[Asterisk-Users] Voicemail file as MP3
Hi * users, Is that possible to make voicemail audio file (that is attached to forwarding email) as MP3 file, rather than WAV? TIA Kuni -- Kuniyoshi Murata English-Japanese Interpreter Macintosh Webcast Specialist [WebSite] www.macwebcaster.com [Email] [EMAIL PROTECTED] [Skype] kuniyoshi_murata [SNS] mixi.jp/show_friend.pl?id=59236 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Errors With Hint
Hello How do you configure Polycom for presence please ? Harry --- Alvaro Parres [EMAIL PROTECTED] a écrit : Hi list, i have the next problem: I create 3 hints.. (111 (SIP/111), 112 (SIP/112), and 102 (ZAP/35) ) the SIP/111 is a GrandStream ATA the SIP/112 is a Polycom 301 the ZAP/35 is a Analogic Phone. The SIP/112 hints works great. But the other 2 no. The ZAP/35 is say is always in USE and as you see en the next console output is not in use. any Idea asterisk*CLI -= Registered Asterisk Dial Plan Hints =- 111 : SIP/111 State:Idle Watchers 4 102 : ZAP/35 State:InUse Watchers 5 112 : SIP/112 State:InUse Watchers 2 - 3 hints registered asterisk*CLI show cha channel channels channeltypes asterisk*CLI show channels Channel Location State Application(Data) Zap/34-1 [EMAIL PROTECTED]:1 Up Bridged Call(SIP/112-1f3d) SIP/112-1f3d [EMAIL PROTECTED]: Up Dial(ZAP/34/3338182842|120|Tt) 2 active channels 1 active call And also the SIP/111 is always in Idle any idea of why ??? thanks ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] receive fax with asterisk
http://www.hylafax.org/ Harry --- Doug Lytle [EMAIL PROTECTED] a écrit : Jason Brashear wrote: Receiving faxes with Asterisk. Is there a good resource for learning how to set this up? www.soft-switch.org Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.2.0-RC1 Crashing with g729 codec and ATA 18
Gervais de Montbrun ha scritto: **keepalive = 5 set the keepalive to 60 or more speeddial = 500,500,[EMAIL PROTECTED] that phone should not be able to display a hint status so speeddial = 500,500 This is what is displayed in the console when I try to call the 12SP from the ATA The log could be more verbose than this. Set debug = 10 in your sccp.conf or in the console sccp debug 10 You should see what is happening with your audio stream Sergio ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] A2Billing Postpay
Hi Seong, The creditlimit mean the amount of credit you authorize the CardHolder to go in negative. You should have try ;-) Rgds, /Areski On 11/11/05, Ah khng [EMAIL PROTECTED] wrote: Hi all, I'm glad to hear that areskicc v3 have been released. But i have a problem to use a2billing as postpay calling card. I have no clear understanding what is meant by credit limit that need to specify when postpay method is selected. I will get the message say that the credit is not enough to make call when the credit of postpay user approaching 0. How to make the postpay user still able to make a call when their credit is in negative? Thanks Seong ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Consultant
Hi: I have been posting this issue for over a month and I am not irritated. I appreciate all the users who helped before and I thank in advance any offer for help including reasonable paid time. You may sent me your price me list at [EMAIL PROTECTED] Regards; chawki --- Rob Lith [EMAIL PROTECTED] wrote: Sounds like a good deal to me. If you want free answers don't sound so irritated that you haven't got a reply in $0 time. :) Rob On 11/9/05, chawki hammoud [EMAIL PROTECTED] wrote: The only pointer I got is a $50/hr Mark phillip offered. I can make VOIP calls between my Asterisk server and my VOIP provider using sip channel without a problem. But when I attempt to make a call using IAX, the call get accepted and then get a hangup message: This is the message I get when I attempt to make an IAX call: Executing Dial(OSS/dsp, IAX2/callshopcompany/0017046872001) in new stack -- Called callshopcompany/0017046872001 -- Call accepted by 213.61.187.150 http://213.61.187.150 (format gsm) -- Format for call is gsm -- Hungup 'IAX2/callshopcompany/1' == No one is available to answeer at this time The call get accepted, but it seems there is no acknowledgement from my server to receive the call from the provider. Thanks; --- Mark Phillips [EMAIL PROTECTED] wrote: He did. And he got pointers to the relevant howto's. Matt Riddell wrote: chawki hammoud wrote: Hi: I posted my problem several times about being unable to make IAX calls from my Asterisk box to another IAX server without luck. So, what's your problem? Post some details. -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com ___ --Bandwidth and Colocation sponsored by Easynews.comhttp://Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Yahoo! FareChase: Search multiple travel sites in one click. http://farechase.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com http://Easynews.com-- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Areski Can you Help ??? We are stuck
Hi Abdock, 1# You can set a context in iax.conf [mytrunkiax] with the username, secret, host, etc.. and then use the name mytrunkiax in A2Billing it will dial using this trunk. This will allow you to configure the willing codec. 2# Directly use in the Edit trunk, username:[EMAIL PROTECTED], I guess this should work. Kinds regards, /Areski On 11/9/05, Abdock [EMAIL PROTECTED] wrote: Hello, I installed the ver3 everything looks ok, i get the prompt, put the pin no and also proceeds with dialing but it fails as it is not able to authenticate with the gateway. How can i configure the IAX2 ? for outgoing, as i require to put in username, password, and the ip address of gateway ? We also have a separate asterisk server, which we connect by IAX2, but that also rejects call from the calling card server ? Anybody done this ? and please if you can share ? Thanks. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IM / presence asterisk-1.2-RC1
Sorry, Here are some files Harry --- BJ Weschke [EMAIL PROTECTED] a écrit : This is good debugging info you've listed below, but this isn't a sip debug/trace. To do that, first verify in your logger.conf file you have the following line: full = notice,warning,error,debug,verbose Then, if you needed to add anything to logger.conf, please first restart Asterisk so those new settings take effect. Then, from the CLI issue set verbose 5 and set debug 5 and finally sip debug. The repeat your dialing steps. The sip debug/trace will then be contained in /var/log/asterisk/full if /var/log/asterisk is where your log files are kept. With that, we can have a better idea of what's happening/not happening to give you the issue you're having. On 11/10/05, harry gaillac [EMAIL PROTECTED] wrote: I did it !? // Connected to Asterisk 1.2.0-rc1 currently running on serveur1 (pid = 1125) Verbosity is at least 4 serveur1*CLI sip show subscriptions Peer UserCall ID Extension Last state Type 192.168.0.21 86 f1682d8d-8f 84 Idle xpidf+xml 192.168.0.21 86 5f32aec-95b 85 Idle xpidf+xml 192.168.0.20 84 cb424ae1-e4 86 Idle xpidf+xml 192.168.0.20 84 715fac66-a9 87 Idle xpidf+xml 4 active SIP subscriptions serveur1*CLI // serveur1*CLI sip show peers Name/username HostDyn Nat ACL Port Status 87/87 192.168.0.21 D N 5060 OK (84 ms) 86/86 192.168.0.21 D N 5060 OK (97 ms) 85/85 192.168.0.20 D N 5060 OK (87 ms) 84/84 192.168.0.20 D N 5060 OK (96 ms) 4 sip peers [4 online , 0 offline] serveur1*CLI /// my sip.conf: [general] context=local ; Default context for incoming calls ; if asterisk was compiled with OSP support. realm=nxs.yi.org; Realm for digest authentication ; defaults to asterisk ; Realms MUST be globally unique according to RFC 3261 ; Set this to your host name or domain name bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=nxs.yi.org ; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ; Enable DNS SRV lookups on outbound calls tos=lowdelay; lowdelay,throughput,reliability,mincost,none maxexpirey=3600 ; Max length of incoming registration we allow defaultexpirey=1000 ; Default length of incoming/outoing registration allow=all ; First disallow all codecs musicclass=default ; Sets the default music on hold class for all SIP calls language=fr ; Default language setting for all users/peers rtptimeout=60 ; Terminate call if 60 seconds of no RTP activity tpholdtimeout=300 ; Terminate call if 300 seconds of no RTP activity useragent=Asterisk PBX ; Allows you to change the user agent string dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833 -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com messages.serveur1.home.net Description: 1676272990-messages.serveur1.home.net debug.serveur1.home.net Description: 3484436676-debug.serveur1.home.net ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Result branching in AEL
Morning all, I'm trying to rewrite my dialplan macros into AEL. How does one handle result-dependent branching (e.g. VoiceMail will branch to n+101 if mailbox doesn't exist) in AEL? Or is there a better way of doing this? Thanks in advance. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited Tel: (01604) 808408 Mobile: (07811) 332969 Skype: minotaur-uk ICQ: 13350579 AIM: MinotaurUK MSN: [EMAIL PROTECTED] Y!: Minotaur_Chris This email is made from 100% recycled electrons ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] A2billing problem.The system disconnects me immediatelly after asking me the PIN
Bukoka Budoka a écrit :
Hi,
i installed A2billing according to instructions. GUI works fine, i
entered a new card and i put the appopriate context in my
extensions.conf:
[callingcard]
exten = _123,1,Answer
exten = _123,2,Wait,2
exten = _123,3,DeadAGI,a2billing.php
exten = _123,4,Wait,2
exten = _123,5,Hangup
However when i enter the Calling card system, it did not ask my to put
my account-code. I found that in the a2billing.conf file there is a
option for the intro_prompt which was empty. I put there the
prepaid-enter-card-num gsm file and now when i enter the calling card
system i have the following behavior:
It asks me to enter the prepaid card number but immediatelly after it
disconnects me with a message of authentication failed - goodbye...
Why the system does not wait for the PIN number to be entered?
cid_enable=yes
Daniel
Any ideas?
Thank you,
Budoka.
My a2billing.conf is as follows:
; config file for the A2Billing Callingcard platform
; Global Database Setup
[database]
hostname=localhost
port=5432
user=a2billinguser
password=a2billing
dbname=mya2billing
;dbtype=postgres
dbtype=mysql
; configuration for the Web interface
[webui]
; Path to store the asterisk configuration files
buddyfilepath = /etc/asterisk/
; Email of the admin (not used yet)
email_admin = [EMAIL PROTECTED]
; Card lenght
len_cardnumber = 4
; Voucher lenght
len_voucher = 5
;amount of MOH class you have created in musiconhold.conf : acc_1,
acc_2... acc_10 classetc...
num_musiconhold_class = 10
;MANAGER CONNECTION PARAMETERS
manager_host = localhost
manager_username = panos
manager_secret = panos123
; Allow to display the help section inside the admin interface (YES -
NO)
show_help=YES
; Parameter of the upload
; PLEASE CHECK ALSO THE VALUE IN YOUR PHP.INI THE LIMIT IF 2MG BY DEFAULT
my_max_file_size_import = 512000
my_max_file_size = 512000 ; in bytes
; Not used yet, goal is to upload files and use them directly in the IVR
dir_store_audio = /var/lib/asterisk/sounds/a2billing
;Parameter of the upload
my_max_file_size_audio=3072000 ; in bytes
; the file type extensions allowed to be uploaded such as gsm, mp3,
wav (separate by ,)
file_ext_allow = gsm, mp3, wav
; the file type extensions allowed to be uploaded for the musiconhold
such as gsm, mp3, wav (separate by ,)
file_ext_allow_musiconhold = mp3
; ENABLE THE CDR VIEWER TO LINK ON THE MONITOR FILES (YES - NO)
link_audio_file = NO
; PATH TO LINK ON THE RECORDED MONITOR FILES
monitor_path = /var/spool/asterisk/monitor
// grant access to apache user on read mode for the directory :
chmod 755 /var/spool/asterisk/monitor/
; FORMAT OF THE RECORDED MONITOR FILE
monitor_formatfile = gsm
; Display the icon in the invoice
show_icon_invoice = YES
; Display the top frame (useful if you want to save space on your
little tiny screen )
show_top_frame = YES
;base currency define the default currency that you want to use to
setup your system (see the file /etc/asterisk/rates.inc to know the
currency code)
base_currency = usd
; currency_choose allow you to great a set of currencies to let the
customer select the most appropriate (all can be used)
currency_choose = usd, eur, cad, hkd
; configuration for the Reccurring process (cront)
[recprocess]
batch_log_file=/tmp/batch-a2billing.log
; configuration for the AGI, different configuration can be defined,
ie agi-conf1, agi-conf2, etc...
; the groupid parameter will define which process_sections to use.
Usage : DeadAGI(a2billing.php|%groupid%)
; by default agi-conf1 is used
[agi-conf1]
; the debug level
; 0=none, 1=low, 2=normal, 3=all
debug=3
; Active the logging of the application
; logging is optimized to write all the logs at once :D
logger_enable=YES
; File to log
log_file=/tmp/a2billing.log
; if YES Use Set(LANGUAGE()=fr) instead, for me it didnt work from AGI
; ### if (SETLANGUAGE_DEPRECATE==YES) $myres = $agi-agi_exec(EXEC
Set('LANGUAGE()=$language'));
setlanguage_deprecate=YES
; play the goodbye message when the user finish
say_goodbye=YES
; enable the menu to choose the language
; press 1 for English, pulsa 2 para el español, Pressez 3 pour Français
play_menulanguage=NO
; force the use of a language, if you dont want to use it leave the
option empty
; Values : ES, EN, FR, etc... (according to the audio you have install)
force_language=EN
; Introduction prompt : to specify an additional prompt to play at the
beginning of the application
; parlezplus-intro_013centimes
intro_prompt=prepaid-enter-card-num
; lenght of the cardnumber (amount of digits)
len_cardnumber=4
; Voucher lenght
len_voucher = 5
; this is the minimum amount of credit to use the application
min_credit_2call=1
; if YES it will catch the DNID and try to dial it out directly
without asking for the phonenumber to call
; value : YES, NO
use_dnid=NO
; list the dnid on which you want to avoid the use of the
Re: [Asterisk-Users] IM / presence asterisk-1.2-RC1
Here are some other files. Why asterisk send sip OPTION message to agents ? Harry 2005-11-11 11:23:10 WARNING[1639]: chan_sip.c:1045 __sip_xmit: sip_xmit of 0x81cf940 (len 477) to 192.168.0.20:-1 returned 5060: Operation not permitted Retransmitting #2 (NAT) to 192.168.0.20:5060: OPTIONS sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 80.119.11.222:5060;branch=z9hG4bK4a119599;rport From: asterisk sip:[EMAIL PROTECTED];tag=as747a6ef0 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 11 Nov 2005 10:23:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- 2005-11-11 11:23:10 WARNING[1639]: chan_sip.c:1045 __sip_xmit: sip_xmit of 0x8194ea0 (len 477) to 192.168.0.20:-1 returned 5060: Operation not permitted /// --- harry gaillac [EMAIL PROTECTED] a écrit : Sorry, Here are some files Harry --- BJ Weschke [EMAIL PROTECTED] a écrit : This is good debugging info you've listed below, but this isn't a sip debug/trace. To do that, first verify in your logger.conf file you have the following line: full = notice,warning,error,debug,verbose Then, if you needed to add anything to logger.conf, please first restart Asterisk so those new settings take effect. Then, from the CLI issue set verbose 5 and set debug 5 and finally sip debug. The repeat your dialing steps. The sip debug/trace will then be contained in /var/log/asterisk/full if /var/log/asterisk is where your log files are kept. With that, we can have a better idea of what's happening/not happening to give you the issue you're having. On 11/10/05, harry gaillac [EMAIL PROTECTED] wrote: I did it !? // Connected to Asterisk 1.2.0-rc1 currently running on serveur1 (pid = 1125) Verbosity is at least 4 serveur1*CLI sip show subscriptions Peer UserCall ID Extension Last state Type 192.168.0.21 86 f1682d8d-8f 84 Idle xpidf+xml 192.168.0.21 86 5f32aec-95b 85 Idle xpidf+xml 192.168.0.20 84 cb424ae1-e4 86 Idle xpidf+xml 192.168.0.20 84 715fac66-a9 87 Idle xpidf+xml 4 active SIP subscriptions serveur1*CLI // serveur1*CLI sip show peers Name/username HostDyn Nat ACL Port Status 87/87 192.168.0.21 D N 5060 OK (84 ms) 86/86 192.168.0.21 D N 5060 OK (97 ms) 85/85 192.168.0.20 D N 5060 OK (87 ms) 84/84 192.168.0.20 D N 5060 OK (96 ms) 4 sip peers [4 online , 0 offline] serveur1*CLI /// my sip.conf: [general] context=local ; Default context for incoming calls ; if asterisk was compiled with OSP support. realm=nxs.yi.org; Realm for digest authentication ; defaults to asterisk ; Realms MUST be globally unique according to RFC 3261 ; Set this to your host name or domain name bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=nxs.yi.org ; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ; Enable DNS SRV lookups on outbound calls tos=lowdelay; lowdelay,throughput,reliability,mincost,none maxexpirey=3600 ; Max length of incoming registration we allow defaultexpirey=1000 ; Default length of incoming/outoing registration allow=all ; First disallow all codecs musicclass=default ; Sets the default music on hold class for all SIP calls language=fr ; Default language setting for all users/peers rtptimeout=60 ; Terminate call if 60 seconds of no RTP activity tpholdtimeout=300 ; Terminate call if 300 seconds of no RTP activity useragent=Asterisk PBX ; Allows you to change the user agent string dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833 -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users
RE: [Asterisk-Users] Intel Desktop MotherBoards Unsuitable for Digium Boards
On Wed, 9 Nov 2005, Nir Simionovich - CTO wrote: but hey, I spend my nights debugging boards and sending back remarks to Intel on how to make their boards better for Asterisk. Heh, A customer of mine discovered that the onboard sound hardware on Intel Desktop boards created an echo - audio sent out came back on the mic side. He knew it was the board because even with the mic side of the headset completely disconnected from the board he could still hear the echo. Adding a separate Soundblaster board completely solved the issue. So he's been trying to report this issue to Intel. They just flatly will not admit that there IS an issue. Its funny to watch because the guy trying to report it is extremely patient, pedantic and thorough. So he's just not giving up. But so far neither are Intel... Steve ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zaptel T1 Timing Source
On Wed, 9 Nov 2005, Kevin Bockman wrote: Waldo Rubinstein wrote: One T1 is with one carrier, who provides timing signal. The other 3 T1s are from a different carrier, all sharing the same timing signal. Based on this, I have in /etc/zaptel.conf something like: I expect that you'll find that the two carriers are actually in clock sync anyway. Steve ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sip ignores context definition?
