RE: [Asterisk-Users] Zaphfc.ko module error
Hi Remco Thanks for the response. I am running Suse 9.3 kernel 2.6.11.4-20a-default. Will check on the auto update, but I don't think so. Cheers Terry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Remco Barende Sent: 18 August 2005 08:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Zaphfc.ko module error Hi! You didn't state what distro you are running but my guess is that you have autoupdate / up2date running. Before the powerfailure the kernel was updated and after the powerfailure the box booted the new kernel for which you need to recompile the module. Cheers! Remco On Thu, 18 Aug 2005, Terry Wade wrote: Hi Guys I have been running a test server for a few days now with * 1.0.9 bristuff RC8n. I had a power failure and the test machine was not on the ups. When power was restored I found the following error: FATAL: Error inserting zaphfc (/lib/modules/2.6.11.4-20a-default/misc/zaphfc.ko): Unknown symbol in module, or unknown parameter (see dmesg) My dmesg output: zaphfc: unsupported module, tainting kernel. ^^ that makes me believe you are now running a newer kernel ___ 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 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] Vonage locked Motorola VT-1000s
Steve Gladden wrote: I have a small pile of them from customers who were too lazy to send them back after switching to our local voice service... Is there any hope of ever using these things with Asterisk? Vonage does not want them back and they won't unlock them either. A terrible shame! Should I just toss them? Steve I wrote a paper on how to 'unlock' them, the short is that without a mot server (similar to the cable modem docsis stuffs) you cant do anything highly meaningful with them. I hope to have my webpage back up soon (it was being physically moved and the people that are doing that broke some stuff in the process, but hey its free). You can see what I did and maybe take it from there. There is a TTL serial port inside the case, I used a TTL-RS232 converter and connected to it, it runs vxworks, and I mapped out the urls that are valid (incl the 2 undocumented ones) and some of the memory addresses the profile info is stored. All I can say is that if you are highly interested in this check my page occasionally over hte next little while, I couldnt find any of this on the net anywhere, maybe google cache has it. http://www.0xdecafbad.com/ I checked while writing this email and the vast majority that was on my site is not cached right now :( -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.10.10/73 - Release Date: 8/15/2005 ___ 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
SV: [Asterisk-Users] Zaphfc.ko module error
-Ursprungligt meddelande- Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För Terry Wade Skickat: den 19 augusti 2005 07:08 Till: 'Asterisk Users Mailing List - Non-Commercial Discussion' Ämne: RE: [Asterisk-Users] Zaphfc.ko module error Hi Remco Thanks for the response. I am running Suse 9.3 kernel 2.6.11.4-20a-default. Will check on the auto update, but I don't think so. Cheers Terry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Remco Barende Sent: 18 August 2005 08:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Zaphfc.ko module error Hi! You didn't state what distro you are running but my guess is that you have autoupdate / up2date running. Before the powerfailure the kernel was updated and after the powerfailure the box booted the new kernel for which you need to recompile the module. Cheers! Remco On Thu, 18 Aug 2005, Terry Wade wrote: Hi Guys I have been running a test server for a few days now with * 1.0.9 bristuff RC8n. I had a power failure and the test machine was not on the ups. When power was restored I found the following error: FATAL: Error inserting zaphfc (/lib/modules/2.6.11.4-20a-default/misc/zaphfc.ko): Unknown symbol in module, or unknown parameter (see dmesg) My dmesg output: zaphfc: unsupported module, tainting kernel. ^^ that makes me believe you are now running a newer kernel ___ 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 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 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] why asterisk starts listening on all ports
hello why asterisk starts listening on all ports and he is trying to listen messages from 5060. /etc/asterisk/sip.conf bindport=5070 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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] Monitoring RTP protocol
Hi all, is it possible to monitor RTP protocol (latency, errors, ...) by Asterisk or other software. Thanks for answer, Bob. ___ 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] sccp help
Hi, I tried to connect cisco 7910 into asterisk system using chan_sccp.so. But I got a major issue : - when I called from 7910 to another sip phone in the same asterisk server, the call took place normally. - when I called from 7910 to another sip phone in different asterisk server, the call is answered but I cannot hear nor say anything. The phone just immediately lose its tone. - when I got a call from another sip phone in the same asterisk server, the phone rang. But after I picked the handset, there were no tone at all.. sccp debug on CLI produced the following messages: SCCP: Alarm Message: Severity: Major (7), 29: DSP Keepalive Timeout [0x5, 0xa, 0x8, 0x2](5) [21/1090360010] I've tried different versions of chan_sccp, yet the result were still the same. Is it time for me to dump this cisco phone to the garbage can ? (I hope not) Anybody had experienced similar issues? Any suggestion will be greatly appreciated.. Thanks Best Regards, Stevanus ___ 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 seems to load but cannot connect using-r?
Still get same: Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) Angus - Original Message - From: Fábio Sakai [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, August 18, 2005 9:18 PM Subject: RES: [Asterisk-Users] asterisk seems to load but cannot connect using-r? Angus, Try this command: asterisk -c -r Fábio Sakai DGX - Digital Express Suporte CosmoCall [EMAIL PROTECTED] +55 11 3049.8109 -Mensagem original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Em nome de Angus Comber Enviada em: quinta-feira, 18 de agosto de 2005 16:58 Para: asterisk-users@lists.digium.com Assunto: [Asterisk-Users] asterisk seems to load but cannot connect using -r? I installed asterisk on SUSE 9.3. Stupidly I loaded selected to load asterisk from the SUSE DVD - then installed latest asterisk head using cvs. At end of asterisk compilation mentioned modules in /modules where from another installation. My telephony cards working ok and if run asterisk just get these warnings: [chan_capi.so]Aug 18 20:43:32 WARNING[13248]: loader.c:314 __load_resource: /usr/lib/asterisk/modules/chan_capi.so: undefined symbol: ast_smoother_feed Aug 18 20:43:32 WARNING[13248]: loader.c:543 load_modules: Loading module chan_capi.so failed! Are they serious? Then I try: linux:/var/run/asterisk # asterisk -r Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) linux:/var/run/asterisk # ls -al total 5 drwxr-x--- 2 asterisk root 112 Aug 18 20:43 . drwxr-xr-x 13 root root 880 Aug 18 18:44 .. srwxr-xr-x 1 root root 0 Aug 18 20:43 asterisk.ctl -rw-r--r-- 1 root root 6 Aug 18 20:43 asterisk.pid linux:/var/run/asterisk # but /var/run/asterisk/asterisk.ctl does exit? how can I fix this? Is it a problem with those modules in /usr/lib/asterisk/modules? Should I delete them? What? Angus ___ 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 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 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] asterick and festival...Help!
John Gruber wrote:
Earlier this afternoon I had this working
exten = 2890,1,Answer
exten = 2890,2,GoTo(12)
exten = 2890,12,Wait(1)
exten = 2890,13,Festival('I can say numbers like')
exten = 2890,14,SayNumber(1230001,f)
exten = 2890,15,Wait(1)
exten = 2890,16,HangUp
I was so very proud of myself...
All of a sudden after a reboot I get the following from the same
call plan
--- (9 headers 0 lines)---
-- Executing Festival(SIP/1000-2915, I can say numbers like) in
new stack
== Parsing '/etc/asterisk/festival.conf': Found
== Spawn extension (mytest, 2890, 13) exited non-zero on 'SIP/1000-2915'
and of course the call exits.
Here is my /etc/asterick/festival.conf
[general]
host=127.0.0.1
port=1314
usecache=no
cachedir=/var/lib/asterisk/festivalcache/
festivalcommand=(tts_textasterisk %s 'file)(quit)\n
Everything is running on the same box. I have rebooted... nothing is
var log messages either.
The local festival_client connects and I can put in (SayText I can say
numbers like) and it works great.
The festival_server log show only this for the calls from asterick:
client(11) Thu Aug 18 17:53:01 2005 : accepted from (my machine name here)
client(11) Thu Aug 18 17:53:01 2005 : disconnected
So it looks like it is connecting right.
Delete the files in /var/lib/asterisk/festivalcache and then try it
again--see if the behavior changes.
___
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] 1-800 number
On Thursday 18 August 2005 22:27, Matt Hess wrote: Just call a milliwatt..? you have a number? I'm also willing to pay my regular fees to my provider for those 3-4 minutes of testing. ___ 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] V.17
Steve Underwood wrote: Tamas J wrote: Hello, I have seen that SpanDSP supports V.17 faxing, however when I tryed to send pages, I eneded with very ugly pages (unreadable). Did anybody else try that? Yes, I checked frame slips and clocking on PRI, everything has to be OK. Regards, Tamas V.17 is disabled in spandsp. There is a reason for that. Regards, Steve What is that reason? ;) How much work is needed to have v.17 working? Yes, depends on who makes the work ;) Do you think a newbie in fax protocols can make it in reasonable time? Regards, Tamas ___ 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] 1-800 number
On Fri, Aug 19, 2005 at 09:15:02AM +0200, Christoph Eicke wrote: On Thursday 18 August 2005 22:27, Matt Hess wrote: Just call a milliwatt..? you have a number? In your dialplan: exten=1800645549288,1,Milliwatt MusicOnHold will also do. In fact it will probably be a better emulation of an 1-800 line ;-) . -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ 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] sccp help
Hi, On 9:04:57 August 19, 2005 stevanus [EMAIL PROTECTED] wrote: Hi, I tried to connect cisco 7910 into asterisk system using chan_sccp.so. But I got a major issue : I've tried different versions of chan_sccp, yet the result were still the same. Which version of chan_sccp did you use? Sourceforge or Berlios? There is a new fork of chan_sccp by Sergio Chersovani who started work some weeks ago and did an almost complete rewrite of the channel. This version supports a lot more features on various phones and has a lot less bugs. You could find it at chan-sccp.berlios.de (official site) or chan-sccp.org (unofficial site). There is a related mailinglist at berlios.de where Sergio does a hell of a lot of support (unless he is one vacation like at the moment :-) ) and gladly accepts bug reports :-). Regards, Stefan -- (o_ Stefan Gofferje| SCLT //\ Reg'd Linux User #247167 | VCP #2263 V_/_ Heckler Koch - the original point and click interface ___ 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 seems to load but cannot connectusing-r?
But when I load Asterisk it doesn't complain. Get 2 warnings: [chan_capi.so]Aug 18 20:43:32 WARNING[13248]: loader.c:314 __load_resource: /usr/lib/asterisk/modules/chan_capi.so: undefined symbol: ast_smoother_feed Aug 18 20:43:32 WARNING[13248]: loader.c:543 load_modules: Loading module chan_capi.so failed! So Asterisk must be crashing after starting? What do I do now? If I look in /var/log/asterisk see this only: Aug 18 21:47:00 WARNING[6079] loader.c: /usr/lib/asterisk/modules/chan_capi.so: undefined symbol: ast_smoother_feed Aug 18 21:47:00 WARNING[6079] loader.c: Loading module chan_capi.so failed! Aug 19 08:48:12 NOTICE[8271] cdr.c: CDR simple logging enabled. Aug 19 08:48:12 WARNING[8271] loader.c: /usr/lib/asterisk/modules/chan_capi.so: undefined symbol: ast_smoother_feed Aug 19 08:48:12 WARNING[8271] loader.c: Loading module chan_capi.so failed! linux:/var/log/asterisk # Angus - Original Message - From: Dave Cotton [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, August 19, 2005 8:22 AM Subject: Re: [Asterisk-Users] asterisk seems to load but cannot connectusing-r? On Fri, 2005-08-19 at 08:08 +0100, Angus Comber wrote: Still get same: Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) The error message says it all. It thinks it's not running. Check with the ps command. -- Dave Cotton [EMAIL PROTECTED] ___ 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 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] sccp help
Hi, I used chan_sccp from ftp://ftp.berlios.de/pub/chan-sccp. Is it the same as chan_sccp from chan-sccp.berlios.de? Best Regards, Stevanus Stefan Gofferje wrote: Hi, On 9:04:57 August 19, 2005 stevanus [EMAIL PROTECTED] wrote: Hi, I tried to connect cisco 7910 into asterisk system using chan_sccp.so. But I got a major issue : I've tried different versions of chan_sccp, yet the result were still the same. Which version of chan_sccp did you use? Sourceforge or Berlios? There is a new fork of chan_sccp by Sergio Chersovani who started work some weeks ago and did an almost complete rewrite of the channel. This version supports a lot more features on various phones and has a lot less bugs. You could find it at chan-sccp.berlios.de (official site) or chan-sccp.org (unofficial site). There is a related mailinglist at berlios.de where Sergio does a hell of a lot of support (unless he is one vacation like at the moment :-) ) and gladly accepts bug reports :-). Regards, Stefan ___ 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: Which AGI Development Software is fastest on Asterisk?
On Fri, Aug 19, 2005 at 01:18:14AM +0100, Matt King wrote: Hello, I'm looking to develop some custom AGI that will be MySQL intensive. It appears Asterisk supports many different development environments. Which would be best suited for Asterisk and MySQL? It's generally fastest to use FastAGI (over TCP/IP), rather than regular AGI as this means the OS isn't starting a new process for each call (just like it's faster to use PHP or Servlets rather than old-school CGI for serving web pages). FastAGI is like FastCGI. PHP can run as CGI/AGI and FastCGI/FastAGI. But it is commonly run (with Apache) as a module an internal PHP interpeter that run in the apache process(es). The equivalent for that on Asterisk is res_LANGNAME . E.g: res_php and res_perl. They run a complete LANGNAME interpeter inside Asterisk. One downside for that: there is some code in the works (written. checked in?) to reduce priority of a spawned process. But interpreted LANGNAME code will probably run with full asterisk scheduling priority and full permissions to hang the system in case of a 100% CPU loop. One nice thing about AGI/CGI: simplicity. No need for a restart/reload. You can easily run them outside of asterisk/apache. Perl's CGI module makes this even simpler. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ 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] Tr: [Asterisk-Dev] Asterisk IM + Presence
Remarque : message transféré en pièce jointe.
___
Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger
Téléchargez cette version sur http://fr.messenger.yahoo.com---BeginMessage---
Hello,
I'v ever posted my problems.
I downloaded asterisk from cvs head I applied patchs
for presence and IM .
I read voip-info for presence unfortunately without
success.
Anybody could help me to configure presence.
