[asterisk-users] Audio lag on SIP connections...
Had something recently on 2 separate sites where there was a lot of audio lag on a call - and by a lot I'm talking 5 seconds or so. Two different sites, but the setup is similar: SIPphone A internet Asterisk IAX/Internet Asterisk - SIPphoneB If the phone called extensions local to it's asterisk there was no lag, but if it called out (separate IAX trunk) or to the SIP phone on another asterisk box then there was lag... Googling doesn't find much (other than others complaining about something similar), so does anyone have any insight? The lag issue was cured by rebooting the ADSL router at SIP phone A's end. The other site that experienced this has an identical router - Draytek 2820, and that also was cured by rebooting, but I'm keen to get to the bottom of it. Two things going through my mind - One - DNS. Could something be doing a DNS lookup which is being stalled by the router (routers are the DNS servers in this case) causing a delay on the audio? Or could the routers themselves be queueing up 5 seconds of traffic? (Unlikely, but I've seen Drayteks do other weird things) Any clues, suggestions, etc. welcome! Thanks, Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with building dahdi-linux RPM
I have some OpenVOX A1200p cards, and driver for them so far works only with dahdi-2.0.0 Sorry, looks like i don't understand, how to correctly rebuild driver: rpmbuild --rebuild http://dl.atrpms.net/all/dahdi-linux-2.1.0.3-59.src.rpm skipped Wrote: /usr/src/redhat/RPMS/i386/dahdi-linux-kmdl-2.6.18-92.1.22.el5-2.1.0.3-59.RHL5.i386.rpm Only kmdl module was wroten, no dahdi-linux and dahdi-linux-devel. Can you explain, what should i do? :-) ___ -- 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] Contact lookup
On Wednesday, February 4, 2009, D Tucny wrote:
I use a slight variant of this...
exten =
s,n,Set(CALLERID(name)=${IF(${ISNULL(${DB(cidname/${CALLERID(num)})})}?Unknown:${DB(cidname/${CALLERID(num)})})})
exten = s,n,NoOp(Caller ID name mapped to ${CALLERID(name)})
Basically the same as yours above (including substitution of Unknown when not
found), but, all on one line...
Once I'd got a handle on it, the task seems almost trivial. Here's
what I've got:
exten = s,1,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})})
I don't need unknown because all my handsets show something similar
(e.g. Unavailable name) by default.
I've been looking into changing it recently such that where I don't
have the name I can substitute something more useful than Unknown,
such as the site, or for external calls, the
country/province/state/city/type/telco/etc, though that won't be in
astdb due to the current 100s of thousands of rows...
That might be a good AGI project...
FWIW, I had trouble loading astdb with the contact list. I dumped the
caller list from my old PBX to a CSV file and then parsed it to give
one line of the following form for each contact:
/usr/sbin/asterisk -rx 'database put cidname 01234567 Caller Name'
I ran the script and it appeared to go well. However, when did
database show cidname at the * CLI prompt, the family and key were
in a right mess. For example, the entry above might have appeared as:
34567: Caller Name
I suspect that it would still have worked since database show
cidname listed these entries but I didn't take chances. There were
only thirty or so entries, so I cleared the cidname family and copied
each database put command from a terminal window and pasted to the *
CLI prompt.
So, this is now sorted for me and I've learned a thing or two about
astdb in the process.
Thanks all.
--
Geoff
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[asterisk-users] BerkeleyTIP Feb 7 Sat Global Meeting - Ekiga3, Asterisk, KDE, GPGPU, Debian Edu, GStreamer
** Great talks this meeting: (live on video) ** Ekiga3, Asterisk, GPGPU, GStreamer, Debian Edu, HowTo Present KDE at meetings http://sites.google.com/site/berkeleytip/ Join from anywhere via VOIP conference, with the friendly, educational, productive, BerkeleyTIP people. :) Join the #berkeleytip freenode.net IRC channel for help getting your VOIP working. http://groups.google.com/group/BerkTIPGlobal/web/irc-voip Programming Party: Whatever you want to work on, or group VOIP technology. = * FEB 7 SCHEDULE (California PacificStandardTime = -8h GMT) * Time Activity Talks - 10 A Setup. Installfest begin. IRC VOIP online 11 Ekiga 3 VOIP HowTo - Live from India 12 N Asterisk Free Software Telephone System 1 P 2 GPGPU - General Purpose computing w/ Graphics Processing Units 3 4 Debian Edu - 100% in main 5 ** 5 Minute Lightning Talks ** GStreamer Multimedia Framework 6 HowTo Rock the Show with KDE 630 - Cleanup [Adjust for your local time: 10AM - 6:30PM PST = 1PM - 9:30PM Eastern = 6PM - 2:30AM GMT] - Ekiga 3 VOIP HowTo Install - Chaitanya Mehandru - 1h Live - Most distros only have version 2 The Asterisk Free Software Telephone System - Paul Charles Leddy - 1h23m http://nylug.org/meetings/index.shtml?20081000 GPGPU - General Purpose GPUs - John Stone - 1h43m http://www.archive.org/details/clug-28-10-2008-gpu-computing Debian Edu 100% in main - Holger Levsen - 1h https://penta.debconf.org/dc8_schedule/events/286.en.html GStreamer Multimedia Framework - Richard Spiers - 42m http://www.archive.org/details/clug-30-09-2008-gstreamer How to rock the show with KDE - Lydia Pintscher - 30m http://akademy.kde.org/conference/presentation/42.php - How to present the KDE project at a conference. ** Download the talk videos you want to see the day _before_ the meeting, so your internet connection is free to do VOIP, not consumed with the video download. :) = = JOIN THE BERKELEY-TIP MAILING LIST Join the mailing list, say Hi, introduce yourself, or just follow the discussions. Click Join this group on the right side of the page: http://groups.google.com/group/BerkTIPGlobal = = BERKELEY-TIP - MONTHLY GNU(LINUX) BSD FREE SW HW CULTURE MEETING BerkeleyTIP is a great monthly meeting about GNU(Linux), BSD all free SW HW Culture. Come learn about, install, use help produce some SW. :) http://sites.google.com/site/berkeleytip/ = = YOU GIVE A 5 MINUTE LIGHTNING TALK? Send an email to the group with the topic you want to talk about. We'll likely approve all talks about any of: GNU(Linux) BSD or any Free SW HW or Culture subject to time constraints. :) = = EDUCATIONAL OUTREACH - SPRING 2009 - COLLEGES UNIVERSITIES Join our effort to bring the BerkeleyTIP monthly meetings to local in-person gatherings at Colleges Universities everywhere.:) You are encouraged to do any of these you want to: 1) Organize a local meeting at a college or university. - A WIFI cafe, or classroom, is a great place to meet. :) 2) Invite attendees by email - you can forward, and add to, this email. 3) Put up meeting announcement posters where appropriate - - see the BTIP site for the current poster 8.5x11 inch ODF file. DO: Join the BTIP mailing list let's discuss share ideas about how to make this a success. :) = = RECORD YOUR LOCAL MEETINGS' TALKS It's easy. Bring a video camera, tripod, lapel pin microphone, microphone cable. Or, just put your camera within about 5-10 feet from the speaker. Put your video online - the internet archive is a great place. Be sure to send me a link. :)I'll try to schedule all newly recorded videos into the next BerkeleyTIP meeting. :) = = FORWARD THIS ANNOUNCEMENT EMAIL WHERE APPROPRIATE You are invited to forward this email wherever appropriate. Pass the word on, encourage other people to attend the meeting, encourage the growth, improvement strengthening of all Public Property, Community, Free as in Freedom FreeSpeech software. :) ___ -- 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] escaping regular expression
Hi!
I am going nuts using REGEXP. I just want to verify if a variable
contains a valid +E164 phone number.
These, the the pattern is ^\+[0-9]+
First I tried:
Set(pattern=^\+[0-9]+);
if (${REGEX(${pattern} ${${var}})})
but that does not work, the backslash is removed, as seen in the log file:
func_strings.c: FUNCTION REGEX (^+[0-9]+)(+4312345
http://www.adaanumber.com/)
So, meanwhile I tried to escape the backslash. I tried:
Set(pattern=^\\+[0-9]+);
Set(pattern=^\\\+[0-9]+);
Set(pattern=^+[0-9]+);
But always the same result:
func_strings.c: FUNCTION REGEX (^+[0-9]+)(+4312345)
How can I solve this problem?
