:59:28] WARNING[6374] chan_zap.c: Detected alarm on channel 9: Red Alarm [Sep 24 12:59:28] WARNING[6374] chan_zap.c: Unable to disable echo cancellation on channel 9 [Sep 24 12:59:28] WARNING[6374] chan_zap.c: Detected alarm on channel 10: Red Alarm [Sep 24 12:59:28] WARNING[6374] chan_zap.c: Unable to disable echo cancellation on channel 10 [Sep 24 12:59:28] WARNING[6374] chan_zap.c: Detected alarm on channel 11: Red Alarm [Sep 24 12:59:28] WARNING[6374] chan_zap.c: Unable to disable echo cancellation on channel 11 [Sep 24 12:59:28] WARNING[6374] chan_zap.c: Detected alarm on channel 12: Red Alarm [Sep 24 12:59:28] WARNING[6374] chan_zap.c: Unable to disable echo cancellation on channel 12 [Sep 24 12:59:28] WARNING[6374] chan_zap.c: Detected alarm on channel 13: Red Alarm [Sep 24 12:59:28] WARNING[6374] chan_zap.c: Unable to disable echo cancellation on channel 13 [Sep 24 12:59:28] WARNING[6374] chan_zap.c: Detected alarm on channel 14: Red Alarm [Sep 24 12:59:28] WARNING[6374] chan_zap.c: Unable to disable echo cancellation on channel 14 [Sep 24 12:59:28] WARNING[6374] chan_zap.c: Detected alarm on channel 15: Red Alarm [Sep 24 12:59:28] WARNING[6374] chan_zap.c: Unable to disable echo cancellation on channel 15 [Sep 24 12:59:28] WARNING[6374] chan_zap.c: Detected alarm on channel 17: Red Alarm [Sep 24 12:59:28] WARNING[6374] chan_zap.c: Unable to disable echo cancellation on channel 17 [Sep 24 12:59:28] WARNING[6374] chan_zap.c: Detected alarm on channel 18: Red Alarm [Sep 24 12:59:28] WARNING[6374] chan_zap.c: Unable to disable echo cancellation on channel 18 [Sep 24 12:59:28] WARNING[6374] chan_zap.c: Detected alarm on channel 19: Red Alarm [Sep 24 12:59:28] WARNING[6374] chan_zap.c: Unable to disable echo cancellation on channel 19 [Sep 24 12:59:28] WARNING[6374] chan_zap.c: Detected alarm on channel 20: Red Alarm [Sep 24 12:59:28] WARNING[6374] chan_zap.c: Unable to disable echo cancellation on channel 20 [Sep 24 12:59:28] WARNING[6374] chan_zap.c: Detected alarm on channel 21: Red Alarm [Sep 24 12:59:28] WARNING[6374] chan_zap.c: Unable to disable echo cancellation on channel 21 [Sep 24 12:59:28] WARNING[6374] chan_zap.c: Detected alarm on channel 22: Red Alarm [Sep 24 12:59:28] WARNING[6374] chan_zap.c: Unable to disable echo cancellation on channel 22 [Sep 24 12:59:28] WARNING[6374] chan_zap.c: Detected alarm on channel 23: Red Alarm [Sep 24 12:59:28] WARNING[6374]
Re: [asterisk-users] Problems Connecting Two Asterisk Installs Via ISDN PRI Cards
On 9/24/07, Brian Alexander [EMAIL PROTECTED] wrote: At this point I do not think the problem is the wiring. What else should I try? Is this the latest Zaptel? Is it 1.4 or 1.2? I may have missed it in a previous message if you mentioned it. I've got a setup where I have a 2 port PRI on our main server. One port goes to telco and the other goes to a test lab asterisk system. Sometimes, I have noticed that I screw the test system up so bad that I have to reboot the main system, and unplug the PRI before doing so. It seems like alarms get hung up or something. This is using a 2 port Sangome card and a 1 port Sangoma card. I can't remember if I have this problem using the 1 port Digium. I'm sure you've already tried rebooting, but if you haven't, see what happens. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Lacy Moore Somewhere I wish I wasn't ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CallerID problem Asterisk 1.4.2
try ${CALLERID(all)}
Peter Kranz wrote:
When receiving inbound calls from a Vonage Softphone extension, I'm unable
to view/maniupulate calledid data. but it shows up in the CDR records and on
called handsets.. any ideas?