Why Asterisk reply method not allowed when IM is sent
even patch is applied
extensions.conf:
[general]
static=yes
writeprotect=no
[globals]
[default]
ignorepat = 0
exten = _0XXX.,1,Dial(Zap/g1/${EXTEN:1})
exten = 80,1,Dial(Zap/g2/)
exten = 84,hint,Dial(Sip/84)
exten = 84,1,Dial(Sip/84)
exten = 85,hint,Dial(Sip/85)
exten = 85,1,Dial(Sip/85)
sip.conf:
[general]
context=default
realm=nxs.yi.org
bindport=5060
bindaddr=192.168.0.50
srvlookup=yes
tos=lowdelay
maxexpirey=3600
defaultexpirey=120
notifymimetype=text/plain
notifyringing=no
checkmwi=10
videosupport=yes
recordhistory=yes
disallow=all
allow=ulaw
allow=ilbc
musicclass=default
language=en
relaxdtmf=yes
rtptimeout=60
rtpholdtimeout=300
trustrpid = no
progressinband=never
useragent=Asterisk PBX
usereqphone = yes
dtmfmode = rfc2833
compactheaders = no
sipdebug = yes
insecure=yes
[84]
type=friend ; Friends place calls and receive calls
context=default ; Context for incoming calls from
this user
secret=84
host=dynamic; This peer register with us
dtmfmode=rfc2833; Choices are inband, rfc2833, or
info
username=84 ; Username to use in INVITE until peer
registers
disallow=all
allow=ulaw ; dtmfmode=inband only
works with ulaw or alaw!
progressinband=no ; Polycom phones don't work
properly with never
incominglimit=1
[85]
type=friend ; Friends place calls
and receive calls
context=default ; Context for incoming
calls from this user
secret=85
host=dynamic; This peer register
with us
dtmfmode=rfc2833; Choices are inband,
rfc2833, or info
username=85 ; Username to use in
INVITE until peer registers
disallow=all
allow=ulaw ; dtmfmode=inband only
works with ulaw or alaw!
progressinband=no ; Polycom phones don't
work properly with never
incominglimit=1
___
Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger
Téléchargez cette version sur http://fr.messenger.yahoo.com
sip_subscription
Description: 1156734228-sip_subscription
___
Asterisk-Dev mailing list
Asterisk-Dev@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-dev
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-dev---End Message---
___
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] Monitoring RTP protocol
Bohuslav Coufal wrote: Hi all, is it possible to monitor RTP protocol (latency, errors, ...) by Asterisk or other software. Try http://tstat.tlc.polito.it/ quote Tstat, a passive sniffer able to provide several insight on the traffic patterns at both the the network and transport levels. /quote I have not tried it myself, just have it in my bookmarks. raj ___ 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] sccp help
On 10:10:54 August 19, 2005 stevanus [EMAIL PROTECTED] wrote: Hi, I used chan_sccp from ftp://ftp.berlios.de/pub/chan-sccp. Jep, it is... If you had problems with this, your chance for a solution is higher at the chan-sccp-users list... :-) Regards, Stefan ___ 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] Help on AGI running
oops, got it. Thanks for the info. thanks Somesh -Original Message- From: Moises Silva [mailto:[EMAIL PROTECTED] Sent: Thursday, August 18, 2005 7:56 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Help on AGI running i guess you are trying to run a non compiled C program, you have to compile it first. # gcc sample.c -o sample then from asterisk run the executable 'sample' without the extension '.c' best regards On 8/18/05, someshwarak [EMAIL PROTECTED] wrote: I am running AGI script written in C. My script gets triggered but there is no action. I have taken the samplec file from http://home.cogeco.ca/~camstuff/agi.html. it says launched AGI script /var/lib/asterisk/agi-bin/sample.c exec format error . Did I miss anything? I am running asterisk1.0.7 can anyone please help me on these. thanks, Somesh ___ 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 -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ 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] sccp help
Hi, Haven't noticed that there exists one :P Thanks for the pointer anyway ;). Gotta sign up pretty soon :) Best Regards, Stevanus Stefan Gofferje wrote: On 10:10:54 August 19, 2005 stevanus [EMAIL PROTECTED] wrote: Hi, I used chan_sccp from ftp://ftp.berlios.de/pub/chan-sccp. Jep, it is... If you had problems with this, your chance for a solution is higher at the chan-sccp-users list... :-) Regards, Stefan ___ 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 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] How many TDM22P Card can be used on thesame PC ?
Actually they have. Interrupt sharing for one. Interrupt overhead for another. Drivers which are optimized for minimum latency instead of a balance between latency and ability to share interrupts and overhead for a third. -A. Can you explain a little bit more? I thought they don't share interrupts. Each TDM400P card writes and reads 4bytes to RAM through DMA in every 125usec. And they generate an interrupt for every 32bytes,that is 1ms. So my suspect for not to use multiple TDM400P card would be that: Is there any data lost on DMA, especially when CPU enters the interrupt_routine? BDM. ___ 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] meetme-icecast2-ice2
I installed icecast-2.2.0.tar.gz and ices-2.0.1.tar.gz and referenced http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20Ices. But I could not succeed to start ices-2.0.1 as follows; -- Attempting call on Local/[EMAIL PROTECTED] for [EMAIL PROTECTED]:1 (Retry 1) -- Executing Answer(Local/[EMAIL PROTECTED],2, ) in new stack Channel Local/[EMAIL PROTECTED],1 was answered. -- Executing Answer(Local/[EMAIL PROTECTED],1, ) in new stack -- Executing Wait(Local/[EMAIL PROTECTED],1, 1) in new stack -- Executing Wait(Local/[EMAIL PROTECTED],2, 1) in new stack -- Executing MeetMe(Local/[EMAIL PROTECTED],1, 104) in new stack -- Executing ICES(Local/[EMAIL PROTECTED],2, /usr/src/asterisk/contrib/asterisk-ices.xml) in new stack Aug 18 21:54:27 WARNING[5929]: app_ices.c:152 ices_exec: Write failed to pipe: Broken pipe == Spawn extension (stream, 33102, 3) exited non-zero on 'Local/[EMAIL PROTECTED],2' == Spawn extension (stream, 33100, 3) exited non-zero on 'Local/[EMAIL PROTECTED],1' Aug 18 21:54:27 NOTICE[5929]: pbx_spool.c:239 attempt_thread: Call completed to Local/[EMAIL PROTECTED] Which is the correct usage of asteriks-icecast 'icecast-2.2.0 and ices-2.0.1 (ogg)' or 'icecast-2.2.0 and ices-0.4(mp3)'? Regards, Zen ___ 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] Optimum online-upload throttling confirmed.
Been there, done that... I was talking to a high level tech for an hour... Basically, they calculate the need for throttle based on the length of time a modem is busy, not the amount of data that is transferred. So for example, asterisk not involved, If I view an axis camera feed remotely, after about 2 minutes the entire network lags. Even though it's only going 10-20k/second, it's the constant traffic that does it. It's a cable thing, probably since they have so many modems up on their nodes now... A year ago the node was nice and empty... Regards, Greg -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brendon Baumgartner Sent: Friday, August 19, 2005 12:45 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Optimum online-upload throttling confirmed. From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Thursday, August 18, 2005 6:08 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Optimum online-upload throttling confirmed. Hello All, I was recently fighting with an optimum online connection in NY. I finally got in touch with someone that confirmed they are throttling my upload connection. I just wanted to make everyone aware of it, so if you have problems if your ping times jump erratically, this could be the cause. Their suggestions were, although you can upload a lot, do not do it constantly. They do not want any constant outgoing connections. Even on business class, they do throttle. All business class primarily does is allow port 25 to pass. Now I am going to look and see if I can get a decent upload speed dsl or something to correct this problem. You might try traffic shaping before going to your ISP. Being that ping is erratic though, is evidence that it may not help. I believe LARTC has some information for you there. -Brendon ___ 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 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] Voipbuster blocking Asterisk/IAX connections?
They're using the same hosted servers with different billin schemes. When I last looked there was a huge difference in ping times and voipbuster when I tested it was very much up and down in responsiveness. I thought they were in Germany (or at least Europe)? ___ 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] Voipbuster blocking Asterisk/IAX connections?
VoipBuster is a service from Finarea SA Po Box 5648 Lugano 6901 CH But you are correct. The servers are supposedly housed in germany. Even accounting is the same as I couldn't get a voipcheap and a voipbuster account with the same username. -Don -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wilson Pickett Sent: Friday, August 19, 2005 2:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Voipbuster blocking Asterisk/IAX connections? They're using the same hosted servers with different billin schemes. When I last looked there was a huge difference in ping times and voipbuster when I tested it was very much up and down in responsiveness. I thought they were in Germany (or at least Europe)? ___ 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 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] Voipbuster blocking Asterisk/IAX connections?
VoipBuster is a service from Finarea SA Po Box 5648 Lugano 6901 CH But you are correct. The servers are supposedly housed in germany. Even accounting is the same as I couldn't get a voipcheap and a voipbuster account with the same username. I must have misunderstood about who is using the same servers. Voipjet and voipbuster couldn't be using the same servers as far as I can see? At the moment, voipbuster is pinging at about 30ms which is excellent among our providers the best is around 10-20 right here in Paris, and the large telcom 9tel/wengo is 33). I dropped voipbuster though, because it jumps in an out of REACHABLEness constantly. I have a 1 eu account as well. Right now, I'm not using it. ___ 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] IPManager now supports SIP, IAX and Zap
2005.08.19 Version 1.3 * IPManager now supports SIP, IAX and Zap extensions and trunks. * Music on Hold Groups can be defined and assigned. * MP3 files can be uploaded directly to Asterisk FREE download: http://ipsoftware.thorben.dk __ IPManager is a configuration tool for Asterisk. It gives you an easy way of configuring Asterisk to perform maintenance and creation of the following: SIP, IAX and Zap Extensions can be configured very easy with Caller ID and Voicemail Virtual Users A user can login at any phone with an Virtual user extension, and (s)he will receive all calls at that extension, the voicemail and Called ID will be moved to that extension as well. This would be very useful if you have people sharing a phone or a person travelling between departments who need to be reached at his own number everywhere. Queues configure Queues and ACD groups very easily. Extension Opening Hours Any extension or Queue can have its own opening hours, say you want to receive calls on your office phone during office hours and then calls will be transferred to your mobile after office hours. You can always force an extension to be open or closed by dialing a code on the phone. Extension Closing Hours Any extension or Queue can have its own Closing hours this can be used for vacations and holidays. IVR Menus can be set up very easily, you can even attach a wav file, which will be uploaded to Asterisk and converted to gsm format automatically. Direct Dial In Map DDI to local extensions Least Cost Routing Configure which calls should use which trunks Conferencing setup a conference room that even outside users can join Virtual Faxes receives faxes and forwards them to an email account DISA Call this number and get a new dial tone where you can call any local extension Music on Hold, make different groups and assign them to queues and upload mp3 files directly from IPM. SIP Channels IAX Channels Zap Channels ___ 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] segfault with chan_capi-cm 0.5.4
Armin Schindler schrieb: Hi, this should already be fixed in current CVS version and will be part of next release. Maybe you want to try it. (Note: capi.conf and dial syntax has changed) Armin Yes, thank you. updating to cvs-version did solve the issue :) tobias wolf ___ 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] Polycom SoundPoint 501 power adapter
Thanks for all the replies! Looks like I was shipped the wrong powersupply. I figured as much, cause when I first plugged it in it took a while to boot, and started to smell something burning. :( Time to RMA it back and get them to ship me the proper parts. PB Paul Belanger wrote: Can somebody who has a SoundPoint 501 please confirm the power adapter input / output settings: Input: 120V AC 60HZ 20W Output: 24V DC 500mA PB ___ 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 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] Nat + Asterisk + Ser (Far end Nat Traversal)
Hello, I have several * serversbehind a SER server (in a local ip range).The SERserveris also publicy reachable. On the other site, I have SIP clients that are behind another NAT or in the same NAT range as the * server. Can someone give me some directions/hints etc. on how to make this work. I think I should be using MediaProxy with SER. But do the SIP clients need to register at the SER server? If not, how will the reach the * server, since they're only reachable VIA the SER router. Here's is scheme: -IP Phone A (Behind NAT router) (ext 100, Asterisk A) - *A-|priv. addr publ. addr| - |--- INTERNET | - SER ---| - |---| - *B-|IP Phone B (Behind NAT router) (ext. 100, Asterisk B) - (Asterisk servers) (10.254.254.x) Phone A can belong to Asterisk A, and B to Asterisk B. Hope this give you enough information. Regards, Ronald Voermans ___ 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] 1-800 number
Just call a milliwatt..? you have a number? I'm also willing to pay my regular fees to my provider for those 3-4 minutes of testing. Milliwatt generators are essentially part of every telephone company's central office switch, and typically are provided by the telco for their installers and technicians to use when diagnosing problems with customer lines. Some telco's require their install technicians to measure the cable loss (amoung other items) for every new install and record those values on service orders, etc. The telephone number assigned to the milliwatt generator is 100% dependent on the local telco engineering selection and is not standard from one telco company to another. Historically, those telephone numbers that end with 98 and 99 use to be a defacto standard, but not any more. The actual telephone number associated with the milliwatt generator is typically not published by the telco. However, in some cases the telco's repair service will give out those numbers when asked. If that doesn't work, then catch a telco installer in the neighborhood (or coffee shop) and ask them for the number. Its always a local toll-free number. Assuming you're not aware, asterisk also provides a milliwatt generator function and it can be programmed in extensions.conf something like: ; Provides a milliwatt tone generator exten = 3911,1,Milliwatt() If you have multiple pstn lines, dial out through one line and back into your asterisk box to hit that extension. ___ 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] Vonage locked Motorola VT-1000s
Very Highly Internested Any chance you could zip or tar your content up and email it to me or give me a link to grab it? Maybe I could help you get it hosted again too ifyou need that. Thanks!!! Steve Steve Gladden wrote: I have a small pile of them from customers who were too lazy to send them back after switching to our local voice service... Is there any hope of ever using these things with Asterisk? Vonage does not want them back and they won't unlock them either. A terrible shame! Should I just toss them? Steve I wrote a paper on how to 'unlock' them, the short is that without a mot server (similar to the cable modem docsis stuffs) you cant do anything highly meaningful with them. I hope to have my webpage back up soon (it was being physically moved and the people that are doing that broke some stuff in the process, but hey its free). You can see what I did and maybe take it from there. There is a TTL serial port inside the case, I used a TTL-RS232 converter and connected to it, it runs vxworks, and I mapped out the urls that are valid (incl the 2 undocumented ones) and some of the memory addresses the profile info is stored. All I can say is that if you are highly interested in this check my page occasionally over hte next little while, I couldnt find any of this on the net anywhere, maybe google cache has it. http://www.0xdecafbad.com/ I checked while writing this email and the vast majority that was on my site is not cached right now :( -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.10.10/73 - Release Date: 8/15/2005 ___ 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 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] Agi Script - sending a message to called party
Hello guys, Can someone help me??? I was wondering to know how to point a agi message to a specific channel?? For example. caller -- * -- agi script(Send message)---called In this above case in my script every thing is all right, it is, I can send the message correctly to the caller. $AGI-send_text(message) But I would like to send the message to the called party like you have a call. How can I point this message to the called party ? The default is to send the message to the caller party and this is working pretty good. I would like to do the opposite. Dou you have a tip??? Thaks in advance Jônatas Amorim ___ 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] Newbie Trying to make 'catch all extension' but is catching voicemail exit!