Thanks
Klaus
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[asterisk-users] question on originate call
I have outgoing call files working. I am trying to get the manager to originate a call. My outgoing call file that works looks like: Channel: SIP/devcentos5x64_to_panel/mediaport SetVar: MEETME_PLAYFILE=/home/silentm/record/pc.1725.wav SetVar: agi_pa_recno=1725 Context: smvoice-dialout Extension: smvoice_single_mediaport Priority: 1 RetryTime: 2 WaitTime: 40 MaxRetries: 0 SetVar: SIPADDHEADER=Alert-Info: Ring Answer CallerID: Jerry Geis 204 317 SetVar: agi_port= SetVar: agi_extension= TESTING SetVar: agi_seconds_to_ring=40 SetVar: agi_dwc_record_num=43 The Manager API session looks like: 04-Feb-09 08:22 am asterisk_command() Action: Login 04-Feb-09 08:22 am asterisk_command() Username: MessageNet 04-Feb-09 08:22 am asterisk_command() Secret: X 04-Feb-09 08:22 am asterisk_command() Events: off 04-Feb-09 08:22 am DEBUG: Response: Success[CR ][LF ]Message: Authentication accepted[CR ][LF ][CR ][LF ] Action: Originate[CR ][LF ] Channel: SIP/devcentos5x64_to_panel/mediaport[CR ][LF ] Variable: MEETME_PLAYFILE=/home/silentm/record/pc.1725.wav[CR ][LF ] Variable: agi_pa_recno=1725[CR ][LF ] Context: smvoice-dialout[CR ][LF ] Exten: smvoice_single_mediaport[CR ][LF ] Priority: 1[CR ][LF ] Timeout: 40[CR ][LF ] Variable: SIPADDHEADER=Alert-Info: Ring Answer[CR ][LF ] Variable: agi_port=3[CR ][LF ] Variable: agi_extension= TESTING[CR ][LF ] Variable: agi_seconds_to_ring=40[CR ][LF ] Variable: agi_dwc_record_num=14[CR ][LF ] [CR ][LF ] 04-Feb-09 08:22 am DEBUG: Response: Error[CR ][LF ]Message: Originate failed[CR ][LF ][CR ][LF ] What am I missing to get the manager API to place the call instead of a call file? Thanks, Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] escaping regular expression
2009/2/4 Klaus Darilion klaus.mailingli...@pernau.at
Hi!
I am going nuts using REGEXP. I just want to verify if a variable
contains a valid +E164 phone number.
These, the the pattern is ^\+[0-9]+
First I tried:
Set(pattern=^\+[0-9]+);
if (${REGEX(${pattern} ${${var}})})
but that does not work, the backslash is removed, as seen in the log file:
func_strings.c: FUNCTION REGEX (^+[0-9]+)(+4312345
http://www.adaanumber.com/)
So, meanwhile I tried to escape the backslash. I tried:
Set(pattern=^\\+[0-9]+);
Set(pattern=^\\\+[0-9]+);
Set(pattern=^+[0-9]+);
But always the same result:
func_strings.c: FUNCTION REGEX (^+[0-9]+)(+4312345)
How can I solve this problem?
Try something like... pattern=^[+]\{1\}[0-9]+
d
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Re: [asterisk-users] escaping regular expression
D Tucny schrieb:
2009/2/4 Klaus Darilion klaus.mailingli...@pernau.at
mailto:klaus.mailingli...@pernau.at
Hi!
I am going nuts using REGEXP. I just want to verify if a variable
contains a valid +E164 phone number.
These, the the pattern is ^\+[0-9]+
First I tried:
Set(pattern=^\+[0-9]+);
if (${REGEX(${pattern} ${${var}})})
but that does not work, the backslash is removed, as seen in the log
file:
func_strings.c: FUNCTION REGEX (^+[0-9]+)(+4312345
http://www.adaanumber.com/)
So, meanwhile I tried to escape the backslash. I tried:
Set(pattern=^\\+[0-9]+);
Set(pattern=^\\\+[0-9]+);
Set(pattern=^+[0-9]+);
But always the same result:
func_strings.c: FUNCTION REGEX (^+[0-9]+)(+4312345)
How can I solve this problem?
Try something like... pattern=^[+]\{1\}[0-9]+
Are you sure? The \ should be in front of the +
klaus
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[asterisk-users] siemens hipath 4000
I am connecting to a siemens hipath 4000 with dahdi 2.1.0.4 and asterisk 1.4.23 using a Te210P card. the phone guy is saying that the lines are reporting always BUSY. however on my end the status shows OK. Anyone seen this? Is there something different about connecting PRI to siemens hipath? system.conf shows: loadzone=us defaultzone=us span=1,1,6,esf,b8zs bchan=1-5 dchan=24 chan_dahdi.conf: [channels] switchtype=national signalling=pri_cpe echocancel=yes echocancelwhenbridged=yes echotraining=400 callerid=asreceived context=smvoice-incoming group=1 channel = 1-5 Any ideas? Also - how do you get a card out of loop mode after you have used the dahdi_tool to select loop mode. there is no UNLOOP or anything. I have reset the machine and it still seems to be in loop mode. Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call parking
All, Quick question that hopefully someone out there will know the answer to... We were previously running Asterisk 1.4.(something) (I forget which one) on Debian. Due to an office move, I am temporarily routing our calls through an Ubuntu box that I have. It runs Asterisk 1.4.17-dfsg-2ubuntu1 (basically, what came with Ubuntu.) Here's the problem I am having: We are using Polycom 500's and 501's.. previously (on the Debian system), to park a call, we could transfer it to extension 7000 (we use 4-digit extensions.) Asterisk would read back the parking space number, then we complete the transfer. No problem. On this new system, Asterisk is not reading back the number. Instead, it simply starts playing hold music. If you complete the transfer, a show parkedcalls will show the call as parked (and you can retrieve it.) However, my users have no way of knowing where their calls are being parked. Anyone have any idea as to why it would stop reading back the parking location? I do have the digit sounds installed (in several formats, also.) No luck there. It's almost as if Asterisk is seeing it as a blind transfer instead of a supervised one. Oh, and I can set up a code for it in features.conf and dial that while on the phone (I set the feature code to *8) .. when I do that, it will read back the location and park the call. However, that feature doesn't seem to work for me for all calls (such as calls coming in via a queue, etc.) I'm sure it's something simple, but I've been pulling my hair out searching for anyone else having this problem and haven't had any luck. Any help would be appreciated. :) Jeremy -- Jeremy G. Gault, KD4NED Network Administrator WinWorld Corporation ___ -- 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] AOC-E pass through
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Klaus Darilion a écrit : Take a look at http://bugs.digium.com/view.php?id=7494 Thanks for the pointer; I'm already monitoring this issue, but there seems to be no progress on that, unfortunately. Unfortunately it is not yet included in Asterisk, as the patch is somehow a workaround (e.g. faking AOC-E based on last AOC-D). Here the telco is not sending AOC-D, just AOC-E. Nevertheless a customer of us uses it for some years now (Astersik 1.2) without any problems. regards klaus Jean-Denis Girard schrieb: Hi, I'd like to know what is the current situation with regard to AOC-E, when Asterisk is inserted between the telco and an existing PBX, using E1 / EuroISDN. Can Asterisk pass the AOC-E information received from the telco to the PBX, so that billing system still works? The system would be for a hotel, so breaking billing system is not possible. Thanks, - -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 50 10 40 / GSM: +689 79 75 27 -BEGIN PGP SIGNATURE- iEYEARECAAYFAkmJuusACgkQuu7Rv+oOo/iHggCghWlXnKBZ+plXZdiHQTM8kyIi QQsAn3+O2kq2jPpcoyMAcReXltDOnQ8t =uh9L -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] escaping regular expression
2009/2/4 Klaus Darilion klaus.mailingli...@pernau.at
D Tucny schrieb:
2009/2/4 Klaus Darilion klaus.mailingli...@pernau.at
mailto:klaus.mailingli...@pernau.at
Hi!
I am going nuts using REGEXP. I just want to verify if a variable
contains a valid +E164 phone number.
These, the the pattern is ^\+[0-9]+
First I tried:
Set(pattern=^\+[0-9]+);
if (${REGEX(${pattern} ${${var}})})
but that does not work, the backslash is removed, as seen in the log
file:
func_strings.c: FUNCTION REGEX (^+[0-9]+)(+4312345
http://www.adaanumber.com/)
So, meanwhile I tried to escape the backslash. I tried:
Set(pattern=^\\+[0-9]+);
Set(pattern=^\\\+[0-9]+);
Set(pattern=^+[0-9]+);
But always the same result:
func_strings.c: FUNCTION REGEX (^+[0-9]+)(+4312345)
How can I solve this problem?
Try something like... pattern=^[+]\{1\}[0-9]+
Are you sure? The \ should be in front of the +
Pretty sure...