Hi All, I've a very strange error. I've configured a Cisco gw with * and when an incoming call is arriving from the Cisco to * asterisk will always put the call in the default context (ignoring the part in the [Cisco]) I'm attaching my conf files: [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) disallow=all allow=alaw allow=gsm allow=ulaw context = from-trunk ; Send unknown SIP callers to this context callerid = Unknown [Cisco] type=user/friend/peer (tried all options) port=5060 host=myip context=from-Cisco disallow=all allow=alaw allow=ulaw qualify=yes autocreatepeer=yes (with and without this option, in here and in the general setting) nat=no canreinvite=no on Asterisk Console I see (with Verbose 9): Executing AbsoluteTimeout(SIP/myip-b6895f10, 15) in new stack -- Set Absolute Timeout to 15 -- Executing Congestion(SIP/myip-b6895f10, ) in new stack -- Executing AbsoluteTimeout(SIP/myip-b6895f10, 15) in new stack -- Set Absolute Timeout to 15 -- Executing Congestion(SIP/myip-b6895f10, ) in new stack which is my default context: [from-trunk] exten = _.,1,AbsoluteTimeout(15) exten = _.,2,Congestion exten = _.,3,Hangup [from-Cisco] exten = s,1,Answer exten = s,2,Dial($bla) exten = s,3,Hangup Thanks! ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Consultant
Im willing to help for free contact me via msn messenger my id is [EMAIL PROTECTED] On 11/11/05, chawki hammoud [EMAIL PROTECTED] wrote: Hi: I have been posting this issue for over a month and I am not irritated. I appreciate all the users who helped before and I thank in advance any offer for help including reasonable paid time. You may sent me your price me list at [EMAIL PROTECTED] Regards; chawki --- Rob Lith [EMAIL PROTECTED] wrote: Sounds like a good deal to me. If you want free answers don't sound so irritated that you haven't got a reply in $0 time. :) Rob On 11/9/05, chawki hammoud [EMAIL PROTECTED] wrote: The only pointer I got is a $50/hr Mark phillip offered. I can make VOIP calls between my Asterisk server and my VOIP provider using sip channel without a problem. But when I attempt to make a call using IAX, the call get accepted and then get a hangup message: This is the message I get when I attempt to make an IAX call: Executing Dial(OSS/dsp, IAX2/callshopcompany/0017046872001) in new stack -- Called callshopcompany/0017046872001 -- Call accepted by 213.61.187.150 http://213.61.187.150 (format gsm) -- Format for call is gsm -- Hungup 'IAX2/callshopcompany/1' == No one is available to answeer at this time The call get accepted, but it seems there is no acknowledgement from my server to receive the call from the provider. Thanks; --- Mark Phillips [EMAIL PROTECTED] wrote: He did. And he got pointers to the relevant howto's. Matt Riddell wrote: chawki hammoud wrote: Hi: I posted my problem several times about being unable to make IAX calls from my Asterisk box to another IAX server without luck. So, what's your problem? Post some details. -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com ___ --Bandwidth and Colocation sponsored by Easynews.comhttp://Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Yahoo! FareChase: Search multiple travel sites in one click. http://farechase.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com http://Easynews.com-- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards, Angelito Manansala www.voicefidelity.net Mobile: +639175425807 DID: (+63) 44 7906770 msn: [EMAIL PROTECTED] skype: bulcrack ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX2 calls being droppped
Hi there I am using an IAX2 softphone built from the IaxClient library dialing into Meetme conferences. It works fine most of the time, but sometimes calls are being dropped and this error is given: Chan_iax2.c:1480 attempt_transmit: Max retries exceeded to host 146.18.3.5 on IAX2/[EMAIL PROTECTED]:4569/3 (type = 2, subclass = 1024, ts=655380, seqno=177) This error is pretty erratic. It mostly happens the first time you try to dial, but also seems to sometimes be happening in the middle of a conversation. Any ideas what the problem could be? Many thanks Steven Langley attachment: winmail.dat___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CAPI problem under gentoo with AVM C2 - asterisk claims CAPI not installed
Hi! I am running Asterisk (1.0.8) on gentoo (2.6.13-gentoo-r5), with AVM C2 ISDN card. capiiinit works OK, capiinfo shows card is up and running with CAPI OK, but asterisk refuses to load the capi-cm module (chan_capi-cm, 0.5.4) giving the warning CAPI not installed, CAPI disabled!. Any hints of where to look next? Julf ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Crashing (high load issues)
On Wed, 9 Nov 2005, Kyle Hagan wrote: We purchased a new Dual Xeon 3ghz, 2gb ram to upgrade our 3ghz Pentium 1gb ram, that has been having load issues due to our growing company. We are having problems... We use a predictive dialer that we custom programmed in perl. It basically drops, moves, files into the callout directory and uses queues to transfer to agents when someone picks up. It has been working pretty good, except we now have 50+ dialers on the system taking calls. The system dials 2-4 per available agent every 3-5 seconds based on, calls ringing and available agents. We can keep them to about 8-20 seconds between calls. But the number of ringing lines is causing load issues. Hence the new server. We put Fedora Core 4 on with now problem. We were running 2 t1's in the beginning of the day just to make sure the system was running good. We finally put it on 8 t1's and the system ran great for about 4 hours. Then the load started going up and up until the server just locked completely. I could not get much information from the server. The lead went to 170+ before it locked. Asterisk was showing 99% cpu usage at crash. I have some information that the log had in it just before the crash. There was something about cpu3 soft lockup and page fault messages. If someone can help I will post the log tomorrow when I get into work. We had to switch back to the old server with the load issues. Some other information about the servers follows: We are running a separate slim server to stream moh. The predictive server is a separate pc connecting via manager interface for agent information, available, busy and callerid of the person they are talking to We have a script (perl) running on the Asterisk server to move the callout files into the callout directory that are created via a web POST via apache, the script checks for files in a temp directory and move the files into the callout directory. Hi Kyle, I'd simply say that you have overloaded that machine. We use boxes like that for a similar outbound dial setup. I don't think I'd attempt to go past 4 E1s (120 lines) which would be 5 T1s. If the box is running hard like that the load average will sit around 7 or so, still fair amount of spare CPU but there is no way an Asterisk box will run well with the CPU anything like maxed out. Our site has 250 agents or so and the work is currently spread over 6 servers with 3 E1 PRIs on each. Each box makes around 3000 to 5000 call attempts per hour. If you are getting very high load average - are you recording calls? It would REALLY not be a good idea to use the m option to Monitor to mix calls on the fly - the soxmix processes will accumulate and accumulate. Your Perl dialler also needs to be more sympathetic to the machine capacity and back off when the server is getting overloaded. Otherwise you are certain to drive it into the ground. Steve ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Siemens optiPoint 420 phone and Asterisk SOS
Stephen Arulraj wrote: Has anyone got these SIP firmwares for the Siemens IP Phones? Would appreciate it. Thanks and regards, Stephen Stephen You can download a copy from my website here:- http://chaz6.com/static/files/sip_v2_3_14.app Regards -- Chris Hills I.T. Services North East Worcestershire College ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CAPI problem under gentoo with AVM C2 - asterisk claims CAPI not installed
Hi! I am running Asterisk (1.0.8) on gentoo (2.6.13-gentoo-r5), with AVM C2 ISDN card. capiiinit works OK, capiinfo shows card is up and running with CAPI OK, but asterisk refuses to load the capi-cm module (chan_capi-cm, 0.5.4) giving the warning CAPI not installed, CAPI disabled!. Any hints of where to look next? Any further messages when starting Asterisk with higher verbose level? Correct permissions to access /dev/capi20 for Asterisk? Armin ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400P + FXO module = PSTN woes
On Friday 11 November 2005 02:25, Jacques Beyers wrote:
I have installed Asterisk 1.0.9 on FC3. I have recently installed a Digium
TDM400P and a red Digium FXO module, which I hope to connect to the PSTN so
I can make outbound calls. The FXO card is installed in port 1, and the
telephone jack is inserted into port 1.
No matter how I try, I cannot get Asterisk to dial out. Please could
someone point me in the right direction.
Here are a bit of information on how my system is configured:
[ very useful, concise and correct data snipped ]
Jaques, your post should be bronzed and placed at the desk of anyone
subscribed to this channel. You've created what I consider the perfect
help, I can't get something working email suitable for a mailing list such
as this one. Now I only hope that my help does indeed help you. :-)
pridialplan = local
prilocaldialplan = local
nationalprefix = 0
internationalprefix = 00
None of that matters, as you don't have a PRI.
exten = _0.,1,Dial(${TRUNK}/${EXTEN:1},70,T,t)
I don't think this is killing you, but the T,t should be Tt -- all flags
just get lumped together, as you did with your SIP example above. Also, if
you want people you call to be able to transfer (very odd scenario, but
possible), then leave the 't' in there. Most people don't allow the people
they've called to be able to use their transfer function.
Now because ZAPTEL moans on boot-up, I have removed ZAPTEL from the startup
services and edited my /etc/rc.d/rc.local file to load everything.
What do you mean that zaptel moans on boot-up? What's the exact error
message?
hisax 598301 0
I've noticed that a lot of Asterisk boards get mistakenly identified as hisax
boards by the hotplug subsystem, and is likely why zaptel is 'moaning' on
boot-up. I would either remove the hisax driver or try to put it in your
hotplug blacklist (/etc/hotplug/blacklist) so that it does not automatically
load.
This *could* be the source of your dialout troubles.
Now Asterisk starts up fine, and when I dial, using X-Ten Lite, Asterisk
shows that it is dialling the ZAP/1, but the number I am dialling never
rings.
I imagine that you are dialing 0{telephone number} then, correct?
I am probably doing something stupid, so if anyone can shed some light, I
will appreciate it.
Nothing stupid... just a couple of other questions:
Do you use any wctdm kernel options? I see that you're from South Africa, so
wctdm may need a different country identification in order to properly
interface to your telephone network? (opermode kernel module parameter, and
also 'loadzone' in zaptel.conf.)
Does the system work correctly when people call in on the line that the FXO
module is plugged in to?
When you attempt to call out, do you hear Asterisk trying to dial if you
listen in on another regular phone extension connected to the same FXO
port?
Again, thank you so much for providing a *shining* example of how to write an
email to a large mailing list asking for help. You described your problem
clearly, provided the RIGHT information concisely and were generally
all-around polite. Thank you.
-A.
___
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Re: [Asterisk-Users] IM / presence asterisk-1.2-RC1
When the polycom ip300 phone (1.6.2) send registration SUBSCRIBE message is sent to buddies from directory file so NOTIFY is received from these one. When I want to change status the ip phone don't send NOTIFY to subscriber unlike SER which is a proxy!!! Why? Harry --- harry gaillac [EMAIL PROTECTED] a écrit : Here are some other files. Why asterisk send sip OPTION message to agents ? Harry 2005-11-11 11:23:10 WARNING[1639]: chan_sip.c:1045 __sip_xmit: sip_xmit of 0x81cf940 (len 477) to 192.168.0.20:-1 returned 5060: Operation not permitted Retransmitting #2 (NAT) to 192.168.0.20:5060: OPTIONS sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 80.119.11.222:5060;branch=z9hG4bK4a119599;rport From: asterisk sip:[EMAIL PROTECTED];tag=as747a6ef0 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 11 Nov 2005 10:23:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- 2005-11-11 11:23:10 WARNING[1639]: chan_sip.c:1045 __sip_xmit: sip_xmit of 0x8194ea0 (len 477) to 192.168.0.20:-1 returned 5060: Operation not permitted /// --- harry gaillac [EMAIL PROTECTED] a écrit : Sorry, Here are some files Harry --- BJ Weschke [EMAIL PROTECTED] a écrit : This is good debugging info you've listed below, but this isn't a sip debug/trace. To do that, first verify in your logger.conf file you have the following line: full = notice,warning,error,debug,verbose Then, if you needed to add anything to logger.conf, please first restart Asterisk so those new settings take effect. Then, from the CLI issue set verbose 5 and set debug 5 and finally sip debug. The repeat your dialing steps. The sip debug/trace will then be contained in /var/log/asterisk/full if /var/log/asterisk is where your log files are kept. With that, we can have a better idea of what's happening/not happening to give you the issue you're having. On 11/10/05, harry gaillac [EMAIL PROTECTED] wrote: I did it !? // Connected to Asterisk 1.2.0-rc1 currently running on serveur1 (pid = 1125) Verbosity is at least 4 serveur1*CLI sip show subscriptions Peer UserCall ID Extension Last state Type 192.168.0.21 86 f1682d8d-8f 84 Idle xpidf+xml 192.168.0.21 86 5f32aec-95b 85 Idle xpidf+xml 192.168.0.20 84 cb424ae1-e4 86 Idle xpidf+xml 192.168.0.20 84 715fac66-a9 87 Idle xpidf+xml 4 active SIP subscriptions serveur1*CLI // serveur1*CLI sip show peers Name/username HostDyn Nat ACL Port Status 87/87 192.168.0.21 D N 5060 OK (84 ms) 86/86 192.168.0.21 D N 5060 OK (97 ms) 85/85 192.168.0.20 D N 5060 OK (87 ms) 84/84 192.168.0.20 D N 5060 OK (96 ms) 4 sip peers [4 online , 0 offline] serveur1*CLI /// my sip.conf: [general] context=local ; Default context for incoming calls ; if asterisk was compiled with OSP support. realm=nxs.yi.org; Realm for digest authentication ; defaults to asterisk ; Realms MUST be globally unique according to RFC 3261 ; Set this to your host name or domain name bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=nxs.yi.org ; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ; Enable DNS SRV lookups on outbound calls tos=lowdelay; lowdelay,throughput,reliability,mincost,none maxexpirey=3600 ; Max length of incoming registration we allow defaultexpirey=1000 ; Default length of incoming/outoing registration allow=all ; First disallow all codecs musicclass=default ; Sets the default music on hold class for all SIP calls language=fr ; Default language setting for all users/peers rtptimeout=60 ; Terminate call if
Re: [Asterisk-Users] asterisk 1.0.10?
Great! tnx matt!On 11/11/05, Matt Florell [EMAIL PROTECTED] wrote: It's CVS v1-0. Digium has said that they will do a release of 1.0.10at the same time they release 1.2.I highly recommend upgrading to this if you are still on the 1.0 tree.It has a lot of bug fixes, and the new v2 firmware telco cards from Digium run much better on it than they do on 1.0.9.If you want it now, just checkout from CVS like this:cvs checkout -r v1-0 zaptel libpri asterisk asterisk-addons asterisk-soundsMATT--- On 11/10/05, Mark Quitoriano [EMAIL PROTECTED] wrote: in the Changelog on http://ftp.digium.com/pub/asterisk/ChangeLog there's a asterisk 1.0.10 which i can't find anywhere, any hints?--snip from ChangeLog--Asterisk 1.0.10-- chan_local-- In releases 1.0.8 and 1.0.9 , the Local channels that are created wouldnot be masqueraded into the new channel type. This has now been fixed.-- chan_sip-- The 'insecure' options have been changed to support matching peersby IP only, not requiring authentication on incoming invites, or both. Before,to not require authentication on incoming invites also required matchingpeers based on IP only.-- chan_zap -- Before, call waiting could occur during the initial ringing on the line.This has now been fixed.-- app_disa-- We will now not set the accountcode if one is not supplied. -- app_meetme-- If the first caller into a conference hangs up while being prompted forthe conference pin number, the conference will no longer be held open.-- app_userevent-- Events created with this application were indicated as a call event instead of a user event. This made the user event permissionsnot work correctly.-- app_voicemail-- When using the externpass option for voicemail, the password will be immediately updated in memory as well, instead of having to wait forthe next time the configuration is reloaded.-- app_zapras-- We now ensure buffer policy is restored after RAS is done with a channel.This could cause audio problems on the channel after zapras is donewith it.-- res_agi-- We now unmask the SIGHUP signal before executing an AGI script. This fixes problems where some AGI scripts would continue running long afterthe call is over.-- extensions-- A potential crash has been fixed when calling LEN() to get the length of a string that was 80 characters or larger.-- logger-- The Asterisk logger will automatically detect when a log file needs tobe rotated. However, this feature could put Asterisk in a nasty loop that would result in a crash.-- general-- Added man pages for astgenkey, autosupport, and safe_asterisk--end of snip-- -- Regards, Mark Quitoriano, CCNA http://www.atamanetworks.com Fan the flame...http://www.spreadfirefox.com/?q=user/registerr=19441 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Regards,Mark Quitoriano, CCNAhttp://www.atamanetworks.comFan the flame... http://www.spreadfirefox.com/?q=user/registerr=19441 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CAPI problem under gentoo with AVM C2 - asterisk claims CAPI not installed
Armin Schindler wrote: Correct permissions to access /dev/capi20 for Asterisk? Duh! Of course it had to be something as trivial as that! Thanks!! Julf ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ASTERISK + POLYCOM IP PHONES
Hello, I try to setup presence with polycom ip phones ip300 (1.6.2) . I added buddies in directory files all is right for registration subscription notification but when i want to change status notify message is not sent to subscribers ? I don't understand ! Regards Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400P + FXO module = PSTN woes
I have installed Asterisk 1.0.9 on FC3. I have recently installed a Digium
TDM400P and a red Digium FXO module, which I hope to connect to the PSTN so
I can make outbound calls. The FXO card is installed in port 1, and the
telephone jack is inserted into port 1.
No matter how I try, I cannot get Asterisk to dial out. Please could someone
point me in the right direction.
Here are a bit of information on how my system is configured:
/proc/interrupts
cat /proc/interrupts
CPU0
0:536844 XT-PIC timer
1:8 XT-PIC i8042
2:0 XT-PIC cascade
3:0 XT-PIC ehci_hcd:usb1
4:0 XT-PIC ohci_hcd:usb3
8:1 XT-PIC rtc
9:0 XT-PIC acpi, ohci_hcd:usb2
10: 504097 XT-PIC wctdm
11: 2963 XT-PIC eth0
12: 110 XT-PIC i8042
14: 4313 XT-PIC ide0
NMI: 0
ERR: 0
/etc/zaptel.conf
fxsks=1
loadzone=za
defaultzone=za
/etc/asterisk/zapata.conf
[channels]
language=en
immediate=no
context=default
usecallerid=yes
callprogress=no
transfer=yes
threewaycalling=yes
callwaitingcallerid=yes
callwaiting=yes
cancallforward=yes
musiconhold=default
pridialplan = local
prilocaldialplan = local
nationalprefix = 0
internationalprefix = 00
;Echo control
echocancel=yes
echotraining=yes
echocancelwhenbridged=yes
;Adjust Volume
rxgain=0.0
txgain=0.0
; This is for the FXS Digium card
signalling=fxs_ks
echocancel=yes
echocancelwhenbridged=yes
echotraining=400
callerid=asreceived
group=1
context=default
channel = 1
/etc/asterisk/extensions.conf
[general]
static=no
writeprotect=no
[globals]
TRUNK=Zap/1
include = daytime|9:00-17:00|mon-fri|*|*
[local]
ignorepat = 0
include = sip
[default]
include = sip
exten = _X.,1,wait(1)
exten = _X.,2,Answer()
exten = _X.,3,Goto(default,s,1)
exten = s,1,Answer()
exten = s,2,NoOp(${CALLERID})
exten = s,3,Goto(sip,2100,1)
exten = t,1,Goto(default,s,4)
exten = i,1,Playback(invalid)
[sip]
exten = 2100,1,Answer
exten = 2100,2,wait(1)
exten = 2100,3,Dial(SIP/2101,20,tr)
exten = 2100,4,Voicemail(u2100)
exten = 2100,103,Voicemail(b2100)
exten = 2100,104,hangup
exten = 2101,1,Answer
exten = 2101,2,wait(1)
exten = 2101,3,Dial(SIP/2101,20,tr)
exten = 2101,4,Voicemail(u2101)
exten = 2101,102,Voicemail(b2101)
exten = 2101,103,hangup
; This is where we handle the outgoing calls
exten = _0.,1,Dial(${TRUNK}/${EXTEN:1},70,T,t)
exten = _0.,2,Hangup
Now because ZAPTEL moans on boot-up, I have removed ZAPTEL from the startup
services and edited my /etc/rc.d/rc.local file to load everything.
/etc/rc.d/rc.local
#!/bin/sh
#
# This script will be executed *after* all the other init scripts.
# You can put your own initialization stuff in here if you don't
# want to do the full Sys V style init stuff.
touch /var/lock/subsys/local
/sbin/modprobe zaptel
/sbin/modprobe wctdm ; I have also tried wcfxs
/sbin/service zaptel start
sleep 1
/sbin/service asterisk start
sleep 1
lsmod
ModuleSizeUsed by
wcfxs 32288 0
zaptel208388 1 wcfxs
md5 40331
ipv6 262977 10
autofs4 28229 0
dm_mod57333 0
video 15685 0
button66090
battery 92850
ac48050
ohci_hcd 26081 0
ehci_hcd 40013 0
i2c_sis96x54450
i2c_core 21313 1 i2c_sis96x
hisax 598301 0
crc_ccitt 21132 zaptel,hisax
isdn 148673 1 hisax
slhc 68491 isdn
r8169 29005 0
ext3 130633 2
jbd 83161 1 ext3
Now Asterisk starts up fine, and when I dial, using X-Ten Lite, Asterisk
shows that it is dialling the ZAP/1, but the number I am dialling never
rings.
I am probably doing something stupid, so if anyone can shed some light, I
will appreciate it.
Based on what you are showing above, the
lsmod
Module SizeUsed by
wcfxs 32288 0
zaptel 208388 1 wcfxs
does not make sense. Your script does /sbin/modprobe wctdm, but the lsmod
shows wcfxs. The wctdm driver _is_ required for the red fxo module to
function, and the wcfxs is associated with the green fxs module. It would
appear your problem is associated with this driver load stuff.
Do the modprobes by had (not a script) and watch the results. If I do
a 'modprobe zaptel' followed by a 'modprobe wctdm', I see:
Module Size Used by
wctdm 33728 0
zaptel209540 1 wctdm
indicating the zaptel driver is used by one wctdm driver.
The do a 'ztcfg -vv' and you should see something like:
Zaptel Configuration
Re: [Asterisk-Users] Softphone with Lotus Notes support?