The catch all extension I use is _. (match everything).
That's a nono, but that is not the problem :-)
and also tried _X. (match any numeric) don't match special extensions.
Much better!
From voip-info.org on the cmd VoiceMail page:
If, during the recording the caller presses:
'#' - or the defined silence limit is exceeded, recording is stopped and
the call continues at priority n+1.
So when the user presses # it tries to find a match of 4102,3
It searches through you're whole dialplan and gets no pattern that matches
4102,3 except for the last thing _X.,3 which matches, therefore it continues
on there. The best thing, as you mentionned is to add the 4102,3 and do what
needs to be done.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Gladden
Sent: August 18, 2005 10:34 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Newbie Trying to make 'catch all extension' but is
catching voicemail exit!
Greetings,
Running CVS HEAD about 3 weeks old,
I have been beating my head trying to get this to work properly..
Or at least figure out what's going on.
Maybe I have done things wrong...
I have created a 'catch all' extension at the end of our last context where
all phones voicemail extension exist.
This catch all is included in all and works quite nicely except when
voicemail is normally exited after leaving a message.
The catch-all is intended to play an error message when someone dials a
wrong extension.
Which it does just fine and that works perfectly!
What does not work is that when someone goes to leave a voice message and
presses # to end and then 1 to save the message
as the voicemail exits, it does not find hangup (h,1) or hangup but goes and
finds the catchall message!
The catch all extension I use is _. (match everything).
and also tried _X. (match any numeric) don't match special extensions.
I put this at the very end of the last context in my dialplan and it does
show up at the end as expected when you do a show dialplan
I've tried matching h t and i to no avail...
when voicemail terminates it still always plays my fatfingers catchall that
is intended only for misdialed numbers.
It's like voicemail is trying to go somewhere that is invalid as it
terminates I just do not know what that somewhere is!
I must be missing some really simple point here :-)
Thanks!
Steve
;normal extension voicemail
exten = 4102,1,Dial(SIP/4102,44,tT)
exten = 4102,2,Voicemail(u4102)
; of course putting a (exten = 4102,3,hangup) ; fixes the problem...
; but I'm trying to learn where the heck it's trying to go when voicemail ;
terminates!
; if there is no 4102,3 in the context why is it not finding ; the h, that
is!?
[last] ;(included at end of all contexts) with an include statement
exten = t,1,hangup
exten = h,1,hangup
exten = #,1,hangup
exten = i,1,hangup; also have tried only the h,1 of course ;-)
exten = _X.,1,answer
exten = _X.,2,wait(1)
exten = _X.,3,playback(vm-extension)
exten = _X.,4,sayalpha(${EXTEN}) ;reads back invalid #
exten = _X.,5,wait(1)
exten = _X.,6,playback(fatfingers);lets them know it was incorrect
exten = _X.,7,Wait,2
exten = _X.,8,playback(fatfingers)
exten = _X.,9,Wait,2
exten = _X.,10,playback(fatfingers)
exten = _X.,11,hangup
;exten = h,1,playback(goodbye)
and a lookie from the prompt:
show dialplan last
[ Context 'last' created by 'pbx_config' ]
'#' =1. hangup()
[pbx_config]
'h' =1. hangup()
[pbx_config]
'i' =1. hangup()
[pbx_config]
't' =1. hangup()
[pbx_config]
'_X.' = 1. answer()
[pbx_config]
2. wait(1)
[pbx_config]
3. playback(vm-extension)
[pbx_config]
4. sayalpha(${EXTEN})
[pbx_config]
5. wait(1)
[pbx_config]
6. playback(fatfingers)
[pbx_config]
7. Wait(2)
[pbx_config]
8. playback(fatfingers)
[pbx_config]
9. Wait(2)
[pbx_config]
10. playback(fatfingers)
[pbx_config]
11. hangup()
[pbx_config]
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
Re: [Asterisk-Users] snom hint
Hi Tom,
thank you. I solved my problem ... but it was really painful because of
the notify process inside the snom phones seem to crash if you send the
wrong commands :-(. So thausends ;-) of reboots were needed...
That's my solution now:
[agents-loginout]
exten = 6011,hint,DS/6011
exten = 6011,1,Macro(agentsloginout,${EXTEN},${CALLERIDNUM})
exten = 6012,hint,DS/6012
exten = 6012,1,Macro(agentsloginout,${EXTEN},${CALLERIDNUM})
exten = 6013,hint,DS/6013
exten = 6013,1,Macro(agentsloginout,${EXTEN},${CALLERIDNUM})
exten = 6014,hint,DS/6014
exten = 6014,1,Macro(agentsloginout,${EXTEN},${CALLERIDNUM})
exten = 6016,hint,DS/6016
exten = 6016,1,Macro(agentsloginout,${EXTEN},${CALLERIDNUM})
exten = 6017,hint,DS/6017
exten = 6017,1,Macro(agentsloginout,${EXTEN},${CALLERIDNUM})
exten = 6018,hint,DS/6018
exten = 6018,1,Macro(agentsloginout,${EXTEN},${CALLERIDNUM})
exten = h,1,GotoIf($[${LED60STATUS}]=]?4)
exten = h,2,DevState(${LED60EXTEN},${LED60STATUS}) ; LED off
exten = h,3,NoOp
[macro-agentsloginout]
exten = s,1,SetVar(LED60EXTEN=${ARG1})
exten = s,2,RemoveQueueMember(zentrale|SIP/${ARG2})
exten = s,3,SetVar(LED60STATUS=0)
exten = s,4,Dial(local/[EMAIL PROTECTED]/n,,D(#))
exten = s,103,SetVar(LED60STATUS=2)
exten = s,104,AddQueueMember(zentrale|SIP/${ARG2})
exten = s,105,AgentCallbackLogin(${ARG2}|[EMAIL PROTECTED])
[agents-loginout-hidden]
exten = 60,1,AgentCallbackLogin(${CALLERIDNUM}|'#')
I do not like the first part of the agents-loginout but there seems to
be no other solution.
Cheers
Gerd
On Wed, 2005-08-17 at 10:25 -0500, Tom Hayden wrote:
It's in the wiki:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20phone%20snom
About halfway down the page where it says: SNOM SUBSCRIBE/NOTIFY
support for monitoring extension states
--
Tom Hayden
Astoria Telecom, LLC
www.astoriatelecom.net
___
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 seems to load but cannot connectusing-r?
It was my own stupid fault for installing the asterisk version available in the SUSE distribution and then downloading and installing the latest version. Another thing not to do! Uninstalled old and re-installed asterisk and it worked! Angus - Original Message - From: Angus Comber [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, August 19, 2005 8:58 AM Subject: Re: [Asterisk-Users] asterisk seems to load but cannot connectusing-r? But when I load Asterisk it doesn't complain. Get 2 warnings: [chan_capi.so]Aug 18 20:43:32 WARNING[13248]: loader.c:314 __load_resource: /usr/lib/asterisk/modules/chan_capi.so: undefined symbol: ast_smoother_feed Aug 18 20:43:32 WARNING[13248]: loader.c:543 load_modules: Loading module chan_capi.so failed! So Asterisk must be crashing after starting? What do I do now? If I look in /var/log/asterisk see this only: Aug 18 21:47:00 WARNING[6079] loader.c: /usr/lib/asterisk/modules/chan_capi.so: undefined symbol: ast_smoother_feed Aug 18 21:47:00 WARNING[6079] loader.c: Loading module chan_capi.so failed! Aug 19 08:48:12 NOTICE[8271] cdr.c: CDR simple logging enabled. Aug 19 08:48:12 WARNING[8271] loader.c: /usr/lib/asterisk/modules/chan_capi.so: undefined symbol: ast_smoother_feed Aug 19 08:48:12 WARNING[8271] loader.c: Loading module chan_capi.so failed! linux:/var/log/asterisk # Angus - Original Message - From: Dave Cotton [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, August 19, 2005 8:22 AM Subject: Re: [Asterisk-Users] asterisk seems to load but cannot connectusing-r? On Fri, 2005-08-19 at 08:08 +0100, Angus Comber wrote: Still get same: Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) The error message says it all. It thinks it's not running. Check with the ps command. -- Dave Cotton [EMAIL PROTECTED] ___ 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 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 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] any ISDN/PRI signaling experts out there?
I have officially engaged in a pissing contest with the local Telco over PRI calling name delivery. The telco publishes their calling name delivery over PRI feature as being bellcore gr-1367-core compliant. The gr-1367-core spec states that the calling name is to be included as a facility IE in the setup message, or sent in a subsequent facility IE message with an indicator in the setup message that the CNAM will follow. Extensive testing and ISDN/PRI protocol analysis shows that the facility IE they are sending out with the CNAM in it comes only after we have sent back PROGRESS and ALERTING in response to the SETUP. If we block the PROGRESS and ALERTING and sit and WAIT for the FACILITY we never get it, the call will time out, so we know they are actually waiting for the call to progress before sending the facility IE CNAM. As far as I can tell the GR-1367-CORE spec does not define a maximum delay in sending the facility IE or whether it is acceptable to wait for PROGRESS and ALERT before sending it. The setup is; Telco PRI Lucent 5ESS Lucent MAX TNT Asterisk The MAX TNT responds to the Facility IE with ISDN error 98, invalid message for call state. The SIP INVITE from the TNT to Asterisk contains no Caller Name information. It seems really odd to me that a Lucent TNT can not translate the caller ID Name info delivered by a Lucent 5ESS switch. On the same setup, if I connect another PRI device to it that emulates switch side signaling and includes the CNAM as a Display IE in the setup, the SIP invite is properly formatted and * receives the calling party name. Does anyone here have enough experience with ISDN PRI signaling to comment with some level of authority on this? Damon ___ 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] How many TDM22P Card can be used on thesame PC ?
Actually they have. Interrupt sharing for one. Interrupt overhead for another. Drivers which are optimized for minimum latency instead of a balance between latency and ability to share interrupts and overhead for a third. -A. Can you explain a little bit more? I thought they don't share interrupts. Each TDM400P card writes and reads 4bytes to RAM through DMA in every 125usec. And they generate an interrupt for every 32bytes,that is 1ms. So my suspect for not to use multiple TDM400P card would be that: Is there any data lost on DMA, especially when CPU enters the interrupt_routine? I'll jump into this dialog one time. There seems to be two outstanding issues (maybe more) with the TDM card. - each card requires 1k interrupts per sec. the reliability of servicing multiple cards without delay is highly dependent on the motherboard and other activities within the OS. Sharing interrupts between any two (or more) devices is oftentimes a problem. Most motherboards do not allow an interrupt per slot, therefore you end up with a limitation of around two or three TDM cards per motherboard. (Two will work with some motherboards, but three TDM cards is pushing the motherboard interrupt structure.) - the transfer of data from the TDM card to the OS happens across the PCI bus, and the PCI bus latency is very very inconsistent from one motherboard to another. The TDM's interaction with the PCI bus is questionable at best, and some folks tend to point to the TigerJet 320 a problematic pci chip. (I don't know of anyone that has actually proven that however.) The motherboard PCI bus issues are well known by those involved with handling audio (as in music editing), and those individuals tend to point out lots of issues with the North and South PCI bridge chips used on motherboards (regardless of manufacturer). If you dig through the archives you'll find comments suggesting one of the older Apple motherboards does in fact support an interrupt per pci slot. Doubtful that's the only difference and highly likely the Apple motherboard uses a very different pci slot design (in addition to a different interrupt design even though the phyical appearence might be the same.) Since a number of well known PC manufacturers purchase motherboards for their systems from various suppliers, it almost imppossible to create a list of what works and what doesn't. Its also a known fact that some manufacturers will use two or more different motherboards in a specific PC model run, making it even more difficult to identify which systems work, etc. If you'd like to write a utility program that would quantify those issues and make that utility available for others to use, lots of asterisk folks would be very very happy. ___ 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] Optimum online-upload throttling confirmed.