exten = *56,1,NoOp(Starting regexp test)
exten = *56,n,Set(pattern=^[+]\{1\}[0-9]+)
exten = *56,n,Set(var=123456789)
exten = *56,n,NoOp(${IF(${REGEX(${pattern} ${var})}?Match:No
Match)}))
exten = *56,n,Set(var=+123456789)
exten = *56,n,NoOp(${IF(${REGEX(${pattern} ${var})}?Match:No
Match)}))
[Feb 4 23:49:21] DEBUG[20518] pbx.c: Launching 'NoOp'
[Feb 4 23:49:21] VERBOSE[20518] logger.c: -- Executing
[...@phonedefault:1] NoOp(SIP/*01-09bd8ff8, Starting regexp test) in
new stack
[Feb 4 23:49:21] DEBUG[20518] pbx.c: Launching 'Set'
[Feb 4 23:49:21] VERBOSE[20518] logger.c: -- Executing
[...@phonedefault:2] Set(SIP/*01-09bd8ff8, pattern=^[+]\{1\}[0-9]+) in
new stack
[Feb 4 23:49:21] DEBUG[20518] pbx.c: Launching 'Set'
[Feb 4 23:49:21] VERBOSE[20518] logger.c: -- Executing
[...@phonedefault:3] Set(SIP/*01-09bd8ff8, var=123456789) in new stack
[Feb 4 23:49:21] DEBUG[20518] func_strings.c: FUNCTION REGEX
(^[+]{1}[0-9]+)(123456789)
[Feb 4 23:49:21] DEBUG[20518] pbx.c: Function result is '0'
[Feb 4 23:49:21] DEBUG[20518] pbx.c: Function result is 'No Match'
[Feb 4 23:49:21] DEBUG[20518] pbx.c: Launching 'NoOp'
[Feb 4 23:49:21] VERBOSE[20518] logger.c: -- Executing
[...@phonedefault:4] NoOp(SIP/*01-09bd8ff8, No Match)) in new stack
[Feb 4 23:49:21] DEBUG[20518] pbx.c: Launching 'Set'
[Feb 4 23:49:21] VERBOSE[20518] logger.c: -- Executing
[...@phonedefault:5] Set(SIP/*01-09bd8ff8, var=+123456789) in new stack
[Feb 4 23:49:21] DEBUG[20518] func_strings.c: FUNCTION REGEX
(^[+]{1}[0-9]+)(+123456789)
[Feb 4 23:49:21] DEBUG[20518] pbx.c: Function result is '1'
[Feb 4 23:49:21] DEBUG[20518] pbx.c: Function result is 'Match'
[Feb 4 23:49:21] DEBUG[20518] pbx.c: Launching 'NoOp'
[Feb 4 23:49:21] VERBOSE[20518] logger.c: -- Executing
[...@phonedefault:6] NoOp(SIP/*01-09bd8ff8, Match)) in new stack
So, the \ is still stripped (ast_app_separate_args removes \), but, it
doesn't matter as the + is bracketed so it's not the first character after
the ^ and so regcomp doesn't fail...
d
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Re: [asterisk-users] siemens hipath 4000
Hello Jerry, I'm using asterisk-1.2.18 with Sangoma A104D interconnect with Siemens HiPath 4000 in Brazil and works fine, no problem. Please, look below my asterisk configurations for your help: zapata.conf [trunkgroups] [channels] context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes overlapdial=yes autofalltrought=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no ;Sangoma A104 port 3 [slot:2 bus:18 span:3] wanpipe3 HIPATH 4000 SIEMENS switchtype=qsig context=default group=2 signalling=pri_net channel =63-77,79-93 Best Regards Josué 2009/2/4 Jerry Geis ge...@pagestation.com: I am connecting to a siemens hipath 4000 with dahdi 2.1.0.4 and asterisk 1.4.23 using a Te210P card. the phone guy is saying that the lines are reporting always BUSY. however on my end the status shows OK. Anyone seen this? Is there something different about connecting PRI to siemens hipath? system.conf shows: loadzone=us defaultzone=us span=1,1,6,esf,b8zs bchan=1-5 dchan=24 chan_dahdi.conf: [channels] switchtype=national signalling=pri_cpe echocancel=yes echocancelwhenbridged=yes echotraining=400 callerid=asreceived context=smvoice-incoming group=1 channel = 1-5 Any ideas? Also - how do you get a card out of loop mode after you have used the dahdi_tool to select loop mode. there is no UNLOOP or anything. I have reset the machine and it still seems to be in loop mode. Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call parking
How is your features.conf set up? Do you have a Parking function in your dialplan? The answer that comes to mind is that you are somehow using parkandannounce instead of park and something is just mis-coded. In my shop, I have hints registered, so core show hints will tell me which lots are in use, but some here consider that a hack. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeremy G. Gault Sent: Wednesday, February 04, 2009 9:53 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Call parking All, Quick question that hopefully someone out there will know the answer to... We were previously running Asterisk 1.4.(something) (I forget which one) on Debian. Due to an office move, I am temporarily routing our calls through an Ubuntu box that I have. It runs Asterisk 1.4.17-dfsg-2ubuntu1 (basically, what came with Ubuntu.) Here's the problem I am having: We are using Polycom 500's and 501's.. previously (on the Debian system), to park a call, we could transfer it to extension 7000 (we use 4-digit extensions.) Asterisk would read back the parking space number, then we complete the transfer. No problem. On this new system, Asterisk is not reading back the number. Instead, it simply starts playing hold music. If you complete the transfer, a show parkedcalls will show the call as parked (and you can retrieve it.) However, my users have no way of knowing where their calls are being parked. Anyone have any idea as to why it would stop reading back the parking location? I do have the digit sounds installed (in several formats, also.) No luck there. It's almost as if Asterisk is seeing it as a blind transfer instead of a supervised one. Oh, and I can set up a code for it in features.conf and dial that while on the phone (I set the feature code to *8) .. when I do that, it will read back the location and park the call. However, that feature doesn't seem to work for me for all calls (such as calls coming in via a queue, etc.) I'm sure it's something simple, but I've been pulling my hair out searching for anyone else having this problem and haven't had any luck. Any help would be appreciated. :) Jeremy -- Jeremy G. Gault, KD4NED Network Administrator WinWorld Corporation ___ -- 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] T1, FoneBRIDGE, and dropped D-Channel
I hope someone can help me out with this issue. It has been dogging me for months and I can't seem to get it to go away. I have a Rhino Ceros box running Asterisk 1.4.21.2 connected via eth0 (nVidia MCP61 Ethernet) to a RedFone FoneBRIDGE2 dual-port with EC. The FB is the latest hardware rev and the latest firmware. I'm running the latest fonulator version and I'm running Zap-1.4.11 sourced from RedFone. Nothing else is on eth0. It is currently connected thru a dedicated switch to the FB and the secondary server although I've observed this problem when connected directly. There are two other eth cards, one for the internal network and one for the DMZ. My problem is that every now and then the D-Channel will drop which will terminate all calls in process. The D-channel will immediately come back up (usually within a second) but that doesn't do any good because the calls are gone by then and users are mad. The log entry at one of these events looks like this: [Feb 3 08:14:02] ERROR[26063] chan_zap.c: Write to 65 failed: Unknown error 500 [Feb 3 08:14:02] ERROR[26063] chan_zap.c: Short write: 0/15 (Unknown error 500) [Feb 3 08:14:02] WARNING[26063] chan_zap.c: Detected alarm on channel 1: Yellow Alarm (same message for other 22 channels) [Feb 3 08:14:02] NOTICE[2660] chan_zap.c: PRI got event: Alarm (4) on Primary D-channel of span 1 [Feb 3 08:14:02] WARNING[2660] chan_zap.c: No D-channels available! Using Primary channel 24 as D-channel anyway! [Feb 3 08:14:02] NOTICE[2662] chan_zap.c: Alarm cleared on channel 1 (same message for other 22 channels) [Feb 3 08:14:02] NOTICE[2660] chan_zap.c: PRI got event: No more alarm (5) on Primary D-channel of span 1 [Feb 3 08:14:02] NOTICE[2660] chan_zap.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 [Feb 3 08:14:02] NOTICE[2660] chan_zap.c: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 You'll notice the timestamps are all within the same 1-second interval which makes me think it is literally missing one packet and causing the drop. I'm sure the configs are fine as they've been reviewed by about 20 people and the system works most of the time. If the machine has been freshly started up, this happens about once every other day. The machine has currently been running for over 36 days and I'm seeing several per day now. ATT has run a stress test on the line from the CO to the smartjack and found no problems. The cable from the smartjack to the FoneBRIDGE is about 18 and I've tried a couple with no difference. I'm convinced this is interrupt related. When I initially commissioned this machine, the FB was connected to eth2 and I couldn't get it to link up with the CO at all. The D-Channel was flapping like crazy. I switched it to eth0 and it worked. You can see from my interrupts that the on-board and the add-in cards are clearly on different busses. CPU0 0: 3266196236IO-APIC-edge timer 1: 2IO-APIC-edge i8042 8: 3IO-APIC-edge rtc 9: 0 IO-APIC-level acpi 169: 207230361 IO-APIC-level ohci_hcd:usb1 177:5080313 IO-APIC-level sata_nv 185: 0 IO-APIC-level sata_nv 193:1632824 IO-APIC-level eth1 201: 39823124 IO-APIC-level eth2 225: 2565938694 PCI-MSI eth0 NMI: 0 LOC: 3266207768 ERR: 1 MIS: 0 So the fact that I couldn't link up when I was on one card and I could when I am on another (with no config changes... other than re-directing ztdynamic) leads me directly to this interrupt issue. Can anyone shed some light here? Has someone seen this before? If so, how did you solve it? Thanks! Jason -- This e-mail message, including any attachments, is only for the use of the intended recipient (s). The information contained may be confidential, in which case its disclosure or reproduction is strictly prohibited. If you are not the intended recipient, please return it immediately to its sender at the above address and delete it. ___ -- 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] Call parking