Message: 5Date: Fri, 11 Nov 2005 08:11:09 +0100From: Stefan-Michael. Guenther (in-put GbR) [EMAIL PROTECTED]Subject: [Asterisk-Users] Softphone with Lotus Notes support?To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED]Content-Type: text/plain;charset=utf-8Hi,has anyone of you heard of a softphone or client that support Lotus Notes? I just want to click on the telephone number of an account and my hard- orsoftphone should get the call.Something similar to the outlook clients fromThirdlane( http://www.thirdlane.com/opensource.htm#dialer)or EyePMedia (http://www.eyepmedia.com/)Thanks for any suggestions,Stefan-- in-put GbR - Das Linux-SystemhausStefan-Michael GuentherMoltkestrasse 49 D-76133 KarlsruheTel./Fax : +49 (0)721 / 83044 - 98/93http://www.in-put.de SchulungenInstallationenBeratung Support As someone who uses and develops Notes and Asterisk on an almost daily basis, I can tell you two things: 1. Technically, all softphones 'support' Lotus Notes- if Notes knew how to pass them a number, they'd dial it. Notes, however, especially in it's address book, doesn't support anyone. 2. Since Notes is one heck of a lot more programmer-friendly than Outlook/Exchange will ever be (I'm not biased, really..), adding such functionality to your address books would be a snap. Simply pick a softphone you like that supports any sort of API to accept dialing, preferably one that supports URI dialing (DIAX comes to mind, but it's really up to you), and modify the design of your address book (personal or system) to turn the Phone Number field into a link hotspot. Click, done. What I have done goes another step farther into the dark side- since Domino natively supports LDAP, I wrote a script to pull all names and numbers (10,000 of them) out of Domino using LDAP, drop them into a MySQL database, then re-present it on my Cisco phones as a directory, and via Apache as a web service, which supports click to dial via call files in Asterisk. I'm now working on an agent for individual user Personal Address Books to 'synchronize' with this directory structure, so I can combine a user's personal contacts with the main 'corporate' directory when they are searching for contacts. I'd offer it here, and someday I might, however, since each corporate Domino enviromnet is so very different, I have to basically restructure the code for each implementation- and I havent made the code mature enough to have anyone other than me do it. So for now, it's a single-client application. But, I'd be happy to share implementation details with anyone who wants to email me offline. -pbd ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk-Users Digest, Vol 16, Issue 85
I have installed Asterisk on SuSE 10.0 with an active firewall/NAT filter. The server has connection to my own Intranet (private IP) and to Internet Everything works well for clients behind and in front-of the firewall but they can not communicate with each other. Signalling gets through but the audio gets blocked by the firewall/NAT. So, I open-up ports 10.000 -to- 20.000 in the fw so that the udp/rtp packages cuold get through but it has not been successful. I am using xlite for clients and have no pot cards installed ( digium fxo,fxs, etc). Does anyone knows what else to do? Has anyone come accross (and solved) this type of problem? Firewall configuration is as follows: FW_DEV_EXT=eth-id-00:0d:87:5c:44:e5 #eth1 FW_DEV_INT=eth-id-00:06:4f:0e:ca:99 eth-id-00:40:f4:9f:12:25 #eth0 wlan0 FW_ROUTE=yes FW_MASQUERADE=yes FW_MASQ_DEV=$FW_DEV_EXT FW_MASQ_NETS=192.168.100.0/255.255.255.0 FW_SERVICES_EXT_TCP=53 http https ssh FW_SERVICES_EXT_UDP=5060 5061 53 FW_SERVICES_INT_TCP=21 3128 5056 53 5801 5901 80 8080 epmap http microsoft-ds netbios-ssn smtp ssh FW_SERVICES_INT_UDP=5060:5075 53 bootps netbios-dgm netbios-ns FW_SERVICES_INT_RPC=mountd nfs nfs_acl nlockmgr portmap status ypbind FW_SERVICES_ACCEPT_EXT=0/0,udp,5060:5075 FW_TRUSTED_NETS=192.168.100.0/255.255.255.0 FW_FORWARD=0/0,192.168.100.0/255.255.255.0,udp,5060 FW_FORWARD=0/0,192.168.100.0/255.255.255.0,udp,1 FW_FORWARD=192.168.100.0/255.255.255.0,0/0,udp,1 Sip Configuration: [general] bindport=5060 bindaddr=0.0.0.0 srvlookup=no externrefresh=10 externip=201.208.246.178 nat=yes localnet=192.168.100.0/255.255.255.0; RTP configuration: [general] rtpstart=1 rtpend=2 rtpchecksums=yes Regards, Enrique Leon ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Digium TDM Revision I Card
We had a Rev I card that did not work. We sent it back to Digium and had it reflashed back to H. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rob LithSent: Friday, November 11, 2005 1:40 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Digium TDM Revision I Card I had a customer have problems with REV I and J cards get snap, crackel pop noise but not on older REV F or H cards. He upgraded to 1.2.0-rc1 and to quote: "Asterisk 1.2.0-rc1was Released on2005-11-08 22:40. as well as zaptel 1.2.0-rc1. (First non Beta version) I compiled it and it works very nicely, without any Snaps,Cracles or Pops, even though zaptel still detects the REV-J as an REV-I." Regards Rob On 11/11/05, Shaun Singh [EMAIL PROTECTED] wrote: Is anyone using version I TDM mothercard? I am currently using 2 revision Hcards and they are working fine. I recently purchased a revision I card from an online vendor which didn't work and the replacement from Digium (anotherrevision I) didn't work either.Shaun Singh, ManagerTravelwave1655 Dufferin Street, Suite 201Toronto, ON M6H 3L9Tel: (416) 652-1212 Ext 101 Fax: (416) 652-7073Website: www.travelwave.ca___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-usersThe contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] A2billing problem.The system disconnects me immediatelly after asking me the PIN
Aíeee Caramba, intro_prompt is really to have a customized message at
the initiation of the call. please read the comment above in the
configuration file.
Try from shell to run the script and press enter until you get back to
the shell,
then you will see perhaps if an error occur on the AGI. if you dont
see anything send me the output debug.
Kind regards,
/Areski
On 11/11/05, Administrator TOOTAI [EMAIL PROTECTED] wrote:
Bukoka Budoka a écrit :
Hi,
i installed A2billing according to instructions. GUI works fine, i
entered a new card and i put the appopriate context in my
extensions.conf:
[callingcard]
exten = _123,1,Answer
exten = _123,2,Wait,2
exten = _123,3,DeadAGI,a2billing.php
exten = _123,4,Wait,2
exten = _123,5,Hangup
However when i enter the Calling card system, it did not ask my to put
my account-code. I found that in the a2billing.conf file there is a
option for the intro_prompt which was empty. I put there the
prepaid-enter-card-num gsm file and now when i enter the calling card
system i have the following behavior:
It asks me to enter the prepaid card number but immediatelly after it
disconnects me with a message of authentication failed - goodbye...
Why the system does not wait for the PIN number to be entered?
cid_enable=yes
Daniel
Any ideas?
Thank you,
Budoka.
My a2billing.conf is as follows:
; config file for the A2Billing Callingcard platform
; Global Database Setup
[database]
hostname=localhost
port=5432
user=a2billinguser
password=a2billing
dbname=mya2billing
;dbtype=postgres
dbtype=mysql
; configuration for the Web interface
[webui]
; Path to store the asterisk configuration files
buddyfilepath = /etc/asterisk/
; Email of the admin (not used yet)
email_admin = [EMAIL PROTECTED]
; Card lenght
len_cardnumber = 4
; Voucher lenght
len_voucher = 5
;amount of MOH class you have created in musiconhold.conf : acc_1,
acc_2... acc_10 classetc...
num_musiconhold_class = 10
;MANAGER CONNECTION PARAMETERS
manager_host = localhost
manager_username = panos
manager_secret = panos123
; Allow to display the help section inside the admin interface (YES -
NO)
show_help=YES
; Parameter of the upload
; PLEASE CHECK ALSO THE VALUE IN YOUR PHP.INI THE LIMIT IF 2MG BY DEFAULT
my_max_file_size_import = 512000
my_max_file_size = 512000 ; in bytes
; Not used yet, goal is to upload files and use them directly in the IVR
dir_store_audio = /var/lib/asterisk/sounds/a2billing
;Parameter of the upload
my_max_file_size_audio=3072000 ; in bytes
; the file type extensions allowed to be uploaded such as gsm, mp3,
wav (separate by ,)
file_ext_allow = gsm, mp3, wav
; the file type extensions allowed to be uploaded for the musiconhold
such as gsm, mp3, wav (separate by ,)
file_ext_allow_musiconhold = mp3
; ENABLE THE CDR VIEWER TO LINK ON THE MONITOR FILES (YES - NO)
link_audio_file = NO
; PATH TO LINK ON THE RECORDED MONITOR FILES
monitor_path = /var/spool/asterisk/monitor
// grant access to apache user on read mode for the directory :
chmod 755 /var/spool/asterisk/monitor/
; FORMAT OF THE RECORDED MONITOR FILE
monitor_formatfile = gsm
; Display the icon in the invoice
show_icon_invoice = YES
; Display the top frame (useful if you want to save space on your
little tiny screen )
show_top_frame = YES
;base currency define the default currency that you want to use to
setup your system (see the file /etc/asterisk/rates.inc to know the
currency code)
base_currency = usd
; currency_choose allow you to great a set of currencies to let the
customer select the most appropriate (all can be used)
currency_choose = usd, eur, cad, hkd
; configuration for the Reccurring process (cront)
[recprocess]
batch_log_file=/tmp/batch-a2billing.log
; configuration for the AGI, different configuration can be defined,
ie agi-conf1, agi-conf2, etc...
; the groupid parameter will define which process_sections to use.
Usage : DeadAGI(a2billing.php|%groupid%)
; by default agi-conf1 is used
[agi-conf1]
; the debug level
; 0=none, 1=low, 2=normal, 3=all
debug=3
; Active the logging of the application
; logging is optimized to write all the logs at once :D
logger_enable=YES
; File to log
log_file=/tmp/a2billing.log
; if YES Use Set(LANGUAGE()=fr) instead, for me it didnt work from AGI
; ### if (SETLANGUAGE_DEPRECATE==YES) $myres = $agi-agi_exec(EXEC
Set('LANGUAGE()=$language'));
setlanguage_deprecate=YES
; play the goodbye message when the user finish
say_goodbye=YES
; enable the menu to choose the language
; press 1 for English, pulsa 2 para el español, Pressez 3 pour Français
play_menulanguage=NO
; force the use of a language, if you dont want to use
Re: [Asterisk-Users] DSL router with QOS
A Linksys WRT54G runs for about $60. I think it supports QoS out of the box, but flash it with 3rd party firmware (i.e. Sveasoft) to get a bunch of extra features. Note: The latest version, version 5, of the WRT54G only has half the memory of the previous versions and there is no 3rd party firmware that yet runs on the platform. (See http://www.sveasoft.com/modules/phpBB2/viewtopic.php?t=9344 ) Get an older version. On Thu, 2005-11-10 at 15:45 -0500, Keith Schmidt wrote: Any recommendations on an ADSL router with QOS for VOIP built in? Anything sub $500 would be great. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DISA multiple calls with single dialup
[DISA-context] exten = 204117733,1,DigitTimeout(8) exten = 204117733,2,ResponseTimeout(15) exten = 204117733,3,DISA(no-password,default) exten = 204117733,4,Hangup We succesfully implement a dialup gateway with the following. What I now wish to do is to be able to make multiple telephone calls. So I would like to terminate an asterisk call with say a * and then be returned to dialtone. How would I define that rule? Thanks. -- Eric Smith ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Cisco DHCP and Polycom boot server
Hi Peter - When you set up the DHCP pool in Cisco you need to use syntax like: -- option 66 ascii a.b.c.d Thanks! I guess maybe I didn't explain very well. I did get this far, and this seems to work well, if I manually set the phone to read an ascii string. I'm being really picky here, though. I want Joe Schmoe user to be able to plug in the phone and have it get provisioned without having to make any changes to the phone (like selecting to use a DCHP string rather than an IP). With all the Cisco phones that I have, the default setting has been to read the tftp-boot-server parameter as an IP rather than as a string, and I can't get this to work with Cisco DHCP. Maybe somebody else has, though? Thanks, Noah -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Noah Miller Sent: Thursday, 10 November 2005 9:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Cisco DHCP and Polycom boot server Hi - I've been trying to set up my Polycom phones to get the boot server info (tftp-server-address) from DHCP on a Cisco router. I've previously just specified it manually on the phone, and that works well enough, but I need to change now (because of the number and geographic locations of the phones). I can actually get it to work just fine (using option 66 on the Cisco router), if I change the DHCP menu on the Polycom phone to show BootSrv Type: String. That's great, but that's not a default setting, and I don't want to have to change any settings on the phone. I want the phones to be able to provision fully, out-of-the-box, with nothing but the info from DHCP. If I leave the default setting (BootSrv Type: IP Address), and tell the Cisco router to send the boot serverinfo as an IP rather than as a string, nothing happens. The phone just says Could not contact boot server, using existing configuration, but according to the FTP logs and ethereal, the phone doesn't actually try to contact the boot server at all. I've tried various version of the bootrom, but nothing has worked so far. Has anybody gotten this to work? (Cisco router DHCP and Polycom boot server) Thanks, Noah ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zyxel omni.net USB ISDN works with Asterisk
Hi Gabor, I'm not sure about USB ISDN adapters but I'm using an AVM Fritz ISDN PCMCIA card with asterisk and chan_capi very sucessfully. The notebook is in a remote location and is solar powered. You'll find that these cards are cheap on Ebay - its a German card and you'll probably find lots of them on www.ebay.de Derek Gabor Horvath wrote: Dear Asterisk users, Can you tell me is the Zyxel omni.net USB ISDN adapter works with Linux, and more specifically, with Asterisk chan_capi? I built an Asterisk PBX test environment on my laptop with Asterisk Management Portal, one hardphone, one ATA, and one softphone. I would connect the whole thing to an ISDN (Euro) line, but because of my laptop, I can use only USB or PCMCIA solutions. Gabor ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] sip ignores context definition?
What version are you running, and is your [Cisco]
definition the last one in the file? I have the same problem with 1.0.7,
and the ugly fix I came up with was to add a dummy entry as the last sip
entry.
B. J.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Sent: Friday, November 11, 2005
4:48To: [EMAIL PROTECTED]Subject:
[Asterisk-Users] sip ignores context definition?
Hi All,
I've a very strange error.
I've configured a Cisco gw with * and when an incoming call is arriving from
the Cisco to * asterisk will always put the call in the default context
(ignoring the part in the [Cisco])
I'm attaching my conf files:
[general] port = 5060
; Port to bind to (SIP is 5060) bindaddr =
0.0.0.0 ; Address to bind to (all addresses on machine)
disallow=all allow=alaw allow=gsm allow=ulaw context =
from-trunk ; Send unknown SIP callers to this context callerid = Unknown
[Cisco] type=user/friend/peer
(tried all options) port=5060 host=myip context=from-Cisco
disallow=all allow=alaw allow=ulaw qualify=yes
autocreatepeer=yes (with and without this option, in here and in the
general setting) nat=no canreinvite=no
on Asterisk Console I see (with
Verbose 9): Executing AbsoluteTimeout("SIP/myip-b6895f10", "15") in new
stack -- Set Absolute Timeout to 15 --
Executing Congestion("SIP/myip-b6895f10", "") in new stack --
Executing AbsoluteTimeout("SIP/myip-b6895f10", "15") in new stack
-- Set Absolute Timeout to 15 -- Executing
Congestion("SIP/myip-b6895f10", "") in new stack
which is my default context:
[from-trunk] exten = _.,1,AbsoluteTimeout(15) exten =
_.,2,Congestion exten = _.,3,Hangup
[from-Cisco] exten =
s,1,Answer exten = s,2,Dial($bla) exten = s,3,Hangup
Thanks!
___
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Re: [Asterisk-Users] Result branching in AEL
n+101 feature is deprecated and is no longer supported in Asterisk.
All applications are modified to set exit status variable. Use something
like
VoiceMail(b${EXTEN});
if(${VMSTATUS} = FAILED) {
Noop(mailbox doesn't exists);
}
On Fri, 2005-11-11 at 10:11 +, Chris Bagnall wrote:
Morning all,
I'm trying to rewrite my dialplan macros into AEL. How does one handle
result-dependent branching (e.g. VoiceMail will branch to n+101 if mailbox
doesn't exist) in AEL? Or is there a better way of doing this?
Thanks in advance.
Regards,
Chris
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Re: [Asterisk-Users] IM / presence asterisk-1.2-RC1
Asterisk sends OPTIONS message if the device have qualify=NNN option set. On Fri, 2005-11-11 at 11:24 +0100, harry gaillac wrote: Here are some other files. Why asterisk send sip OPTION message to agents ? Harry 2005-11-11 11:23:10 WARNING[1639]: chan_sip.c:1045 __sip_xmit: sip_xmit of 0x81cf940 (len 477) to 192.168.0.20:-1 returned 5060: Operation not permitted Retransmitting #2 (NAT) to 192.168.0.20:5060: OPTIONS sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 80.119.11.222:5060;branch=z9hG4bK4a119599;rport From: asterisk sip:[EMAIL PROTECTED];tag=as747a6ef0 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 11 Nov 2005 10:23:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- 2005-11-11 11:23:10 WARNING[1639]: chan_sip.c:1045 __sip_xmit: sip_xmit of 0x8194ea0 (len 477) to 192.168.0.20:-1 returned 5060: Operation not permitted /// --- harry gaillac [EMAIL PROTECTED] a écrit : Sorry, Here are some files Harry --- BJ Weschke [EMAIL PROTECTED] a écrit : This is good debugging info you've listed below, but this isn't a sip debug/trace. To do that, first verify in your logger.conf file you have the following line: full = notice,warning,error,debug,verbose Then, if you needed to add anything to logger.conf, please first restart Asterisk so those new settings take effect. Then, from the CLI issue set verbose 5 and set debug 5 and finally sip debug. The repeat your dialing steps. The sip debug/trace will then be contained in /var/log/asterisk/full if /var/log/asterisk is where your log files are kept. With that, we can have a better idea of what's happening/not happening to give you the issue you're having. On 11/10/05, harry gaillac [EMAIL PROTECTED] wrote: I did it !? // Connected to Asterisk 1.2.0-rc1 currently running on serveur1 (pid = 1125) Verbosity is at least 4 serveur1*CLI sip show subscriptions Peer UserCall ID Extension Last state Type 192.168.0.21 86 f1682d8d-8f 84 Idle xpidf+xml 192.168.0.21 86 5f32aec-95b 85 Idle xpidf+xml 192.168.0.20 84 cb424ae1-e4 86 Idle xpidf+xml 192.168.0.20 84 715fac66-a9 87 Idle xpidf+xml 4 active SIP subscriptions serveur1*CLI // serveur1*CLI sip show peers Name/username HostDyn Nat ACL Port Status 87/87 192.168.0.21 D N 5060 OK (84 ms) 86/86 192.168.0.21 D N 5060 OK (97 ms) 85/85 192.168.0.20 D N 5060 OK (87 ms) 84/84 192.168.0.20 D N 5060 OK (96 ms) 4 sip peers [4 online , 0 offline] serveur1*CLI /// my sip.conf: [general] context=local ; Default context for incoming calls ; if asterisk was compiled with OSP support. realm=nxs.yi.org; Realm for digest authentication ; defaults to asterisk ; Realms MUST be globally unique according to RFC 3261 ; Set this to your host name or domain name bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=nxs.yi.org ; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ; Enable DNS SRV lookups on outbound calls tos=lowdelay; lowdelay,throughput,reliability,mincost,none maxexpirey=3600 ; Max length of incoming registration we allow defaultexpirey=1000 ; Default length of incoming/outoing registration allow=all ; First disallow all codecs musicclass=default ; Sets the default music on hold class for all SIP calls language=fr ; Default language setting for all users/peers rtptimeout=60 ; Terminate call if 60 seconds of no RTP activity tpholdtimeout=300 ; Terminate call if 300 seconds of no RTP activity useragent=Asterisk PBX ; Allows you to change the user
[Asterisk-Users] command returns a result code of -1 (indicating failure)
In the wiki it states: When an extension is dialed, the command tagged with a priority of 1 is executed, followed by command priority 2, and so on. This goes on until: the call is hung up, a command returns a result code of -1 (indicating failure), a command with the next higher priority doesn't exist (note: Asterisk will not skip over missing priorities), or the call is routed to a new extension. In the case of a return code of -1, where do we go when a -1 is encountered? How can I retain control? For instance, on a Read, if the user enters nothing, I need to continue execution in a known state. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MOH/Media Server
Is there a way to have a separate MOH/Media server for playing music and/or audio prompts/files? I have an * box where calls come in and sit in a queue until an agent is available. I noticed that at the end of the day, I end up with a bunch of zombie mpg123 processes for calls that were once on hold and this seems to be eating up memory. I thought that I could just have a media server that when a call is placed on queue in the * box for the agents, it would use minimum resources from that box and just use the resources of another box. Is it possible? Is there a more efficient/better approach? Thanks, Waldo ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: libbluetooth
Hi,
Thanks Dave, gracias Jose
Luis ;-).