On Thu, 2005-08-18 at 21:08, [EMAIL PROTECTED] wrote: Hello All, I was recently fighting with an optimum online connection in NY. I finally got in touch with someone that confirmed they are throttling my upload connection. I know they watch for people doing peer to peer file sharing and throttle those connections quite severely, but I wasn't aware that they do general throttling. I just wanted to make everyone aware of it, so if you have problems if your ping times jump erratically, this could be the cause. Their suggestions were, although you can upload a lot, do not do it constantly. They do not want any constant outgoing connections. Even on business class, they do throttle. All business class primarily does is allow port 25 to pass. Now I am going to look and see if I can get a decent upload speed dsl or something to correct this problem. I have a friend who uses Vonage (on a ComCast cable modem, not Cablevision) and many times when I talk to him, the voice quality is bad. The reason is the way that _all_ cable companies deploy their data services (it's a CableLabs DOCSIS standard that they all use). Remember that cable modem networks are shared media. The downstream to the cable modem is a broadcast and each cable modem listens for traffic to it. No latency problems here. However, on the upstream, each cable modem requests permission to send and then the cable modem termination system (CMTS) grants it a token to send. Very significant latency and jitter problems here for VoIP. For their own service offerings, the cable companies solve the problem by identifying the VoIP call flows during the call setup and scheduling the media packet stream (RTP packets) grants in advance. Therefore, when I use my Cablevision optimum voice line, my RTP packets are given special priority and the latency and jitter problem is solved. But if you are using Vonage or your own Asterisk box, your RTP packets are treated the same as any other data packet. Not good for VoIP quality. And the problem becomes very bad when the network gets busy. The bottom line is that I wouldn't try to use any cable modem service (ComCast, Cox, Time Warner, Cablevision, doesn't matter) for a VoIP service where voice quality really matters a lot. Regards, Greg ___ 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 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] Monitoring RTP protocol
don't know if Asterisk can do it, but ethereal can. Ethereal is an open source protocol analyzer. Download it from www.etheral.com On Fri, 2005-08-19 at 02:55, Bohuslav Coufal wrote: Hi all, is it possible to monitor RTP protocol (latency, errors, ...) by Asterisk or other software. Thanks for answer, Bob. ___ 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 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] tdm400 and hfc card problem after ztcfg
HI, I installed a tdm440p and a monoBRI in the same Dell machine (PowerEdge 600SC) but after typing ztcfg -vv the server screen is filled with tons of the following lines: Aug 19 11:54:42 pippo kernel: zaphfc: bchan rx fifo not enough bytes to receive! (z1=3200, z2=3193, wanted 8 got 7), probably a buffer overrun. Aug 19 11:54:44 pippo kernel: zaphfc: dropped audio (z1=6972, z2=6955, wanted 8 got 17, dropped 9). while in the linux log I see: Aug 19 11:54:36 pippo kernel: PCI: Enabling device 00:04.0 ( - 0003) Aug 19 11:54:36 pippo kernel: zaphfc: CCD/Billion/Asuscom 2BD0 configured at mem 0xe0ffd000 fifo 0xdef88000(0x1ef88000) IRQ 20 HZ 100 Aug 19 11:54:36 pippo kernel: zaphfc: Card 0 configured for TE mode Aug 19 11:54:36 pippo devfsd[145]: error copying: /lib/dev-state/zap/1 to /dev/zap/1 Aug 19 11:54:36 pippo devfsd[145]: error copying: /lib/dev-state/zap/2 to /dev/zap/2 Aug 19 11:54:36 pippo devfsd[145]: error copying: /lib/dev-state/zap/3 to /dev/zap/3 Aug 19 11:54:36 pippo kernel: zaphfc: 1 hfc-pci card(s) in this box. Aug 19 11:54:39 pippo kernel: Freshmaker version: 71 Aug 19 11:54:39 pippo kernel: Freshmaker passed register test Aug 19 11:54:39 pippo kernel: Uhhuh. NMI received for unknown reason 31. Aug 19 11:54:39 pippo kernel: Dazed and confused, but trying to continue Aug 19 11:54:39 pippo kernel: Do you have a strange power saving mode enabled? Aug 19 11:54:39 pippo kernel: Module 0: Installed -- AUTO FXS/DPO Aug 19 11:54:39 pippo kernel: Module 1: Installed -- AUTO FXO (FCC mode) Aug 19 11:54:39 pippo kernel: Module 2: Installed -- AUTO FXO (FCC mode) Aug 19 11:54:39 pippo kernel: Module 3: Installed -- AUTO FXO (FCC mode) Aug 19 11:54:39 pippo devfsd[145]: error copying: /lib/dev-state/zap/4 to /dev/zap/4 Aug 19 11:54:39 pippo devfsd[145]: error copying: /lib/dev-state/zap/5 to /dev/zap/5 Aug 19 11:54:39 pippo devfsd[145]: error copying: /lib/dev-state/zap/6 to /dev/zap/6 Aug 19 11:54:39 pippo devfsd[145]: error copying: /lib/dev-state/zap/7 to /dev/zap/7 ... snip ... Anybody has any idea about what error copying means?? TIA Giorgio -- GIORGIO INCANTALUPO Tel. +39 02 9350 4780 (104) FGA Software 20017 Rho - Via Puccini, 8 E-Mail : [EMAIL PROTECTED] Internet: http://www.fgasoftware.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: IPManager now supports SIP, IAX and Zap
Just 2 questions: 1. Is there a plan for supporting mISDN, CAPI and SCCP exts. and trunks ? 2. Is it compatible with asterisk STABLE 1.0.X ? regards, Nenad Message: 13 Date: Fri, 19 Aug 2005 12:40:22 +0200 From: Thorben Jensen [EMAIL PROTECTED] Subject: [Asterisk-Users] IPManager now supports SIP, IAX and Zap To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii 2005.08.19 Version 1.3 * IPManager now supports SIP, IAX and Zap extensions and trunks. * Music on Hold Groups can be defined and assigned. * MP3 files can be uploaded directly to Asterisk FREE download: http://ipsoftware.thorben.dk/ http://ipsoftware.thorben.dk ___ 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] V.17
Tamas Jalsovszky wrote: Steve Underwood wrote: Tamas J wrote: Hello, I have seen that SpanDSP supports V.17 faxing, however when I tryed to send pages, I eneded with very ugly pages (unreadable). Did anybody else try that? Yes, I checked frame slips and clocking on PRI, everything has to be OK. Regards, Tamas V.17 is disabled in spandsp. There is a reason for that. Regards, Steve What is that reason? ;) How much work is needed to have v.17 working? Yes, depends on who makes the work ;) Do you think a newbie in fax protocols can make it in reasonable time? Being a fax protocol newbie is no problem at all. Being a DSP newbie might be. :-) The missing bit is making the V.17 modem's carrier and symbol sync crisp enough to acquire the signal on the short training sequence. I think everything else is basically in place. Of course, it needs extensive testing once the missing piece is in place. I haven't touched that modem code for ages, as it hasn't bubbled up the priority list enough. IBM had a patent related to the trellis coding in V.17, but I think that has expired. As far as I know there are no longer patent issues with V.17. If someone knows otherwise, please tell me. V.17 is basically one half of V.32bis, so the patent issues should be similar. Regards, Steve ___ 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] Out of G.729 Decoder Licenses!
Innocent Evil wrote: Hello, I have SIP and Asterisk. On Asterisk I have 2 liscenses of g729 (from digium website) SIP user (100) is calling another SIP user (101). As 101 is not online, my SIP server is redirecting that call to Asterisk. Asterisk forward it to 101's voice mail box. SIP user 100's phone have g729 codec. I havn't buy any codec for SIP server itself. But when 100 reach at 101's voice mail, I get this: Aug 18 18:31:36 WARNING[14511]: codec_g729.c:180 g729tolin_framein: Out of G.729 Decoder Licenses! I didn't get it. Would anybody please explain it. Are the licenses installed? Do show g729 from CLI. You will need a g729 license to access asterisk voicemail from a g729 phone. -Matthew ___ 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] Out of G.729 Decoder Licenses!
Matthew, thanks for answering me. I think, I have found the problem. Yes, the 2 liscenses was intalled. If I make a phone call from SIP phone, asterisk use 1/1 encoder/decoder.. If I make a call and asterisk forward to voice mailbox.. just before it starts recording voice mail, it use 1/1 encode/decode.. but right after recording voice mail, i start getting that liscence violation error. May be I need more channel. And I need to understand 'channel' properly.. Would anybody please explain on channel.. when channel number increase based on uses, link, interportaion. Thanks, -Original Message- From: [EMAIL PROTECTED] Sent: Fri, 19 Aug 2005 09:02:21 -0500 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Out of G.729 Decoder Licenses! Innocent Evil wrote: Hello, I have SIP and Asterisk. On Asterisk I have 2 liscenses of g729 (from digium website) SIP user (100) is calling another SIP user (101). As 101 is not online, my SIP server is redirecting that call to Asterisk. Asterisk forward it to 101's voice mail box. SIP user 100's phone have g729 codec. I havn't buy any codec for SIP server itself. But when 100 reach at 101's voice mail, I get this: Aug 18 18:31:36 WARNING[14511]: codec_g729.c:180 g729tolin_framein: Out of G.729 Decoder Licenses! I didn't get it. Would anybody please explain it. Are the licenses installed? Do show g729 from CLI. You will need a g729 license to access asterisk voicemail from a g729 phone. -Matthew ___ 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 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] Sudenly unable to get incoming from Broadvoice
So it was all working well and then suddenly I'm unable to get incoming calls from BV. Outgoing is fine. I'm using AAH. I have the following settings; [EMAIL PROTECTED]:PASSWORD-GOES-HERE:[EMAIL PROTECTED]/2208 [broadvoice] username=9738281625 user=phone type=peer secret=PASSWORD-GOES-HERE qualify=1000 port=5060 nat=yes insecure=very host=sip.broadvoice.com fromuser=9738281625 fromdomain=sip.broadvoice.com dtmfmode=inband dtmf=inband disallow=all context=ext-local canreinvite=no authname=9738281625 allow=ulaw allow=g726 allow=g729 -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.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] Persistent variables disappear when dialingLocal extension
Kevin P. Fleming wrote: Falck Kenneth wrote: My persistent variables (_XXX or __XXX) don't persist when I dial a Local extension. I'm doing a forked dial where the other channel is SIP and the other Local. Is this a known problem? Using Asterisk 1.0.9. Variable inheritance is a CVS HEAD feature, it is not supported in 1.0.x. Thanks, I was misguided by http://www.voip-info.org/wiki-Asterisk+Variables which didn't mention this. I guess there is no way to achieve what I want to do with the stable version? I.e. pass call-specific variables when dialling through a Local channel. Now I can only see the original Caller ID and the destination extension, but not the other destination extension of the forked call, which might suffice for my purposes. -- Kenneth Falck, SWelcom Oy, Ludviginkatu 6-8, 00130 Helsinki, Finland Private: [EMAIL PROTECTED] Business: [EMAIL PROTECTED] GSM: +358405103121 Balance is the essential component. ___ 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] Persistent variables disappear when dialingLocal extension
Falck Kenneth wrote: Thanks, I was misguided by http://www.voip-info.org/wiki-Asterisk+Variables which didn't mention this. Yeah. Nobody ever seems to mention on the Wiki when a specific feature became available. ___ 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] Plantronics USB Headsets Audio 45
No memory leaks or choppy sound? |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Wiley Siler |Sent: Martes, 16 de Agosto de 2005 07:00 a.m. |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: RE: [Asterisk-Users] Plantronics USB Headsets Audio 45 | |I use a DSP 500 and I love it. Great sound, good price. | |IaxComm is hands down the best softphone I have found. | |As you can guess it is for IAX though... | |Cheers, |W | | |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Anton Krall |Sent: Monday, August 15, 2005 10:20 PM |To: 'Asterisk Users Mailing List - Non-Commercial Discussion' |Subject: [Asterisk-Users] Plantronics USB Headsets Audio 45 | |Anybody using Plantronics USB headsets? What softphone are you |using and whats your overall experience? Any comments/suggestions? | |___ |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 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 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] Persistent variables disappear when dialingLocal extension
Falck Kenneth wrote: Thanks, I was misguided by http://www.voip-info.org/wiki-Asterisk+Variables which didn't mention this. You are more than welcome to edit the page to make it obvious to the next reader :-) I guess there is no way to achieve what I want to do with the stable version? I.e. pass call-specific variables when dialling through a Local channel. Now I can only see the original Caller ID and the destination extension, but not the other destination extension of the forked call, which might suffice for my purposes. No, I don't know of a simple way to what you want to do. ___ 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] Plantronics USB Headsets Audio 45
BTW, any sip or iax softphones with skin support, for example, for putting you logo in for semi-branding for internal use? |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Anton Krall |Sent: Viernes, 19 de Agosto de 2005 09:54 a.m. |To: 'Asterisk Users Mailing List - Non-Commercial Discussion' |Subject: RE: [Asterisk-Users] Plantronics USB Headsets Audio 45 | |No memory leaks or choppy sound? | | | ||-Original Message- ||From: [EMAIL PROTECTED] ||[mailto:[EMAIL PROTECTED] On Behalf Of Wiley ||Siler ||Sent: Martes, 16 de Agosto de 2005 07:00 a.m. ||To: Asterisk Users Mailing List - Non-Commercial Discussion ||Subject: RE: [Asterisk-Users] Plantronics USB Headsets Audio 45 || ||I use a DSP 500 and I love it. Great sound, good price. || ||IaxComm is hands down the best softphone I have found. || ||As you can guess it is for IAX though... || ||Cheers, ||W || || ||-Original Message- ||From: [EMAIL PROTECTED] ||[mailto:[EMAIL PROTECTED] On Behalf Of Anton ||Krall ||Sent: Monday, August 15, 2005 10:20 PM ||To: 'Asterisk Users Mailing List - Non-Commercial Discussion' ||Subject: [Asterisk-Users] Plantronics USB Headsets Audio 45 || ||Anybody using Plantronics USB headsets? What softphone are you using ||and whats your overall experience? Any comments/suggestions? || ||___ ||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 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 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 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] Overlap digits...