Danny, I have parkext set to 7000, parkpos set to 7060-7069, context is set to parkedcalls. In extensions.conf I just include = parkedcalls When I dial 7000 from my desk phone (which used to render a parking location and then play hold music), I get this on the CLI: -- Executing [7...@from-local-sip:1] Park(SIP/7411b-081e28b8, ) in new stack -- Started music on hold, class 'default', on SIP/7411b-081e28b8 == Parked SIP/7411b-081e28b8 on 7...@parkedcalls. Will timeout back to extension [from-local-sip] s, 1 in 3600 seconds -- Added extension '7060' priority 1 to parkedcalls == Spawn extension (from-local-sip, s, 1) exited KEEPALIVE on 'SIP/7411b-081e28b8' So, it seems it is using Park() but for some reason it just doesn't read back the location. Jeremy On Wed, Feb 4, 2009 at 11:08 AM, Danny Nicholas da...@debsinc.com wrote: How is your features.conf set up? Do you have a Parking function in your dialplan? The answer that comes to mind is that you are somehow using parkandannounce instead of park and something is just mis-coded. In my shop, I have hints registered, so core show hints will tell me which lots are in use, but some here consider that a hack. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jeremy G. Gault *Sent:* Wednesday, February 04, 2009 9:53 AM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] Call parking All, Quick question that hopefully someone out there will know the answer to... We were previously running Asterisk 1.4.(something) (I forget which one) on Debian. Due to an office move, I am temporarily routing our calls through an Ubuntu box that I have. It runs Asterisk 1.4.17-dfsg-2ubuntu1 (basically, what came with Ubuntu.) Here's the problem I am having: We are using Polycom 500's and 501's.. previously (on the Debian system), to park a call, we could transfer it to extension 7000 (we use 4-digit extensions.) Asterisk would read back the parking space number, then we complete the transfer. No problem. On this new system, Asterisk is not reading back the number. Instead, it simply starts playing hold music. If you complete the transfer, a show parkedcalls will show the call as parked (and you can retrieve it.) However, my users have no way of knowing where their calls are being parked. Anyone have any idea as to why it would stop reading back the parking location? I do have the digit sounds installed (in several formats, also.) No luck there. It's almost as if Asterisk is seeing it as a blind transfer instead of a supervised one. Oh, and I can set up a code for it in features.conf and dial that while on the phone (I set the feature code to *8) .. when I do that, it will read back the location and park the call. However, that feature doesn't seem to work for me for all calls (such as calls coming in via a queue, etc.) I'm sure it's something simple, but I've been pulling my hair out searching for anyone else having this problem and haven't had any luck. Any help would be appreciated. :) Jeremy -- Jeremy G. Gault, KD4NED Network Administrator WinWorld Corporation ___ -- 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 -- Jeremy G. Gault, KD4NED Network Administrator WinWorld Corporation ___ -- 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] Call parking
I think you need to use ParkAndAnnounce instead of Park to get the call back. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeremy G. Gault Sent: Wednesday, February 04, 2009 11:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call parking Danny, I have parkext set to 7000, parkpos set to 7060-7069, context is set to parkedcalls. In extensions.conf I just include = parkedcalls When I dial 7000 from my desk phone (which used to render a parking location and then play hold music), I get this on the CLI: -- Executing [7...@from-local-sip:1] Park(SIP/7411b-081e28b8, ) in new stack -- Started music on hold, class 'default', on SIP/7411b-081e28b8 == Parked SIP/7411b-081e28b8 on 7...@parkedcalls. Will timeout back to extension [from-local-sip] s, 1 in 3600 seconds -- Added extension '7060' priority 1 to parkedcalls == Spawn extension (from-local-sip, s, 1) exited KEEPALIVE on 'SIP/7411b-081e28b8' So, it seems it is using Park() but for some reason it just doesn't read back the location. Jeremy On Wed, Feb 4, 2009 at 11:08 AM, Danny Nicholas da...@debsinc.commailto:da...@debsinc.com wrote: How is your features.conf set up? Do you have a Parking function in your dialplan? The answer that comes to mind is that you are somehow using parkandannounce instead of park and something is just mis-coded. In my shop, I have hints registered, so core show hints will tell me which lots are in use, but some here consider that a hack. From: asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeremy G. Gault Sent: Wednesday, February 04, 2009 9:53 AM To: asterisk-users@lists.digium.commailto:asterisk-users@lists.digium.com Subject: [asterisk-users] Call parking All, Quick question that hopefully someone out there will know the answer to... We were previously running Asterisk 1.4.(something) (I forget which one) on Debian. Due to an office move, I am temporarily routing our calls through an Ubuntu box that I have. It runs Asterisk 1.4.17-dfsg-2ubuntu1 (basically, what came with Ubuntu.) Here's the problem I am having: We are using Polycom 500's and 501's.. previously (on the Debian system), to park a call, we could transfer it to extension 7000 (we use 4-digit extensions.) Asterisk would read back the parking space number, then we complete the transfer. No problem. On this new system, Asterisk is not reading back the number. Instead, it simply starts playing hold music. If you complete the transfer, a show parkedcalls will show the call as parked (and you can retrieve it.) However, my users have no way of knowing where their calls are being parked. Anyone have any idea as to why it would stop reading back the parking location? I do have the digit sounds installed (in several formats, also.) No luck there. It's almost as if Asterisk is seeing it as a blind transfer instead of a supervised one. Oh, and I can set up a code for it in features.conf and dial that while on the phone (I set the feature code to *8) .. when I do that, it will read back the location and park the call. However, that feature doesn't seem to work for me for all calls (such as calls coming in via a queue, etc.) I'm sure it's something simple, but I've been pulling my hair out searching for anyone else having this problem and haven't had any luck. Any help would be appreciated. :) Jeremy -- Jeremy G. Gault, KD4NED Network Administrator WinWorld Corporation ___ -- 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 -- Jeremy G. Gault, KD4NED Network Administrator WinWorld Corporation ___ -- 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] Call parking
You could try adding this to the default section of your dialplan (extensions.conf) ; park a call in the lot exten = 7000,1,Answer exten = 7000,n,Park() exten = 7000,n,Playback(vm-goodbye) exten = 7000,n,Hangup() Without this, * makes an implicit Park in your dialplan, with it you have some degree of control. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeremy G. Gault Sent: Wednesday, February 04, 2009 10:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call parking Danny, I have parkext set to 7000, parkpos set to 7060-7069, context is set to parkedcalls. In extensions.conf I just include = parkedcalls When I dial 7000 from my desk phone (which used to render a parking location and then play hold music), I get this on the CLI: -- Executing [7...@from-local-sip:1] Park(SIP/7411b-081e28b8, ) in new stack -- Started music on hold, class 'default', on SIP/7411b-081e28b8 == Parked SIP/7411b-081e28b8 on 7...@parkedcalls. Will timeout back to extension [from-local-sip] s, 1 in 3600 seconds -- Added extension '7060' priority 1 to parkedcalls == Spawn extension (from-local-sip, s, 1) exited KEEPALIVE on 'SIP/7411b-081e28b8' So, it seems it is using Park() but for some reason it just doesn't read back the location. Jeremy On Wed, Feb 4, 2009 at 11:08 AM, Danny Nicholas da...@debsinc.com wrote: How is your features.conf set up? Do you have a Parking function in your dialplan? The answer that comes to mind is that you are somehow using parkandannounce instead of park and something is just mis-coded. In my shop, I have hints registered, so core show hints will tell me which lots are in use, but some here consider that a hack. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeremy G. Gault Sent: Wednesday, February 04, 2009 9:53 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Call parking All, Quick question that hopefully someone out there will know the answer to... We were previously running Asterisk 1.4.(something) (I forget which one) on Debian. Due to an office move, I am temporarily routing our calls through an Ubuntu box that I have. It runs Asterisk 1.4.17-dfsg-2ubuntu1 (basically, what came with Ubuntu.) Here's the problem I am having: We are using Polycom 500's and 501's.. previously (on the Debian system), to park a call, we could transfer it to extension 7000 (we use 4-digit extensions.) Asterisk would read back the parking space number, then we complete the transfer. No problem. On this new system, Asterisk is not reading back the number. Instead, it simply starts playing hold music. If you complete the transfer, a show parkedcalls will show the call as parked (and you can retrieve it.) However, my users have no way of knowing where their calls are being parked. Anyone have any idea as to why it would stop reading back the parking location? I do have the digit sounds installed (in several formats, also.) No luck there. It's almost as if Asterisk is seeing it as a blind transfer instead of a supervised one. Oh, and I can set up a code for it in features.conf and dial that while on the phone (I set the feature code to *8) .. when I do that, it will read back the location and park the call. However, that feature doesn't seem to work for me for all calls (such as calls coming in via a queue, etc.) I'm sure it's something simple, but I've been pulling my hair out searching for anyone else having this problem and haven't had any luck. Any help would be appreciated. :) Jeremy -- Jeremy G. Gault, KD4NED Network Administrator WinWorld Corporation ___ -- 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 -- Jeremy G. Gault, KD4NED Network Administrator WinWorld Corporation ___ -- 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] Call parking