exten = asda,n,NoOp(callerID is ${CALLERID})
exten = asda,n,NoOp(CallerID is ${CALLERIDNAME})
exten = asda,n,NoOp(CallerID is ${CALLERIDNUM})
-- Executing [EMAIL PROTECTED]:2] Wait(SIP/asd1-086775b8, 1) in new
stack
-- Executing [EMAIL PROTECTED]:3] NoOp(SIP/asd-086775b8, callerID is
) in
new stack
-- Executing [EMAIL PROTECTED]:4] NoOp(SIP/asd-086775b8, CallerID is
) in
new stack
-- Executing [EMAIL PROTECTED]:5] NoOp(SIP/asd-086775b8, CallerID is
) in
new stack
But CDR data is there.
WIRELESS CALLER
asd,asd,100,pstn-in,SIP/asd-086775b8,SIP/peter5-08689458,Dial
,SIP/peterS
IP/peter2SIP/peter3SIP/peter4SIP/peter5|25|m,2007-09-24
11:19:27,2007-09-24 11:19:27,2007-09-24 11:19:44,17,17,ANSWER
ED,DOCUMENTATION,,1190657967.4165,
Peter Kranz
Founder/CEO - Unwired Ltd
www.UnwiredLtd.com
Desk: 510-868-1614 x100
Mobile: 510-207-
[EMAIL PROTECTED]
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Re: [asterisk-users] CallerID problem Asterisk 1.4.2
Those variables were deprecated in 1.2 and removed in 1.4. You should
read both the 1.2 and 1.4 UPGRADE.txt files. Also read README.variables.
Peter Kranz wrote:
When receiving inbound calls from a Vonage Softphone extension, I'm unable
to view/maniupulate calledid data. but it shows up in the CDR records and on
called handsets.. any ideas?
exten = asda,n,NoOp(callerID is ${CALLERID})
exten = asda,n,NoOp(CallerID is ${CALLERIDNAME})
exten = asda,n,NoOp(CallerID is ${CALLERIDNUM})
-- Executing [EMAIL PROTECTED]:2] Wait(SIP/asd1-086775b8, 1) in new
stack
-- Executing [EMAIL PROTECTED]:3] NoOp(SIP/asd-086775b8, callerID is
) in
new stack
-- Executing [EMAIL PROTECTED]:4] NoOp(SIP/asd-086775b8, CallerID is
) in
new stack
-- Executing [EMAIL PROTECTED]:5] NoOp(SIP/asd-086775b8, CallerID is
) in
new stack
But CDR data is there.
WIRELESS CALLER
asd,asd,100,pstn-in,SIP/asd-086775b8,SIP/peter5-08689458,Dial
,SIP/peterS
IP/peter2SIP/peter3SIP/peter4SIP/peter5|25|m,2007-09-24
11:19:27,2007-09-24 11:19:27,2007-09-24 11:19:44,17,17,ANSWER
ED,DOCUMENTATION,,1190657967.4165,
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Re: [asterisk-users] CallerID problem Asterisk 1.4.2
Peter Kranz wrote:
When receiving inbound calls from a Vonage Softphone extension, I'm unable
to view/maniupulate calledid data. but it shows up in the CDR records and on
called handsets.. any ideas?
exten = asda,n,NoOp(callerID is ${CALLERID})
exten = asda,n,NoOp(CallerID is ${CALLERIDNAME})
exten = asda,n,NoOp(CallerID is ${CALLERIDNUM})
read the UPGRADE.txt, it mentions that callerid is standardized as
basically, you need to change to CALLERID(num) or (name) or etc...
you should be able to do a 'core show function callerid' on the CLI
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Re: [asterisk-users] Anyone use the Linksys phones?
