Re: [Asterisk-Users] Asterisk and firewall
El vie, 07-10-2005 a las 00:12, Arjan van Eersel escribió: I have installed an asterisk server at my office, the server is behind a firewall. On the firewall I’ve set NAT a rule for incoming traffic on port 5060 to be forwarded to the server. Connecting from home with my sip client doesn’t work at all. The asterisk server itself is ok, when I make a local connection at my office, 10.0.0.129 (client) to 10.0.0.6 (asterisk server) it works all perfect. Should I perhaps open more ports in the NAT settings? Kindest Regards, Arjan http://www.voip-info.org/tiki-index.php?page=Asterisk+firewall+rules and http://www.voip-info.org/tiki-index.php?page=Asterisk+SIP+externip Regards, -- Ing CIP Alejandro Celi Mariátegui [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Tux/Asterisk logo for Cisco phones
El jue, 22-09-2005 a las 19:04, David McNett escribió: I made http://slacker.com/~nugget/stuff/asterisk-cow-real.bmp for my non-Linux asterisk servers. I made my * + tux + office logo http://www.cipher.com.pe/central/asterisk-tux-cipher.bmp Regards, -- Ing CIP Alejandro Celi Mariátegui [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fedora Core 4 not recognizing X100P cards
I have the same problem. I've been having a bit of trouble getting the cards to work with asterisk, and I thought perhaps you might know what I might be doing wrong. I installed them in a linux box, and when I check to see if the OS has recognized them it looks fine: They show up as HSP56 MicroModem (rev 04) [EMAIL PROTECTED] lspci 00:00.0 Host bridge: Intel Corporation 82850 850 (Tehama) Chipset Host Bridge (MCH) (rev 04) 00:01.0 PCI bridge: Intel Corporation 82850 850 (Tehama) Chipset AGP Bridge (rev 04) 00:1e.0 PCI bridge: Intel Corporation 82801 PCI Bridge (rev 04) 00:1f.0 ISA bridge: Intel Corporation 82801BA ISA Bridge (LPC) (rev 04) 00:1f.1 IDE interface: Intel Corporation 82801BA IDE U100 (rev 04) 00:1f.2 USB Controller: Intel Corporation 82801BA/BAM USB (Hub #1) (rev 04) 00:1f.3 SMBus: Intel Corporation 82801BA/BAM SMBus (rev 04) 00:1f.5 Multimedia audio controller: Intel Corporation 82801BA/BAM AC'97 Audio (rev 04) 01:00.0 VGA compatible controller: nVidia Corporation NV18 [GeForce4 MX 4000 AGP 8x] (rev c1) 02:01.0 USB Controller: NEC Corporation USB (rev 41) 02:01.1 USB Controller: NEC Corporation USB (rev 41) 02:01.2 USB Controller: NEC Corporation USB 2.0 (rev 02) 02:09.0 Modem: PCTel Inc HSP56 MicroModem (rev 04) 02:0a.0 Modem: PCTel Inc: Unknown device 2181 (rev 04) 02:0b.0 Modem: PCTel Inc HSP56 MicroModem (rev 04) 02:0c.0 Ethernet controller: 3Com Corporation 3c905C-TX/TX-M [Tornado] (rev 78) 02:0d.0 Modem: PCTel Inc HSP56 MicroModem (rev 04) But when I try to start the wcfxo module it doesn't work: [EMAIL PROTECTED] modprobe wcfxo Notice: Configuration file is /etc/zaptel.conf line 146: Unable to open master device '/dev/zap/ctl' And the linux kernel doesn't quite recognize them: Sep 6 18:58:45 asterisk2 kernel: zaptel: no version for struct_module found: kernel tainted. Sep 6 18:58:45 asterisk2 kernel: Zapata Telephony Interface Registered on major 196 When I try to configure the ztcfg it doesn't find anything on channel 1: [EMAIL PROTECTED] /sbin/ztcfg -vvv Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) Channel 03: FXS Kewlstart (Default) (Slaves: 03) Channel 04: FXS Kewlstart (Default) (Slaves: 04) 4 channels configured. ZT_CHANCONFIG failed on channel 1: No such device or address (6) And Asterisk can't get them working: [chan_zap.so] = (Zapata Telephony) == Parsing '/etc/asterisk/zapata.conf': Found Sep 6 19:01:34 WARNING[2549]: chan_zap.c:778 zt_open: Unable to specify channel 1: No such device or address Sep 6 19:01:34 ERROR[2549]: chan_zap.c:6239 mkintf: Unable to open channel 1: No such device or address here = 0, tmp-channel = 1, channel = 1 Sep 6 19:01:34 ERROR[2549]: chan_zap.c:9191 setup_zap: Unable to register channel '1-4' Sep 6 19:01:34 WARNING[2549]: loader.c:345 ast_load_resource: chan_zap.so: load_module failed, returning -1 == Unregistered channel type 'Tor' == Unregistered channel type 'Zap' Sep 6 19:01:34 WARNING[2549]: loader.c:440 load_modules: Loading module chan_zap.so failed! I've tried most of the pci cards and they all give the same result. When using a different type of card in that same PC i don't get those errors. I thought perhaps there is some software upgrade necessary for them to work, or something along those lines? Any help you could mention would be very appreciated. Thanks -- Ing CIP Alejandro Celi Mariátegui [EMAIL PROTECTED] El mar, 13-09-2005 a las 15:01, Mojo with Horan Company, LLC escribió: hisax seems to be a loadable module for an ISDN card. if: # lsmod | grep hisax prints any output, try # rmmod hisax; modprobe zaptel ? hth Mojo Shawn Porter wrote: I am getting quite frustrated today, so please bear with me. I just installed Fedora Core 4 (was running RedHat 9 with a working Asterisk) now my Fedora does not appear to be recognizing my X100P (clone) at all. Hardware browser just shows them as unknown device. driver: hisax So, of course, my zaptel drivers do not work and therefore my asterisk does not work. any help would be greatly appreciated….. Shawn ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] [Fwd: [SA16438] Grandstream BudgeTone Denial of Service Vulnerability]
-- Ing CIP Alejandro Celi Mariátegui [EMAIL PROTECTED] -Mensaje reenviado- From: Secunia Security Advisories [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: [SA16438] Grandstream BudgeTone Denial of Service Vulnerability Date: Mon, 15 Aug 2005 12:49:44 +0200 -- Bist Du interessiert an einem neuen Job in IT-Sicherheit? Secunia hat zwei freie Stellen als Junior und Senior Spezialist in IT- Sicherheit: http://secunia.com/secunia_vacancies/ -- TITLE: Grandstream BudgeTone Denial of Service Vulnerability SECUNIA ADVISORY ID: SA16438 VERIFY ADVISORY: http://secunia.com/advisories/16438/ CRITICAL: Less critical IMPACT: DoS WHERE: From local network OPERATING SYSTEM: Grandstream BudgeTone 100 Series SIP Phones http://secunia.com/product/5537/ DESCRIPTION: Pierre Kroma has reported a vulnerability in Grandstream BudgeTone 100 Series SIP Phones, which can be exploited by malicious people to cause a DoS (Denial of Service). The vulnerability is caused due to an error when processing large UDP datagrams and can be exploited by sending a large UDP datagram (more than 65534 bytes) to port 5060/udp. Successful exploitation causes the phone to stop working by aborting active calls, blank the display, and make the integrated HTTP server become inaccessible. The vulnerability has been reported in firmware release 1.0.6.7. Other versions may also be affected. SOLUTION: Use the phones on trusted networks only. PROVIDED AND/OR DISCOVERED BY: Pierre Kroma, SySS. -- About: This Advisory was delivered by Secunia as a free service to help everybody keeping their systems up to date against the latest vulnerabilities. Subscribe: http://secunia.com/secunia_security_advisories/ Definitions: (Criticality, Where etc.) http://secunia.com/about_secunia_advisories/ Please Note: Secunia recommends that you verify all advisories you receive by clicking the link. Secunia NEVER sends attached files with advisories. Secunia does not advise people to install third party patches, only use those supplied by the vendor. -- Unsubscribe: Secunia Security Advisories http://secunia.com/sec_adv_unsubscribe/?email=alex%40linux.org.pe -- ___ 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] X100P FXO PCI Card + Incoming Fax
El lun, 04-07-2005 a las 09:53, Paul Goodyear escribió:
Is the X100P FXO PCI Card capable of detecting a fax, answering the
call, and then emailing the fax content to an email address?