Once everything is
configured, the mobile phone connected via bluetooth.. I've got a segmentation
fault when trying to dial from sip to bluetooth:
CLI Nov 11 16:53:34 NOTICE[]:
/usr/src/chan_bluetooth/chan_bluetooth.c:2041 try_connect: Initialised bluetooth
link to device Nokia[AG] Nokia
AT+BRSF=23[AG] Nokia
AT+BRSF=23Nov 11 16:53:34 WARNING[]:
/usr/src/chan_bluetooth/chan_bluetooth.c:2399 handle_rd_data: Device Nokia:
Unhandled Unsolicited: +BRSF: 47[AG]
Nokia +BRSF: 47[AG] Nokia
OK[AG] Nokia
AT+CIND=?[AG] Nokia
AT+CIND=?[AG] Nokia +CIND:
("call",(0,1)),("service",(0,1)),("call_setup",(0-3)),("callsetup",(0-3))[AG]
Nokia OK[AG] Nokia
AT+CIND?[AG] Nokia AT+CIND?Nov
11 16:53:34 NOTICE[]: /usr/src/chan_bluetooth/chan_bluetooth.c:417 set_cind:
Audio Gateway Nokia got signal[AG] Nokia
+CIND: 0,1,0,0[AG] Nokia
OK[AG] Nokia
AT+CMER=3,0,0,1[AG] Nokia
AT+CMER=3,0,0,1[AG] Nokia
OK[AG] Nokia
AT+CLIP=1[AG] Nokia
AT+CLIP=1[AG] Nokia
OK[AG] Nokia
AT+CGMI=?[AG] Nokia
AT+CGMI=?[AG] Nokia
OK[AG] Nokia
AT+CGMI[AG] Nokia
AT+CGMI[AG] Nokia
Nokia[AG] Nokia OKbluetooth
show
peersBDAddr
Name Role
Status A/C SCOCon/Fd/Th Sig-
-- --- --- ---00:12:62:E1:E5:45
Nokia AG
Ready Yes
-1/-1/0 Yes -- Executing
Dial("SIP/01-25d3", "BLT/Nokia") in new stackSegmentation fault
Version of asterisk: CVS-v1-0-01/08/05-16:05:25 .
There is something hereI don't quite grasp. If I use the command cvs checkout r v1-0 zaptel libpri asterisk asterisk-addons with the
purpose of download the last stable version, why doIget a version
dated january 2005?? (because that's an american date format 8th January 2005,
right?).
If I try last CVS HEAD the result is even worse. Can't start up
asterisk:
[chan_bluetooth.so]Nov 11 20:43:28 WARNING[15987]: loader.c:325
__load_resource: /usr/lib/asterisk/modules/chan_bluetooth.so: undefined symbol:
ast_pthread_createNov 11 20:43:28 WARNING[15987]: loader.c:554 load_modules:
Loading module chan_bluetooth.so failed!
line 654 of chan_bluetooth.c: if
(ast_pthread_create((dev-sco_thread), NULL, sco_thread, dev) 0)
{
I wonder if there is any
patch or anyother explanationfor that segmentation fault. In any
case, thanks for your attention and support.
Victor.
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Re: [Asterisk-Users] Asterisk Lucent TNT w/11.0.2
On Sat, 5 Nov 2005, Shane DeRidder wrote: I've been scouring the mailing list archives for an answer to this, and cannot find one. I'm hoping someone else out there has run into this. Communication between the TNT and Asterisk seems to be operating properly, but I'm unable to accept or originate calls. When I attempt to dial out, I see the following in the TNT's syslog: 10.0.0.10 = TNT 10.0.0.103 = Asterisk new MEDIA-GATEWAY set name = voip set active = yes set protocol-type = sip set mg-sig-address type = specific set mg-sig-address ip-address = 10.0.0.10 set mg-rtp-address ip-address = 10.0.0.10 I have these set to what would be 10.0.0.103 on my TNT and it's working. dave set transport-options type = udp set transport-options heartbeat = yes set voip-options codec-options g711-ulaw dtmf-tone-passing = rtp set voip-options codec-options g711-ulaw silence-det-cng = yes set sip-options primary-proxy ip-address = 10.0.0.103 set sip-options primary-proxy transport-options heartbeat = yes set sip-options registration-proxy ip-address = 10.0.0.103 set sip-options unknown-ani = 00 set sip-options unknown-name = Unknown set sip-options blocked-ani = 00 set sip-options blocked-name = Blocked write -f My 12 T1/PRI are configured exactly alike: new T1 set name = PRI-0 set physical-address shelf = shelf-1 set physical-address slot = slot-1 set physical-address item-number = 1 set line-interface enabled = yes set line-interface frame-type = esf set line-interface encoding = b8zs set line-interface signaling-mode = isdn set line-interface default-call-type = dnis-or-voip set line-interface switch-type = nat-isdn-2-pri set line-interface front-end-type = csu set line-interface channel-config 24 channel-usage = d-channel set line-interface collect-incoming-digits = yes set line-interface voip-gain-control output-pad = 9db-loss set line-interface media-gateway = voip set line-interface egress-ani-dnis-format = dnis write -f Asterisk sip.conf: [maxtnt] type=friend host=10.0.0.10 dtmfmode=inband callerid=MaxTNT maxtnt context=toll-access qualify=yes reinvite=no canreinvite=no disallow=all allow=g729 allow=ulaw [xxx] type=friend host=dynamic nat=yes callerid=Name xxx context=toll-access dtmfmode=info call-limit=1 [EMAIL PROTECTED] disallow=all allow=g729 allow=ulaw Asterisk extensions.conf: [toll-trunks] exten = _1NXXNXX,1,Dial(SIP/[EMAIL PROTECTED],60) exten = _1NXXNXX,2,Hangup [local-trunks] exten = _NXX,1,Dial(SIP/[EMAIL PROTECTED],60) exten = _NXX,2,Hangup [local-access] include = extensions include = local-trunks [toll-access] include = local-access include = toll-trunks I apologize if this is considered off-topic. My thoughts are that I have a problem with the configuration of my TNT and not Asterisk itself. -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Digium TDM400 on freebsd
Trying to get this working on FreeBSD 5.4. zaptel-0.10_1 driver from ports. Digium TDM400 ( don't remember all these codes, but the card has fxs module in the 1st socket, and fxo module in the 4th one. /usr/local/etc/zaptel.conf loadzone=us defaultzone=us fxsks=4 fxoks=1 When I start the system, both modules get detected but I get TDM PCI Master abort message, which goes away if I run /usr/local/etc/rc.d/zaptel.sh stop, unloading the modules. Running this again with start argument, loads the modules without the above mentioned message, also detecting both modules, but ztcfg shows ZT_CHANCONFIG failed on channel 1: Device not configured (6) output of ztcfg -vv Zaptel Configuration == Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) Channel 04: FXS Kewlstart (Default) (Slaves: 04) 2 channels configured. ZT_CHANCONFIG failed on channel 1: Device not configured (6) what would be the basic troubleshooting steps ? thanx a lot in advance. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Needed - Pager notification script
I have a script that doesn't quite fit your needs, but does send out email reminders for on a regular basis, and runs as a daemon. If you are interested, please let me know and I will send it to you. A little warning, this was one of my first major perl scripts, so it may be a little ugly and crude. :) B. J. -Original Message- From: Tom Rymes [mailto:[EMAIL PROTECTED] Sent: Thursday, November 10, 2005 16:50 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Needed - Pager notification script This is a second post on this subject from me: A while back, someone posted to the list about a script that they had created that would handle paging and escalation for on-call mailboxes. Basically, it would monitor the voicemail directories and if a message was left and not retrieved by the on-call tech within X minutes, the system would page the tech again. If after Y minutes the message had still not been retrieved, the script would then page his/ her supervisor, and so on. Unfortunately, the original poster did not include the script body in his post to the list and it is not available at the wiki. My question: Does anyone have such a script that they have already created? Would you be willing to share? If not, what if I chipped in some $$$. I'm really trying to avoid reinventing the wheel here Tom ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] GPS data from cell phones
Hi, Does anyone know if GPS data is available from a cell phone (GPS cell phone) in a similar fashion as CallerID. I saw a past posting where the GPS data is emailed - which just seems strange... Being able to integrate such data into a dial plan could lead to all sorts of applications. Anyone have experience with this. Thanks ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: libbluetooth
Hi.
I`m using asterisk 1.0.9 and it works fine.
I didn`t patch the asterisk, only I followed the README of
chan_bluetooth.
Regards,
José Luis
El vie, 11-11-2005 a las 14:58 +, Victor Alvarez escribió:
Hi,
Thanks Dave, gracias Jose Luis ;-).
Once everything is configured, the mobile phone connected via
bluetooth.. I've got a segmentation fault when trying to dial from sip
to bluetooth:
CLI Nov 11 16:53:34
NOTICE[]: /usr/src/chan_bluetooth/chan_bluetooth.c:2041
try_connect: Initialised bluetooth link to device Nokia
[AG] Nokia AT+BRSF=23
[AG] Nokia AT+BRSF=23
Nov 11 16:53:34
WARNING[]: /usr/src/chan_bluetooth/chan_bluetooth.c:2399
handle_rd_data: Device Nokia: Unhandled Unsolicited: +BRSF: 47
[AG] Nokia +BRSF: 47
[AG] Nokia OK
[AG] Nokia AT+CIND=?
[AG] Nokia AT+CIND=?
[AG] Nokia +CIND:
(call,(0,1)),(service,(0,1)),(call_setup,(0-3)),(callsetup,(0-3))
[AG] Nokia OK
[AG] Nokia AT+CIND?
[AG] Nokia AT+CIND?
Nov 11 16:53:34
NOTICE[]: /usr/src/chan_bluetooth/chan_bluetooth.c:417 set_cind:
Audio Gateway Nokia got signal
[AG] Nokia +CIND: 0,1,0,0
[AG] Nokia OK
[AG] Nokia AT+CMER=3,0,0,1
[AG] Nokia AT+CMER=3,0,0,1
[AG] Nokia OK
[AG] Nokia AT+CLIP=1
[AG] Nokia AT+CLIP=1
[AG] Nokia OK
[AG] Nokia AT+CGMI=?
[AG] Nokia AT+CGMI=?
[AG] Nokia OK
[AG] Nokia AT+CGMI
[AG] Nokia AT+CGMI
[AG] Nokia Nokia
[AG] Nokia OK
bluetooth show peers
BDAddrName Role Status A/C SCOCon/Fd/Th Sig
- -- --- --- ---
00:12:62:E1:E5:45 Nokia AG Ready Yes -1/-1/0 Yes
-- Executing Dial(SIP/01-25d3, BLT/Nokia) in new stack
Segmentation fault
Version of asterisk: CVS-v1-0-01/08/05-16:05:25 . There is something
here I don't quite grasp. If I use the command cvs checkout –r v1-0
zaptel libpri asterisk asterisk-addons with the purpose of download
the last stable version, why do I get a version dated january 2005??
(because that's an american date format 8th January 2005, right?).
If I try last CVS HEAD the result is even worse. Can't start up
asterisk:
[chan_bluetooth.so]Nov 11 20:43:28 WARNING[15987]: loader.c:325
__load_resource: /usr/lib/asterisk/modules/chan_bluetooth.so:
undefined symbol: ast_pthread_create
Nov 11 20:43:28 WARNING[15987]: loader.c:554 load_modules: Loading
module chan_bluetooth.so failed!
line 654 of chan_bluetooth.c: if
(ast_pthread_create((dev-sco_thread), NULL, sco_thread, dev) 0) {
I wonder if there is any patch or any other explanation for that
segmentation fault. In any case, thanks for your attention and
support.
Victor.
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--
José Luis Gómez
Qualis Information Technology
Av. Rivadavia 2553, PB Of. 43 EP
0342-4565684 int 102
Bs. As. 011-51990896
www.qualis.com.ar
Soporte 0810-8880022
Santa Fe - Argentina
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Re: [Asterisk-Users] MOH/Media Server
I have an * box where calls come in and sit in a queue until an agent is available. I noticed that at the end of the day, I end up with a bunch of zombie mpg123 processes for calls that were once on hold and this seems to be eating up memory. There should not be several zombies remaining, there is something wrong with your configuration. Is it possible? Is there a more efficient/better approach? An simple solution would be replaying the MP3 files by RAW files and use cat as player. There are instructions as www.voip-info.org available how to do that. Elmar ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Needed - Pager notification script
I have one also that just does nag paging. It looks up the extension in the db and gets the pagers to notify. Sets x number of attempts and if a user checks his messages it will clear the remainder of the pager attempts. Written in perl. Not a daemon, uses the run_external_notify. Sends one message immediately with the message and caller id info, the nag pages are sent with just 'Mailbox xxx has y messages'. I would be interested in looking at the daemon version. - James B. J. Bomar wrote: I have a script that doesn't quite fit your needs, but does send out email reminders for on a regular basis, and runs as a daemon. If you are interested, please let me know and I will send it to you. A little warning, this was one of my first major perl scripts, so it may be a little ugly and crude. :) B. J. -Original Message- From: Tom Rymes [mailto:[EMAIL PROTECTED] Sent: Thursday, November 10, 2005 16:50 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Needed - Pager notification script This is a second post on this subject from me: A while back, someone posted to the list about a script that they had created that would handle paging and escalation for on-call mailboxes. Basically, it would monitor the voicemail directories and if a message was left and not retrieved by the on-call tech within X minutes, the system would page the tech again. If after Y minutes the message had still not been retrieved, the script would then page his/ her supervisor, and so on. Unfortunately, the original poster did not include the script body in his post to the list and it is not available at the wiki. My question: Does anyone have such a script that they have already created? Would you be willing to share? If not, what if I chipped in some $$$. I'm really trying to avoid reinventing the wheel here Tom ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] missing name part in to field of SIP header
Hi everyone.
I have a small problem with my Asterisk setup?!?
I am trying to connect to another endpoint through my asterisk server.
The packet going in is just like i want it, but the packet going out of
asterisk at to the other endpoint is missing a part in the header?
it looks like this:
To: sip:x.x.x.x;tag=.
where is the phone2@ part in my SIP URI??
I want it to look like:
To: sip:[EMAIL PROTECTED];tag=.
I have my own very simple dialplan using:
exten = s,2,Dial(${ARG2},20,Cf) where ARG2 is SIP/phone2
The reason i need this is to have several conferences going on at the
same time at the same ip-address.
Any ideas ?
Trond
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SV: [Asterisk-Users] Call p2p
Do you know anywhere to find information about this? MVH Amund Nygaard A NOVO Norge AS Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne av Dean Collins Sendt: 10. november 2005 15:27 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: RE: [Asterisk-Users] Call p2p yes From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Amund Nygaard Sent: Thursday, November 10, 2005 8:19 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Call p2p Hello I am still new to Asterisk, but looking at some products to offer small and medium sized buisnesses. Is it possibel to have the sip ends talk directly to eachother? Have authorisation and call setup on the asterisk, but leave the actual conversation p2p? BR Amund Nygaard ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.2.0-RC1 Crashing with g729 codec and ATA 1
Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com on November 11, 2005 at 10:10 AM -0400 wrote: set the keepalive to 60 or more OK. I set this to 120 that phone should not be able to display a hint status so speeddial = 500,500 Thanks. I've made the change The log could be more verbose than this. Set debug = 10 in your sccp.conf or in the console sccp debug 10 You should see what is happening with your audio stream I did this in the console and the output is below. It does not seem to say much to me about audio. Cheers, Gervais --- Asterisk Ready. *CLI sccp debug 10 -- SEP003080629796: Old session marked down -- SEP003080629796: Killing Session 192.168.1.440|20|tr) in new stack -- SCCP: Looking for line 140eate a channel type=SCCP, format=256, data="" options= -- SCCP: Asterisk asked for the state (5) of the line 140 -- SEP003080629796: found line 140 -- SEP003080629796: New channel number: 1 on line 140 -- SEP003080629796: Global Capabilities: 268 -- SEP003080629796: format request: 4/4 -- SEP003080629796: Channel SCCP/140-0001, capabilities: DEVICE 0x4 (ulaw) NATIVE 0x4 (ulaw) BEST 4 (ulaw) -- SEP003080629796: Allocated asterisk channel 140-1 -- SEP003080629796: Asterisk request to call SCCP/140-0001 -- SEP003080629796: Set callingParty Name TLS Group on channel 1 -- SEP003080629796: Set callingParty Number 500 on channel 1 -- SEP003080629796: Set calledParty Name TLS Group on channel 1 -- SEP003080629796: Set calledParty Number 140 on channel 1 -- SEP003080629796: getting the active channel on device -- SEP003080629796: Indicate SCCP state (Ringing) on call 140-1 -- SEP003080629796: Send and Set the call state Ringing(4) for 140-1 -- SEP003080629796: Send callinfo for Inbound channel 1 -- SEP003080629796: Send lamp mode LampBlink(5) on line 1 -- SEP003080629796: Send ringer mode Outside(3) on device -- SEP003080629796: Set asterisk state Ringing (5) for call 1 -- SEP003080629796: Finish to indicate state SCCP (Ringing), SKINNY (Ringing) on call 140-1 -- Called 140 -- SCCP: Looking for line 140 -- SEP003080629796: found line 140 -- SEP003080629796: Looking for a channel with state Ringing (4) on device -- SEP003080629796: Looking for a channel with state Ringing (4) on line 140 -- SEP003080629796: Found channel (1) with state Ringing (4) on line 140 -- SEP003080629796: Found channel (1) with state Ringing (4) on device -- SCCP: Asterisk asked for the state (6) of the line 140 -- SCCP/140-0001 is ringing -- SEP003080629796: Got message OffHookMessage -- SEP003080629796: getting the active channel on device -- SEP003080629796: Taken Offhook -- SEP003080629796: Looking for a channel with state Ringing (4) on device -- SEP003080629796: Looking for a channel with state Ringing (4) on line 140 -- SEP003080629796: Found channel (1) with state Ringing (4) on line 140 -- SEP003080629796: Found channel (1) with state Ringing (4) on device -- SEP003080629796: getting the active channel on device -- SEP003080629796: Answer the channel 140-1 -- SEP003080629796: Set the active channel 1 on device -- SEP003080629796: Send the active line 140 -- SEP003080629796: Indicate SCCP state (Connected) on call 140-1 -- SEP003080629796: Send ringer mode RingOff(1) on device -- SEP003080629796: Send speaker mode 1 -- SEP003080629796: Stop tone on device -- SEP003080629796: Send lamp mode LampOn(2) on line 1 -- SEP003080629796: Send and Set the call state Connected(5) for 140-1 -- SEP003080629796: Send callinfo for Inbound channel 1 -- SEP003080629796: readformat 4, payload 4 -- SEP003080629796: Ask the device to open a RTP port on channel 1. Codec: G.711 u-law 64k, echocancel: ON -- SEP003080629796: Starting RTP on channel 140-1 -- SEP003080629796: Creating rtp server connection at 192.168.1.125 -- SEP003080629796: Set asterisk state Up (6) for call 1 -- SEP003080629796: Finish to indicate state SCCP (Connected), SKINNY (Connected) on call 140-1 -- SCCP: Looking for line 140 -- SEP003080629796: found line 140 -- SEP003080629796: Looking for a channel with state Ringing (4) on device -- SEP003080629796: Looking for a channel with state Ringing (4) on line 140 -- SCCP: Asterisk asked for the state (2) of the line 140 -- SCCP/140-0001 answered SIP/500-59ab -- SCCP: Asterisk request to hangup Inbound channel SCCP/140-0001 -- SEP003080629796: Close openreceivechannel on channel 1 -- SEP003080629796: Stopping RTP on channel 140-1 -- SEP003080629796: Stop media transmission on channel 1 -- SEP003080629796: Requesting CallStatisticsAndClear from Phone -- SEP003080629796: Current channel 140-1 state Connected(5) -- SEP003080629796: getting the active channel on device -- SEP003080629796: Sending tone Zip (50) -- SEP003080629796: Indicate SCCP state (OnHook) on call 140-1 -- SEP003080629796: Send speaker mode 2 -- SEP003080629796: Send and Set the call
[Asterisk-Users] Comand Read issue (Asterisk rel. 1.0.9)
Hi everybody,
I have this issue: one particular Read command seems not work and return an
empty string immediatelly.
This is CLI output(partial)...
-- Goto (ask_aster,s,1)
-- Executing Read(SIP/2000-0b6d, aster|asterisco|2|skip) in new stack
-- Accepting a maximum of 2 digits.
-- Playing 'asterisco' (language 'en')
-- User entered '**'
-- Executing GotoIf(SIP/2000-0b6d, 1?ask_service|s|1) in new stack
-- Goto (ask_service,s,1)
-- Executing SetVar(SIP/2000-0b6d, aster=) in new stack
-- Executing Read(SIP/2000-0b6d, aster|menu|1|skip) in new stack
-- Accepting a maximum of 1 digits.