I'm wondering how the speakerphone on the 962 compares to the Cisco 7960? I found that the SPA-941 did not work well in noisy situations, unlike the Cisco 7960 which performs flawlessly. I find that the automatic gain circuit with regards to how it functions with speakerphone VOX on the Cisco 7960 is far superior to the SPA-941. However, I have not a 962 to compare with... Anyone have any comparisons/opinions?? What I'm most interested in is that with the Cisco, if it's in a noisy situation, like say lots of fans and HVAC running, the person on the other can get their voice through, whereas the SPA-941 speakerphone seems to lack the ability to distinguish a constant level of background noise from audio. This all has to do with the VOX capabilities, but more so in the analog section of the device... This isn't a software/firmware issue. Craig Guy wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Bagnall Sent: Monday, 24 September 2007 6:15 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Anyone use the Linksys phones? Is anyone out there using any of the newer linksys phones since Cisco took over? I am more specifically looking at the spa-941 942's. Just curious about call quality, programability, and functionality with asterisk. I've replaced the Cisco 7960 on my desk with the colour SPA-962. It's about half the buy price of the Cisco, takes less desk space, has more features, and a vastly superior screen. -- I'd like to second the SPA962 - I've deployed a couple of them now and they're great, clients get a kick out of sticking the company logo in colour on the screen and as of fw 5.1.15 the SPA932 sidecar supports asterisk for BLF and speed dial. They're also supposed to to support RSS for stock ticker type scrollies but haven't played with this yet. The only nasty thing I've found is that whenever the handsets resync they reboot even if no settings have changed. When this occurs anything connected to the phones second Ethernet port will drop connection for a few seconds. Craig ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users signature.asc Description: OpenPGP digital signature ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DTMF dropping digits
We have a Te410P with 3 Telco T1's (D4 SF ) with DID's (non-PRI). ANI DNIS is received in-band DTMF in a format such as *7145551212*8002* What happens when there are 30 or more calls, asterisk might see is DNIS = 802 or ANI = 4551212 for examples, where parts of the numbers are dropped. All the traffic arrives into a simple IVR script where a message is played. We are using Asterisk 1.2 and Server is 2.8 Dual Xeon SuperMicro with 2 GB RAM. Any clues what I can do to fix this? Bart ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TDM2400 answer detection
Hello All I have a TDM2400 card with 4 FXO, and the the following problem: This card answered all the calls, but for the caller, the call is ringing and I don't hear nothing when it has picked up. Here is piece of my log: Thanks for any help. == Starting post polarity CID detection on channel 21 -- Starting simple switch on 'Zap/21-1' -- Executing [EMAIL PROTECTED]:1] Answer(Zap/21-1, ) in new stack -- Executing [EMAIL PROTECTED]:2] Dial(Zap/21-1, SIP/ramal01SIP/ramal02SIP/ramal03|30|tT|r) in new stack -- Called ramal01 -- Called ramal02 -- Called ramal03 -- SIP/ramal03-0070e020 is ringing -- SIP/ramal01-006fd4f0 is ringing -- SIP/ramal02-00705d70 is ringing -- SIP/ramal01-006fd4f0 answered Zap/21-1 == Spawn extension (entrada, s, 2) exited non-zero on 'Zap/21-1' -- Hungup 'Zap/21-1' Asterisk 1.4.11 / Zaptel 1.4.5.1 Regards, McCoy Silva ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk as Media Server
Hi Guys, I want to configure asterisk to act as media server on my network. I have one specific situation descibed bellow. Collect Calls 1. Subscriber A call to subscriber B 2. Gateway in A side, send this call to media server (asterisk) and the asterisk send the call to subscriber B 3. When B answer the call, MS should play prompt1 for B and prompt2 for A (prompts are differents but with same duration). 4, I will give a free time for this call during the time of prompt1 message How can I do this ? Thanks. -- Frederico Madeira [EMAIL PROTECTED] www.madeira.eng.br ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk with multi-line appearence? How?