Hi, Just so you know, some parking bugs were fixed in 1.4.23.1, so it might be a good idea to update. Regards, Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeremy G. Gault Sent: Wednesday, February 04, 2009 10:53 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Call parking All, Quick question that hopefully someone out there will know the answer to... We were previously running Asterisk 1.4.(something) (I forget which one) on Debian. Due to an office move, I am temporarily routing our calls through an Ubuntu box that I have. It runs Asterisk 1.4.17-dfsg-2ubuntu1 (basically, what came with Ubuntu.) Here's the problem I am having: We are using Polycom 500's and 501's.. previously (on the Debian system), to park a call, we could transfer it to extension 7000 (we use 4-digit extensions.) Asterisk would read back the parking space number, then we complete the transfer. No problem. On this new system, Asterisk is not reading back the number. Instead, it simply starts playing hold music. If you complete the transfer, a show parkedcalls will show the call as parked (and you can retrieve it.) However, my users have no way of knowing where their calls are being parked. Anyone have any idea as to why it would stop reading back the parking location? I do have the digit sounds installed (in several formats, also.) No luck there. It's almost as if Asterisk is seeing it as a blind transfer instead of a supervised one. Oh, and I can set up a code for it in features.conf and dial that while on the phone (I set the feature code to *8) .. when I do that, it will read back the location and park the call. However, that feature doesn't seem to work for me for all calls (such as calls coming in via a queue, etc.) I'm sure it's something simple, but I've been pulling my hair out searching for anyone else having this problem and haven't had any luck. Any help would be appreciated. :) Jeremy -- Jeremy G. Gault, KD4NED Network Administrator WinWorld Corporation ___ -- 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] AOC-E pass through
Take a look at http://bugs.digium.com/view.php?id=7494 Unfortunately it is not yet included in Asterisk, as the patch is somehow a workaround (e.g. faking AOC-E based on last AOC-D). Nevertheless a customer of us uses it for some years now (Astersik 1.2) without any problems. regards klaus Jean-Denis Girard schrieb: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, I'd like to know what is the current situation with regard to AOC-E, when Asterisk is inserted between the telco and an existing PBX, using E1 / EuroISDN. Can Asterisk pass the AOC-E information received from the telco to the PBX, so that billing system still works? The system would be for a hotel, so breaking billing system is not possible. Thanks, - -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 50 10 40 / GSM: +689 79 75 27 -BEGIN PGP SIGNATURE- iEYEARECAAYFAkmJMs0ACgkQuu7Rv+oOo/gFwwCgkO0LFaJ4uOQXifeGajhZAXOe pDkAoJDnClPDX16ZuT27XXYUU02n5Uw1 =L5/q -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- 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] TAPI and Asterisk
If you just want to trigger click2dial you can use SIPTAPI. (make sure to specify type=friend in sip.conf for this account) klaus Jeff LaCoursiere schrieb: Funny how a topic will come up that you have never dealt with before, and suddenly it comes up from multiple directions at the same time. I was recently involved in a meeting where TAPI (which I understand only vaguely) was proposed as way to link a custom application to Asterisk for outbound and inbound call processing, much like SugarCRM and probably others are doing. Today I was asked by an existing client if I knew a way to synch their mobile device contacts with the system in some way so that they would have quick access to speed dial or otherwise call up a personal directory on their (Polycom) phones that could be synched in this manner. It struck me that the Polycom directory interface is a bit kludgy for such things, having no search capability and no sorting capability once loaded that I am aware of. Given the meeting last week it seems that a more elegant solution would be to link Outlook itself with Asterisk, so right clicking a contact makes it possible to launch an outbound call. That would take care of integrating a WHOLE LOT of devices, as (sadly) the MS contact database would be the go-between that all of these devices synch with in one way or another already. Is TAPI the right protocol to investigate for this purpose? Would something like Fonality's HUD software bridge this gap? Has this wheel already been invented? Hoping for some thoughts! Cheers, j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call parking
Mike, Okay. That seems to be the answer. I was able to compile it from source (couldn't find any .deb packages) and parking works as it should. However, upgrading broke the ability to use any of our Zap channels (even using --with-zaptel/usr/src/modules/zaptel when doing ./configure in Asterisk wouldn't work..) It doesn't install chan_zap.so or chan_dahdi.so :( Looks like I will ahve to find a download of DAHDI and do a major overhaul from source. I reverted everything back to what it was and I can tackle that one after-hours if needed. Thanks for the help :) Jeremy On Wed, Feb 4, 2009 at 11:32 AM, Mike l...@virtutel.ca wrote: Hi, Just so you know, some parking bugs were fixed in 1.4.23.1, so it might be a good idea to update. ___ -- 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 -- Jeremy G. Gault, KD4NED Network Administrator WinWorld Corporation ___ -- 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] Stopping chanspy followup
I am still trying to figure out a reasonable way to exit the chanspy application in a dialplan. For the most part I understand how things are working and there is one change I would like to propose. The way the 1.4.23.1 code seems to work is that if there are no channels that match the chanprefix argument the chanspy code stays in a loop waiting for a new channel to come into being that matches chanprefix and spying will start. I would like it if there are no channels to spy on that the chanspy application exit. This can be done by changing line 673 of chanspy.c in the following way Old: if (res == -1 || ast_check_hangup(chan)) New: if (res == -1 || ast_check_hangup(chan) || !peer_chanspy_ds) Otherwise, as best I can tell, unless there is some error chanspy never exits unless the channel running the chanspy application hangs up, which I do not particularly want to do. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ ___ -- 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] siemens hipath 4000
I found out more information.. the OTHER end is configured for OPS - off premise switch. What settings does that correlate to in asterisk? It sounds like is basically T1, b8zs, em wink... However I changed my side to the above (switchtype is still national) singalling is em_w I have not heard of OPS before nor do I see a setting in the dahdi config files for it. Any suggestions? Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] question on originate call