-- Playing 'menu' (language 'en')
-- User entered '1'
-- Executing SetVar(SIP/2000-0b6d, try=3) in new stack
-- Executing Wait(SIP/2000-0b6d, .5) in new stack
-- Executing GotoIf(SIP/2000-0b6d, 1?ask_codice|s|1) in new stack
-- Goto (ask_codice,s,1)
-- Executing Wait(SIP/2000-0b6d, .5) in new stack
-- Executing Read(SIP/2000-0b6d, codicez|codice|1|skip) in new stack
-- Accepting a maximum of 1 digits.
-- Playing 'codice' (language 'en')
-- User entered '
Extensions file:
[general]
static=yes
writeprotect=yes
[home]
exten = 2000,1,Answer
exten = 2000,2,Goto(start-con,s,1)
[start-con]
exten = s,1,DigitTimeout(6)
exten = s,2,ResponseTimeout(6)
exten = s,3,Goto(start-connect,s,1)
[start-connect]
exten = s,1,Answer
exten = s,2,Wait(2)
exten = s,3,Playback(benvenuto)
exten = s,4,SetVar(try=3)
exten = s,5,Goto(ask_aster,s,1)
[ask_aster]
exten = s,1,Read(aster,asterisco,2,skip)
exten = s,2,GotoIf($[${aster} = **]?ask_service,s,1)
exten = s,3,SetVar(try=${try}-1)
exten = s,4,GotoIf($[${try} = 0]?ask_aster,s,1:numero_verde,s,1)
[ask_service]
exten = s,1,SetVar(aster=)
exten = s,2,Read(aster,menu,1,skip)
exten = s,3,SetVar(try=3)
exten = s,4,Wait(.5)
exten = s,5,GotoIf($[${aster} = 1]?ask_codice,s,1) ; this is last
branch to failing instruction
exten = s,6,GotoIf($[${aster} = 2]?numero_verde,s,1:ask_service,s,1)
[numero_verde]
exten = s,1,Dial(zap/g1/800366466,20)
exten = s,2,Goto(verde_occupato)
exten = s,102,Goto(verde_occupato)
[verde_occupato]
exten = s,1,Playback(grazie)
exten = s,2,Hangup
[ask_codice]
exten = s,1,Wait(.5)
exten = s,2,Read(codicez,codice,1,skip) ; this is the failing instruction
...
...
exten = s,3,MYSQL(Connect connection localhost mydb user telelettura)
exten = s,4,MYSQL(Query resultid ${connection} Select\
lettura_precedente\,lettura_corrente\ from\ lettura_contatori\ where\
codiceutente=${codicez})
exten = s,5,MYSQL(Fetch fetchid ${resultid} precedente corrente)
exten = s,6,MYSQL(Clear ${resultid})
exten = s,7,MYSQL(Disconnect ${connection})
exten = s,8,GotoIf($[${resultid} = 1]?controlla_date,s,1) ; se trovato
va a controllo data
exten = s,9,SetVar(try=${try}-1)
exten = s,10,GotoIf($[${try} = 0]?chiama_operatore,s,1:ask_codice,s,1) ;
chiede dell'operatore
Thank you for help...
Ciao
Mauro
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[Asterisk-Users] New Asterisk WEB Interface ( astwebmgr )
Hello List! I wrote something to allow me to easily interact/configure Asterisk thru a WEB interface. Over time I added several things to it. I thought 'you all' might get some use from it. I call it 'astwebmgr'. You can get it here: http://www.micpc.com/astwebmgr It is written in PHP (with a little JavaScript). Functions are listed below. earl HOMEReturn to the Main Menu AGI AGI Documentation CDR List or Search Asterisk CDR records (CSV only, not SQL) DB Database Functions Add/Delete/Deltree/Get/Show EDITAccess various system and Asterisk configuration files [Edit/Delete] FAX Access/view FAX files FW Turn ON/OFF IPTABLES FireWall Rules for Asterisk functions LOGSList or Search System or Asterisk Log files MAILBOX Add or Delete a VoiceMail Mailbox MANAGER Interact with the Asterisk Manager ORIGINATE Create call files or create a call through the manager interface PHPInfo PHP Configuration Information SOUND Access/View Sound files TC Traffic Control functions VOICEMAIL View Voice MailBox and Listen via the WEB (uses the asterisk provided script) Information Information on how to obtain the latest version ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NvFaxDetect , rxfax, Quantumvoice SIP : Dropping incompatible voice frame
Hello,
I forgot to mention the version details..
Asterisk CVS-v1-0-11/08/05-01:22:43
spandsp-0.0.2pre21
libtiff-devel-3.6.1-8
libtiff-3.6.1-8
Could this be a problem with my provider cos they support only alaw and
ulaw ?
Regards
Dushyanth
Hey all,
Iam trying to receive fax's over a Quantumvoice SIP account. Found some
posts about this same error on google, asterisk-users with little or no
answers. Could this be a codec translation problem specific to my
configuration ?. Iam able to send faxes very fine on this SIP link.
I get the below error when rxfax tries to receive the fax..
Nov 11 03:55:17 NOTICE[3633]: channel.c:1317 ast_read: Dropping
incompatible voice frame on SIP/415xxx-8a80 of format slin since our
native format has changed to ulaw
Regards
Dushyanth
sip.conf
[general]
port=5060
allowguest=yes
bindaddr=192.168.1.235
context=default
disallow=all
allow=ulaw
;allow=slin
;allow=alaw
dtmfmode=rfc2833
register = 415xxx:[EMAIL PROTECTED]/415xxx
[quantumvoice]
context=in-qvoice
nat=yes
disallow=all
allow=ulaw
allow=alaw
allow=slin
fromuser=415xxx
insecure=very
username=415xxx
secret=Secret
type=friend
host=sipdr.quantumvoice-sip.com
extensions.conf
[in-qvoice]
exten = 4152361970,1,wait(2)
exten = 4152361970,2,NoOp(${EXTEN});
exten = 4152361970,3,NoOp(${CALLERID});
exten = 4152361970,4,NoOp(${CALLERIDNAME});
exten = 4152361970,5,SetVar(CALLEDFAX=${EXTEN})
;exten = 4152361970,6,Macro(trfrpana,${PANA_IN_TRUNK});
exten = 4152361970,6,Answer
exten = 4152361970,7,Playtones(ring)
exten = 4152361970,8,NVFaxDetect(6)
exten = 4152361970,9,hangup
exten = fax,1,Goto(sendfax,2201,1);
exten = fax,2,hangup
Logs
*CLI
-- Executing Wait(SIP/415xxx-8a80, 2) in new stack
-- Executing NoOp(SIP/415xxx-8a80, 415xxx) in new stack
-- Executing NoOp(SIP/415xxx-8a80, 216.144.xxx.xxx) in new stack
-- Executing NoOp(SIP/415xxx-8a80, 216.144.xxx.xxx) in new stack
-- Executing SetVar(SIP/415xxx-8a80, CALLEDFAX=415xxx)
in new stack
-- Executing Answer(SIP/415xxx-8a80, ) in new stack
-- Executing PlayTones(SIP/415xxx-8a80, ring) in new stack
-- Executing NVFaxDetect(SIP/415xxx-8a80, 6) in new stack
*CLI Nov 11 03:55:17 NOTICE[3633]: app_nv_faxdetect.c:215
nv_detectfax_exec: Redirecting SIP/415xxx-8a80 to fax extension
-- Executing Goto(SIP/415xxx-8a80, sendfax|2201|1) in new stack
-- Goto (sendfax,2201,1)
-- Executing Macro(SIP/415xxx-8a80, faxreceive) in new stack
-- Executing SetVar(SIP/415xxx-8a80,
FAXFILE=/var/spool/asterisk/fax/1131661512.2) in new stack
-- Executing DBget(SIP/415xxx-8a80, EXTEMAIL=2201/xEmail) in
new stack
-- DBget: varname=EXTEMAIL, family=2201, key=xEmail
-- DBget: set variable EXTEMAIL to [EMAIL PROTECTED]
-- Executing NoOp(SIP/415xxx-8a80, ) in new stack
-- Executing DBget(SIP/415xxx-8a80, EXTNAME=2201/xName) in
new stack
-- DBget: varname=EXTNAME, family=2201, key=xName
-- DBget: set variable EXTNAME to firstname.lastname
-- Executing NoOp(SIP/415xxx-8a80, ) in new stack
-- Executing DBget(SIP/415xxx-8a80,
EXTCOMPANY=2201/xCompany) in new stack
-- DBget: varname=EXTCOMPANY, family=2201, key=xCompany
-- DBget: set variable EXTCOMPANY to Company
-- Executing RxFAX(SIP/415xxx-8a80,
/var/spool/asterisk/fax/1131661512.2.tif|debug) in new stack
Nov 11 03:55:17 NOTICE[3633]: channel.c:1317 ast_read: Dropping
incompatible voice frame on SIP/415xxx-8a80 of format slin since our
native format has changed to ulaw
FLOW Changed from phase 1 to 4
FLOW ???:
FLOW Real-time Internet fax (T.38)
FLOW V.8 capable
FLOW Prefer 64 octet blocks
FLOW Reserved: 0x90
FLOW Supported data signalling rates: V.27ter fallback mode
FLOW 2D coding
FLOW Scan line length: 215mm
FLOW Recording length: A4 (297mm)
FLOW Receiver's minimum scan line time: 20ms at 3.85 l/mm: T7.7 = T3.85
FLOW Reserved: 0x1
FLOW Minimum scan line time for higher resolutions: T15.4 = T7.7
FLOW Character mode
FLOW Reserved: 0x10
FLOW DIS: 80 00 ce f4 80 80 81 80 80 80 18
FLOW HDLC underflow in state 9
FLOW Changed from phase 4 to 3
FLOW Slow carrier up
FLOW Slow carrier down
FLOW T4 timeout in state 9
FLOW Changed from phase 3 to 4
FLOW ???:
FLOW Real-time Internet fax (T.38)
FLOW V.8 capable
FLOW Prefer 64 octet blocks
FLOW Reserved: 0x90
FLOW Supported data signalling rates: V.27ter fallback mode
FLOW 2D coding
FLOW Scan line length: 215mm
FLOW Recording length: A4 (297mm)
FLOW Receiver's minimum scan line time: 20ms at 3.85 l/mm: T7.7 = T3.85
FLOW Reserved: 0x1
FLOW Minimum scan line time for higher resolutions: T15.4 = T7.7
FLOW Character mode
FLOW Reserved: 0x10
FLOW DIS: 80 00 ce f4 80 80 81 80 80 80 18
FLOW T2 timeout
FLOW Start receiving document
FLOW
[Asterisk-Users] sip.ld for a SoundStation IP 4000
Hi does any one have the sip.ld file of a SoundStatios IP 4000 Thanks. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problems with MFC/R2 in Brazil
People, Im tring to use 2 e-1 in Brazil. In order to get R@ signaling, I compilled libdsp, unicall and stuff following www.soft-switch.org comparing with another site (Dezert of Zazamora, in Mexico). Asterisk is running fine with Asterisk but I cant make calls. The guys on the telco company tells me that I have a LOMF (Loss of Multi Frame) error in their end (the far-end) and we can exchange digits. They expect R2-Digital signaling and they think the implementation I use is not quite right. When I try to make a call the result is as follows: Nov 11 12:02:13 VERBOSE[3451]: -- Executing Dial(SIP/200-3ced, UNICALL/g2/55431100) in new stack Nov 11 12:02:13 DEBUG[3451]: Using channel 1 Nov 11 12:02:13 DEBUG[3451]: unicall_call called - 'g2/55431100' Nov 11 12:02:13 DEBUG[3451]: unicall_call caller id - 'Mabio Coelho 200' Nov 11 12:02:13 WARNING[3451]: MFC/R2 UniCall/1 Call control(1) Nov 11 12:02:13 WARNING[3451]: MFC/R2 UniCall/1 Make call Nov 11 12:02:13 WARNING[3451]: Make call failed - Blocked Nov 11 12:02:13 DEBUG[3451]: ast call on peer returned -1 Nov 11 12:02:13 DEBUG[3451]: Hanging up channel 'UniCall/1-1' Nov 11 12:02:13 DEBUG[3451]: unicall_hangup(UniCall/1-1) Nov 11 12:02:13 WARNING[3451]: MFC/R2 UniCall/1 Channel gains Nov 11 12:02:13 WARNING[3451]: MFC/R2 UniCall/1 Channel switching Nov 11 12:02:13 DEBUG[3451]: Hangup: channel: 1 index = 0, normal = 18, callwait = -1, thirdcall = -1 Nov 11 12:02:13 DEBUG[3451]: Updated conferencing on 1, with 0 conference users Nov 11 12:02:13 VERBOSE[3451]: -- Hungup 'UniCall/1-1' Nov 11 12:02:13 VERBOSE[3451]: == Everyone is busy/congested at this time Im attaching the three most relevant configuration files. Sorry if is kind of messy (a lot of lines commented out), that is because I tried a lot of things before posting. Best regards, Mabio Coelho ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems with MFC/R2 in Brazil
Mabio Coelho wrote: People, I’m tring to use 2 e-1 in Brazil. In order to get R@ signaling, I compilled libdsp, unicall and stuff following www.soft-switch.org http://www.soft-switch.org/ comparing with another site (Dezert of Zazamora, in Mexico). Asterisk is running fine with Asterisk but I can’t make calls. The guys on the telco company tells me that I have a LOMF (Loss of Multi Frame) error in their end (the far-end) and we can exchange digits. They expect R2-Digital signaling and they think the implementation I use is not quite right. When I try to make a call the result is as follows: Nov 11 12:02:13 VERBOSE[3451]: -- Executing Dial(SIP/200-3ced, UNICALL/g2/55431100) in new stack Nov 11 12:02:13 DEBUG[3451]: Using channel 1 Nov 11 12:02:13 DEBUG[3451]: unicall_call called - 'g2/55431100' Nov 11 12:02:13 DEBUG[3451]: unicall_call caller id - 'Mabio Coelho 200' Nov 11 12:02:13 WARNING[3451]: MFC/R2 UniCall/1 Call control(1) Nov 11 12:02:13 WARNING[3451]: MFC/R2 UniCall/1 Make call Nov 11 12:02:13 WARNING[3451]: Make call failed - Blocked Nov 11 12:02:13 DEBUG[3451]: ast call on peer returned -1 Nov 11 12:02:13 DEBUG[3451]: Hanging up channel 'UniCall/1-1' Nov 11 12:02:13 DEBUG[3451]: unicall_hangup(UniCall/1-1) Nov 11 12:02:13 WARNING[3451]: MFC/R2 UniCall/1 Channel gains Nov 11 12:02:13 WARNING[3451]: MFC/R2 UniCall/1 Channel switching Nov 11 12:02:13 DEBUG[3451]: Hangup: channel: 1 index = 0, normal = 18, callwait = -1, thirdcall = -1 Nov 11 12:02:13 DEBUG[3451]: Updated conferencing on 1, with 0 conference users Nov 11 12:02:13 VERBOSE[3451]: -- Hungup 'UniCall/1-1' Nov 11 12:02:13 VERBOSE[3451]: == Everyone is busy/congested at this time I’m attaching the three most relevant configuration files. Sorry if is kind of messy (a lot of lines commented out), that is because I tried a lot of things before posting. If the telco sees a loss of multi-frame sync from your end the problem is nothing to so with the MFC/R2 code. Either you have the zaptel.conf set incorrectly, or your line is faulty, or a card is faulty. Steve ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problems with MFC/R2 in Brazil
Steve, The line is good, because we looped in my end and everything is ok. Regarding the card, it might be faulty, but I doubt it because I've tested with two cards. My config files might be broken, that is why I've posted my config files in my previous post but it got stripped by the list. So there we go: Zaptel.conf: # 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: WPE1/0 wanpipe1 card 0 HDB3/ RED span=1,1,0,cas,hdb3 span=2,2,0,cas,hdb3 # cas=1-15:1101 #dchan=16 cas=17-31:1101 #bchan=1-15,17-31:1101 # cas=32-46:1101 #dchan=47 cas=48-62:1101 # Span 3: WCTDM/0 Wildcard TDM400P REV I Board 1 fxoks=63 fxoks=64 fxsks=65 fxsks=66 ##alaw=63-66 ##fxoks=32 ##fxoks=33 ##fxsks=34 ##fxsks=35 # Global data loadzone= br defaultzone = br I think that is the only file needed right? Thanks for the quick feedback. Mabio Coelho -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Underwood Sent: sexta-feira, 11 de novembro de 2005 14:37 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Problems with MFC/R2 in Brazil Mabio Coelho wrote: People, I'm tring to use 2 e-1 in Brazil. In order to get R@ signaling, I compilled libdsp, unicall and stuff following www.soft-switch.org http://www.soft-switch.org/ comparing with another site (Dezert of Zazamora, in Mexico). Asterisk is running fine with Asterisk but I can't make calls. The guys on the telco company tells me that I have a LOMF (Loss of Multi Frame) error in their end (the far-end) and we can exchange digits. They expect R2-Digital signaling and they think the implementation I use is not quite right. When I try to make a call the result is as follows: Nov 11 12:02:13 VERBOSE[3451]: -- Executing Dial(SIP/200-3ced, UNICALL/g2/55431100) in new stack Nov 11 12:02:13 DEBUG[3451]: Using channel 1 Nov 11 12:02:13 DEBUG[3451]: unicall_call called - 'g2/55431100' Nov 11 12:02:13 DEBUG[3451]: unicall_call caller id - 'Mabio Coelho 200' Nov 11 12:02:13 WARNING[3451]: MFC/R2 UniCall/1 Call control(1) Nov 11 12:02:13 WARNING[3451]: MFC/R2 UniCall/1 Make call Nov 11 12:02:13 WARNING[3451]: Make call failed - Blocked Nov 11 12:02:13 DEBUG[3451]: ast call on peer returned -1 Nov 11 12:02:13 DEBUG[3451]: Hanging up channel 'UniCall/1-1' Nov 11 12:02:13 DEBUG[3451]: unicall_hangup(UniCall/1-1) Nov 11 12:02:13 WARNING[3451]: MFC/R2 UniCall/1 Channel gains Nov 11 12:02:13 WARNING[3451]: MFC/R2 UniCall/1 Channel switching Nov 11 12:02:13 DEBUG[3451]: Hangup: channel: 1 index = 0, normal = 18, callwait = -1, thirdcall = -1 Nov 11 12:02:13 DEBUG[3451]: Updated conferencing on 1, with 0 conference users Nov 11 12:02:13 VERBOSE[3451]: -- Hungup 'UniCall/1-1' Nov 11 12:02:13 VERBOSE[3451]: == Everyone is busy/congested at this time I'm attaching the three most relevant configuration files. Sorry if is kind of messy (a lot of lines commented out), that is because I tried a lot of things before posting. If the telco sees a loss of multi-frame sync from your end the problem is nothing to so with the MFC/R2 code. Either you have the zaptel.conf set incorrectly, or your line is faulty, or a card is faulty. Steve ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] HNT PROBLEM
Hi list, i have the next problem: I have conifgured hint for all my extension ( SIP and ZAP) but at the console i send show hints and always all the channels are idle.. My config files: at extension.conf ... [sip-test] exten = 101,hint,ZAP/35 exten = 101,1,Dial(ZAP/35) exten = 102,hint,ZAP/35 exten = 102,1,Dial(ZAP/23) exten = 111,hint,SIP/111 exten = 111,1,Dial(SIP/111) exten = 112,hint,SIP/112 exten = 112,1,Dial(SIP/112) exten = 113,hint,SIP/113 exten = 113,1,Dial(SIP/113) exten = 121,hint,SIP/121 exten = 121,1,Dial(SIP/121) exten = 122,hint,SIP/122 exten = 122,1,Dial(SIP/122) exten = 132,hint,ZAP/36 exten = 132,1,Dial(ZAP/36) exten = 141,hint,SIP/311 exten = 141,1,Dial(SIP/141) ... all the phones have as context = home (where are more extension). and only the SIP/116 (and snom phone) have the context and subscription context as sip-test thanks ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems with MFC/R2 in Brazil
Mabio Coelho wrote: Steve, The line is good, because we looped in my end and everything is ok. Regarding the card, it might be faulty, but I doubt it because I've tested with two cards. My config files might be broken, that is why I've posted my config files in my previous post but it got stripped by the list. So there we go: Zaptel.conf: # 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: WPE1/0 wanpipe1 card 0 HDB3/ RED span=1,1,0,cas,hdb3 span=2,2,0,cas,hdb3 # cas=1-15:1101 #dchan=16 cas=17-31:1101 #bchan=1-15,17-31:1101 # cas=32-46:1101 #dchan=47 cas=48-62:1101 # Span 3: WCTDM/0 Wildcard TDM400P REV I Board 1 fxoks=63 fxoks=64 fxsks=65 fxsks=66 ##alaw=63-66 ##fxoks=32 ##fxoks=33 ##fxsks=34 ##fxsks=35 # Global data loadzone= br defaultzone = br I think that is the only file needed right? Thanks for the quick feedback. Mabio Coelho If you are using a wanpipe card you need to get the wanpipe config files right too. Steve ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IM / presence asterisk-1.2-RC1
Hello, Asterisk don't support IM presence because of no proxy function in chan_sip ! Regards Harry --- harry gaillac [EMAIL PROTECTED] a écrit : When the polycom ip300 phone (1.6.2) send registration SUBSCRIBE message is sent to buddies from directory file so NOTIFY is received from these one. When I want to change status the ip phone don't send NOTIFY to subscriber unlike SER which is a proxy!!! Why? Harry --- harry gaillac [EMAIL PROTECTED] a écrit : Here are some other files. Why asterisk send sip OPTION message to agents ? Harry 2005-11-11 11:23:10 WARNING[1639]: chan_sip.c:1045 __sip_xmit: sip_xmit of 0x81cf940 (len 477) to 192.168.0.20:-1 returned 5060: Operation not permitted Retransmitting #2 (NAT) to 192.168.0.20:5060: OPTIONS sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 80.119.11.222:5060;branch=z9hG4bK4a119599;rport From: asterisk sip:[EMAIL PROTECTED];tag=as747a6ef0 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 11 Nov 2005 10:23:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- 2005-11-11 11:23:10 WARNING[1639]: chan_sip.c:1045 __sip_xmit: sip_xmit of 0x8194ea0 (len 477) to 192.168.0.20:-1 returned 5060: Operation not permitted /// --- harry gaillac [EMAIL PROTECTED] a écrit : Sorry, Here are some files Harry --- BJ Weschke [EMAIL PROTECTED] a écrit : This is good debugging info you've listed below, but this isn't a sip debug/trace. To do that, first verify in your logger.conf file you have the following line: full = notice,warning,error,debug,verbose Then, if you needed to add anything to logger.conf, please first restart Asterisk so those new settings take effect. Then, from the CLI issue set verbose 5 and set debug 5 and finally sip debug. The repeat your dialing steps. The sip debug/trace will then be contained in /var/log/asterisk/full if /var/log/asterisk is where your log files are kept. With that, we can have a better idea of what's happening/not happening to give you the issue you're having. On 11/10/05, harry gaillac [EMAIL PROTECTED] wrote: I did it !? // Connected to Asterisk 1.2.0-rc1 currently running on serveur1 (pid = 1125) Verbosity is at least 4 serveur1*CLI sip show subscriptions Peer UserCall ID Extension Last state Type 192.168.0.21 86 f1682d8d-8f 84 Idle xpidf+xml 192.168.0.21 86 5f32aec-95b 85 Idle xpidf+xml 192.168.0.20 84 cb424ae1-e4 86 Idle xpidf+xml 192.168.0.20 84 715fac66-a9 87 Idle xpidf+xml 4 active SIP subscriptions serveur1*CLI // serveur1*CLI sip show peers Name/username Host Dyn Nat ACL Port Status 87/87 192.168.0.21 D N 5060 OK (84 ms) 86/86 192.168.0.21 D N 5060 OK (97 ms) 85/85 192.168.0.20 D N 5060 OK (87 ms) 84/84 192.168.0.20 D N 5060 OK (96 ms) 4 sip peers [4 online , 0 offline] serveur1*CLI /// my sip.conf: [general] context=local ; Default context for incoming calls ; if asterisk was compiled with OSP support. realm=nxs.yi.org; Realm for digest authentication ; defaults to asterisk ; Realms MUST be globally unique according to RFC 3261 ; Set this to your host name or domain name bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=nxs.yi.org ; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ; Enable DNS SRV lookups on outbound calls tos=lowdelay; lowdelay,throughput,reliability,mincost,none maxexpirey=3600 ; Max length of incoming registration we allow === message truncated ===