Hi, anybody can give me instruction on how to setup multi-line appearence in Asterisk. I have a Polycom soundpoint IP650 phone and want to provision multi-line for it. Thanks ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Extensions Configuration
Hi all, I am building out a new platform and I need help with a couple of items. I need to have an extension 101 that is public (on business cards, in the directory, etc...) however I want this extension to exist as a hunt group with a ring all strategy so two phones (107 which is the private extension for the 101 user is run, and the 102 extension). The 107 extension should not have a separate voicemail and when the user at 107 presses the messages button they need to log into the 101 mailbox. When 107 dials other users internally it should show the callerid as 101. What is the best way to configure asterisk to to this? Second question, for the hunt groups I want to change the callerid display for incoming calls so the phone displays Boss's Line:123456789, but I want to make sure that when the user redials via the phone directory the number 123456789 is dialed directly. How do I change the caller id display for inbound calls and still have the directory work properly? Thanks in advance, Max ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CID spill after second ring
I have an Asterisk 1.4 setup behind a TDM800P with 5 incoming lines, which happen to be inside my university's PBX. I have a problem wherein the university's PBX is apparently sending its CID spill after the second ring for a certain class of incoming calls (those from outside the PBX). I've temporarily hacked around this with a goto in chan_zap.c to restart CID processing if a good result wasn't obtained, but would it be possible to get this sort of workaround (with a better max retry safety count) integrated into the mainline code? I'd hate to think my successor would need to keep patching code to have CID work in our environment. jeff -- Jeff Bachtel ([EMAIL PROTECTED],TAMU) http://www.cepheid.org/~jeff The sciences, each straining in [finger [EMAIL PROTECTED] for PGP key] its own direction, have hitherto harmed us little; - HPL, TCoC ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Backports to 1.2.14 of 1.4.0 app_queue features.
Sorry to drag up an old thread, but the backport of ringinuse is a godsend for those of use stuck using asterisk 1.2 (trixbox 2.2). Many thanks, Gavin GTG On 1/21/07, Gavin Hamill [EMAIL PROTECTED] wrote: Nothing much to be said.. I backported ringinuse, autofill and the QueueLog application from 1.4.0 to 1.2.14. Any or all may be applied - order doesn't matter. They have received minimal testing but appear to function correctly. As always with these things, don't blame me if they connect your callers to a phonesex line, etc. http://bum.net/patches/ Cheers, Gavin. ___ --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 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Yikes! Polycom 501 chokes on BootRom 4.0.0?
I was progressively upgrading this phone from 3.1.2 to 3.2.3, then to 4.0.0. v3.2.3 worked fine, but when I went to 4.0.0 (Even adding the more specific 2345-11500-040.bootrom.ld), it won't run, and just keeps rebooting. Now I've got a really nice doorstop unless someone knows how to get out of this predicament. Help! 0925003705|cfg |3|00|Beginning to provision phone 0925003705|dns |3|00|DNS lookup for 'somedomain.com'(66.16.26.106) TTL=83485 0925003705|copy |3|00|'ftp://someuser:[EMAIL PROTECTED]/2345-11500-040.bootrom.ld' from 'somedomain.com(66.16.26.106)' 0925003706|cfg |3|00|Image 2345-11500-040.bootrom.ld has not changed 0925003706|copy |3|00|Download of '2345-11500-040.bootrom.ld' succeeded on attempt 1 (addr 1 of 1) 0925003706|cfg |3|00|Downloaded bootROM is identical to current version 4.0.0 0925003706|copy |3|00|'ftp://someuser:[EMAIL PROTECTED]/0004f210.cfg' from 'somedomain.com(66.16.26.106)' 0925003707|copy |3|00|Download of '0004f210.cfg' succeeded on attempt 1 (addr 1 of 1) 0925003708|cfg |5|00|Could not get the list of CONFIG_FILES 0925003708|cfg |5|00|Could not get the list of MISC_FILES 0925003709|cfg |5|00|Couldn't get parameter APP_FILE_PATH 0925003709|cfg |3|00|Unspecified error occured with downloaded application image 0925003709|app1 |6|00|Error in saving application. 0925003709|app1 |6|00|Uploading boot log, time is TUE SEP 25 00:37:10 2007 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Yikes! Polycom 501 chokes on BootRom 4.0.0?