Hello, I'm again there I have also a Problem with Overlap Digits... I'm getting a Call from my Telco to the extension 1234 and i will forward it with exten = 1234,1,Dial(Zap/g1/987654), but asterisk is not dialing 987654, asterisk is dialing 987654 and as overlap digits 1234. so i see on the CDR's from my telco as dialed number 9876541234 and thats not what i want. Do anybody know a solution for this? Nico ___ 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] Persistent variables disappear when dialingLocal extension
On 8/19/05, Falck Kenneth [EMAIL PROTECTED] wrote: I guess there is no way to achieve what I want to do with the stable version? I.e. pass call-specific variables when dialling through a Local channel. Now I can only see the original Caller ID and the destination extension, but not the other destination extension of the forked call, which might suffice for my purposes. We struggled with this a couple years back before CVS_head had that function. What we ended up doing was using the CallerIDName field for a 20 character unique identifier and we used the callerIDnum as usual (telcos in the US only use callerIDnum anyway). That way we were able to track a call from creation as Local/ channel to it's end as whatever channel type it ended up being when it's hung up by the callerIDname. This is certainly not the most elegant solution but is the only way to do it and I guarantee it works, we have been using it for over 2 years now in the astGUIclient project(http://astguiclient.sf.net) MATT--- ___ 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] Automatic start with SuSe linux
Hi, In this link is the script Suse http://www.leals.com/~mm/asterisk/asterisk_suse.sh On 8/18/05, James Oakley [EMAIL PROTECTED] wrote: On Wednesday 17 August 2005 3:04 pm, Tzafrir Cohen wrote: On Wed, Aug 17, 2005 at 01:27:08PM +0300, [EMAIL PROTECTED] wrote: Hi! I'm trying to start asterisk at boottime. Since SuSe It was SuSE (the old way). Now it is SUSE. Was it ever SuSe? Nope, but it was S.u.S.E. before SuSE: http://en.wikipedia.org/wiki/SuSE#History has no rc.local like in Redhat linux, I need asterisk starting script to /etc/init.d/rc3.d -directory (I assume it is like that if i want automated asterisk startup). Do you have any experience how this is implemented in SuSe, and if you have some useful script for starting asterisk, I would be very, i mean VERY pleased? Thank you all in advance! One nice thing SuSE has and most other distros lack is service dependencies: you can define in your init.d script which services your script needs and let insserv sert out the load order. For instance, asterisk needs to load after zaptel. The flash operator panel's daemon needs to start after asterisk. Also, if you install from RPMs, note that the init.d dir of SuSE is actually different than the one od RH. Or at least it was last time I looked. Everything you just described is part of the Linux Standard Base: http://www.linuxbase.org/ SUSE was the first to truly embrace the specification, but Red Hat still only supports the bare minimum, which is why chkconfig still sucks. -- James Oakley Engineering - SolutionInc Ltd. [EMAIL PROTECTED] http://www.solutioninc.com ++ This e-mail is CONFIDENTIAL and contains information intended only for the person(s) named. Any other distribution, copying or disclosure is strictly prohibited. If you have received this e-mail in error, please notify me immediately at 902 420 0077 or reply by e-mail to the sender and destroy the original communication. Thank You. ++ ___ 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 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] Persistent variables disappear when dialingLocalextension
Kevin P. Fleming wrote: Falck Kenneth wrote: Thanks, I was misguided by http://www.voip-info.org/wiki-Asterisk+Variables which didn't mention this. You are more than welcome to edit the page to make it obvious to the next reader :-) You're quite right - I added a little note there to warn others now. It's a great Wiki anyway, just these little inaccuracies or misunderstandings every now and then... -- Kenneth Falck, SWelcom Oy, Ludviginkatu 6-8, 00130 Helsinki, Finland Private: [EMAIL PROTECTED] Business: [EMAIL PROTECTED] GSM: +358405103121 Balance is the essential component. ___ 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] TDM04B, trunk group
During the install [EMAIL PROTECTED] will add all the lines on your card to group 0. Make a trunk for g0 and it will use all the lines. --- Sascha Ferley [EMAIL PROTECTED] wrote: Hi, I am just trying to figure out how to setup a TDM04B card for incoming/outgoing calls. I have 4 lines, which are provided as a rotary trunk group, currently hooked into a Nortel system, which asterisk will replace. I have setup a Dell 1800 (Tower) system with the TDM04B card, which seems to work. The question is how do I set it up that all 4 lines are part of a trunk group, such that all 4 lines can be used for incoming aswell as outgoing calls? I am using [EMAIL PROTECTED] 1.4. Please let me know Thanks S. ___ 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 Start your day with Yahoo! - make it your home page http://www.yahoo.com/r/hs ___ 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] DTMF on Zap / PBX Transfer
Hello, I am hoping someone might be able to help me with this issue. Right now I am testing a X100P card in asterisk connected to a Lucent Partner ACS 3.0 PBX. The card, obviously, acts as an SLT to the system. The card interacts wonderfully with the system. The problem I am having, however, is transferring calls that came in via the PBX, answered on VoIP, to another user on the PBX. The proper way to transfer is to send a hookflash, followed by the extension number, and then hang up. I can send the hookflash via defining a special AGI for transferring (they transfer to extension **24 instead of 24). The hookflash works, however when I try to dial the extension to send the call to, I get a message saying that all cards are currently congested. Does anyone have either a) a way to do this type of transfer or b) can tell me a way, in the AGI, to dial DTMF digits on the current circuit? Any help is appreciated. Thank you, Matt Brennan ___ 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] Persistent variables disappear when dialingLocalextension
Matt Florell wrote: We struggled with this a couple years back before CVS_head had that function. What we ended up doing was using the CallerIDName field for a 20 character unique identifier and we used the callerIDnum as usual (telcos in the US only use callerIDnum anyway). Thanks! I applied this to our case, and I can now store the recipient's CallerID in CALLERIDNAME, and then restore it in the Local context into CALLERIDNUM. The solution is still a little dirty, since CALLERIDNAME will show up in the SIP From header. But I think I can work around that by dialling both calls in the fork through a Local channel that clears CALLERIDNAME before connecting to the real destination. -- Kenneth Falck, SWelcom Oy, Ludviginkatu 6-8, 00130 Helsinki, Finland Private: [EMAIL PROTECTED] Business: [EMAIL PROTECTED] GSM: +358405103121 Balance is the essential component. ___ 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] Sudenly unable to get incoming from Broadvoice
Have you restarted Asterisk to see if that helps? What does 'sip show registry' show? Tom On Aug 19, 2005, at 10:42 AM, Mark Phillips wrote: So it was all working well and then suddenly I'm unable to get incoming calls from BV. Outgoing is fine. I'm using AAH. I have the following settings; [EMAIL PROTECTED]:PASSWORD-GOES-HERE: [EMAIL PROTECTED]/2208 [broadvoice] username=9738281625 user=phone type=peer secret=PASSWORD-GOES-HERE qualify=1000 port=5060 nat=yes insecure=very host=sip.broadvoice.com fromuser=9738281625 fromdomain=sip.broadvoice.com dtmfmode=inband dtmf=inband disallow=all context=ext-local canreinvite=no authname=9738281625 allow=ulaw allow=g726 allow=g729 -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.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 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] Sudenly unable to get incoming from Broadvoice
OK, this seems to be their side at first look. My friend whom has the same setup as me is also having the same problem. Opinions? Mark Phillips wrote: So it was all working well and then suddenly I'm unable to get incoming calls from BV. Outgoing is fine. I'm using AAH. I have the following settings; [EMAIL PROTECTED]:PASSWORD-GOES-HERE:[EMAIL PROTECTED]/2208 [broadvoice] username=9738281625 user=phone type=peer secret=PASSWORD-GOES-HERE qualify=1000 port=5060 nat=yes insecure=very host=sip.broadvoice.com fromuser=9738281625 fromdomain=sip.broadvoice.com dtmfmode=inband dtmf=inband disallow=all context=ext-local canreinvite=no authname=9738281625 allow=ulaw allow=g726 allow=g729 -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.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] Cisco 7960 Line rollover for secretary's phone.
Hi folks. I attempting to set up a Cisco 7960 (SIP) so that if the user is on a call, other incoming calls will ring through to her phone and can be answered. So far I have only been able to get this working by using the call-waiting function, which is cumbersome and does not properly allow the first call to be retreived. Is there a better way to do this? Thanks in advance. John Mensel ___ 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] Cisco 7960 Line rollover for secretary's phone.
On 12:07, Fri 19 Aug 05, John Mensel wrote: Hi folks. I attempting to set up a Cisco 7960 (SIP) so that if the user is on a call, other incoming calls will ring through to her phone and can be answered. So far I have only been able to get this working by using the call-waiting function, which is cumbersome and does not properly allow the first call to be retreived. Is there a better way to do this? Thanks in advance. Try chan_sccp. I had my devices working with SIP, but the SCCP support is great and the SCCP image is way better. -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Why is it drug addicts and computer afficionados are both called users? ___ 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] Cisco 7960 Line rollover for secretary's phone.
If you register 2 line buttons with the same SIP account, then the second call will go to the second button. Also search the list for this. On 8/19/05, John Mensel [EMAIL PROTECTED] wrote: Hi folks. I attempting to set up a Cisco 7960 (SIP) so that if the user is on a call, other incoming calls will ring through to her phone and can be answered. So far I have only been able to get this working by using the call-waiting function, which is cumbersome and does not properly allow the first call to be retreived. Is there a better way to do this? Thanks in advance. John Mensel ___ 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 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] Is there a way in dialplan to determine if call is incoming or outgoing if callerid presentation not enabled on line?
Using contexts, and making sure which device is coming in to where. On 8/19/05, Angus Comber [EMAIL PROTECTED] wrote: Hello If callerid is not available on an external line, how can you tell if call is incoming or outgoing? Angus ___ 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 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] SPA-2100 Analog Telephone Adapter
Sean Rima wrote: Does anyone have any experience of these, I have been offered one and am thinking of adding sticking it onto the back of my Asterisk box and just ignore the WAN port if possible, It would be to stick my exisiting phones onto the asterisk box Sean I just bought 12 of them to link 5 offices PBX systems together. so far in my testing they work extremmly well with asterisk. you will want to modify the dial plan on it otherwise you will get a delay when calling extentions. Dennis ___ 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] OT: autoresponders
Too many people with misconfigured autoresponders... Latest is Make Zuzlak, who has announced he'll be annoying everyone until August 22nd. If people are going on holiday please do one of 3 things: 1. Don't use an autoresponder or 2. Use one that isn't broken.. ie. knows what the Precedence: header is for. or 3. Unsubscribe. Tony ___ 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] SPA-2100 Analog Telephone Adapter
Dennis Gilmore wrote: Sean Rima wrote: Does anyone have any experience of these, I have been offered one and am thinking of adding sticking it onto the back of my Asterisk box and just ignore the WAN port if possible, It would be to stick my exisiting phones onto the asterisk box Sean I just bought 12 of them to link 5 offices PBX systems together. so far in my testing they work extremmly well with asterisk. you will want to modify the dial plan on it otherwise you will get a delay when calling extentions. Excellent, I am still waiting on the bloke to get back to me or else it is ebay :) Sean -- ++ |VOIP: FreeWorldDial 689482 VOIPBuster thecivvie | |GPG Key: http://thecivvie.fastmail.fm/thecivvie.asc | ++ smime.p7s Description: S/MIME Cryptographic Signature ___ 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] initiating Monitor during call
Il Neofita wrote: I put these lines on features.conf in asterisk CVS-v1-0-08/16/05 [featuremap] blindxfer= ## automon = *1 atxfer = *2 You need to use CVS-HEAD for those features. You are using 1.0.x CVS ___ 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] Unexpected hangups when calling Dialogic D/41JTC-LS
Has anyone tried attaching calling a Dialogic D/41JTC-LS (analog) device on another system from an asterisk system with TDM10B? Calling to asterisk from the outside, asterisk correctly dials the internal line and makes the connection to the Dialogic system. A few seconds later Asterisk debug info says it had an On Hook event and hangs up Zap-2-1. I have worked on this problem for over a week--no luck so far. Rollin Weeks ___ 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] Sending digits from SIP to Asterisk's VoiceMailMain
Hi, I am using Asterisk cmd VoiceMailMain to manage voice mail. Problem is, voice mail box can't read password sent from SIP phone, but I don't have any problem to read password from the handset attached to my asterisk box. Your help will be greatly appreciated. Thanks,___ 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] Vonage locked Motorola VT-1000s
Steve Gladden wrote: Very Highly Internested Any chance you could zip or tar your content up and email it to me or give me a link to grab it? Maybe I could help you get it hosted again too ifyou need that. Thanks!!! Steve I would love to have a tarball of my web stuff. I didnt know it was getting moved, and it got moved earlier than expected. I will see if I can get a tarball myself (I should have kept my own backups but ...) -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.10.13/78 - Release Date: 8/19/2005 ___ 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] Sending digits from SIP to Asterisk's VoiceMailMain
It sounds to me like an issue of transmitting DTMF tones from the SIP phones. There are several methods that can be used to accomplish DTMF from SIP phones. Of course, you may ask why it isn't just sent as audio (like a regular POTS phone would.) What happens if you are using a SIP phone, hold down the number 4 button for two seconds (so it sends 2 seconds worth of DTMF on the audio stream) and there is some packet loss during that time? You'll have an audio dropout (thus, tone followed by brief silence and tone again.) The remote end will see this as two tones, not one, which obviously can cause undesired results (and is why it's not a good idea to send DTMF in the audio stream.) That being said, look in your sip.conf for a dtmfmode parameter. You can use inband (in the audio stream, not recommended), RFC2833, or SIP INFO. Your SIP phone should also allow you to set how DTMF is sent (although it may not support all of these formats.) Preferably, use RFC2833 or SIP INFO. Find a setting that is available on your phone and on *, and make sure they're set to match. Once you do that, it should work. Jeremy Innocent Evil wrote: Hi, I am using Asterisk cmd VoiceMailMain to manage voice mail. Problem is, voice mail box can't read password sent from SIP phone, but I don't have any problem to read password from the handset attached to my asterisk box. Your help will be greatly appreciated. Thanks,___ 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 -- Jeremy Gault, KD4NED[EMAIL PROTECTED] Network Administrator, WinWorld Corporation Member: Bradley County ACS/RACES/SkyWarn voice: +1.423.473.8084 fax: +1.423.472.9465 fwd: 461771 msn msgr: [EMAIL PROTECTED] ___ 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] Unexpected hangups when calling Dialogic D/41JTC-LS