RE: [Asterisk-Users] Problems with MFC/R2 in Brazil
I've tried to leave the wanpipe configuration as vanilla as possible. I just turned of the hardware HDLC (that is because I've been told that if Hardware HDLC turned off, Sangoma cards are 100% compatible with digium/tormenta2 cards). Here is my wanpipe configuration: [devices] wanpipe1 = WAN_AFT_TE1, Comment [interfaces] w1g1 = wanpipe1, , TDM_VOICE, Comment [wanpipe1] CARD_TYPE = AFT S514CPU = A CommPort= PRI AUTO_PCISLOT= NO PCISLOT = 14 PCIBUS = 0 FE_MEDIA= E1 FE_LCODE= HDB3 FE_FRAME= NCRC4 FE_LINE = 2 TE_CLOCK= NORMAL ACTIVE_CH = ALL TE_HIGHIMPEDANCE= NO INTERFACE = V35 CLOCKING= EXTERNAL BaudRate= 0 MTU = 1500 UDPPORT = 9000 TTL = 255 IGNORE_FRONT_END = NO [w1g1] PROTOCOL= HDLC HDLC_STREAMING = NO ACTIVE_CH = ALL IDLE_FLAG = 0x7E MTU = 1500 MRU = 1500 TDMV_SPAN = 1 TDMV_ECHO_OFF = NO MULTICAST = NO TRUE_ENCODING_TYPE = NO Any hints? Regards, Mabio Coelho -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Underwood Sent: sexta-feira, 11 de novembro de 2005 15:14 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Problems with MFC/R2 in Brazil Mabio Coelho wrote: Steve, The line is good, because we looped in my end and everything is ok. Regarding the card, it might be faulty, but I doubt it because I've tested with two cards. My config files might be broken, that is why I've posted my config files in my previous post but it got stripped by the list. So there we go: Zaptel.conf: # 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: WPE1/0 wanpipe1 card 0 HDB3/ RED span=1,1,0,cas,hdb3 span=2,2,0,cas,hdb3 # cas=1-15:1101 #dchan=16 cas=17-31:1101 #bchan=1-15,17-31:1101 # cas=32-46:1101 #dchan=47 cas=48-62:1101 # Span 3: WCTDM/0 Wildcard TDM400P REV I Board 1 fxoks=63 fxoks=64 fxsks=65 fxsks=66 ##alaw=63-66 ##fxoks=32 ##fxoks=33 ##fxsks=34 ##fxsks=35 # Global data loadzone = br defaultzone= br I think that is the only file needed right? Thanks for the quick feedback. Mabio Coelho If you are using a wanpipe card you need to get the wanpipe config files right too. Steve ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] sip ignores context definition?
Hi, Asterisk is 1.09, I've tried to change that like you suggested but no luck. When I'm doing sip debug, its look like it always go to the default sip context. I've a second sip host definition and that works, exactly the same configuration just different IP. Could that be a bug? How can I make sure, and if its a bug, how do I submit it? Thanks again, Ohad What version are you running, and is your [Cisco] definition the last one in the file? I have the same problem with 1.0.7, and the ugly fix I came up with was to add a dummy entry as the last sip entry. B. J. _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] Sent: Friday, November 11, 2005 4:48 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] sip ignores context definition? Hi All, I've a very strange error. I've configured a Cisco gw with * and when an incoming call is arriving from the Cisco to * asterisk will always put the call in the default context (ignoring the part in the [Cisco]) I'm attaching my conf files: [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) disallow=all allow=alaw allow=gsm allow=ulaw context = from-trunk ; Send unknown SIP callers to this context callerid = Unknown [Cisco] type=user/friend/peer (tried all options) port=5060 host=myip context=from-Cisco disallow=all allow=alaw allow=ulaw qualify=yes autocreatepeer=yes (with and without this option, in here and in the general setting) nat=no canreinvite=no on Asterisk Console I see (with Verbose 9): Executing AbsoluteTimeout(SIP/myip-b6895f10, 15) in new stack -- Set Absolute Timeout to 15 -- Executing Congestion(SIP/myip-b6895f10, ) in new stack -- Executing AbsoluteTimeout(SIP/myip-b6895f10, 15) in new stack -- Set Absolute Timeout to 15 -- Executing Congestion(SIP/myip-b6895f10, ) in new stack which is my default context: [from-trunk] exten = _.,1,AbsoluteTimeout(15) exten = _.,2,Congestion exten = _.,3,Hangup [from-Cisco] exten = s,1,Answer exten = s,2,Dial($bla) exten = s,3,Hangup Thanks! ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk high load high availability servers
anyone using a high availability server set up for Asterisk ? I saw IBM had some kind of solution at VON but was too busy exhibiting to check it out. :( ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GPS data from cell phones
On 11/11/05, Chuck Bunn [EMAIL PROTECTED] wrote: Hi, Does anyone know if GPS data is available from a cell phone (GPS cell phone) in a similar fashion as CallerID. I saw a past posting where the GPS data is emailed - which just seems strange... Being able to integrate such data into a dial plan could lead to all sorts of applications. Anyone have experience with this. Thanks ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I've heard that it's not publicly accessible, but Nextel/Sprint apparently lets you get at it with your applications you develop for their phones that have GPS support on them. You must be part of their developer program though. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Needed - Pager notification script
I'd love to see both of these scripts, if only to help me get started crafting my own. Can you guys post them to the list so that others will be able to find them in the archives? Tom On Nov 11, 2005, at 10:24 AM, James Armstrong wrote: I have one also that just does nag paging. It looks up the extension in the db and gets the pagers to notify. Sets x number of attempts and if a user checks his messages it will clear the remainder of the pager attempts. Written in perl. Not a daemon, uses the run_external_notify. Sends one message immediately with the message and caller id info, the nag pages are sent with just 'Mailbox xxx has y messages'. I would be interested in looking at the daemon version. - James B. J. Bomar wrote: I have a script that doesn't quite fit your needs, but does send out email reminders for on a regular basis, and runs as a daemon. If you are interested, please let me know and I will send it to you. A little warning, this was one of my first major perl scripts, so it may be a little ugly and crude. :) B. J. -Original Message- From: Tom Rymes [mailto:[EMAIL PROTECTED] Sent: Thursday, November 10, 2005 16:50 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Needed - Pager notification script This is a second post on this subject from me: A while back, someone posted to the list about a script that they had created that would handle paging and escalation for on-call mailboxes. Basically, it would monitor the voicemail directories and if a message was left and not retrieved by the on-call tech within X minutes, the system would page the tech again. If after Y minutes the message had still not been retrieved, the script would then page his/ her supervisor, and so on. Unfortunately, the original poster did not include the script body in his post to the list and it is not available at the wiki. My question: Does anyone have such a script that they have already created? Would you be willing to share? If not, what if I chipped in some $ $$. I'm really trying to avoid reinventing the wheel here Tom ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk high load high availability servers
It's more like a research project going to proof of concept. Was very interesting tho. -bill On 11-Nov-05, at 12:23 PM, Matthew Simpson wrote: anyone using a high availability server set up for Asterisk ? I saw IBM had some kind of solution at VON but was too busy exhibiting to check it out. :( ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX2 multiple audio frames per UDP packet?
Hi, I am wondering if it is possible to tweak IAX2 protocol to packetize audio data more efficiently. I would like to try setups where multiple audio frames (gsm) are combined into single UDP packet. I know that it will incur delay in audio streams but I don't care. Primary concern is to lower bandwidth so that communication can go over slow dialup link (33.6kbps). Also, it looks to me that trunkfreq parameter might be of interest to try. Am I on good track? Any advice/help is appreciated. Regards, Branko S. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.2.0-RC1 Crashing with=?ISO-8859-1?Q? g729_?= codec and ATA 1
Gervais de Montbrun ha scritto: **I did this in the console and the output is below. It does not seem to say much to me about audio. Dunno why, but the phone is not sending an open receive channel ack. In fact it does ot open the rtp media port so the channel don't know where to send (udp port) the rtp packets What firmware are you running? Sergio ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2 multiple audio frames per UDP packet?
On Friday 11 November 2005 13:00, Branko Samardzic wrote: I am wondering if it is possible to tweak IAX2 protocol to packetize audio data more efficiently. I would like to try setups where multiple audio frames (gsm) are combined into single UDP packet. I know that it will incur delay in audio streams but I don't care. Primary concern is to lower bandwidth so that communication can go over slow dialup link (33.6kbps). Take a look at IAX2 trunking, this is *exactly* what it's for. Also, it looks to me that trunkfreq parameter might be of interest to try. Am I on good track? Any advice/help is appreciated. Trunking frequency won't do a thing for you. Take a look at the IAX2 spec and poke through the code a little if you like. Please note that IAX2 trunking in the 1.0.x series is ... iffy. CVS HEAD and 1.2.x trunking should be much, much better. -A. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GPS data from cell phones
On Fri, 2005-11-11 at 12:41 -0500, BJ Weschke wrote: On 11/11/05, Chuck Bunn [EMAIL PROTECTED] wrote: Hi, Does anyone know if GPS data is available from a cell phone (GPS cell phone) in a similar fashion as CallerID. I saw a past posting where the GPS data is emailed - which just seems strange... Being able to integrate such data into a dial plan could lead to all sorts of applications. Anyone have experience with this. I've heard that it's not publicly accessible, but Nextel/Sprint apparently lets you get at it with your applications you develop for their phones that have GPS support on them. You must be part of their developer program though. The Nextel/Sprint/Boost phones do allow access to the GPS data via the JAVA apps. However, there is a security feature in the phones that allows the phone's user to disable JAVA access to the GPS data. The GPS data is also available to E911, but as far as I'm aware, that is a proprietary system. Regards, Ozz. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk T100P to Merlin Legend
I am running into issues with this same setup and would like to update the wiki with information on connecting an avaya legend to an asterisk server via a T100P. Please post your experiences with the legend and asterisk so we can compile a great list of step by step instructions for the wiki. Some questions I have pertain to the protocol used between the MLPBX and the T100P card. We are running ML Release 7.0 with a 100D DS1 Card (hardware 1B, firmware 90) and a Cross Over Cable (1,2,4,5 - 4,5,1,2) to the T100P. We have selected the Legend-Ntwk type on the ML side. Should we be running something like 5ESS rather than legend-ntwk? Again, if anyone has semi-detailed to detailed instructions for the ML Legend or ML Magix please post them so we can update the wiki. Sterling. Spectro, I have a T100P connected to my Merlin Magix and it works like a champ. I am using an older T1 card (one pulled from a Merlin Legend) without any problems. The card is a 100D 1 T1 Trunk blade. I have my system configured as a PRI (23B+D). The T100P is configured to signal with pri_net and to provide timing. I used the 5ESS protocol, but I suppose any of the PRI variants supported by both devices would work. I do not have (or need) any CSU/DSU equipment in between the two devices. I simply built a crossover T1 cable (warning - not the same as an Ethernet cross-over cable) and connected the two devices. The card and the switch synched up and worked immediately. The hardest part was programming the ARS table to properly make use of the connection. As for the insertion of the new card -- I had to do a board renumber and it did mess with the dial plan on the switch a bit. Stations did not change, but the trunk numbering did. (This may be specific to my installation -- Magix/Legend are a field unto themselves.) Good luck with the install. My number is below if you have additional questions. Thanks, Steve Steven Sokol Owner/Manager Sokol Associates, LLC Phone: 816.822.1807 IaxTel: 700.613.9004 Web:http://www.sokol-associates.com ASK ME ABOUT AstriCon 2004! http://www.astricon.net/ -Original Message- From: asterisk-users-admin at lists.digium.com [mailto:asterisk- users- admin at lists.digium.com] On Behalf Of spectro Sent: Sunday, August 22, 2004 11:25 AM To: asterisk-users at lists.digium.com Subject: [Asterisk-Users] asterisk T100P to Merlin Legend Management just approved purchase of a Digium T100P and a T1 card for our Merlin Legend Switch. I will appreciate comments from anyone performing this installation before: - Which T1 card did you use in the Merlin Legend? - Did you require any special interface? (CSU/DSU, etc) - Any items to watch during installation. Also we will need to remove one of our old analog Trunk cards to accomodate the new T1 interface in the switch. This switch already has two T1 cards in it. I don't know much about restriction in the Merlin Legend Switch, but our phone tech told me some horror stories about taking a card out of the switch. I know this is OT and we will probably have our switch part supplier to take care of that but I will appreciate any comments. Thanks in advance ___ Asterisk-Users mailing list Asterisk-Users at lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] GPS data from cell phones
In Canada, Bell is pushing a CDMA-based geolocation service as a subscription add on to your plan. Unfortunately, you are required to use their crappy web app although one could probably hook the data with some well-crafted wget's and grep's -Original Message- From: Austin Denyer [mailto:[EMAIL PROTECTED] Sent: Friday, November 11, 2005 11:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] GPS data from cell phones On Fri, 2005-11-11 at 12:41 -0500, BJ Weschke wrote: On 11/11/05, Chuck Bunn [EMAIL PROTECTED] wrote: Hi, Does anyone know if GPS data is available from a cell phone (GPS cell phone) in a similar fashion as CallerID. I saw a past posting where the GPS data is emailed - which just seems strange... Being able to integrate such data into a dial plan could lead to all sorts of applications. Anyone have experience with this. I've heard that it's not publicly accessible, but Nextel/Sprint apparently lets you get at it with your applications you develop for their phones that have GPS support on them. You must be part of their developer program though. The Nextel/Sprint/Boost phones do allow access to the GPS data via the JAVA apps. However, there is a security feature in the phones that allows the phone's user to disable JAVA access to the GPS data. The GPS data is also available to E911, but as far as I'm aware, that is a proprietary system. Regards, Ozz. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Crashing (high load issues)
Why would you have all those modules loaded on an asterisk server?? Do you *REALLY* even have a PCMCIA slot on your server? Do you need USB? Or parallel port? Do you use IPv6 with asterisk (not supported AFAIK)?? even bluetooth and stuff is running! I know I need to remove alot of things, and am working on it.. It was installed when I installed Fedora, even though I removed alot of things, some were installed anyway. I'd also not use any crappy fake hardware raid, I've found either proper hardware raid cards, or else linux software raid (MD driver) works best. And it is REALhardware RAID. Not crappy fake.. Also, keep in mind that the fact that your old server can cope with the load (albeit slowly) yet your new server crashes, then the fault is (imho) clearly somewhere other than digium hardware/asterisk. I would be looking at areas such as hardware faults, linux kernel faults with your specific hardware, etc. I'd be doing various stress tests on the various components to try to make it crash. I talked with Digium support today, their support is great. And found a few things we can change to help. They are sending me new updated cards with newer firmware to fix some load issues. We have tried these 2 cards in several NEWLY build servers and all have crashed. We are going to test them by swapping them into the server that doesnot crash to see if it is in fact the cards, even though they are replacing them, just to be sure. Have you tried it with a straight linux kernel from kernel.org ?? What versions? No. Have you tried it with a non-SMP kernel from kernel.org and/or your distro? No Have you tried a nice, simple, distro like debian? IMHO, I found redhat, etc make too many customisations even to simple things like the kernel, so even when I used to use redhat, I always used my own kernel without any of their patches etc. One thing I always did was to not compile anything into the kernel unless it was needed for the system, and usually I'd disable module loading completely (though you can't do this with asterisk unfortunately). I have tried Debian, same crash, on another server. I will try other kernels to see if that helps. We did find an IRQ conflict on the server with the vidoe card, but we are not running xserver, we are working out that problem too. Just my 10c worth ... I really appreciate your help, it will be used. Thanks, Kyle ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] asterisk T100P to Merlin Legend
Might be sharing a brain today... Here is my config as it stands: /etc/zapata.conf span=1,1,1,esf,b8zs bchan=1-23 dchan=24 loadzone= us defaultzone = us /etc/asterisk/zaptel.conf switchtype = 5ess signalling = pri_net channel = 1-23 On the merlin side, I have: ADS1 SLOT ATTRIBUTES ASlot Type Format Supp Signal LineComp A 5 PRI ESF B8ZS DMI-MOS 2 A BchnlGrp #: Slot: TestTelNum: NtwkServ:Incoming Routing: A 80 5 ElecTandNtwk Route Directly to UDP I am trying to finish my install up ... -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Sterling Moses Sent: Friday, November 11, 2005 1:17 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] asterisk T100P to Merlin Legend I am running into issues with this same setup and would like to update the wiki with information on connecting an avaya legend to an asterisk server via a T100P. Please post your experiences with the legend and asterisk so we can compile a great list of step by step instructions for the wiki. Some questions I have pertain to the protocol used between the MLPBX and the T100P card. We are running ML Release 7.0 with a 100D DS1 Card (hardware 1B, firmware 90) and a Cross Over Cable (1,2,4,5 - 4,5,1,2) to the T100P. We have selected the Legend-Ntwk type on the ML side. Should we be running something like 5ESS rather than legend-ntwk? Again, if anyone has semi-detailed to detailed instructions for the ML Legend or ML Magix please post them so we can update the wiki. Sterling. Spectro, I have a T100P connected to my Merlin Magix and it works like a champ. I am using an older T1 card (one pulled from a Merlin Legend) without any problems. The card is a 100D 1 T1 Trunk blade. I have my system configured as a PRI (23B+D). The T100P is configured to signal with pri_net and to provide timing. I used the 5ESS protocol, but I suppose any of the PRI variants supported by both devices would work. I do not have (or need) any CSU/DSU equipment in between the two devices. I simply built a crossover T1 cable (warning - not the same as an Ethernet cross-over cable) and connected the two devices. The card and the switch synched up and worked immediately. The hardest part was programming the ARS table to properly make use of the connection. As for the insertion of the new card -- I had to do a board renumber and it did mess with the dial plan on the switch a bit. Stations did not change, but the trunk numbering did. (This may be specific to my installation -- Magix/Legend are a field unto themselves.) Good luck with the install. My number is below if you have additional questions. Thanks, Steve Steven Sokol Owner/Manager Sokol Associates, LLC Phone: 816.822.1807 IaxTel: 700.613.9004 Web:http://www.sokol-associates.com ASK ME ABOUT AstriCon 2004! http://www.astricon.net/ -Original Message- From: asterisk-users-admin at lists.digium.com [mailto:asterisk- users- admin at lists.digium.com] On Behalf Of spectro Sent: Sunday, August 22, 2004 11:25 AM To: asterisk-users at lists.digium.com Subject: [Asterisk-Users] asterisk T100P to Merlin Legend Management just approved purchase of a Digium T100P and a T1 card for our Merlin Legend Switch. I will appreciate comments from anyone performing this installation before: - Which T1 card did you use in the Merlin Legend? - Did you require any special interface? (CSU/DSU, etc) - Any items to watch during installation. Also we will need to remove one of our old analog Trunk cards to accomodate the new T1 interface in the switch. This switch already has two T1 cards in it. I don't know much about restriction in the Merlin Legend Switch, but our phone tech told me some horror stories about taking a card out of the switch. I know this is OT and we will probably have our switch part supplier to take care of that but I will appreciate any comments. Thanks in advance ___ Asterisk-Users mailing list Asterisk-Users at lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com