I read something the other day about 4.0.0 being not-quite-right. PaulH On Mon, 2007-09-24 at 19:49 -0500, Doug wrote: I was progressively upgrading this phone from 3.1.2 to 3.2.3, then to 4.0.0. v3.2.3 worked fine, but when I went to 4.0.0 (Even adding the more specific 2345-11500-040.bootrom.ld), it won't run, and just keeps rebooting. Now I've got a really nice doorstop unless someone knows how to get out of this predicament. Help! 0925003705|cfg |3|00|Beginning to provision phone 0925003705|dns |3|00|DNS lookup for 'somedomain.com'(66.16.26.106) TTL=83485 0925003705|copy |3|00|'ftp://someuser:[EMAIL PROTECTED]/2345-11500-040.bootrom.ld' from 'somedomain.com(66.16.26.106)' 0925003706|cfg |3|00|Image 2345-11500-040.bootrom.ld has not changed 0925003706|copy |3|00|Download of '2345-11500-040.bootrom.ld' succeeded on attempt 1 (addr 1 of 1) 0925003706|cfg |3|00|Downloaded bootROM is identical to current version 4.0.0 0925003706|copy |3|00|'ftp://someuser:[EMAIL PROTECTED]/0004f210.cfg' from 'somedomain.com(66.16.26.106)' 0925003707|copy |3|00|Download of '0004f210.cfg' succeeded on attempt 1 (addr 1 of 1) 0925003708|cfg |5|00|Could not get the list of CONFIG_FILES 0925003708|cfg |5|00|Could not get the list of MISC_FILES 0925003709|cfg |5|00|Couldn't get parameter APP_FILE_PATH 0925003709|cfg |3|00|Unspecified error occured with downloaded application image 0925003709|app1 |6|00|Error in saving application. 0925003709|app1 |6|00|Uploading boot log, time is TUE SEP 25 00:37:10 2007 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ExternNotify Voicemail
I have googled and can seem to find the answer to this one Does anyone here have experience with externnotify in voicemail.conf? The sample states that it will run when a message is delivered and retrieved. Does asterisk pass any arguments to the script? Thanks. Forrest Beck [EMAIL PROTECTED] www.shift8.biz ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Yikes! Polycom 501 chokes on BootRom 4.0.0?
At 20:48 9/24/2007, Paul Hales wrote: I read something the other day about 4.0.0 being not-quite-right. Hmmm. Should I downgrade? If so, what versions? BootRom? sip.ld? etc? PaulH On Mon, 2007-09-24 at 19:49 -0500, Doug wrote: I was progressively upgrading this phone from 3.1.2 to 3.2.3, then to 4.0.0. v3.2.3 worked fine, but when I went to 4.0.0 (Even adding the more specific 2345-11500-040.bootrom.ld), it won't run, and just keeps rebooting. Now I've got a really nice doorstop unless someone knows how to get out of this predicament. Help! 0925003705|cfg |3|00|Beginning to provision phone 0925003705|dns |3|00|DNS lookup for 'somedomain.com'(66.16.26.106) TTL=83485 0925003705|copy |3|00|'ftp://someuser:[EMAIL PROTECTED]/2345-11500-040.bootrom.ld' from 'somedomain.com(66.16.26.106)' 0925003706|cfg |3|00|Image 2345-11500-040.bootrom.ld has not changed 0925003706|copy |3|00|Download of '2345-11500-040.bootrom.ld' succeeded on attempt 1 (addr 1 of 1) 0925003706|cfg |3|00|Downloaded bootROM is identical to current version 4.0.0 0925003706|copy |3|00|'ftp://someuser:[EMAIL PROTECTED]/0004f210.cfg' from 'somedomain.com(66.16.26.106)' 0925003707|copy |3|00|Download of '0004f210.cfg' succeeded on attempt 1 (addr 1 of 1) 0925003708|cfg |5|00|Could not get the list of CONFIG_FILES 0925003708|cfg |5|00|Could not get the list of MISC_FILES 0925003709|cfg |5|00|Couldn't get parameter APP_FILE_PATH 0925003709|cfg |3|00|Unspecified error occured with downloaded application image 0925003709|app1 |6|00|Error in saving application. 0925003709|app1 |6|00|Uploading boot log, time is TUE SEP 25 00:37:10 2007 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Analog Telephone Adapter
Hai, Is the Asterisk supports PMC-Sierra Analogue Telephone Adapter? Thanks Regards Bincy K Philip ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] prepaid application recommendation
a2billing so far seems to be quite comprehensive compared to the other freeware asterisk-based billing solutions available out there. We are building our own billing solution(due to the very peculiar requirements, one of which is to bill the callee, rather than the caller). We are achieving this so far, using AGI scripts, tho we plan to migrate to Asterisk Manager APIs soon. I haven't been able to go thru a2billing in detail(just skimmed thru the code n sample dialplans). It seems , if I am not mistaken, in a2billing every call invokes an AGI script. So this sounds a lil inefficient, where db connections, data structures etc, are created every time. Any experiences on the performance of both a2billing and AstMan API based solutions? cheerz - Ben. Sarfaraz Chougule wrote: I would recomend using Areski's billing solution : http://www.areski.net/a2billing On 9/21/07, *Rilawich Ango* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi all, I am looking for a prepaid application. I found that there are many applications in the page http://www.voip-info.org/wiki/view/Asterisk+Prepaid+Applications. Anyone recommendation among them? ango ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- With Best Regards, ** Sarfaraz Chougule ** ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users EMAIL DISCLAIMER : This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. Any unauthorised distribution or copying is strictly prohibited. If you receive this transmission in error, please notify the sender by reply email and then destroy the message. Opinions, conclusions and other information in this message that do not relate to official business of Mascon shall be understood to be neither given nor endorsed by Mascon. Any information contained in this email, when addressed to Mascon clients is subject to the terms and conditions in governing client contract. Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we can not guarantee that any email or attachment is free from computer viruses and you are strongly advised to undertake your own anti-virus precautions. Mascon grants no warranties regarding performance, use or quality of any e-mail or attachment and undertakes no liability for loss or damage, howsoever caused. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Yikes! Polycom 501 chokes on BootRom 4.0.0?