Rollin Weeks wrote: Has anyone tried attaching calling a Dialogic D/41JTC-LS (analog) device on another system from an asterisk system with TDM10B? Calling to asterisk from the outside, asterisk correctly dials the internal line and makes the connection to the Dialogic system. A few seconds later Asterisk debug info says it had an On Hook event and hangs up Zap-2-1. set busydetect=no and callprogress=no ___ 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] CVS-HEAD Compile Problem
I ran into the same problem the other day and just went back to non head version It would be nice to figure out why it does this. - Original Message - From: Nico Giefing To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Friday, August 19, 2005 9:20 AM Subject: [Asterisk-Users] CVS-HEAD Compile Problem I have a little Problem, I will compile asterisk CVS-HEAD but after 20 second of compiling i get the message as shown at http://pastebin.com/340654 about 1000 times. Do anybody know a solution for this? Thanks a lot Nico ___ 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] Sending digits from SIP to Asterisk's VoiceMailMain
my sip phone have dtmf relay: rfc2833 asterisk sip.conf have dtmf relay: rfc2833 in associated context. I tried with Inband.. but g729 doesn't support it. I have g729 liscence from digium I havn't try with INFO yet. I prefer to have rfc2833 as dtmf relay. Is there any other thing that can cause this issue? Thanks, -Original Message- From: [EMAIL PROTECTED] Sent: Fri, 19 Aug 2005 14:21:27 -0400 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Sending digits from SIP to Asterisk's VoiceMailMain It sounds to me like an issue of transmitting DTMF tones from the SIP phones. There are several methods that can be used to accomplish DTMF from SIP phones. Of course, you may ask why it isn't just sent as audio (like a regular POTS phone would.) What happens if you are using a SIP phone, hold down the number 4 button for two seconds (so it sends 2 seconds worth of DTMF on the audio stream) and there is some packet loss during that time? You'll have an audio dropout (thus, tone followed by brief silence and tone again.) The remote end will see this as two tones, not one, which obviously can cause undesired results (and is why it's not a good idea to send DTMF in the audio stream.) That being said, look in your sip.conf for a dtmfmode parameter. You can use inband (in the audio stream, not recommended), RFC2833, or SIP INFO. Your SIP phone should also allow you to set how DTMF is sent (although it may not support all of these formats.) Preferably, use RFC2833 or SIP INFO. Find a setting that is available on your phone and on *, and make sure they're set to match. Once you do that, it should work. Jeremy Innocent Evil wrote: Hi, I am using Asterisk cmd VoiceMailMain to manage voice mail. Problem is, voice mail box can't read password sent from SIP phone, but I don't have any problem to read password from the handset attached to my asterisk box. Your help will be greatly appreciated. Thanks,___ 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 -- Jeremy Gault, KD4NED[EMAIL PROTECTED] Network Administrator, WinWorld Corporation Member: Bradley County ACS/RACES/SkyWarn voice: +1.423.473.8084 fax: +1.423.472.9465 fwd: 461771 msn msgr: [EMAIL PROTECTED] ___ 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 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 not conforming to the RFC?/Aastra phone delay issue
Fellow list members, I have run into an issue where I encounter a delay at the beginning of a phone conversation when I make outgoing calls through Asterisk with an Aastra 9133i hardphone. This is most noticable when I call a voicemail system with the Aasta and then with a land line or other VoIP phone. The first word or two of the voicemail message is generally cut off. According to Aastra's engineering this is because Asterisk does not confromt o the RFC, setting FTP voice stream before getting the ACK. They have not seen this with other call servers besides Asterisk. Has anyone else seen this sort of behaviour or is aware of this? Right now we are in the process of switching over the business edition, and we are wondering if we will see a difference in this problem. Thanks, Franklin Webb ___ 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] static noise with this hardware any advice
Follow-up on this We have tried several things without success. Digium responded that the problem was NMI (non-maskable interrupts) and told me to boot linux with the nmi_watchdog=0 option It did not solve the problem. Finally I replaced the TDM card with an older one (revision F) 2FXO 2FXS And the static noise is gone !! Anobody have an idea why this happened Of course it doesn't solve my problem because we have one old card and several new card but it may give digium an idea of where is my problem Patrick Hi We have static noise problem on our asterisk server. latest stable release. The card is a new TDM04B We have it installed on the following hardware Motherboard Intel SE7520BD2SCSI 2x POWER SUPPLY 730W INTEL I will not mention the other hardware because we have desactivated/changed all the other items The only 2 items that we have not changed is the mobo and the power supply. At first it was on scsi drives but we re-installed using a IDE drive We deactivated the two onboard nic and tried two different brand. We have deactivated hyper-treading We have deactivated USB We have deactivated SATA We have tried a noise-cancelling power-bar We have tried two different phones lines We have tried several IP phones, Cisco, Snom, Gnet (There is no noise for a call between two phones) The phone is connected directly in the nic card so there is no network problem possible. We have tried several TDM Card Anybody knows if the motherboard or the power-supply could be the problem ? ___ 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 and Vonage - Can't call out but can receive calls
Hi, I'm trying to get Asterisk to connect to Vonage (softphone acct) to allow me to place and receive calls. I have successfully configured Asterisk to route inbound calls and send them to the correct extension, but I can't get outbound calls to work. I have Asterisk successfully registering with Vonage, but when an INVITE is sent out, I get a 404 Not Found back from Vonage and thus Asterisk shows SIP/atlas-east.vonage.net is circuit-busy Below are my sip.conf and extensions.conf files that I am using. I saw several other sample configuration files that claimed connectivity to Vonage, but I have not been able to get them to work. My Asterisk server is behind a NAT firewall. The files below are the closest I have gotten to complete connectivity. Any feedback is appreciated! One note: I was able to receive calls only if I used atlas-east as the Vonage server. If I used sphone.vopr.vonage.net, no calls came in. sip.conf: [general] externip=X.X.X.X port=5060 bindaddr=X.X.X.X context=vonage-out disallow=all allow=ulaw allow=alaw nat=yes register=:[EMAIL PROTECTED]:5060/201 [vonage] username= type=peer secret=PASSWORD port=5060 nat=yes host=atlas-east.vonage.net fromdomain=vonage.net canreinvite=no fromuser= dtmfmode=rfc2833 context=vonage-out [201] type=friend username=201 secret=PASSWORD host=dynamic dtmfmode=rfc2833 defaultip=X.X.X.X mailbox=201 callerid=NAME progressinband=no context=from-sip extensions.conf (relevant part) [vonage-out] exten = ,1,Goto(from-sip,201,1) [from-sip] exten = _9.,1,Dial(SIP/[EMAIL PROTECTED]) exten = 201,1,Dial(SIP/201) exten = 202,1,Dial(SIP/202) Thanks, -- Brian ___ 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 and Vonage - Can't call out but can receive calls
Brian Deep wrote:
[from-sip]
exten = _9.,1,Dial(SIP/[EMAIL PROTECTED])
exten = 201,1,Dial(SIP/201)
exten = 202,1,Dial(SIP/202)
try exten = _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
___
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] Sending digits from SIP to Asterisk's VoiceMailMain
If you can get an rtp debug while your pressing digits I can see if maybe your device is sending the digits incorrectly. /b On Aug 19, 2005, at 1:46 PM, Innocent Evil wrote: my sip phone have dtmf relay: rfc2833 asterisk sip.conf have dtmf relay: rfc2833 in associated context. I tried with Inband.. but g729 doesn't support it. I have g729 liscence from digium I havn't try with INFO yet. I prefer to have rfc2833 as dtmf relay. Is there any other thing that can cause this issue? Thanks, -Original Message- From: [EMAIL PROTECTED] Sent: Fri, 19 Aug 2005 14:21:27 -0400 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Sending digits from SIP to Asterisk's VoiceMailMain It sounds to me like an issue of transmitting DTMF tones from the SIP phones. There are several methods that can be used to accomplish DTMF from SIP phones. Of course, you may ask why it isn't just sent as audio (like a regular POTS phone would.) What happens if you are using a SIP phone, hold down the number 4 button for two seconds (so it sends 2 seconds worth of DTMF on the audio stream) and there is some packet loss during that time? You'll have an audio dropout (thus, tone followed by brief silence and tone again.) The remote end will see this as two tones, not one, which obviously can cause undesired results (and is why it's not a good idea to send DTMF in the audio stream.) That being said, look in your sip.conf for a dtmfmode parameter. You can use inband (in the audio stream, not recommended), RFC2833, or SIP INFO. Your SIP phone should also allow you to set how DTMF is sent (although it may not support all of these formats.) Preferably, use RFC2833 or SIP INFO. Find a setting that is available on your phone and on *, and make sure they're set to match. Once you do that, it should work. Jeremy Innocent Evil wrote: Hi, I am using Asterisk cmd VoiceMailMain to manage voice mail. Problem is, voice mail box can't read password sent from SIP phone, but I don't have any problem to read password from the handset attached to my asterisk box. Your help will be greatly appreciated. Thanks,___ 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 -- Jeremy Gault, KD4NED[EMAIL PROTECTED] Network Administrator, WinWorld Corporation Member: Bradley County ACS/RACES/SkyWarn voice: +1.423.473.8084 fax: +1.423.472.9465 fwd: 461771 msn msgr: [EMAIL PROTECTED] ___ 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 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 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 and Vonage - Can't call out but can receive calls
You are calling a sip host that you do not have defined in sip.conf.
I think the line should look like this
exten = _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
This will force * to look in its sip.conf file for a stanza called
[vonage] which you have rather than [atlas-east.vonage.net] which you
don't have.
Brian Deep wrote:
Hi,
I'm trying to get Asterisk to connect to Vonage (softphone acct) to allow me to
place and receive
calls. I have successfully configured Asterisk to route inbound calls and send
them to the correct extension, but I can't get outbound calls to work. I have
Asterisk successfully registering with Vonage, but when an INVITE is sent out, I
get a 404 Not Found back from Vonage and thus Asterisk shows
SIP/atlas-east.vonage.net is circuit-busy
Below are my sip.conf and extensions.conf files that I am using. I saw several other sample configuration
files that claimed connectivity to Vonage, but I have not been able to get them
to work. My Asterisk server is behind a NAT firewall.
The files below are the closest I have gotten to complete connectivity. Any
feedback is appreciated!
One note:
I was able to receive calls only if I used atlas-east as the Vonage server. If
I used sphone.vopr.vonage.net, no calls came in.
sip.conf:
[general]
externip=X.X.X.X
port=5060
bindaddr=X.X.X.X
context=vonage-out
disallow=all
allow=ulaw
allow=alaw
nat=yes
register=:[EMAIL PROTECTED]:5060/201
[vonage]
username=
type=peer
secret=PASSWORD
port=5060
nat=yes
host=atlas-east.vonage.net
fromdomain=vonage.net
canreinvite=no
fromuser=
dtmfmode=rfc2833
context=vonage-out
[201]
type=friend
username=201
secret=PASSWORD
host=dynamic
dtmfmode=rfc2833
defaultip=X.X.X.X
mailbox=201
callerid=NAME
progressinband=no
context=from-sip
extensions.conf (relevant part)
[vonage-out]
exten = ,1,Goto(from-sip,201,1)
[from-sip]
exten = _9.,1,Dial(SIP/[EMAIL PROTECTED])
exten = 201,1,Dial(SIP/201)
exten = 202,1,Dial(SIP/202)
Thanks,
-- Brian
___
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
--
Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.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] Sudenly unable to get incoming from Broadvoice
Well, some smarty pants lady at broadvoice, claim that the problem is in our end, well, I have taking my box out of the picture, I went to bv control panel and have forwarded the calls to my home phone number, she STILL insist that the problem is my asterisk box, the one I deleted the Broadvoice trunk.. ;) Maybe I should just leave the trunk deleted and don't fight it anymore... :( The real funny part is the if I call from teliax to my 10 digit number the call get forwarded to my home, NO problem.. Is only when the number is called from a real PSTN number that the person get fast busy, well fast busy today, yesterday was this number has been disconnected... Maybe next she is going to say bv will not work with asterisk box out of the picture...jajajajaja Manny -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Phillips Sent: Friday, August 19, 2005 1:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Sudenly unable to get incoming from Broadvoice Yes, I've restarted asterisk and even rebooted my machine. sip show registry shows pbx*CLI sip show registry HostUsername Refresh State sip.varphonex.com:5060 8281625105 Registered sip.broadvoice.com:5060 [EMAIL PROTECTED] 3495 Registered pbx*CLI I did the same on my friends machine and it show the same thing. Why is the refresh period so large and what can I do to shorten it? I've ruled out any ISP issues. I can receive calls on my other VoIP services just fine. Mark Tom Rymes wrote: Have you restarted Asterisk to see if that helps? What does 'sip show registry' show? Tom On Aug 19, 2005, at 10:42 AM, Mark Phillips wrote: So it was all working well and then suddenly I'm unable to get incoming calls from BV. Outgoing is fine. I'm using AAH. I have the following settings; [EMAIL PROTECTED]:PASSWORD-GOES-HERE: [EMAIL PROTECTED]/2208 [broadvoice] username=9738281625 user=phone type=peer secret=PASSWORD-GOES-HERE qualify=1000 port=5060 nat=yes insecure=very host=sip.broadvoice.com fromuser=9738281625 fromdomain=sip.broadvoice.com dtmfmode=inband dtmf=inband disallow=all context=ext-local canreinvite=no authname=9738281625 allow=ulaw allow=g726 allow=g729 ___ 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] Ascend Pipeline POTS to TDM400P FXO Question..
I have a TDM400P with some FXO ports, and I wanted to connect the two POTS lines from my Pipeline-75 ISDN router into the FXO interfaces on my Asterisk server. Hooked it up, seemed fine, called in and it answered. The problem is when the call is hung up on, the FXO port never drops. So of course then the P75 just holds the line off hook and you get a busy. So it's good for the first call, and then it's done. Does anyone know of any adjustments that will make this work? Figured maybe someone here has run into such an issue before. Hooking it to a normal POTS line works great, but out of the P75 seems to be a no-go outside of the first call. --- Howard Leadmon http://www.leadmon.net ___ 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] Ascend Pipeline POTS to TDM400P FXO Question..
Do you need a hangup in your dialplan? -Jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Howard Leadmon Sent: Friday, August 19, 2005 4:06 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Ascend Pipeline POTS to TDM400P FXO Question.. I have a TDM400P with some FXO ports, and I wanted to connect the two POTS lines from my Pipeline-75 ISDN router into the FXO interfaces on my Asterisk server. Hooked it up, seemed fine, called in and it answered. The problem is when the call is hung up on, the FXO port never drops. So of course then the P75 just holds the line off hook and you get a busy. So it's good for the first call, and then it's done. Does anyone know of any adjustments that will make this work? Figured maybe someone here has run into such an issue before. Hooking it to a normal POTS line works great, but out of the P75 seems to be a no-go outside of the first call. --- Howard Leadmon http://www.leadmon.net ___ 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 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] Ascend Pipeline POTS to TDM400P FXO Question..