For me work fine this card, the spanDSP and the
Follow these steps:
/etc/asterisk/zapata.conf
faxdetect=incoming
/etc/asterisk/extensions.conf
exten = s,1,Wait,1
Then:
http://www.soft-switch.org/installing-spandsp.html
http://scottstuff.net/blog/articles/2004/03/28/faxing-with-asterisk
And here:
http://lists.digium.com/pipermail/asterisk-users/2005-April/103817.html
here, I made these changes:
/usr/bin/metasend -b -F $SENDER -t $RECIPIENT \
-s Fax de $FAXSENDER \
-S 1 \
-m 'text/plain' -f ${TMPFILE} -n \
-m 'application/pdf;name=fax'${FAXID}'.pdf' -f ${TMPFILE_A} \
-D 'PDF Fax Document'
(because, if the *.pdf file is too large, the metasend begin to split
it.)
Hope that this help you.
Regards,
--
Ing CIP Alejandro Celi Mariátegui
[EMAIL PROTECTED]
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Re: [Asterisk-Users] Spanish doc
Leonardo: If you need a hand, only drop me an email. Regards, -- Ing CIP Alejandro Celi Mariátegui [EMAIL PROTECTED] El mié, 22-06-2005 a las 04:28, Leonardo F. Bauchwitz escribió: Hi: We have finished the translation of the FAQ of Digium to spanish. They are already (in Spanish) available for download (in http://ourproject.org/projects/asterix/): * FAQ Frequently Asked Questions * Features * Hardware compatibility list * Fast Installation Zaptel All the documentation is available for download Soon the following documents will be finished: Volume one and Asterisk Gateway Interface (AGI) Bye Leonardo Federico Bauchwitz Coordinator of Asterisk documentation in Spanish https://ourproject.org/projects/asterix/ [EMAIL PROTECTED] ___ 1GB gratis, Antivirus y Antispam Correo Yahoo!, el mejor correo web del mundo http://correo.yahoo.com.ar ___ 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] Cheap Asterisk FXO PCI cards
Hi, Does anybody know a website or company where I can buy cheap Asterisk and SIP compatible PCI cards that have 2, 3 or 4 FXO ports? Digium cards that have 2 or more FXO ports work great, but they are a bit over my budget at the moment. I have found digium compatible clone cards on the internet that are cheap, but haven't found any that have more than 1 port. Any help would be appreciated. Regards, -- Ing CIP Alejandro Celi Maritegui [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] faxdetect config issues
I'm trying to make the fax detect to work, without luck Regards, -- Ing CIP Alejandro Celi Maritegui [EMAIL PROTECTED] El jue, 16-06-2005 a las 13:35, Greg Blakely escribi: My Asterisk fax detection used to work, but no longer does. OK. So, here's the deal: 1. It appears that the faxdetect command cannot be applied channel-by-channel in zapata.conf anymore, as Asterisk appears to the last faxdetect= command to ALL channels. 2. My stations are detected and sent to the proper extension; i.e., when I send a fax from one zap extension to a zap voice extension, it is intercepted and sent to my fax machine (which is on a SIP ATA). HOWEVER, my ZAP trunks are NOT detected. A call from an outside FAX machine goes to voice mail, and I get a message full of CNG tone. My questions are: 1. How can I make faxdetect apply on a per-channel basis again? (It USED to work that way) 2. How can I make my outside lines have CNG tone detected on them? Here is my config: In ZAPATA.CONF: ; A typical trunk Faxdetect=incoming ; have tried also both and outgoing context = fromqwest group= 9 channel = 1 ; ; A typical station signalling = fxo_ks musiconhold=default usecallerid=yes hidecallerid=no restrictcid=no usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes group = 1 mailbox = 777 callerid = 777 context = internal channel = 7 ; In EXTENSIONS.CONF [fromqwest] exten = s,1,Answer exten = s,2,Wait(1) ;exten = s,3,Zapateller(answer|nocallerid) exten = s,3,NoOp ;exten = s,4,PrivacyManager exten = s,4,NoOp exten = s,5,Goto(internal,s,1) ; exten = fax,1,NoOp(Fax Detected) exten = fax,2,Dial(SIP/222-5000,20,tr) exten = fax,3,Congestion ; [internal] exten = s,1,Answer exten = s,2,Dial(ZAP/g1ZAP/10ZAP/11ZAP/17ZAP/38SIP/20,16,tr) exten = s,3,Goto(vm,s,1) exten = s,4,Hangup exten = s,103,Playtones(busy) exten = s,104,Wait(20) exten = s,105,Hangup ; ; exten = 10,1,Dial(ZAP/21r2,18,tr) ; exten = fax,1,Dial(SIP/222-5000,20,tr) exten = fax,2,Congestion exten = fax,102,Congestion ; ___ 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] pressing a key to get in of voicemail?