[Asterisk-Users] Setting up IP PBX
Hi all iam new to this VoIP iam just looking to deploy IP PBX Services I have 4 port gateway to call Around the world i want to setup with Asterix the followings local user authentication and billing user authenticated need to route to 4 port gateway call record track call start time, end time duration of call so how can i intregrate to test first level with my 4 port gateway with NAT, using SIP my users can be public IP or Private.. please suggest me integration docs and installations thanks ram ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] groupware + unified messagerie +Asterisk
On Wed, 2005-11-09 at 12:45 +, Are wrote: We want to intergrate AstBill with a Groupeware or CRM but want input what people will prefeer. On our list today we have http://www.sugarcrm.com/crm/ http://www.vtiger.com/ http://www.egroupware.org/ A couple more worth looking at. Don't remember which one but one of these projects is planning or working on Asterisk integration. CentraView - http://www.centraview.com Centric CRM - http://www.centriccrm.com Regards, Patrick ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] groupware + unified messagerie +Asterisk
hello, http://www.egroupware.org/ would be a good choice ( open source). --- Patrick [EMAIL PROTECTED] a écrit : On Wed, 2005-11-09 at 12:45 +, Are wrote: We want to intergrate AstBill with a Groupeware or CRM but want input what people will prefeer. On our list today we have http://www.sugarcrm.com/crm/ http://www.vtiger.com/ http://www.egroupware.org/ A couple more worth looking at. Don't remember which one but one of these projects is planning or working on Asterisk integration. CentraView - http://www.centraview.com Centric CRM - http://www.centriccrm.com Regards, Patrick ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Non-numerical caller id in Budgetone 101 Ip Phon
Hi everyone ! I'm running Asterisk 1.0.9 and testing it with a GrandStream BudgeTone 101, a Linksys PAP2-NA Gateway, and 2 X-Ten Lite clients. Everything works fine, except the BudgeTone is not showing the name of the calling extension only shows the extension number. In the sip.conf file i have defined: callerid = User Name ext # for every extension (Budgetone, Linksys and X-Ten) When i call to a X-Ten Lite extension, the phone shows me the User Name of the calling extension. But, when i call the BudgeTone phone, the LCD display only shows me the ext # and not the User Name The BudgeTone is running the last firmware available. I don't know if it's an Asterisk or BudgeTone issue. Did anyone experienced something like that? Thanks for your help. Kind regards. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Setting up IP PBX
Hi,My information is that Asterisk/Portaone Radius behind a NAT cannotsend start accounting packet to SIP, so no call accounting...confirm, anyone? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Cisco DHCP and Polycom boot server
Hmmm, I tested this quite a bit as per below... Sorry if this seems lame, but you are using FTP right? Because FTP is the default, not TFTP (even though you use the DHCP TFTP option to set the FTP server address). Peter Case 1 (by FTP from current bootrom and application versions): == DHCP server model: cisco WS-C3560-24PS DHCP server firmware: 12.1(19)EA1c DHCP server IP address: 192.168.0.30 Phone model:Polycom IP500 Phone firmware: Bootrom 3.0.1.0023, Application SIP 1.5.2.0054 Phone reset to factory defaults prior to test. (CDP is disabled by default on phone) DHCP server settings: ip dhcp excluded-address 192.168.0.1 192.168.0.49 ip dhcp excluded-address 192.168.0.56 192.168.0.255 ! ip dhcp pool peter network 192.168.0.0 255.255.255.0 default-router 192.168.0.27 dns-server 192.168.0.4 192.168.0.5 domain-name testme.com option 42 ip 192.168.0.4 option 66 ascii ftp://user:[EMAIL PROTECTED] lease 0 4 ! The phone booted, obtained an ip address, created the boot file on the ftp server, loaded its config files, set the time correctly, and registered with the sip server. Case 2 (by FTP from old bootrom and old application versions): == DHCP server model: cisco WS-C3560-24PS DHCP server firmware: 12.1(19)EA1c DHCP server IP address: 192.168.0.30 Phone model:Polycom IP500 Phone firmware: Bootrom 2.5.0.0006, Application SIP 1.4.0 Phone reset to factory defaults prior to test. (CDP is disabled by default on phone) DHCP server settings: ip dhcp excluded-address 192.168.0.1 192.168.0.49 ip dhcp excluded-address 192.168.0.56 192.168.0.255 ! ip dhcp pool peter network 192.168.0.0 255.255.255.0 default-router 192.168.0.27 dns-server 192.168.0.4 192.168.0.5 domain-name testme.com option 42 ip 192.168.0.4 option 66 ascii 192.168.0.2 lease 0 4 ! The phone booted, obtained an ip address, created the boot file on the ftp server, loaded the new application, rebooted, loaded its config files, set the time correctly, and registered with the sip server. -Original Message- From: Noah Miller [mailto:[EMAIL PROTECTED] Sent: Friday, 11 November 2005 1:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: [EMAIL PROTECTED] Subject: Re: Cisco DHCP and Polycom boot server Hi Peter - When you set up the DHCP pool in Cisco you need to use syntax like: -- option 66 ascii a.b.c.d Thanks! I guess maybe I didn't explain very well. I did get this far, and this seems to work well, if I manually set the phone to read an ascii string. I'm being really picky here, though. I want Joe Schmoe user to be able to plug in the phone and have it get provisioned without having to make any changes to the phone (like selecting to use a DCHP string rather than an IP). With all the Cisco phones that I have, the default setting has been to read the tftp-boot-server parameter as an IP rather than as a string, and I can't get this to work with Cisco DHCP. Maybe somebody else has, though? Thanks, Noah -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Noah Miller Sent: Thursday, 10 November 2005 9:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Cisco DHCP and Polycom boot server Hi - I've been trying to set up my Polycom phones to get the boot server info (tftp-server-address) from DHCP on a Cisco router. I've previously just specified it manually on the phone, and that works well enough, but I need to change now (because of the number and geographic locations of the phones). I can actually get it to work just fine (using option 66 on the Cisco router), if I change the DHCP menu on the Polycom phone to show BootSrv Type: String. That's great, but that's not a default setting, and I don't want to have to change any settings on the phone. I want the phones to be able to provision fully, out-of-the-box, with nothing but the info from DHCP. If I leave the default setting (BootSrv Type: IP Address), and tell the Cisco router to send the boot serverinfo as an IP rather than as a string, nothing happens. The phone just says Could not contact boot server, using existing configuration, but according to the FTP logs and ethereal, the phone doesn't actually try to contact the boot server at all. I've tried various version of the bootrom, but nothing has worked so far. Has anybody gotten this to work? (Cisco router DHCP and Polycom boot server) Thanks, Noah
[Asterisk-Users] MINNESOTA: TwinCities Asterisk Users Group - Saturday 11/12/2005
Good Afternoon, The next Asterisk Users Group meeting has been scheduled for tomorrow, November 12th at 11:30am. Meetings are held monthly on the second Saturday of each month, excluding July and December. Sound Choice Communcations is located in Bloomington Minnesota, just 1/2 mile west of the Mall of America. The address is: 7839 12th Ave S, Bloomington Minnesota 55425. We are just South of Hwy494 on 12th Ave. -12th Aveune is one exit West of Hwy 77 (Ceder Ave). Meetings are held at Sound Choice Communications LLC... http://maps.google.com/maps?oi=mapq=7839%2012th%20Ave%20S%2055425 This month, we'll take a look at what's new in 1.2 and how to upgrade your system. The RC is out and we need folks to try this version out and submit bugs if any can be found. We'll also hear from Eric Osterberg who attended the Astricon event in CA last month. We are always looking for help with meeting topics. If you feel like taking the lead, please do and simply let me know if you need anything. Meeting starts at 11:30am and parking is available in the rear of the building. Runs about 2 hours or less, and we'll order Pizza to the meeting for lunch. Look forward to seeing you there. http://www.voip-info.org/tiki-index.php?page=Asterisk%20User%20Group%20TwinCities%20Minnesota%20USA There is no meeting in December, so don't miss this, the last meeting for 2005. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Cisco DHCP and Polycom boot server
Hi Peter - Hmmm, I tested this quite a bit as per below... Sorry if this seems lame, but you are using FTP right? Because FTP is the default, not TFTP (even though you use the DHCP TFTP option to set the FTP server address). Thanks Again! I haven't tried yet with the 3.x bootrom series. I've just tested on 2.5.0, 2.6.1, and 2.6.2. I also haven't actually done the reset to factory defaults before (though I don't think I had changed any of the bootrom settings previously). More to try. Thanks! Noah ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX2 phones
Can anyone recommend a source for IAX2 phones located in the USA? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Softphone with Lotus Notes support?
On 11 Nov 2005, at 12:49, Paul Davidson wrote: ** As someone who uses and develops Notes and Asterisk on an almost daily basis, I can tell you two things: 1. Technically, all softphones 'support' Lotus Notes- if Notes knew how to pass them a number, they'd dial it. Notes, however, especially in it's address book, doesn't support anyone. 2. Since Notes is one heck of a lot more programmer-friendly than Outlook/Exchange will ever be (I'm not biased, really..), adding such functionality to your address books would be a snap. Simply pick a softphone you like that supports any sort of API to accept dialing, preferably one that supports URI dialing (DIAX comes to mind, but it's really up to you), and modify the design of your address book (personal or system) to turn the Phone Number field into a link hotspot. Click, done. What I have done goes another step farther into the dark side- since Domino natively supports LDAP, I wrote a script to pull all names and numbers (10,000 of them) out of Domino using LDAP, drop them into a MySQL database, then re-present it on my Cisco phones as a directory, and via Apache as a web service, which supports click to dial via call files in Asterisk. I'm now working on an agent for individual user Personal Address Books to 'synchronize' with this directory structure, so I can combine a user's personal contacts with the main 'corporate' directory when they are searching for contacts. I'd offer it here, and someday I might, however, since each corporate Domino enviromnet is so very different, I have to basically restructure the code for each implementation- and I havent made the code mature enough to have anyone other than me do it. So for now, it's a single-client application. But, I'd be happy to share implementation details with anyone who wants to email me offline. -pbd ___ --Bandwidth and Colocation sponsored by Easynews.com -- Do you think there would be any interest in a softphone that supports LDAP ? T. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 2 SIP phones on Y data connector on 1 ethernet
2 SIP phones on Y data connector on 1 ethernet - will that cause problems ? thx in advance _ Dont just search. Find. Check out the new MSN Search! http://search.msn.click-url.com/go/onm00200636ave/direct/01/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 2 SIP phones on Y data connector on 1 ethernet
At least use a hub or switch (preferred) But if you MUST use a Y connector make sure the adapter meets the International Data 10T Standard -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of A_ Navone Sent: Friday, November 11, 2005 3:54 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] 2 SIP phones on Y data connector on 1 ethernet 2 SIP phones on Y data connector on 1 ethernet - will that cause problems ? thx in advance _ Don't just search. Find. Check out the new MSN Search! http://search.msn.click-url.com/go/onm00200636ave/direct/01/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Quantumvoice vs Broadvoice - Multiline
We are considering Quantumvoice as a provider - They are telling us they will give us 1 line number but we can have 5 concurrent incoming and outgoing line numbers. Charge is about $45 + extras - this seems considerable less expensive than the competition which seem to focus on. My second choice is BroadVoice $29.99 + $9.99 per additional line (in state only?) - more expensive, less features, and they don't seem loved by many ? Is anyone else using Quantum Voice? It was mentioned earlier that it requires an ATA connection and Asterisk support/compatibility is sketchy at best - I've contacted BV and they responded saying they need 24hrs to look into it? Seems like a popular topic but I'm looking for 2-3 lines - I only need one number but need to be able to make or receive several calls at a time? Any advice or recommendations appreciated - I want to port my number but I'm running out of time and must make a decision very soon. Thanks, Dane Reugger Crescent City Technologies New Orleans, LA 70112 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Quantumvoice vs Broadvoice - Multiline
I was testing Broadvoice few weeks before Hurricane Wilma here in FL. Since then, I had been since the landline (Bellsouth), and I had to 'remote callfwd' the BS # to my broadvoice #. So, from my impression, is ok for my needs (I got a weird no ringback problem that I kind of solved with a Background trick), and no surprises yet regarding the bill (my mother in law call Brazil a lot from my house, no, she is not aware of the 'unlimited' plan. So I may be in for a surprise in a couple of months). I've no tried several calls at the same time, you may want to ask them.. PS: I'm running Asterisk 1.0.9 Dane Reugger wrote: We are considering Quantumvoice as a provider - They are telling us they will give us 1 line number but we can have 5 concurrent incoming and outgoing line numbers. Charge is about $45 + extras - this seems considerable less expensive than the competition which seem to focus on. My second choice is BroadVoice $29.99 + $9.99 per additional line (in state only?) - more expensive, less features, and they don't seem loved by many ? Is anyone else using Quantum Voice? It was mentioned earlier that it requires an ATA connection and Asterisk support/compatibility is sketchy at best - I've contacted BV and they responded saying they need 24hrs to look into it? Seems like a popular topic but I'm looking for 2-3 lines - I only need one number but need to be able to make or receive several calls at a time? Any advice or recommendations appreciated - I want to port my number but I'm running out of time and must make a decision very soon. Thanks, Dane Reugger Crescent City Technologies New Orleans, LA 70112 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Quantumvoice vs Broadvoice - Multiline
Julio Arruda wrote: I was testing Broadvoice few weeks before Hurricane Wilma here in FL. Since then, I had been since the landline (Bellsouth), and I had to 'remote callfwd' the BS # to my broadvoice #. So, from my impression, is ok for my needs (I got a weird no ringback problem that I kind of solved with a Background trick), and no surprises yet regarding the bill (my mother in law call Brazil a lot from my house, no, she is not aware of the 'unlimited' plan. So I may be in for a surprise in a couple of months). I've no tried several calls at the same time, you may want to ask them.. PS: I'm running Asterisk 1.0.9 Dane Reugger wrote: We are considering Quantumvoice as a provider - They are telling us they will give us 1 line number but we can have 5 concurrent incoming and outgoing line numbers. Charge is about $45 + extras - this seems considerable less expensive than the competition which seem to focus on. My second choice is BroadVoice $29.99 + $9.99 per additional line (in state only?) - more expensive, less features, and they don't seem loved by many ? Is anyone else using Quantum Voice? It was mentioned earlier that it requires an ATA connection and Asterisk support/compatibility is sketchy at best - I've contacted BV and they responded saying they need 24hrs to look into it? Seems like a popular topic but I'm looking for 2-3 lines - I only need one number but need to be able to make or receive several calls at a time? Any advice or recommendations appreciated - I want to port my number but I'm running out of time and must make a decision very soon. Thanks, Dane Reugger Crescent City Technologies New Orleans, LA 70112 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Broadvoice only allows only the normal 3 way calling so is 2 channels for # about BV i got a lot of water under the bridge every works ok supper ok for times. then BV brokes without you make a single change in your asterisk server and stop working.. if u call support you are the guy with the problem.. yes BV support sucks, and it took me 9 phone calls, 12 emails, 3 chargeback and 2 call to my bank to remove myself from their billing all them well documented... so my advice nothing can be worts than BV. regards Saul ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Non-numerical caller id in Budgetone 101 Ip Phon
I have the same problem but I did not think it was a problem. I don't think the display supports alpha characters. Carlos Prieto wrote: Hi everyone ! I'm running Asterisk 1.0.9 and testing it with a GrandStream BudgeTone 101, a Linksys PAP2-NA Gateway, and 2 X-Ten Lite clients. Everything works fine, except the BudgeTone is not showing the name of the calling extension only shows the extension number. In the sip.conf file i have defined: callerid = User Name ext # for every extension (Budgetone, Linksys and X-Ten) When i call to a X-Ten Lite extension, the phone shows me the User Name of the calling extension. But, when i call the BudgeTone phone, the LCD display only shows me the ext # and not the User Name The BudgeTone is running the last firmware available. I don't know if it's an Asterisk or BudgeTone issue. Did anyone experienced something like that? Thanks for your help. Kind regards. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Non-numerical call er id in Budgetone 101 Ip Phon