OK, I nailed it, it's working now. If any are curious, seems in the P75 there is an option called Forward Disconnect and by default it's set to NO, and needed to be set to YES so it sends the disconnect to the TDM card. --- Howard Leadmon http://www.leadmon.net -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Howard Leadmon Sent: Friday, August 19, 2005 4:06 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Ascend Pipeline POTS to TDM400P FXO Question.. I have a TDM400P with some FXO ports, and I wanted to connect the two POTS lines from my Pipeline-75 ISDN router into the FXO interfaces on my Asterisk server. Hooked it up, seemed fine, called in and it answered. The problem is when the call is hung up on, the FXO port never drops. So of course then the P75 just holds the line off hook and you get a busy. So it's good for the first call, and then it's done. Does anyone know of any adjustments that will make this work? Figured maybe someone here has run into such an issue before. Hooking it to a normal POTS line works great, but out of the P75 seems to be a no-go outside of the first call. --- Howard Leadmon http://www.leadmon.net ___ 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 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] Sudenly unable to get incoming from Broadvoice
Same here. When I turn on their forwarding or their VM my inbound calls complete. Thing is though, why suddenly now does it not work. I can't believ that 3 of us have been messing with our boxes at the same time? I reckon they made a change at their end and won't fess up. Manny A. Wise wrote: Well, some smarty pants lady at broadvoice, claim that the problem is in our end, well, I have taking my box out of the picture, I went to bv control panel and have forwarded the calls to my home phone number, she STILL insist that the problem is my asterisk box, the one I deleted the Broadvoice trunk.. ;) Maybe I should just leave the trunk deleted and don't fight it anymore... :( The real funny part is the if I call from teliax to my 10 digit number the call get forwarded to my home, NO problem.. Is only when the number is called from a real PSTN number that the person get fast busy, well fast busy today, yesterday was this number has been disconnected... Maybe next she is going to say bv will not work with asterisk box out of the picture...jajajajaja Manny -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Phillips Sent: Friday, August 19, 2005 1:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Sudenly unable to get incoming from Broadvoice Yes, I've restarted asterisk and even rebooted my machine. sip show registry shows pbx*CLI sip show registry HostUsername Refresh State sip.varphonex.com:5060 8281625105 Registered sip.broadvoice.com:5060 [EMAIL PROTECTED] 3495 Registered pbx*CLI I did the same on my friends machine and it show the same thing. Why is the refresh period so large and what can I do to shorten it? I've ruled out any ISP issues. I can receive calls on my other VoIP services just fine. Mark Tom Rymes wrote: Have you restarted Asterisk to see if that helps? What does 'sip show registry' show? Tom On Aug 19, 2005, at 10:42 AM, Mark Phillips wrote: So it was all working well and then suddenly I'm unable to get incoming calls from BV. Outgoing is fine. I'm using AAH. I have the following settings; [EMAIL PROTECTED]:PASSWORD-GOES-HERE: [EMAIL PROTECTED]/2208 [broadvoice] username=9738281625 user=phone type=peer secret=PASSWORD-GOES-HERE qualify=1000 port=5060 nat=yes insecure=very host=sip.broadvoice.com fromuser=9738281625 fromdomain=sip.broadvoice.com dtmfmode=inband dtmf=inband disallow=all context=ext-local canreinvite=no authname=9738281625 allow=ulaw allow=g726 allow=g729 ___ 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 -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.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] Sudenly unable to get incoming from Broadvoice
OK, now I know of 5 peeps that suddenly are having this problem. It has to be them right? Mark (in the rainy end of NNJ) Manny A. Wise wrote: Well, some smarty pants lady at broadvoice, claim that the problem is in our end, well, I have taking my box out of the picture, I went to bv control panel and have forwarded the calls to my home phone number, she STILL insist that the problem is my asterisk box, the one I deleted the Broadvoice trunk.. ;) Maybe I should just leave the trunk deleted and don't fight it anymore... :( The real funny part is the if I call from teliax to my 10 digit number the call get forwarded to my home, NO problem.. Is only when the number is called from a real PSTN number that the person get fast busy, well fast busy today, yesterday was this number has been disconnected... Maybe next she is going to say bv will not work with asterisk box out of the picture...jajajajaja Manny -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Phillips Sent: Friday, August 19, 2005 1:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Sudenly unable to get incoming from Broadvoice Yes, I've restarted asterisk and even rebooted my machine. sip show registry shows pbx*CLI sip show registry HostUsername Refresh State sip.varphonex.com:5060 8281625105 Registered sip.broadvoice.com:5060 [EMAIL PROTECTED] 3495 Registered pbx*CLI I did the same on my friends machine and it show the same thing. Why is the refresh period so large and what can I do to shorten it? I've ruled out any ISP issues. I can receive calls on my other VoIP services just fine. Mark Tom Rymes wrote: Have you restarted Asterisk to see if that helps? What does 'sip show registry' show? Tom On Aug 19, 2005, at 10:42 AM, Mark Phillips wrote: So it was all working well and then suddenly I'm unable to get incoming calls from BV. Outgoing is fine. I'm using AAH. I have the following settings; [EMAIL PROTECTED]:PASSWORD-GOES-HERE: [EMAIL PROTECTED]/2208 [broadvoice] username=9738281625 user=phone type=peer secret=PASSWORD-GOES-HERE qualify=1000 port=5060 nat=yes insecure=very host=sip.broadvoice.com fromuser=9738281625 fromdomain=sip.broadvoice.com dtmfmode=inband dtmf=inband disallow=all context=ext-local canreinvite=no authname=9738281625 allow=ulaw allow=g726 allow=g729 ___ 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 -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.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 not conforming to the RFC?/Aastra phonedelay issue
I don't believe that this issue is with Asterisk. The issue is that the phone does not set up the RTP stream until it receives the "200 OK". Asterisk sets up the RTP stream when it receives, or sends,the message with SDP (either INVITE message, 180 response or 183 response), as per the RFC. The firmware for the 91XXi phones were branched from the 480i version back in late 2004. Since this branching, the 480i firmware has been fixed, and the phone now sets up the "early media". The following is from the "KNOWN ISSUES" section of the 480i release notes from November 2004, v1.0.0.50. 2.6 FAILURE TO PROCESS "REMOTE RINGBACK" A device may send an invitation that includes an SDP message. When the terminator responds with an alert message, it many also contain SDP message. This is known as "early media" or "remote ring-back" and indicates that the terminator will provide ring-back tone. The originating device should provide this tone to the originating user. Currently, the firmware does not process this remote ringback. Currently under investigation. I would suspect that the 91XXi firmware was never updated to correct this problem. Also, The Asterisk voicemail application will answer the channel if it hasn't already been answered, before playing any prompts. Answering the channel send a "200 OK" message to the phone. The phone will then set up the RTP voice stream for the call. So, if the phone is "clipping" the begining of the voice prompts, it is an indication that the phone is taking a long time to set up the RTP after receiving the "200 OK" message. This behaviour is not seen on the other call servers that Aastra test the phone against, due to the fact that this issue was known a long time ago, and these call servers have incorporated a workaround to deal with it. Fortunately, there is a workaround for Asterisk as well. In your dialplan, issue an "Answer" and "wait(1)" berfore sending the call to voicemail. Regards, Derek - Original Message - From: Franklin Webb To: Asterisk-Users@lists.digium.com Sent: Friday, August 19, 2005 12:49 PM Subject: [Asterisk-Users] Asterisk not conforming to the RFC?/Aastra phonedelay issue Fellow list members, I have run into an issue where I encounter a delay at the beginning of a phone conversation when I make outgoing calls through Asterisk with an Aastra 9133i hardphone. This is most noticable when I call a voicemail system with the Aasta and then with a land line or other VoIP phone. The first word or two of the voicemail message is generally cut off. According to Aastra's engineering this is because Asterisk does not confromt o the RFC, setting FTP voice stream before getting the ACK. They have not seen this with other call servers besides Asterisk. Has anyone else seen this sort of behaviour or is aware of this? Right now we are in the process of switching over the business edition, and we are wondering if we will see a difference in this problem. Thanks, Franklin Webb ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ 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[2]: [Asterisk-Users] Asterisk and Vonage - Can't call out but can receive calls
Mark,
Thank you very much!!! That was exactly it. My config files now look like the
following and I can send and receive calls using Vonage.
sip.conf:
[general]
externip=X.X.X.X
port=5060
bindaddr=X.X.X.X
context=vonage-out
disallow=all
allow=ulaw
allow=alaw
nat=yes
register=:[EMAIL PROTECTED]:5060/201
[vonage]
username=
type=peer
secret=PASSWORD
port=5060
nat=yes
host=atlas-east.vonage.net
fromdomain=vonage.net
canreinvite=no
fromuser=
dtmfmode=rfc2833
context=vonage-out
[201]
type=friend
username=201
secret=PASSWORD
host=dynamic
dtmfmode=rfc2833
defaultip=X.X.X.X
mailbox=201
callerid=NAME
progressinband=no
context=from-sip
extension.conf:
[vonage-out]
exten = ,1,Goto(from-sip,201,1)
[from-sip]
exten = _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
Thanks,
-- Brian
Friday, August 19, 2005, 3:50:12 PM, you wrote:
You are calling a sip host that you do not have defined in sip.conf.
I think the line should look like this
exten = _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
This will force * to look in its sip.conf file for a stanza called
[vonage] which you have rather than [atlas-east.vonage.net] which you
don't have.
extensions.conf (relevant part)
[vonage-out]
exten = ,1,Goto(from-sip,201,1)
[from-sip]
exten = _9.,1,Dial(SIP/[EMAIL PROTECTED])
exten = 201,1,Dial(SIP/201)
exten = 202,1,Dial(SIP/202)
Thanks,
-- Brian
___
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 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] Sound warnings bringing asterisk down.
Does anybody know what would be causing the errors below? I get these errors continuously until asterisk finally quits. This happens when I make 20 simultaneous SIP calls with the Dial Command. chan_oss.c:291 sound_thread: Failed to write sound chan_oss.c:200 send_sound: Unable to read output space __ Do you Yahoo!? Yahoo! Mail - Find what you need with new enhanced search. http://info.mail.yahoo.com/mail_250 ___ 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] Sending digits from SIP to Asterisk's VoiceMailMain
Other than below: Got RTP packet from x.y.z.sip_phone:10006 (type 18, seq 25407, ts 191360, len 40) Sent RTP packet to x.y.z.asterisk:10006 (type 18, seq 63928, ts 193744, len 20) I dont see any message while sending digits. -Original Message- From: [EMAIL PROTECTED] Sent: Fri, 19 Aug 2005 14:33:14 -0500 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Sending digits from SIP to Asterisk's VoiceMailMain If you can get an rtp debug while your pressing digits I can see if maybe your device is sending the digits incorrectly. /b On Aug 19, 2005, at 1:46 PM, Innocent Evil wrote: my sip phone have dtmf relay: rfc2833 asterisk sip.conf have dtmf relay: rfc2833 in associated context. I tried with Inband.. but g729 doesn't support it. I have g729 liscence from digium I havn't try with INFO yet. I prefer to have rfc2833 as dtmf relay. Is there any other thing that can cause this issue? Thanks, -Original Message- From: [EMAIL PROTECTED] Sent: Fri, 19 Aug 2005 14:21:27 -0400 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Sending digits from SIP to Asterisk's VoiceMailMain It sounds to me like an issue of transmitting DTMF tones from the SIP phones. There are several methods that can be used to accomplish DTMF from SIP phones. Of course, you may ask why it isn't just sent as audio (like a regular POTS phone would.) What happens if you are using a SIP phone, hold down the number 4 button for two seconds (so it sends 2 seconds worth of DTMF on the audio stream) and there is some packet loss during that time? You'll have an audio dropout (thus, tone followed by brief silence and tone again.) The remote end will see this as two tones, not one, which obviously can cause undesired results (and is why it's not a good idea to send DTMF in the audio stream.) That being said, look in your sip.conf for a dtmfmode parameter. You can use inband (in the audio stream, not recommended), RFC2833, or SIP INFO. Your SIP phone should also allow you to set how DTMF is sent (although it may not support all of these formats.) Preferably, use RFC2833 or SIP INFO. Find a setting that is available on your phone and on *, and make sure they're set to match. Once you do that, it should work. Jeremy Innocent Evil wrote: Hi, I am using Asterisk cmd VoiceMailMain to manage voice mail. Problem is, voice mail box can't read password sent from SIP phone, but I don't have any problem to read password from the handset attached to my asterisk box. Your help will be greatly appreciated. Thanks,___ 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 -- Jeremy Gault, KD4NED[EMAIL PROTECTED] Network Administrator, WinWorld Corporation Member: Bradley County ACS/RACES/SkyWarn voice: +1.423.473.8084 fax: +1.423.472.9465 fwd: 461771 msn msgr: [EMAIL PROTECTED] ___ 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 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 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 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] Overriding Caller ID
Hello list,
We have some kind of a problem with our Asterisk installation. We
want to be able to publish different caller id when placing outbound
calls through the PSTN. We have Asterisk with TE410P and T1 from FDN
Communications. The problem is that all our outbound calls show our
main number, regardless of what we set with SetCallerID, even using
CallingPres with all possible combinations. When speaking with FDN,
they say they have set their T1 to show our main number for outbound
calls, but that we should be able to override that with no problem.
As I said, I have tried all possible combinations, yet, nothing seems
to work. Below are snippets of some of our configs:
extensions.conf
;
; Local calls
;
exten = _NXXNXX,1,CallingPres(32)
exten = _NXXNXX,2,SetCallerID(2125551234)
exten = _NXXNXX,3,Dial(${TRUNK_LO}/${EXTEN})
zapata.conf
[channels]
usecallerid=yes
cidsignalling=bell
cidstart=ring
hidecallerid=no
restrictcid=no
usecallingpres=yes
callerid=asreceived
switchtype = dms100
signalling = em_w
group = 1
context=inbound
callerid=asreceived
channel = 1-24
Does anyone have any suggestions?