I've currently got Asterisk configured to take incoming calls, ask for extension, ring the phone and send them directly to the voicemail. What I want to be able to do is first a message press 1 for voicemail or hangup before voicemail come up. Any ideas? regards, -- Ing CIP Alejandro Celi Mariátegui [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] Asterisk can't create Zap channel - nightmare
Ahhh, OIC, I see my n00b error, thank you for the help.
Regards,
Alex
--
Ing CIP Alejandro Celi Mariátegui
[EMAIL PROTECTED]
El jue, 19-05-2005 a las 10:09, [EMAIL PROTECTED] escribió:
Hi,
You must dial Zap/g1/${EXTEN}, not Zap/1/${EXTEN}.
Otherwise you are just dialling on channel one, not on the group.
Steve
On Tue, 17 May 2005, Ing CIP Alejandro Celi [ISO-8859-1] Mariátegui wrote:
I have the same problem, I have 4 X100P (FXO) with 4 external lines
(hunting lines), and Asterisk runing on FC3.
Sometimes when 1 or 2 persons are using 2 lines, when I try to make a
call, I receive this message.
May 17 13:18:53 NOTICE[1857]: app_dial.c:749 dial_exec: Unable
to create channel of type 'Zap'
== Everyone is busy/congested at this time
I look on zap show channels and zap show channel X and no problem. The
idea is that you can take any of the 4 lines.
But when i can't make a outbound call, if we receive a call, the call
was received good by the Asterisk.
May 17 14:20:37 NOTICE[6969]: chan_zap.c:5367 ss_thread: Got
event 2 (Ring/Answered)...
-- Executing Wait(Zap/4-1, 1) in new stack
-- Executing Answer(Zap/4-1, ) in new stack
-- Executing DigitTimeout(Zap/4-1, 5) in new stack
-- Set Digit Timeout to 5
-- Executing ResponseTimeout(Zap/4-1, 10) in new stack
-- Set Response Timeout to 10
-- Executing BackGround(Zap/4-1, vm-inicio) in new stack
-- Playing 'vm-inicio' (language 'en')
and the call success!
/etc/zaptel.conf
loadzone = us
defaultzone=us
fxsks=1-4
/etc/asterisk/zapata.conf
#
[channels]
busydetect=yes
busycount=5
relaxdtmf=yes
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
usecallerid=yes
hidecallerid=no
echotraining=800
echocancel=yes
echocancelwhenbridged=yes
rxgain=8.0
txgain=-5.5
immediate=no
context=local-trunks
signalling=fxs_ks
callerid=asreceived
musiconhold=yes
group=1
callgroup=1
pickupgroup=1
channel = 1
signalling=fxs_ks
context=local-trunks
callerid=asreceived
musiconhold=yes
channel = 2
signalling=fxs_ks
context=local-trunks
callerid=asreceived
musiconhold=yes
channel = 3
signalling=fxs_ks
context=local-trunks
callerid=asreceived
musiconhold=yes
channel = 4
/etc/asterisk/extensions.conf
#
[local-trunks]
exten = _9XXX,1,Dial,Zap/1/${EXTEN:1} ;
exten = _8X,1,Dial,Zap/1/${EXTEN:1} ; 0800
Hope that you can help us.