I am running sip 2.2 with an older bootrom, and the phone is running fine. PaulH On Mon, 2007-09-24 at 22:32 -0500, Doug wrote: At 20:48 9/24/2007, Paul Hales wrote: I read something the other day about 4.0.0 being not-quite-right. Hmmm. Should I downgrade? If so, what versions? BootRom? sip.ld? etc? PaulH On Mon, 2007-09-24 at 19:49 -0500, Doug wrote: I was progressively upgrading this phone from 3.1.2 to 3.2.3, then to 4.0.0. v3.2.3 worked fine, but when I went to 4.0.0 (Even adding the more specific 2345-11500-040.bootrom.ld), it won't run, and just keeps rebooting. Now I've got a really nice doorstop unless someone knows how to get out of this predicament. Help! 0925003705|cfg |3|00|Beginning to provision phone 0925003705|dns |3|00|DNS lookup for 'somedomain.com'(66.16.26.106) TTL=83485 0925003705|copy |3|00|'ftp://someuser:[EMAIL PROTECTED]/2345-11500-040.bootrom.ld' from 'somedomain.com(66.16.26.106)' 0925003706|cfg |3|00|Image 2345-11500-040.bootrom.ld has not changed 0925003706|copy |3|00|Download of '2345-11500-040.bootrom.ld' succeeded on attempt 1 (addr 1 of 1) 0925003706|cfg |3|00|Downloaded bootROM is identical to current version 4.0.0 0925003706|copy |3|00|'ftp://someuser:[EMAIL PROTECTED]/0004f210.cfg' from 'somedomain.com(66.16.26.106)' 0925003707|copy |3|00|Download of '0004f210.cfg' succeeded on attempt 1 (addr 1 of 1) 0925003708|cfg |5|00|Could not get the list of CONFIG_FILES 0925003708|cfg |5|00|Could not get the list of MISC_FILES 0925003709|cfg |5|00|Couldn't get parameter APP_FILE_PATH 0925003709|cfg |3|00|Unspecified error occured with downloaded application image 0925003709|app1 |6|00|Error in saving application. 0925003709|app1 |6|00|Uploading boot log, time is TUE SEP 25 00:37:10 2007 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Completing my Configuration
Hallo Group, I have basically set up a small asterisk system, which ahs 4 peers: * registers at 2 Sipgates * 2 hardware phones connected to it Both Hardware phones can phone outwards(cheaper sipgate is selected with dialplan) Calls from both sipgates make my hardware phones ring But here comes the challenges: Is it possible to configure asterisk in such a way that in the phone: * there are names instead of numbers in my hardware phone displayed * The Ringtone is different for special call numbers * it is displayed, in which sipgate the call came from * using an extension in my call number redirects the call just to one sip phone ? And What about Asterisk web server: I was told you can sue it to configure asterisk via web. I turned it on an connected to it, but I can only read 404 Object not found Asterisk Webserver Whats wrong ? Thank you very much for your inspirations! -- Ist Ihr Browser Vista-kompatibel? Jetzt die neuesten Browser-Versionen downloaden: http://www.gmx.net/de/go/browser ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