In my experience, no it does not support alpha only digits -Original Message- From: Paul [mailto:[EMAIL PROTECTED] Sent: Friday, November 11, 2005 2:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Non-numerical caller id in Budgetone 101 Ip Phon I have the same problem but I did not think it was a problem. I don't think the display supports alpha characters. Carlos Prieto wrote: Hi everyone ! I'm running Asterisk 1.0.9 and testing it with a GrandStream BudgeTone 101, a Linksys PAP2-NA Gateway, and 2 X-Ten Lite clients. Everything works fine, except the BudgeTone is not showing the name of the calling extension only shows the extension number. In the sip.conf file i have defined: callerid = User Name ext # for every extension (Budgetone, Linksys and X-Ten) When i call to a X-Ten Lite extension, the phone shows me the User Name of the calling extension. But, when i call the BudgeTone phone, the LCD display only shows me the ext # and not the User Name The BudgeTone is running the last firmware available. I don't know if it's an Asterisk or BudgeTone issue. Did anyone experienced something like that? Thanks for your help. Kind regards. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MFC/R2
Turn on full logging with loglevel=255 in unicall.conf, and send me a log when a channel locks up. Steve Thank you for your answer. In the below log, loglevel=255, the Unicall/2 is locked up, it stay in Bad State. It starts work well, but at about 8:26:48 it's locked up until the next reload CLI command. It's only happing with the outbound calls. I have verified that always have a Timed out waiting for grou B before it locked up.. Nov 11 08:24:16 WARNING[13812] chan_unicall.c: MFC/R2 UniCall/2 Call control(1) Nov 11 08:24:16 WARNING[13812] chan_unicall.c: MFC/R2 UniCall/2 Make call Nov 11 08:24:16 WARNING[13812] chan_unicall.c: MFC/R2 UniCall/2 Making a new call with CRN 32769 Nov 11 08:24:16 WARNING[13812] chan_unicall.c: MFC/R2 UniCall/2 0001 - [1/ 1/Idle /Idle ] Nov 11 08:24:16 WARNING[13812] chan_unicall.c: Unicall/2 event Dialing Nov 11 08:24:16 WARNING[13812] chan_unicall.c: MFC/R2 UniCall/2 - 1101 [1/ 40/Seize /Idle ] Nov 11 08:24:16 WARNING[13812] chan_unicall.c: MFC/R2 UniCall/2 9 on - [2/ 40/Group I /Idle ] Nov 11 08:24:16 WARNING[13812] chan_unicall.c: MFC/R2 UniCall/2 - 1 on [2/ 40/Group I /DNIS ] Nov 11 08:24:16 WARNING[13812] chan_unicall.c: MFC/R2 UniCall/2 9 off - [2/ 40/Group I /DNIS ] Nov 11 08:24:16 WARNING[13812] chan_unicall.c: MFC/R2 UniCall/2 - 1 off [2/ 40/Group I /DNIS ] Nov 11 08:24:16 WARNING[13812] chan_unicall.c: MFC/R2 UniCall/2 1 on - [2/ 40/Group I /DNIS ] Nov 11 08:24:16 WARNING[13812] chan_unicall.c: MFC/R2 UniCall/2 - 1 on [2/ 40/Group I /DNIS ] Nov 11 08:24:16 WARNING[13812] chan_unicall.c: MFC/R2 UniCall/2 1 off - [2/ 40/Group I /DNIS ] Nov 11 08:24:17 WARNING[13812] chan_unicall.c: MFC/R2 UniCall/2 - 1 off [2/ 40/Group I /DNIS ] Nov 11 08:24:17 WARNING[13812] chan_unicall.c: MFC/R2 UniCall/2 3 on - [2/ 40/Group I /DNIS ] Nov 11 08:24:17 WARNING[13812] chan_unicall.c: MFC/R2 UniCall/2 - 1 on [2/ 40/Group I /DNIS ] Nov 11 08:24:17 WARNING[13812] chan_unicall.c: MFC/R2 UniCall/2 3 off - [2/ 40/Group I /DNIS ] Nov 11 08:24:17 WARNING[13812] chan_unicall.c: MFC/R2 UniCall/2 - 1 off [2/ 40/Group I /DNIS ] Nov 11 08:24:17 WARNING[13812] chan_unicall.c: MFC/R2 UniCall/2 3 on - [2/ 40/Group I /DNIS ] Nov 11 08:24:17 WARNING[13812] chan_unicall.c: MFC/R2 UniCall/2 - 1 on [2/ 40/Group I /DNIS ] Nov 11 08:24:17 WARNING[13812] chan_unicall.c: MFC/R2 UniCall/2 3 off - [2/ 40/Group I /DNIS ] Nov 11 08:24:17 WARNING[13812] chan_unicall.c: MFC/R2 UniCall/2 - 1 off [2/ 40/Group I /DNIS ] Nov 11 08:24:17 WARNING[13812] chan_unicall.c: MFC/R2 UniCall/2 1 on - [2/ 40/Group I /DNIS ] Nov 11 08:24:17 WARNING[13812] chan_unicall.c: MFC/R2 UniCall/2 - 1 on [2/ 40/Group I /DNIS ] Nov 11 08:24:17 WARNING[13812] chan_unicall.c: MFC/R2 UniCall/2 1 off - [2/ 40/Group I /DNIS ] Nov 11 08:24:17 WARNING[13812] chan_unicall.c: MFC/R2 UniCall/2 - 1 off [2/ 40/Group I /DNIS ] Nov 11 08:24:17 WARNING[13812] chan_unicall.c: MFC/R2 UniCall/2 7 on - [2/ 40/Group I /DNIS ] Nov 11 08:24:17 WARNING[13812] chan_unicall.c: MFC/R2 UniCall/2 - 1 on [2/ 40/Group I /DNIS ] Nov 11 08:24:17 WARNING[13812] chan_unicall.c: MFC/R2 UniCall/2 7 off - [2/ 40/Group I /DNIS ] Nov 11 08:24:17 WARNING[13812] chan_unicall.c: MFC/R2 UniCall/2 - 1 off [2/ 40/Group I /DNIS ] Nov 11 08:24:17 WARNING[13812] chan_unicall.c: MFC/R2 UniCall/2 3 on - [2/ 40/Group I /DNIS ] Nov 11 08:24:17 WARNING[13812] chan_unicall.c: MFC/R2 UniCall/2 - 1 on [2/ 40/Group I /DNIS ] Nov 11 08:24:17 WARNING[13812] chan_unicall.c: MFC/R2 UniCall/2 3 off - [2/ 40/Group I /DNIS ] Nov 11 08:24:17 WARNING[13812] chan_unicall.c: MFC/R2 UniCall/2 - 1 off [2/ 40/Group I /DNIS ] Nov 11 08:24:17 WARNING[13812] chan_unicall.c: MFC/R2 UniCall/2 7 on - [2/ 40/Group I /DNIS ] Nov 11 08:24:17 WARNING[13812] chan_unicall.c: MFC/R2 UniCall/2 - 3 on [2/ 40/Group I /DNIS ] Nov 11 08:24:17 WARNING[13812] chan_unicall.c: MFC/R2 UniCall/2 7 off - [2/ 40/Group I /DNIS ] Nov 11 08:24:17 WARNING[13812] chan_unicall.c: MFC/R2 UniCall/2 - 3 off [2/ 40/Group I /DNIS ] Nov 11 08:24:17 WARNING[13812] chan_unicall.c: MFC/R2 UniCall/2 1 on - [2/ 40/Group I /DNIS ] Nov 11 08:24:19 WARNING[13812] chan_unicall.c: MFC/R2 UniCall/2 Channel gains Nov 11 08:24:19 WARNING[13812] chan_unicall.c: MFC/R2 UniCall/2 Channel switching Nov
[Asterisk-Users] Asterisk behind a NAT
Second post I have installed Asterisk on SuSE 10.0 with an active firewall/NAT filter. The server has connection to my own Intranet (private IP) and to Internet Everything works well for clients behind and in front-of the firewall but they can not communicate with each other. Signalling gets through but the audio gets blocked by the firewall/NAT. So, I open-up ports 10.000 -to- 20.000 in the fw so that the udp/rtp packages cuold get through but it has not been successful. I am using xlite for clients and have no pot cards installed ( digium fxo,fxs, etc). Does anyone knows what else to do? Has anyone come accross (and solved) this type of problem? Firewall configuration is as follows: FW_DEV_EXT=eth-id-00:0d:87:5c:44:e5 #eth1 FW_DEV_INT=eth-id-00:06:4f:0e:ca:99 eth-id-00:40:f4:9f:12:25 #eth0 wlan0 FW_ROUTE=yes FW_MASQUERADE=yes FW_MASQ_DEV=$FW_DEV_EXT FW_MASQ_NETS=192.168.100.0/255.255.255.0 FW_SERVICES_EXT_TCP=53 http https ssh FW_SERVICES_EXT_UDP=5060 5061 53 FW_SERVICES_INT_TCP=21 3128 5056 53 5801 5901 80 8080 epmap http microsoft-ds netbios-ssn smtp ssh FW_SERVICES_INT_UDP=5060:5075 53 bootps netbios-dgm netbios-ns FW_SERVICES_INT_RPC=mountd nfs nfs_acl nlockmgr portmap status ypbind FW_SERVICES_ACCEPT_EXT=0/0,udp,5060:5075 FW_TRUSTED_NETS=192.168.100.0/255.255.255.0 FW_FORWARD=0/0,192.168.100.0/255.255.255.0,udp,5060 FW_FORWARD=0/0,192.168.100.0/255.255.255.0,udp,1 FW_FORWARD=192.168.100.0/255.255.255.0,0/0,udp,1 Sip Configuration: [general] bindport=5060 bindaddr=0.0.0.0 srvlookup=no externrefresh=10 externip=201.208.246.178 nat=yes localnet=192.168.100.0/255.255.255.0; RTP configuration: [general] rtpstart=1 rtpend=2 rtpchecksums=yes Regards, Enrique Leon ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] GoToIf Regular Expression
I am trying to test whether a callerid number is a valid ten digit
number. I'm a total novice with regular expressions.
I've tried:
exten = s,n,GotoIf($[${CALLERIDNUM} : \d{10,10}]?label)
But CLI gives an error. Can someone please show me what the correct
syntax would be to do this?
Thanks,
Adam
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[Asterisk-Users] [Announce] Web-MeetMe v1.4.0
Title: [Announce] Web-MeetMe v1.4.0 New Features- - Weekly recurring meetings with the same room and pin numbers. Any conflict in the conference number as identified before the conference is added, allowing the submitter to change the conference room number - Database storage of MeetMe flags This requires a db update to add the columns and a new version of app_cbmysql. In this release the flags are hard coded in the UI. I will be making a configuration option for the number of flags, and which flags are exposed. For now the Admin has only 'Announce name' as and option, and the User has 'Announce name' and 'Listen mode' options This may be the last update to app_cbmysql. There is a recent bug opened on Mantis to make MeetMe use the Realtime architecture. If it is merged, I will port the scheduling functions to app_meetme. The web interface will need minimal changes to be compatible, and I will continue to work on refining it. [Location] http://www.fitawi.com/Asterisk [Files] Web-MeetMe_v1.4.0.tgz (required) app_cbmysql.c (required) cbmysql.conf (required) cb-extensions.conf (suggested) README (suggested) [Installation] See the README [Features] 1. Schedule new conferences a. Control start and end times b. Set conference pin # i. Generate one if the requester leaves it blank ii. Identify pin # conflicts (another conference with the same pin is scheduled at the same time) c. Set Admin and User passwords i. Generate a user password if an Admin pw is set but the User pw is blank d. Weekly recurring conferences with the same settings e. Select MeetMe flags per conference for Admins and Users 2. Email the details for a successfully scheduled conference 3. Separate views for Current, Past and Future conferences 4. Ability to modify the end time of a running conference a. Can also reschedule a past or future conference. 5. Monitor realtime conference activity a. Mute/Kick participants 6. Optional authentication a. Currently Active Directory or LDAP based b. Authentication is abstracted so unix/PAM/DB/RADIUS support could be easily added 7. Users can only monitor, update or delete their conferences 8. Verified administrators can monitor, update or delete any conferences. 9. Updated to Asterisk 1.2.0-beta1 a. Changes to the Manager interface may have caused support for 1.0.X to slip, I cannot test that) Thanks and enjoy, Dan ***Developer help/guidence request*** The day/month/year code needs to be rewritten in _javascript_ to allow the fields to dynamically update. Changing from a month that has 31 day to one with 30 should update the day field if it is set to 31. Similar logic is needed for dealing with February in leap/non-leap years. This is well outside my experience and if anyone would care to contribute the code, I'd appreciate it. Or if someone can point out a way to do it in PHP, even better. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX2 multiple audio frames per UDP packet?
Hi Andrew, thanks for your prompt response. However, I am not sure whether IAX trunking can be of any benefit on 33.6kbps link. It shows significant bandwith reduction on 2 and more simultaneous calls. My question relates to single call over such link. Current measurement say that gsm call consumes approx 32kbps in each direction. So, there is no question that single call is only possibility for such link. I was able to tweak some old H323 implementations to consume far less bandwith by grouping 2 or more audio frames into single UDP packet (at the cost of additional delay and bigger jitter buffer). My question is related to this matter. Is it possible to force Asterisk to package multiple audio frames in single UDP packet even in case of single IAX call. If that is the case then some overhead can be prevented thus enabling conversations even over such a slow link. Regards, Branko S. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with CallerIDNum
I've been jacking with this for a while but don't
understand all thatI'm reading...
The problem is sometimes I get ANI II digits from
the phone company. These will be two digits that prefix ANI- so some
callerid might arrive as only "00" or "007147391234", "00714", "714"
or normal"7147391234".
The prefix digits I get are
"00", "23", "61", "62", "63" - see http://www.nanpa.com/number_resource_info/ani_ii_assignments.htmlfor
info on ANI II digits.
I need ascriptdeals with thisby
normalizes the ANI as received at the beginning of the call.
What I would like to do is ( ANI =
${CALLERIDNUM} ):
if the ANI is a 10 digit number - do noting
if ANI is greater than 10 digits and the first two digits are one of these:
"00", "23", "61", "62", "63" or might be others not found
yet-Then strip the first two digits and make CALLERIDNUM =
corrected ANI
if ANI is less than 10 digit and the first two digits are one of these:
"00", "23", "61", "62", "63" ormight be others not found yet - then
strip these digits and make CALLERIDNUM = corrected ANI
I was wondering if you could show me an example of
how you would do this?
TIA
Bart
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Re: [Asterisk-Users] IAX2 multiple audio frames per UDP packet?
How about you stop pulling your hair out and let me send you one of the 56k modems I have sitting on my desk. heh On Fri, 11 Nov 2005 13:00:15 -0500, Branko Samardzic [EMAIL PROTECTED] wrote: Hi, I am wondering if it is possible to tweak IAX2 protocol to packetize audio data more efficiently. I would like to try setups where multiple audio frames (gsm) are combined into single UDP packet. I know that it will incur delay in audio streams but I don't care. Primary concern is to lower bandwidth so that communication can go over slow dialup link (33.6kbps). Also, it looks to me that trunkfreq parameter might be of interest to try. Am I on good track? Any advice/help is appreciated. Regards, Branko S. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Wits end with echo
On Wednesday, November 09, 2005 5:57 PM, Jon Reynolds wrote Hello, I have an AAH-1.5 with a TMD400P with four lines, 8 Grandstream GXP-2000 phones, I am having echo issues on the GXP-2000 side. I have evaluated a similar setup as yours involving the Granstream 2000. I was able to isolate two sources of echo. 1. The Grandstream 2000 when the volume is up will cause echo because the microphone picks up the speaker on the handset. Don't even attempt to use speakerphone as you will cause full echo that will drive the remote party nuts. This problem is specific to the phone and doesn't relate to Asterisk. (Perhaps a newer firmware will resolve this?) Here is what I have tried so far: The server has everything in the bios turned off except what is needed, USB, LPT, Serial etc,etc. I have uncommented Echo Suppresion in zconfig.h and shutdown and turned back on the asterisk box. I have updated the phones to 1.0.12 firmware, I have echotraining=800, echocancel=yes, echowhenbridged=yes, in my sip.conf file. I am using Mark2 as the echo suppresion and still I have echo. 2. Try the following settings in your zapata.conf. These seem to work well for me. echocancel=yes echocancelwhenbridged=yes echotraining=yes ; Use ztmonitor to adjust your gain to levels that work for you. rxgain=-4.0 txgain=-4.0 All the phones have been wired straight to the cisco 2950 switch and all cables have been tested and found to be good. I am completely at a loss at this point as to where to start looking and working to fix the problem. I would like to switch from Mark2 to MG1 but I don't know how I would acomplish that with AAH. I have played with the rx and tx gain but after reading multiple docs on it am still unsure how this would help and how to adjust it using /usr/bin/ztmonitor 1 -v. When you place a call outbound, launch it and watch your gain as you speak. If you can humm a tone at around normal speaking voice to the far side, you can adjust the tx gain up or down to get it about halfway. Have the far end party do the same for the rx gain. It is trial and error. I was surprised to find that my setup worked best by turning the gain down. Check out this link for more info: http://www.voip-info.org/wiki/view/Asterisk+x100p+echotraining If anybody could point me in a new direction or something else to look at or something more to read that I may have missed I would be very appreciative. Thanks for any help, Jon BTW, Digium recently released a new card with hardware-based echo cancellation. It may be worth a try. http://www.digium.com/index.php?menu=product_detailcategory=hardwarepr oduct=TE411Ptab=details You may still hear echo at the first moment a call is placed, but it should completely disappear in a few seconds. -- Shawn ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 7940 paperweight
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Still looking for any advice with this. I had given up with the upgrade process (to SIP.. tftp won't send the files for some reason) but I can't even get this to work with sccp. It doesn't seep to ever finish booting. My understanding is that after the hunt is exhausted through tftp, the phone will boot it's current image, but this isn't the case for me. The display shows Configuring IP Requesting Configuration Opening 192.168.1.104 (tftp server i assigned) Defaulting CM to TFTP Server infinite loop Here is my phone info and below that is a tcpdump. If you have any ideas, please let me know. If this phone is bricked, I need to get my money back before it's too late. MAC Address 00XXBD4D Host Name SEP00XXD4D Phone DN App Load ID P00306000400 Boot Load IDPC0303010100 Version 6.0(4.0) Expansion Module 1 Expansion Module 2 Hardware Revision 4.3 Serial Number INMXXT Model NumberCP-7940G Codec ADLCodec Amps5V Amp C3PO Revision 2 Message Waiting NO excerpt from network settings... CallManager 1 CiscoCM1 CallManager 2 TFTP 192.168.1.104 DHCP EnabledYes DHCP Address Released No Alternate TFTP Yes Erase Configuration NO Forwarding DelayNO GARP EnabledYes Voice VLAN Enabled Yes Auto Line Select EnabledNo Video Capability EnabledNo DSCP For Call Control default DSCP For Configuration default DSCP For Services default Device Security ModeNon Secure Web Access Enabled Yes Tx Excessive Collisions 0 Tx Frames 232 Tx Broadcasts 28 Tx Multicasts 13 Tx Collisions 0 Tx Deferred Abort 0 Rx Overruns 0 Rx Long/CRC 0 Rx Frames 54 Debug display: 0x8103, 0x0, 0x12310044 0x8103, 0x0, 0x12310044 0x8103, 0x0, 0x12310044 0x8103, 0x0, 0x12310044 0x8103, 0x0, 0x12310044 0x8103, 0x0, 0x12310044 Socket Task 616 of 1200 Phone Task 916 of 4000 RTP Task104 of 1200 TLS Task104 of 6000 Config Task 1592 of 6000 Display Task472 of 1300 CAST Task 144 of 1600 Sidecar Task348 of 1500 Audit Task 436 of 1600 Undefined Mode 0 of 64 SVC Mode12 of 64 IRQ Mode28 of 128 FIQ Mode0 of 64 Domain snmpUDPDomain Remote Address /0 Local Address /0 Sender Joins0 Receiver Joins 0 Byes0 Start Time 0 Row Status Not Ready NameSEP00XXBD4D Sender Packets 0 Sender Octets 0 Sender Tool None Sender Reports 1 Sender Report Time 0 Sender Start Time 0 Rcvr Lost Packets 0 Rcvr Jitter 0,0 Receiver Tool None Rcvr Reports1 Rcvr Report Time0 Rcvr Packets0 Rcvr Octets 0 Rcvr Start Time 0 Here is a tcpdump (mac changed): 15:48:31.501856 IP 192.168.1.105.50170 mulbman.tftp: 31 RRQ CTLSEP00XXBD4D.tlv o 15:48:35.501998 IP 192.168.1.105.50170 mulbman.tftp: 31 RRQ CTLSEP00XXBD4D.tlv o 15:48:39.502162 IP 192.168.1.105.50170 mulbman.tftp: 31 RRQ CTLSEP00XXBD4D.tlv o 15:48:43.502293 IP 192.168.1.105.50170 mulbman.tftp: 31 RRQ CTLSEP00XXBD4D.tlv o 15:48:47.504194 IP 192.168.1.105.50171 mulbman.tftp: 32 RRQ SEP00XXBD4D.cnf.xml 15:48:51.502542 IP 192.168.1.105.50171 mulbman.tftp: 32 RRQ SEP00XXBD4D.cnf.xml 15:48:55.502685 IP 192.168.1.105.50171 mulbman.tftp: 32 RRQ SEP00XXBD4D.cnf.xml 15:48:59.502815 IP 192.168.1.105.50171 mulbman.tftp: 32 RRQ SEP00XXBD4D.cnf.xml 15:49:03.502961 IP 192.168.1.105.50171 mulbman.tftp: 32 RRQ SEP00XXBD4D.cnf.xml 15:49:07.544093 IP 192.168.1.105.50077 mulbman.cisco-sccp: S 2491131163:2491131163(0) win 1400 mss 1400 15:49:08.033496 IP 192.168.1.105.50077 mulbman.cisco-sccp: S 2491131163:2491131163(0) win 1400 mss 1400 15:49:09.033501 IP 192.168.1.105.50077 mulbman.cisco-sccp: S 2491131163:2491131163(0) win 1400 mss 1400 15:49:11.033569 IP 192.168.1.105.50077 mulbman.cisco-sccp: S 2491131163:2491131163(0) win 1400 mss 1400 15:49:19.864393 CDPv2, ttl: 180s, Device-ID 'SEP00XXBD4D'[|cdp] 15:49:22.922753 IP 192.168.1.105.50078 mulbman.cisco-sccp: S 1785338396:1785338396(0) win 1400 mss 1400 15:49:23.404027 IP 192.168.1.105.50078 mulbman.cisco-sccp: S 1785338396:1785338396(0) win 1400 mss 1400 15:49:24.404061 IP 192.168.1.105.50078 mulbman.cisco-sccp: S 1785338396:1785338396(0)