Saludos desde PERU
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Re: [Asterisk-Users] FREE music for downloading
Paul, your web is down: Error 400 - Bad Request Regards, -- Ing CIP Alejandro Celi Mariátegui [EMAIL PROTECTED] El mié, 18-05-2005 a las 17:54, Paul Mahler escribió: Need new Music on Hold for your PBX? Signate is happy to make a variety of classical music selections available, sampled at rates that are appropriate for telephony. There is no charge. The selections feature Elena Kuschnerova, pianist, and Lev Guelbard, violinist, playing public domain pieces that will give callers a classic impression of you or your company . Click on the link to see a list of the available music and download what you want from our ftp site. http://www.signate.com/moh.php Thanks to Greg Camp, who graciously provided us with the original files. We plan to add other types of music over time. Legal Stuff Follows SIGNATE MAKES NO WARRANTIES, EXPRESS OR IMPLIED, REGARDING THE FREE MUSIC ON HOLD FILES, INCLUDING, WITHOUT LIMITATION, ANY IMPLIED WARRANTIES OF MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE. SIGNATE SHALL NOT BE LIABLE TO YOU OR ANY OTHER PERSON OR ENTITY FOR ANY GENERAL, PUNITIVE, SPECIAL, DIRECT, INDIRECT, CONSEQUENTIAL OR INCIDENTAL DAMAGES, OR LOST PROFITS OR ANY OTHER DAMAGES, COSTS OR LOSSES ARISING OUT OF YOUR USE OF THE FREE MUSIC ON HOLD FILES, EVEN IF SIGNATE HAS BEEN ADVISED OF THE POSSIBILITY OF SUCH DAMAGES, COSTS OR LOSSES. Paul Mahler www.signate.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] mpg123 zombie processes ...
I had this problem with the asterisk manual start, with /etc/init.d/asterisk no problem -- Ing CIP Alejandro Celi Mariátegui [EMAIL PROTECTED] El jue, 05-05-2005 a las 08:18, C F escribió: yeah, have cron running every night that issues this command: killall -9 mpg123 On 5/5/05, Vamsi Pottangi [EMAIL PROTECTED] wrote: Hi All, I had noticed that MOH's mpg123 processes are not killed when asterisk is killed. Eventually after many restarts I see many of these zombie processes eating up CPU. Any Idea how could I make asterisk to clean up these properly. Thanks, ~Vamsi ___ 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] Asterisk can't create Zap channel - nightmare
I have the same problem, I have 4 X100P (FXO) with 4 external lines
(hunting lines), and Asterisk runing on FC3.
Sometimes when 1 or 2 persons are using 2 lines, when I try to make a
call, I receive this message.
May 17 13:18:53 NOTICE[1857]: app_dial.c:749 dial_exec: Unable
to create channel of type 'Zap'
== Everyone is busy/congested at this time
I look on zap show channels and zap show channel X and no problem. The
idea is that you can take any of the 4 lines.
But when i can't make a outbound call, if we receive a call, the call
was received good by the Asterisk.
May 17 14:20:37 NOTICE[6969]: chan_zap.c:5367 ss_thread: Got
event 2 (Ring/Answered)...
-- Executing Wait(Zap/4-1, 1) in new stack
-- Executing Answer(Zap/4-1, ) in new stack
-- Executing DigitTimeout(Zap/4-1, 5) in new stack
-- Set Digit Timeout to 5
-- Executing ResponseTimeout(Zap/4-1, 10) in new stack
-- Set Response Timeout to 10
-- Executing BackGround(Zap/4-1, vm-inicio) in new stack
-- Playing 'vm-inicio' (language 'en')
and the call success!