Thanks,
Waldo
___
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] Unexpected hangups when calling Dialogic D/41JTC-LS
Thanks Eric, I tried the changes to zapata.conf. I still get the hangup. It makes me wonder if the Dialogic card is sending a hangup tone to the FXO module. It seems to work OK if I use an analog phone instead of linking to the Dialogic card. RollinOn 8/19/05, Eric Wieling aka ManxPower [EMAIL PROTECTED] wrote: Rollin Weeks wrote: Has anyone tried attaching calling a Dialogic D/41JTC-LS (analog) device on another system from an asterisk system with TDM10B? Calling to asterisk from the outside, asterisk correctly dials the internal line and makes the connection to the Dialogic system. A few seconds later Asterisk debug info says it had an On Hook event and hangs up Zap-2-1.set busydetect=no and callprogress=no___ 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-users ___ 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 X100P clone
i am install asterisk in gentoo linux, #emerge zaptel #emerge asterisk #modprobe zaptel #modprobe wcfxo #asterisk -vvvc localhost ~ # asterisk -vvvc == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Found Asterisk 1.0.8, Copyright (C) 1999-2004 Digium. Written by Mark Spencer [EMAIL PROTECTED] = == Parsing '/etc/asterisk/logger.conf': Found Asterisk Event Logger Started /var/log/asterisk/event_log == Manager registered action Ping == Manager registered action Events == Manager registered action Logoff == Manager registered action Hangup == Manager registered action Status == Manager registered action Setvar == Manager registered action Getvar == Manager registered action Redirect == Manager registered action Originate == Manager registered action Command == Manager registered action ExtensionState == Manager registered action AbsoluteTimeout == Manager registered action MailboxStatus == Manager registered action MailboxCount == Manager registered action ListCommands == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/rtp.conf': Found == RTP Allocating from port range 1 - 2 Asterisk PBX Core Initializing Registering builtin applications: [AbsoluteTimeout] == Registered application 'AbsoluteTimeout' [Answer] == Registered application 'Answer' [BackGround] == Registered application 'BackGround' [Busy] == Registered application 'Busy' [Congestion] == Registered application 'Congestion' [DigitTimeout] == Registered application 'DigitTimeout' [Goto] == Registered application 'Goto' [GotoIf] == Registered application 'GotoIf' [GotoIfTime] == Registered application 'GotoIfTime' [Hangup] == Registered application 'Hangup' [NoOp] == Registered application 'NoOp' [Prefix] == Registered application 'Prefix' [Progress] == Registered application 'Progress' [ResetCDR] == Registered application 'ResetCDR' [ResponseTimeout] == Registered application 'ResponseTimeout' [Ringing] == Registered application 'Ringing' [SayNumber] == Registered application 'SayNumber' [SayDigits] == Registered application 'SayDigits' [SayAlpha] == Registered application 'SayAlpha' [SayPhonetic] == Registered application 'SayPhonetic' [SetAccount] == Registered application 'SetAccount' [SetAMAFlags] == Registered application 'SetAMAFlags' [SetGlobalVar] == Registered application 'SetGlobalVar' [SetLanguage] == Registered application 'SetLanguage' [SetVar] == Registered application 'SetVar' [StripMSD] == Registered application 'StripMSD' [Suffix] == Registered application 'Suffix' [Wait] == Registered application 'Wait' [WaitExten] == Registered application 'WaitExten' Asterisk Dynamic Loader Starting: == Parsing '/etc/asterisk/modules.conf': Found [chan_modem.so] = (Generic Voice Modem Driver) == Parsing '/etc/asterisk/modem.conf': Found == Loading modem driver chan_modem_aopen.so = (A/Open (Rockwell Chipset) ITU-2 VoiceModem Driver) == Registered channel type 'Modem' (Generic Voice Modem Channel Driver) [res_musiconhold.so] = (Music On Hold Resource) == Parsing '/etc/asterisk/musiconhold.conf': Found == Registered application 'MusicOnHold' == Registered application 'WaitMusicOnHold' == Registered application 'SetMusicOnHold' [res_adsi.so] = (ADSI Resource) == Parsing '/etc/asterisk/adsi.conf': Found [res_features.so] = (Call Parking Resource) == Parsing '/etc/asterisk/features.conf': Found -- Registered extension context 'parkedcalls' -- Added extension '700' priority 1 to parkedcalls == Registered application 'ParkedCall' == Registered application 'Park' == Manager registered action ParkedCalls [res_crypto.so] = (Cryptographic Digital Signatures) -- Loaded PUBLIC key 'iaxtel' -- Loaded PUBLIC key 'freeworlddialup' [res_indications.so] = (Indications Configuration) == Parsing '/etc/asterisk/indications.conf': Found -- Registered indication country 'cl' -- Registered indication country 'tw' -- Registered indication country 'us' -- Registered indication country 'au' -- Registered indication country 'fr' -- Registered indication country 'de' -- Registered indication country 'nl' -- Registered indication country 'uk' -- Registered indication country 'fi' -- Registered indication country 'no' -- Registered indication country 'br' -- Registered indication country 'za' -- Registered indication country 'it' -- Registered indication country 'us-o' -- Registered indication country 'gr' -- Registered indication country 'ru' -- Registered indication country 'nz' -- Registered indication country 'sg' -- Registered indication country 'hu' -- Registered indication country 'lt' -- Registered indication country 'pl' -- Registered indication country
Re: [Asterisk-Users] Overriding Caller ID
Waldo Rubinstein wrote: switchtype = dms100 signalling = em_w group = 1 context=inbound callerid=asreceived channel = 1-24 You cannot set your own Caller ID on anything except PRI. ___ 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 and Vonage - Can't call out but can receive calls
OK, now that you have it working put it into the WIKI!!
Mark
Brian Deep wrote:
Mark,
Thank you very much!!! That was exactly it. My config files now look like the
following and I can send and receive calls using Vonage.
sip.conf:
[general]
externip=X.X.X.X
port=5060
bindaddr=X.X.X.X
context=vonage-out
disallow=all
allow=ulaw
allow=alaw
nat=yes
register=:[EMAIL PROTECTED]:5060/201
[vonage]
username=
type=peer
secret=PASSWORD
port=5060
nat=yes
host=atlas-east.vonage.net
fromdomain=vonage.net
canreinvite=no
fromuser=
dtmfmode=rfc2833
context=vonage-out
[201]
type=friend
username=201
secret=PASSWORD
host=dynamic
dtmfmode=rfc2833
defaultip=X.X.X.X
mailbox=201
callerid=NAME
progressinband=no
context=from-sip
extension.conf:
[vonage-out]
exten = ,1,Goto(from-sip,201,1)
[from-sip]
exten = _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
Thanks,
-- Brian
Friday, August 19, 2005, 3:50:12 PM, you wrote:
You are calling a sip host that you do not have defined in sip.conf.
I think the line should look like this
exten = _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
This will force * to look in its sip.conf file for a stanza called
[vonage] which you have rather than [atlas-east.vonage.net] which you
don't have.
extensions.conf (relevant part)
[vonage-out]
exten = ,1,Goto(from-sip,201,1)
[from-sip]
exten = _9.,1,Dial(SIP/[EMAIL PROTECTED])
exten = 201,1,Dial(SIP/201)
exten = 202,1,Dial(SIP/202)
Thanks,
-- Brian
___
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 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
--
Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.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] Sudenly unable to get incoming from Broadvoice
I'm using their dca proxy and have not had any problems at all today with them. I've got 201 and 212 DID's with them and both have completed incoming calls throughout the day today. On 8/19/05, Mark Phillips [EMAIL PROTECTED] wrote: OK, now I know of 5 peeps that suddenly are having this problem. It has to be them right? Mark (in the rainy end of NNJ) Manny A. Wise wrote: Well, some smarty pants lady at broadvoice, claim that the problem is in our end, well, I have taking my box out of the picture, I went to bv control panel and have forwarded the calls to my home phone number, she STILL insist that the problem is my asterisk box, the one I deleted the Broadvoice trunk.. ;) Maybe I should just leave the trunk deleted and don't fight it anymore... :( The real funny part is the if I call from teliax to my 10 digit number the call get forwarded to my home, NO problem.. Is only when the number is called from a real PSTN number that the person get fast busy, well fast busy today, yesterday was this number has been disconnected... Maybe next she is going to say bv will not work with asterisk box out of the picture...jajajajaja Manny -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Phillips Sent: Friday, August 19, 2005 1:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Sudenly unable to get incoming from Broadvoice Yes, I've restarted asterisk and even rebooted my machine. sip show registry shows pbx*CLI sip show registry HostUsername Refresh State sip.varphonex.com:5060 8281625105 Registered sip.broadvoice.com:5060 [EMAIL PROTECTED] 3495 Registered pbx*CLI I did the same on my friends machine and it show the same thing. Why is the refresh period so large and what can I do to shorten it? I've ruled out any ISP issues. I can receive calls on my other VoIP services just fine. Mark Tom Rymes wrote: Have you restarted Asterisk to see if that helps? What does 'sip show registry' show? Tom On Aug 19, 2005, at 10:42 AM, Mark Phillips wrote: So it was all working well and then suddenly I'm unable to get incoming calls from BV. Outgoing is fine. I'm using AAH. I have the following settings; [EMAIL PROTECTED]:PASSWORD-GOES-HERE: [EMAIL PROTECTED]/2208 [broadvoice] username=9738281625 user=phone type=peer secret=PASSWORD-GOES-HERE qualify=1000 port=5060 nat=yes insecure=very host=sip.broadvoice.com fromuser=9738281625 fromdomain=sip.broadvoice.com dtmfmode=inband dtmf=inband disallow=all context=ext-local canreinvite=no authname=9738281625 allow=ulaw allow=g726 allow=g729 ___ 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 -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.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 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] Overriding Caller ID
I may be wrong here, so if anyone else here knows contrary, please feel
free to jump in and correct me. ::dons his asbestos armor::
When we first deployed * we were coming from an analog channel bank
setup (hooked into our old PBX as analog lines.) I was able to connect
* to the T1 and use EM Wink signaling to make things work. However, we
couldn't control our caller ID. The number that appeared would depend
on which channel the call took.
Not long thereafter, we migrated to a PRI. Once we were on the PRI, we
were able to have control of the CID.
As far as I know, you can't control your outbound CID on a T1 setup the
way yours is. You probably need to switch to a PRI instead if you want
this ability. But again, that's based on my knowledge and experience,
so I could be wrong. If so, hopefully someone else here will clear it
up for both of us.
Jeremy
Waldo Rubinstein wrote:
Hello list,
We have some kind of a problem with our Asterisk installation. We
want to be able to publish different caller id when placing outbound
calls through the PSTN. We have Asterisk with TE410P and T1 from FDN
Communications. The problem is that all our outbound calls show our
main number, regardless of what we set with SetCallerID, even using
CallingPres with all possible combinations. When speaking with FDN,
they say they have set their T1 to show our main number for outbound
calls, but that we should be able to override that with no problem.
As I said, I have tried all possible combinations, yet, nothing seems
to work. Below are snippets of some of our configs:
extensions.conf
;
; Local calls
;
exten = _NXXNXX,1,CallingPres(32)
exten = _NXXNXX,2,SetCallerID(2125551234)
exten = _NXXNXX,3,Dial(${TRUNK_LO}/${EXTEN})
zapata.conf
[channels]
usecallerid=yes
cidsignalling=bell
cidstart=ring
hidecallerid=no
restrictcid=no
usecallingpres=yes
callerid=asreceived
switchtype = dms100
signalling = em_w
group = 1
context=inbound
callerid=asreceived
channel = 1-24
Does anyone have any suggestions?
Thanks,
Waldo
___
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
--
Jeremy Gault, KD4NED[EMAIL PROTECTED]
Network Administrator, WinWorld Corporation
Member: Bradley County ACS/RACES/SkyWarn
voice: +1.423.473.8084 fax: +1.423.472.9465
fwd: 461771 msn msgr: [EMAIL PROTECTED]
___
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] Sudenly unable to get incoming from Broadvoice
Working fine here in the Northwest. Actually I haven't had a single problem with them since the dreaded Global Crossing fiasco... -Chris ___ 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] Overriding Caller ID
There are 2 possibilities:
1) Your PRI provider does not have the overide settings correctly set on
your PRI.
2) you are not setting the callerid correctly in your dialplan.
You indicate that your provider indicates that they have it set up
correctly. You have a 50/50 chance that this is indeed correct. Many
providers will set up a PRI to allow override of CallerID, but only for
numbers that are specifically assigned to the PRI.
Ie: your main number is 212555, and you have DIDs 2125551233 and
2125551235. If you try to set the CallerID to 2125551234, it will default to
212555 since the CallerID you set is not assigned to the PRI. If this is
the case, you will need to ask your provider to enable unrestricted
override. Be aware that not all providers will allow unrestricted override,
or may require sufficient justification to allow it.
If the override is set correctly, then you need to look at the second
possibility.
You do not indicate which version of Asterisk that you are running.
If you are running an older version of Asterisk, you need to set the
CallerID like this:
exten = _NXXNXX,1,CallingPres(32)
exten = _NXXNXX,2,SetCallerID(Caller Name 2125551234)
exten = _NXXNXX,3,Dial(${TRUNK_LO}/${EXTEN})
If you are running a newer version of Asterisk, try this:
exten = _NXXNXX,1,CallingPres(32)
exten = _NXXNXX,2,SetCIDName(Caller Name)
exten = _NXXNXX,3,SetCIDNum(2125551234)
exten = _NXXNXX,4,Dial(${TRUNK_LO}/${EXTEN})
Regards,
Derek
- Original Message -
From: Waldo Rubinstein [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, August 19, 2005 2:51 PM
Subject: [Asterisk-Users] Overriding Caller ID
Hello list,
We have some kind of a problem with our Asterisk installation. We
want to be able to publish different caller id when placing outbound
calls through the PSTN. We have Asterisk with TE410P and T1 from FDN
Communications. The problem is that all our outbound calls show our
main number, regardless of what we set with SetCallerID, even using
CallingPres with all possible combinations. When speaking with FDN,
they say they have set their T1 to show our main number for outbound
calls, but that we should be able to override that with no problem.
As I said, I have tried all possible combinations, yet, nothing seems
to work. Below are snippets of some of our configs:
extensions.conf
;
; Local calls
;
exten = _NXXNXX,1,CallingPres(32)
exten = _NXXNXX,2,SetCallerID(2125551234)
exten = _NXXNXX,3,Dial(${TRUNK_LO}/${EXTEN})
zapata.conf
[channels]
usecallerid=yes
cidsignalling=bell
cidstart=ring
hidecallerid=no
restrictcid=no
usecallingpres=yes
callerid=asreceived
switchtype = dms100
signalling = em_w
group = 1
context=inbound
callerid=asreceived
channel = 1-24
Does anyone have any suggestions?
Thanks,
Waldo
___
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 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