Jerry Geis wrote: I have outgoing call files working. I am trying to get the manager to originate a call. My outgoing call file that works looks like: Channel: SIP/devcentos5x64_to_panel/mediaport SetVar: MEETME_PLAYFILE=/home/silentm/record/pc.1725.wav SetVar: agi_pa_recno=1725 Context: smvoice-dialout Extension: smvoice_single_mediaport Priority: 1 RetryTime: 2 WaitTime: 40 MaxRetries: 0 SetVar: SIPADDHEADER=Alert-Info: Ring Answer CallerID: Jerry Geis 204 317 SetVar: agi_port= SetVar: agi_extension= TESTING SetVar: agi_seconds_to_ring=40 SetVar: agi_dwc_record_num=43 The Manager API session looks like: 04-Feb-09 08:22 am asterisk_command() Action: Login 04-Feb-09 08:22 am asterisk_command() Username: MessageNet 04-Feb-09 08:22 am asterisk_command() Secret: X 04-Feb-09 08:22 am asterisk_command() Events: off 04-Feb-09 08:22 am DEBUG: Response: Success[CR ][LF ]Message: Authentication accepted[CR ][LF ][CR ][LF ] Action: Originate[CR ][LF ] Channel: SIP/devcentos5x64_to_panel/mediaport[CR ][LF ] Variable: MEETME_PLAYFILE=/home/silentm/record/pc.1725.wav[CR ][LF ] Variable: agi_pa_recno=1725[CR ][LF ] Context: smvoice-dialout[CR ][LF ] Exten: smvoice_single_mediaport[CR ][LF ] Priority: 1[CR ][LF ] Timeout: 40[CR ][LF ] Variable: SIPADDHEADER=Alert-Info: Ring Answer[CR ][LF ] Variable: agi_port=3[CR ][LF ] Variable: agi_extension= TESTING[CR ][LF ] Variable: agi_seconds_to_ring=40[CR ][LF ] Variable: agi_dwc_record_num=14[CR ][LF ] [CR ][LF ] 04-Feb-09 08:22 am DEBUG: Response: Error[CR ][LF ]Message: Originate failed[CR ][LF ][CR ][LF ] What am I missing to get the manager API to place the call instead of a call file? Thanks, Jerry Seems like the first call to Channel is being MADE successfully. Then it goes to do Context and Exten: I get failed... [smvoice-dialout] exten = smvoice_single_mediaport,1,agi(smvoice) exten = smvoice_single_mediaport,n,Hangup I am running an AGI at that point. Can the Mangager API not handle that All the AGI is doing looking at the variables and playing a wave file (at this time). This works fine using a call file. Any thoughts? Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with MOH and streaming music on 1.6.0.5
I am having a problem getting MOH to work with mpg123 on 1.6. I created a bug ticket and I am not getting any where so I am looking here for help. Please see http://bugs.digium.com/view.php?id=14387 for details. -- Jonn Taylor Taylor Telephone Systems, Inc 8334 Argenta Trail Inver Grove Heights, MN 55077 http://www.taylortelephone.com/ ___ -- 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] Problems with 9133i config
I am unable to get my 9133i to register with my asterisk server. I am including config files below, this a simple test network so there's nothing secret in the config files. I have upgraded the phone to the latest software version (1.4.3) I'm not sure what the problem is. I can call the phone from a softphone, but the 9133i says no service on the screen and I can't dial anything on it. configs: Aastra.cfg dhcp: 1 # DHCP enabled. sip silence suppression: 2# 0 = off, 1 = on, 2 = default sip proxy port: 5060 # 5060 is set by default. sip registrar ip: 192.168.0.94# IP of registrar sip registrar port: 5060 # 5060 is set by default. sip digit time out: 6 time server disabled: 0 # Time server disabled. time server1: 192.168.0.90# Enable time server and enter at mac.cfg - this is the correct mac address in all uppercase sip line1 auth name: phone1 sip line1 password: 1234 sip line1 registrar ip: 192.168.0.94 sip line1 user name: phone1 sip line1 display name: myname sip line1 screen name: myname sip.conf [general] port = 5060 bindaddr = 0.0.0.0 context=tutorial [phone1] type=friend username=phone1 secret=1234 host=dynamic canreinvite=no permit=192.168.0.0/24 allow=all qualify=yes extensions.conf [tutorial] exten = 1234,1,Answer exten = 1234,n,SayDigits(123456789) exten = 3001,1,Dial(SIP/phone1,18) exten = 3002,1,Dial(SIP/phone2,18) sip debug output --- SIP read from 192.168.0.11:5060 --- REGISTER sip:192.168.0.94 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.11:5060;branch=z9hG4bKcd3373869 Max-Forwards: 70 Content-Length: 0 To: myname sip:phone1@ From: myname sip:phone1@;tag=24b6354e352ab62 Call-ID: a85fb0e71a3e7def4bd7d7f158933...@192.168.0.11 CSeq: 1006065354 REGISTER Contact: myname sip:pho...@192.168.0.11:5060;transport=udp Allow-Events: talk,hold,conference Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO User-Agent: Aastra 9133i/1.4.3.23 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45 - --- (12 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.0.11 : 5060 (no NAT) --- Transmitting (no NAT) to 192.168.0.11:5060 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.11:5060;branch=z9hG4bKcd3373869;received=192.168.0.11 From: myname sip:phone1@;tag=24b6354e352ab62 To: myname sip:phone1@ Call-ID: a85fb0e71a3e7def4bd7d7f158933...@192.168.0.11 CSeq: 1006065354 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:pho...@192.168.0.94 Content-Length: 0 --- Transmitting (no NAT) to 192.168.0.11:5060 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.0.11:5060;branch=z9hG4bKcd3373869;received=192.168.0.11 From: myname sip:phone1@;tag=24b6354e352ab62 To: myname sip:phone1@;tag=as51ded290 Call-ID: a85fb0e71a3e7def4bd7d7f158933...@192.168.0.11 CSeq: 1006065354 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=2f250e11 Content-Length: 0 Scheduling destruction of SIP dialog 'a85fb0e71a3e7def4bd7d7f158933...@192.168.0.11' in 32000 ms (Method: REGISTER) --- SIP read from 192.168.0.11:5060 --- REGISTER sip:192.168.0.94 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.11:5060;branch=z9hG4bKcd3373869 Max-Forwards: 70 Content-Length: 0 To: myname sip:phone1@ From: myname sip:phone1@;tag=24b6354e352ab62 Call-ID: a85fb0e71a3e7def4bd7d7f158933...@192.168.0.11 CSeq: 1006065354 REGISTER Contact: myname sip:pho...@192.168.0.11:5060;transport=udp Allow-Events: talk,hold,conference Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO User-Agent: Aastra 9133i/1.4.3.23 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45 - --- (12 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.0.11 : 5060 (no NAT) --- Transmitting (no NAT) to 192.168.0.11:5060 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.11:5060;branch=z9hG4bKcd3373869;received=192.168.0.11 From: myname sip:phone1@;tag=24b6354e352ab62 To: myname sip:phone1@ Call-ID: a85fb0e71a3e7def4bd7d7f158933...@192.168.0.11 CSeq: 1006065354 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:pho...@192.168.0.94 Content-Length: 0 --- Transmitting (no NAT) to 192.168.0.11:5060 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.0.11:5060;branch=z9hG4bKcd3373869;received=192.168.0.11 From: myname sip:phone1@;tag=24b6354e352ab62 To: myname sip:phone1@;tag=as51ded290 Call-ID: a85fb0e71a3e7def4bd7d7f158933...@192.168.0.11 CSeq: 1006065354 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=2f250e11 Content-Length: 0 Scheduling destruction of SIP dialog
Re: [asterisk-users] [asterisk-dev] RFC 2833 DTMF w/ Level 3 Sonus
On Wed, Feb 4, 2009 at 4:00 PM, Gregory Boehnlein da...@nacs.net wrote: Hello, Is anyone running Asterisk 1.4 w/ RFC2833 to Level3's SONUS network? We are unable to get reliable RFC 2833 DTMF working, and have instead had to use G711ULAW w/ INBAND DTMF to get around the issue. Looks like an issue on the SONUS side. Anyone else have this issue? Welcome to the club! ;) I'll be blogging about this later today. Look out for that post... -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- 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] early dial: asterisk and ATA
On Tue, Feb 3, 2009 at 5:04 PM, Vieri rentor...@yahoo.com wrote: I did but apparently, there's nothing in the guides that lets me do this. It's something about supporting 484 responses that Grandstream GXW4008 seems to do and Linksys SPA8000 doesn't (or at least it's not documented). In other words, the SPA8000's L1-L8 Dial Plan parameter only allows for matches to be performed entirely on the device and not via 484 ADDRESS INCOMPLETE responses with Asterisk's dial patterns. Sorry, I failed to fully understand your question. I'm not sure if the SPAs will dispatch partial numbers and manage 484 responses like the GS gear seems to do. -- exvito ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] question on originate call
On Wed, Feb 4, 2009 at 7:40 PM, Jerry Geis ge...@pagestation.com wrote: Seems like the first call to Channel is being MADE successfully. Then it goes to do Context and Exten: I get failed... [smvoice-dialout] exten = smvoice_single_mediaport,1,agi(smvoice) exten = smvoice_single_mediaport,n,Hangup I can't identify nothing specific, apart from the fact that you're running non-standard Asterisk applications. My humble suggestions: 1. Increase log verbosity and check logs 2. Break the problem into something simpler (example: Grab one extension and Play a file into it) When you get to a working setup, build up from there, one step at a time... -- exvito ___ -- 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] Contact lookup
For a simple (but flexible) case I would consider ODBC + func_odbc.
Here is the idea (in case you aren't aware of how it goes...)
- Make a DB available (your choice as long as it is accessible via ODBC)
- Create table in it with your contacts (say columns number and
name, maybe more)
- Setup an ODBC connection for asterisk so that it can connect to that DB
(res_odbc.conf)
- Setup an ODBC func.This is basically an SQL query which will be
mapped into a dialplan function. (func_odbc.conf) It is essentially
something that states my function ODBC_LOOKUP(arg) will give me
the results of SELECT name FROM contactsTable WHERE number=${arg}
into the dialplan.
- Then use it in the dialplan
exten = _x.,n,Set(CALLERID(name)=${ODBC_LOOKUP(${EXTEN})})
There! Your dialplan is almost directly executing SQL queries. :)
Check both the sample asterisk configs + Asterisk TFOT, chapter 12.
It may be a bit more work than using the Ast DB or other means, but it
has the advantage of allowing the easy setup of any kind of frontend for
contact management.
Note: Check for the correctness of my filenames/syntax... They're shown
just to fill in the idea with something resembing the reality!
My 2c,
--
exvito
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Re: [asterisk-users] siemens hipath 4000
Any suggestions? Jerry Are you sure asterisk is to behave as signalling=pri_cpe or should it be pri_net ? -- exvito ___ -- 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] Stopping chanspy followup
Jim Dickenson wrote: I am still trying to figure out a reasonable way to exit the chanspy application in a dialplan. For the most part I understand how things are working and there is one change I would like to propose. The way the 1.4.23.1 code seems to work is that if there are no channels that match the chanprefix argument the chanspy code stays in a loop waiting for a new channel to come into being that matches chanprefix and spying will start. I would like it if there are no channels to spy on that the chanspy application exit. This can be done by changing line 673 of chanspy.c in the following way Old: if (res == -1 || ast_check_hangup(chan)) New: if (res == -1 || ast_check_hangup(chan) || !peer_chanspy_ds) Otherwise, as best I can tell, unless there is some error chanspy never exits unless the channel running the chanspy application hangs up, which I do not particularly want to do. In the interim I would recommend you make chat change and recompile. -- Thank you and have any kind of day you want, Anthony Francis ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] question on originate call - solved
Jerry Geis wrote: Jerry Geis wrote: I have outgoing call files working. I am trying to get the manager to originate a call. My outgoing call file that works looks like: Channel: SIP/devcentos5x64_to_panel/mediaport SetVar: MEETME_PLAYFILE=/home/silentm/record/pc.1725.wav SetVar: agi_pa_recno=1725 Context: smvoice-dialout Extension: smvoice_single_mediaport Priority: 1 RetryTime: 2 WaitTime: 40 MaxRetries: 0 SetVar: SIPADDHEADER=Alert-Info: Ring Answer CallerID: Jerry Geis 204 317 SetVar: agi_port= SetVar: agi_extension= TESTING SetVar: agi_seconds_to_ring=40 SetVar: agi_dwc_record_num=43 The Manager API session looks like: 04-Feb-09 08:22 am asterisk_command() Action: Login 04-Feb-09 08:22 am asterisk_command() Username: MessageNet 04-Feb-09 08:22 am asterisk_command() Secret: X 04-Feb-09 08:22 am asterisk_command() Events: off 04-Feb-09 08:22 am DEBUG: Response: Success[CR ][LF ]Message: Authentication accepted[CR ][LF ][CR ][LF ] Action: Originate[CR ][LF ] Channel: SIP/devcentos5x64_to_panel/mediaport[CR ][LF ] Variable: MEETME_PLAYFILE=/home/silentm/record/pc.1725.wav[CR ][LF ] Variable: agi_pa_recno=1725[CR ][LF ] Context: smvoice-dialout[CR ][LF ] Exten: smvoice_single_mediaport[CR ][LF ] Priority: 1[CR ][LF ] Timeout: 40[CR ][LF ] Variable: SIPADDHEADER=Alert-Info: Ring Answer[CR ][LF ] Variable: agi_port=3[CR ][LF ] Variable: agi_extension= TESTING[CR ][LF ] Variable: agi_seconds_to_ring=40[CR ][LF ] Variable: agi_dwc_record_num=14[CR ][LF ] [CR ][LF ] 04-Feb-09 08:22 am DEBUG: Response: Error[CR ][LF ]Message: Originate failed[CR ][LF ][CR ][LF ] What am I missing to get the manager API to place the call instead of a call file? Thanks, Jerry Seems like the first call to Channel is being MADE successfully. Then it goes to do Context and Exten: I get failed... [smvoice-dialout] exten = smvoice_single_mediaport,1,agi(smvoice) exten = smvoice_single_mediaport,n,Hangup I am running an AGI at that point. Can the Mangager API not handle that All the AGI is doing looking at the variables and playing a wave file (at this time). This works fine using a call file. Any thoughts? Jerry Found it The timeout in a call file (which worked) is in seconds. The timeout on the manager api is in milliseconds. my timeout was kicking me! Have a great evening! jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No Reply to Our Critical Packet SIP Calls Dropped in Voicemail
Hi Steve, Thanks again for the response-- the answer you gave was more or less the answer that I was expecting. I was logging all packets to and from the phone, and I never saw an ACK from the phone for the OK to Asterisk on the VM calls -- not an ACK directed to a different location, just no ACK period. I noted in my other reply that as a test I had added a call to Ringing() followed by Wait(1) before dropping into Voicemail for the voicemail extension in the dialplan, since I noticed that the only difference that appeared to exist between a SIP-POTS or SIP-SIP call and a SIP-Voicemail call, aside from the missing ACK from the phone is that Asterisk reported session progress of 100 Trying and 180 Ringing to the phone, where it didn't report either of these when calling Voicemail, instead jumping straight to 200 OK with session description. In the 24 hours since I did that we haven't been able to get any of dozens of calls to Voicemail to fail, when normally it would borderline on greater than one in every two call. I'm still not convinced it's fixed, but I'm feeling fairly good about the solution, so it seems to my untrained eye like there may be an issue in the Cisco 79x1 firmware if the PBX accepts a call without providing any intermediate status? That seems like it would manifest itself in other places, and I'm kind of grasping at straws but... Thanks again to everyone who took the time to read and or respond to this issue -- I'll post again if it turns out that that wasn't actually the fix, but for now management is happy that they can actually listen to their entire voicemail messages. Lincoln -- Lincoln King-Cliby, CTS Applications Engineer ControlWorks Consulting, LLC V: 440.729.4640 x1107 F: 440.729.0884 I:http://www.controlworks.com Crestron Authorized Independent Programmer -Original Message- From: Steven J. Douglas [mailto:stev...@moij.biz] Sent: Wednesday, February 04, 2009 12:43 AM To: Lincoln King-Cliby Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] No Reply to Our Critical Packet SIP Calls Dropped in Voicemail Hi Lincoln, The fact that you can hear and respond to the voice mail (even if its for the first 20 seconds), means that your phone has received the OK message properly. The problem is the missing ACK after receiving OK. When asterisk did not receive the ACK after a few retries of the OK, it terminated the call. This resulted in your RTP streams getting the icmp error messages. Assuming that you are capturing every packet that goes on between Asterisk and the phone, there are two possibilities. 1. The phone has a bug. 2. The ACK was sent somewhere else. Normally the ACK message destination is constructed from the response to the INVITE. In this case, it will be the OK message. If you suspect its the second case, you can capture the traffic for both a good voicemail call and a failed voicemail call. Then by comparing the messages, you might get a hint. If you need help, you can send the packet capture to me privately (not through the list as it might be a large file) and I can help vet it for you. Unfortunately there is no flag that you can set to confirm a session based on OK being transmitted and not wait for ACK. Regards, Steve snipped my original reply ___ -- 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] hardware that can accomondate 2 TDM24
Hi guys, I'm building a server that need to host 2 digium TDM24 cards. I know any 3U server with 2 PCI-E slots would do. Since I do prefer supermicro server, but getting one configured is pretty darn hard. Any suggestions here? Cheers, Kelvin Chan | Positronics Ent. Product Development | | unit 272 604-628-9330 (direct) | 8128 128th St. 604-585-2...@104 (main) | Surrey, BC 604-585-3056 (fax)| Canada, V3W 1R1 ___ -- 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] hardware that can accomondate 2 TDM24
Are you locked into the 3U form factor? We're running Asterisk on a Dell PowerEdge 1950 (1U, 2 full height PCI-E slots [one home to an AEX-804E], 3 drive bays, redundant power). I both the 2950 and 2970 (both are 2U, variable number of drive bays based on the config you choose, the 2950 shares firmware with the 1950) can be ordered with PCI-E risers because we have a handful in our datacenter, but I have no idea how many slots -- I want to say 3. I think the TDM24 is too long to fit in a 1950, but I'm pretty sure (you'd have to check) that the 2950/70 has at least two full-length slots. HTH, Lincoln -- Lincoln King-Cliby, CTS Applications Engineer ControlWorks Consulting, LLC V: 440.729.4640 x1107 F: 440.729.0884 I:http://www.controlworks.com Crestron Authorized Independent Programmer -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kelvin Chan Sent: Wednesday, February 04, 2009 7:17 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] hardware that can accomondate 2 TDM24 Hi guys, I'm building a server that need to host 2 digium TDM24 cards. I know any 3U server with 2 PCI-E slots would do. Since I do prefer supermicro server, but getting one configured is pretty darn hard. Any suggestions here? Cheers, Kelvin Chan | Positronics Ent. Product Development | | unit 272 604-628-9330 (direct) | 8128 128th St. 604-585-2...@104 (main) | Surrey, BC 604-585-3056 (fax)| Canada, V3W 1R1 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hardware that can accomondate 2 TDM24
Saw your post...let me know what suggestions arise (I do not watch the list that closely -- your was flagged because my monitoring software spotted your email address). g. -- George Pajari (dCAP), netVOICE communications 604 484 VOIP(8647) x102 www.netvoice.ca www.ip-centrex.ca www.ip-pbx.ca www.vpas.ca www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with building dahdi-linux RPM
On Wed, Feb 04, 2009 at 01:04:47PM +0300, bee-beeep wrote: I have some OpenVOX A1200p cards, and driver for them so far works only with dahdi-2.0.0 Sorry, looks like i don't understand, how to correctly rebuild driver: rpmbuild --rebuild http://dl.atrpms.net/all/dahdi-linux-2.1.0.3-59.src.rpm skipped Wrote: /usr/src/redhat/RPMS/i386/dahdi-linux-kmdl-2.6.18-92.1.22.el5-2.1.0.3-59.RHL5.i386.rpm Only kmdl module was wroten, no dahdi-linux and dahdi-linux-devel. Can you explain, what should i do? :-) Use the packages at http://atrpms.net/dist/el5/dahdi-tools/ in addition to the kmdl you built. Unless you also need a specific older dahdi-tools packafe, which you would have to rebuild, too. -- Axel.Thimm at ATrpms.net pgp95mZaLW0RD.pgp Description: PGP signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Autodialler query
Hi Everybody I've a requirement for one of my operators for an autodialler for which i plan to deploy asterisk (I already have 3 asterisk servers on PRI running very well ! ). The scene is like : Asterisk will call a customer and play a prompt that prompts him to press 1 if he wishes to talk to an agent , If the customer presses 1 then the call gets connected to one of my proffessional agents who talk on certain subject - but the challenge here is that the moment he presses 1 - the customer should be billed a premium rate ex, Rs.9 per minute.. Is that possible ? If yes then can anyone guide me as to what all points i need to focus on during my discussion with operator ? Thanks Sriram___ -- 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] TDM400P Circuit/channel congestion problem
Hello, I have an issue with Digium TDM 400 card series. When I try to make outgoing call (PSTN call) for example, the Zap channel could not be created and busy channel message appeared. Below is the full log : [Feb 5 09:26:17] VERBOSE[3047] logger.c: -- Executing [...@macro- dialout-trunk:20] Dial(SIP/213-09648720, ZAP/g1/08170709XXX|300|) in new stack [Feb 5 09:26:17] WARNING[3047] app_dial.c: Unable to create channel of type 'ZAP' (cause 34 - Circuit/channel congestion) [Feb 5 09:26:17] VERBOSE[3047] logger.c: == Everyone is busy/ congested at this time (1:0/1/0) [Feb 5 09:26:17] DEBUG[3047] app_macro.c: Executed application: Dial [Feb 5 09:26:17] VERBOSE[3047] logger.c: -- Executing [...@macro- dialout-trunk:21] Goto(SIP/213-09648720, s-CONGESTION|1) in new stack [Feb 5 09:26:17] VERBOSE[3047] logger.c: -- Goto (macro-dialout- trunk,s-CONGESTION,1) The problem is fixed (outgoing call will work fine) when the PSTN cable attached to card are _manually_ unplugged and then plugged back to card. Of course, I don't want to do this job everytime when server restarted :-). Searching over internet, it say that I must disable echotraining, but the problem still persist. System: Asterisk 1.4.22-rc5 (Elastix 1.3-2) OS: Centos 5.2 Core 2 Duo Processor E6750 @ 2.66GHz $ dmesg | egrep '(echo|tone|Zap|Zap|TDM|Module)' zaptel: no version for oslec_echo_can_traintap found: kernel tainted. Zapata Telephony Interface Registered on major 196 Zaptel Version: 1.4.11 Zaptel Echo Canceller: OSLEC Zaptap registered 'sample' char driver on major 33 Module 0: Installed -- AUTO FXO (FCC mode) Module 1: Installed -- AUTO FXO (FCC mode) Module 2: Installed -- AUTO FXO (FCC mode) Module 3: Installed -- AUTO FXO (FCC mode) Found a Wildcard TDM: Wildcard TDM400P REV I (4 modules) Registered tone zone 0 (United States / North America) -- Setting echo registers: -- Set echo registers successfully -- Setting echo registers: -- Set echo registers successfully no echo canceller being monitored - make a new call - File zaptel.conf: fxsks=1 fxsks=2 #fxsks=3 #fxsks=4 loadzone= us defaultzone = us --- File zapata.conf: [trunkgroups] [channels] context=from-zaptel signalling=fxs_ks rxwink=300 ; Atlas seems to use long (250ms) winks usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no faxdetect=incoming echotraining=800 ;rxgain=0.0 ;txgain=0.0 group=0 callgroup=1 pickupgroup=1 ;Uncomment these lines if you have problems with the disconection of your analog lines ;busydetect=yes ;busycount=3 immediate=yes #include zapata_additional.conf #include zapata-channels.conf Thank you. Rgds, Asfihani ___ -- 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] Autodialler query
Hi Sriram, the customer should be billed a premium rate ex, Rs.9 per minute.. Will be billed by you or by telecomm company? Regards David - Original Message - From: Sriram To: asterisk-users@lists.digium.com Sent: Thursday, February 05, 2009 1:46 PM Subject: [asterisk-users] Autodialler query Hi Everybody I've a requirement for one of my operators for an autodialler for which i plan to deploy asterisk (I already have 3 asterisk servers on PRI running very well ! ). The scene is like : Asterisk will call a customer and play a prompt that prompts him to press 1 if he wishes to talk to an agent , If the customer presses 1 then the call gets connected to one of my proffessional agents who talk on certain subject - but the challenge here is that the moment he presses 1 - the customer should be billed a premium rate ex, Rs.9 per minute.. Is that possible ? If yes then can anyone guide me as to what all points i need to focus on during my discussion with operator ? Thanks Sriram -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Contact lookup
On Wednesday, February 4, 2009, Ex Vito wrote: For a simple (but flexible) case I would consider ODBC + func_odbc. Here is the idea (in case you aren't aware of how it goes...) [... snip ...] It may be a bit more work than using the Ast DB or other means, but it has the advantage of allowing the easy setup of any kind of frontend for contact management. Thanks for the reply. The nice thing about that is that if I use MySQL I can run the management application on another machine, and so don't need to run a web server on the Asterisk box. However, I wonder whether the overhead necessary to run MySQL on the Asterisk box is more than that required to run Apache to provide a web interface to astdb. I'm not running either at present, which is probably as well since my Asterisk machine is low-spec by todays standards. At the moment it's academic since I don't have a large or extremely dynamic contact list and so can handle it with commands in the * CLI. However, it'll be an interesting exercise when I eventually upgrade the hardware and also move to Asterisk 1.6. Thanks again, -- Geoff ___ -- 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] Autodialler query
Sriram: whats going on here?? unless you are developing a vas, in which case, the provider for whom you are doing this will have to help you. each provider would be doing this differently. regards Kinjal Dixit On Thu, Feb 5, 2009 at 7:20 AM, da...@iaxtalk.com wrote: Hi Sriram, the customer should be billed a premium rate ex, Rs.9 per minute.. Will be billed by you or by telecomm company? Regards David - Original Message - *From:* Sriram d_r_sri...@hotmail.com *To:* asterisk-users@lists.digium.com *Sent:* Thursday, February 05, 2009 1:46 PM *Subject:* [asterisk-users] Autodialler query Hi Everybody I've a requirement for one of my operators for an autodialler for which i plan to deploy asterisk (I already have 3 asterisk servers on PRI running very well ! ). The scene is like : Asterisk will call a customer and play a prompt that prompts him to press 1 if he wishes to talk to an agent , If the customer presses 1 then the call gets connected to one of my proffessional agents who talk on certain subject - but the challenge here is that the moment he presses 1 - the customer should be billed a premium rate ex, Rs.9 per minute.. Is that possible ? If yes then can anyone guide me as to what all points i need to focus on during my discussion with operator ? Thanks Sriram -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- http://www.linkedin.com/in/kinjaldixit open networker ___ -- 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] escaping regular expression
D Tucny schrieb:
2009/2/4 Klaus Darilion klaus.mailingli...@pernau.at
mailto:klaus.mailingli...@pernau.at
D Tucny schrieb:
2009/2/4 Klaus Darilion klaus.mailingli...@pernau.at
mailto:klaus.mailingli...@pernau.at
mailto:klaus.mailingli...@pernau.at
mailto:klaus.mailingli...@pernau.at
Hi!
I am going nuts using REGEXP. I just want to verify if a variable
contains a valid +E164 phone number.
These, the the pattern is ^\+[0-9]+
First I tried:
Set(pattern=^\+[0-9]+);
if (${REGEX(${pattern} ${${var}})})
but that does not work, the backslash is removed, as seen in
the log
file:
func_strings.c: FUNCTION REGEX (^+[0-9]+)(+4312345
http://www.adaanumber.com/)
So, meanwhile I tried to escape the backslash. I tried:
Set(pattern=^\\+[0-9]+);
Set(pattern=^\\\+[0-9]+);
Set(pattern=^+[0-9]+);
But always the same result:
func_strings.c: FUNCTION REGEX (^+[0-9]+)(+4312345)
How can I solve this problem?
Try something like... pattern=^[+]\{1\}[0-9]+
Are you sure? The \ should be in front of the +
Pretty sure...
exten = *56,1,NoOp(Starting regexp test)
exten = *56,n,Set(pattern=^[+]\{1\}[0-9]+)
exten = *56,n,Set(var=123456789)
exten = *56,n,NoOp(${IF(${REGEX(${pattern} ${var})}?Match:No
Match)}))
exten = *56,n,Set(var=+123456789)
exten = *56,n,NoOp(${IF(${REGEX(${pattern} ${var})}?Match:No
Match)}))
[Feb 4 23:49:21] DEBUG[20518] pbx.c: Launching 'NoOp'
[Feb 4 23:49:21] VERBOSE[20518] logger.c: -- Executing
[...@phonedefault:1] NoOp(SIP/*01-09bd8ff8, Starting regexp test)
in new stack
[Feb 4 23:49:21] DEBUG[20518] pbx.c: Launching 'Set'
[Feb 4 23:49:21] VERBOSE[20518] logger.c: -- Executing
[...@phonedefault:2] Set(SIP/*01-09bd8ff8, pattern=^[+]\{1\}[0-9]+)
in new stack
[Feb 4 23:49:21] DEBUG[20518] pbx.c: Launching 'Set'
[Feb 4 23:49:21] VERBOSE[20518] logger.c: -- Executing
[...@phonedefault:3] Set(SIP/*01-09bd8ff8, var=123456789) in new stack
[Feb 4 23:49:21] DEBUG[20518] func_strings.c: FUNCTION REGEX
(^[+]{1}[0-9]+)(123456789)
[Feb 4 23:49:21] DEBUG[20518] pbx.c: Function result is '0'
[Feb 4 23:49:21] DEBUG[20518] pbx.c: Function result is 'No Match'
[Feb 4 23:49:21] DEBUG[20518] pbx.c: Launching 'NoOp'
[Feb 4 23:49:21] VERBOSE[20518] logger.c: -- Executing
[...@phonedefault:4] NoOp(SIP/*01-09bd8ff8, No Match)) in new stack
[Feb 4 23:49:21] DEBUG[20518] pbx.c: Launching 'Set'
[Feb 4 23:49:21] VERBOSE[20518] logger.c: -- Executing
[...@phonedefault:5] Set(SIP/*01-09bd8ff8, var=+123456789) in new stack
[Feb 4 23:49:21] DEBUG[20518] func_strings.c: FUNCTION REGEX
(^[+]{1}[0-9]+)(+123456789)
[Feb 4 23:49:21] DEBUG[20518] pbx.c: Function result is '1'
[Feb 4 23:49:21] DEBUG[20518] pbx.c: Function result is 'Match'
[Feb 4 23:49:21] DEBUG[20518] pbx.c: Launching 'NoOp'
[Feb 4 23:49:21] VERBOSE[20518] logger.c: -- Executing
[...@phonedefault:6] NoOp(SIP/*01-09bd8ff8, Match)) in new stack
So, the \ is still stripped (ast_app_separate_args removes \), but, it
doesn't matter as the + is bracketed so it's not the first character
after the ^ and so regcomp doesn't fail...
Ah, the trick is to put the + into [], so it need not be escaped.
Thanks
Klaus
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