/etc/zaptel.conf
loadzone = us
defaultzone=us
fxsks=1-4
/etc/asterisk/zapata.conf
#
[channels]
busydetect=yes
busycount=5
relaxdtmf=yes
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
usecallerid=yes
hidecallerid=no
echotraining=800
echocancel=yes
echocancelwhenbridged=yes
rxgain=8.0
txgain=-5.5
immediate=no
context=local-trunks
signalling=fxs_ks
callerid=asreceived
musiconhold=yes
group=1
callgroup=1
pickupgroup=1
channel = 1
signalling=fxs_ks
context=local-trunks
callerid=asreceived
musiconhold=yes
channel = 2
signalling=fxs_ks
context=local-trunks
callerid=asreceived
musiconhold=yes
channel = 3
signalling=fxs_ks
context=local-trunks
callerid=asreceived
musiconhold=yes
channel = 4
/etc/asterisk/extensions.conf
#
[local-trunks]
exten = _9XXX,1,Dial,Zap/1/${EXTEN:1} ;
exten = _8X,1,Dial,Zap/1/${EXTEN:1} ; 0800
Hope that you can help us.
Saludos desde PERU
--
Ing CIP Alejandro Celi Mariátegui
[EMAIL PROTECTED]
El jue, 05-05-2005 a las 15:19, Matthew Boehm escribió:
Before you jump ahead, yes I do have chan_zap.so loaded..
Call Flow: Asterisk 1 --IAX2-- Asterisk 2 --- PRI
-- Accepting AUTHENTICATED call from 22.22.22.22:
requested format = ulaw,
requested prefs = (),
actual format = ulaw,
host prefs = (ulaw|alaw|gsm),
priority = mine
-- Executing Dial(IAX2/[EMAIL PROTECTED],
Zap/R1d/18005551212|60) in new stack
May 5 15:21:37 NOTICE[16153]: app_dial.c:968 dial_exec_full: Unable to
create channel of type 'Zap' (cause 0)
== Everyone is busy/congested at this time (1:0/0/1)
-- Hungup 'IAX2/[EMAIL PROTECTED]'
pri debug is irrelevent because the call never makes it to pri. What is
cause 0? Its not listed in the header files.
Nothing is busy on that span.
Any ideas?
-Matthew
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Re: [Asterisk-Users] How to display info from Asterisk on/to the phone ?
El mar, 03-05-2005 a las 03:43, Deborah MALKA escribió: Hello, I wanted to know if there is a way to dissplay infos from Asterisk on a SIP phone ? Because I know Asterisk is very powerfull, so I'm nearly sure that there is a way to do it. Using XML on Directory.xml and services.xml with a Cisco 7960/7940 phone. I combine it with PHP Regqrds, -- Ing CIP Alejandro Celi Mariátegui [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] Fedora Core 3 Shorewall Install
El lun, 02-05-2005 a las 16:14, Anonymous Account escribió: My questions concern the installation of the latest/greatest Asterisk on Fedora Core 3 with a Shorewall (Shoreline) Firewall installed. For me * works fine, with FC3 and iptables (Shorewall is an IPtables too) I haven't been able to find a step-by-step howto that is CURRENT that addresses this particular configuration. The config is similar for all distros http://www.voip-info.org/wiki-Asterisk+Linux+Fedora Does anyone have a link they could point me to? Please keep in mind the word current and by that I mean something that takes into account that I am using a Kernel that is 2.6+ and that Shorewall is version 2.2+ You only need to open the ports, but is better to use iptables directly ;) http://www.voip-info.org/wiki-Asterisk+firewall+rules Mucho Gracias, amigos! De nada, Saludos desde Peru, -- Ing CIP Alejandro Celi Mariátegui [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] IP Softphone Recommendations
El mar, 26-04-2005 a las 09:42, Guillermo Salas M escribió: I´m using X-lite on windows and linux, looks pretty well. Do you have the link of the X-Lite Linux version? Not found in the xlite website. Regards from PERU... -- Ing CIP Alejandro Celi Mariátegui [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] Cisco 7970
Lol, my grandstreams (about 6) work fine with *, but I have about 1 week trying to configure a 7960G without luck. -- Ing CIP Alejandro Celi Mariátegui [EMAIL PROTECTED] El mar, 19-04-2005 a las 16:08, C F escribió: I don't think so, its one of the cheaper phones out there, and doens't realy work, it's a scam from some small company called Cisco. I think you should try Grandstream. On 4/19/05, Paul A Brown [EMAIL PROTECTED] wrote: Has anyone got the Cisco 7970 working with asterisk? ___ 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
[Fwd: [Asterisk-Users] newbie DNS problem with BT100
No idea for this problem? Alex -Mensaje reenviado- From: Ing CIP Alejandro Celi Mariátegui [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [Asterisk-Users] [Fwd: newbie DNS problem with BT100] Date: Tue, 22 Mar 2005 19:42:30 -0500 (Sorry, but my english is very bad) Hi I'm newbie with Asterisk, but i was able to install and configure Asterisk with 3 budgetone 102 and 2 Handytone 206 and works fine for me. I have a problem and i don't see answer in forums: DNS resolution: First Day: == In configuration menu of the BT100 I use: DHCP SIP server: central.mydomain.com or 192.168.100.180 Use DNS SRV: Yes NTP Server: time.nist.gov Symptoms: DHCP works fine for me. All config like IP, Mask, DNS, router etc. LED: 1900-01-01 SIP server: With 192.168.100.180 work fine with asterisk but with central.mydomain.com not work. Looking in configuration, I understood that the problem (perhaps) was the DNS server. I saw my old firewall that had a RH7.2 with a RPM bind9, DHCP etc. and think that is time to format this machine. All my stations worked fine with this DNS, but not the phones: I thought that could be a good option. Second Day: === Format firewall, Fedora Core 3, bind etc. In configuration menu of the BT100 I use: DHCP SIP server: central.mydomain.com or 192.168.100.180 Use DNS SRV: Yes NTP Server: time.nist.gov Symptoms: DHCP works fine for me. All config like IP, Mask, DNS, router etc. LED: 2005-03-22 (excelent! nslookup with time.nist.gov) SIP server: With 192.168.100.180 work fine with asterisk but with central.mydomain.com not work. Then, only lookup with the time server works fine, but not with my Authoritative domain, only with the IP. Perhaps BT100 need an special configuration for bind servers or perhaps is a configuration problem, can you help me? Best regards, Alex -- Ing CIP Alejandro Celi Mariátegui [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] [Fwd: newbie DNS problem with BT100]
(Sorry, but my english is very bad) Hi I'm newbie with Asterisk, but i was able to install and configure Asterisk with 3 budgetone 102 and 2 Handytone 206 and works fine for me. I have a problem and i don't see answer in forums: DNS resolution: First Day: == In configuration menu of the BT100 I use: DHCP SIP server: central.mydomain.com or 192.168.100.180 Use DNS SRV: Yes NTP Server: time.nist.gov Symptoms: DHCP works fine for me. All config like IP, Mask, DNS, router etc. LED: 1900-01-01 SIP server: With 192.168.100.180 work fine with asterisk but with central.mydomain.com not work. Looking in configuration, I understood that the problem (perhaps) was the DNS server. I saw my old firewall that had a RH7.2 with a RPM bind9, DHCP etc. and think that is time to format this machine. All my stations worked fine with this DNS, but not the phones: I thought that could be a good option. Second Day: === Format firewall, Fedora Core 3, bind etc. In configuration menu of the BT100 I use: DHCP SIP server: central.mydomain.com or 192.168.100.180 Use DNS SRV: Yes NTP Server: time.nist.gov Symptoms: DHCP works fine for me. All config like IP, Mask, DNS, router etc. LED: 2005-03-22 (excelent! nslookup with time.nist.gov) SIP server: With 192.168.100.180 work fine with asterisk but with central.mydomain.com not work. Then, only lookup with the time server works fine, but not with my Authoritative domain, only with the IP. Perhaps BT100 need an special configuration for bind servers or perhaps is a configuration problem, can you help me? Best regards, Alex -- Ing CIP Alejandro Celi Mariátegui [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
