[Asterisk-Users] SIP phones, Asterisk and bandwidth
I have not studied SIP yet. I have an Asterisk server, and SIP phones somewhere on the globe, not locally connected to the Asterisk server. What is the bandwidth I need to the Asterisk server? I assume that only the registration is the bandwidth to the Asterisk server, while the phone call will be made from phone to phone directly. How much bandwidth do I need to server as a PBX for 100 phones, 1000 phones, How many phones are usually on at the same time from 100% registered phones? I understand that if I operate a gateway, than all the bandwith of the voice stream will also go through this gateway, ... bye Ronald ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] supposable timing problem with TE100P
Hi list, every now and then I get the following message in my * logs: chan_zap.c:7379 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 As this is only a notice and voice worked quite well, despite the messages, I didn't bother. _But_, I wanted to try spandsp/fax (latest version) lately and whenever I send a fax to * the pages get chopped, which according to Steve Underwood points at a timing problem, either in * or Hardware. I don't know if the messages about FCS errors have smth. to do with the fax problems, but hope that someone out there has a clue ;-)) BTW, I see a lot of the following messages too: !! Unknown IE 49 (cs5, Unknown Information Element) !! Unknown IE 50 (cs5, Unknown Information Element) If any further information is needed to narrow the problem down please let me know. Thanks a lot in advance, best regards, Kurt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] h323 dundi problems with 11/04/04 CVS
Hi list, I face 2 problems with a today (11/04/04) CVS version: H323: Nufone h323: calling Dial(H323/extension) give in logs Host: extension Username: placing outgoing call to :1721 - this is my GK port h323_make_call failed(H323/extension) [...] dialstatus=chanunavail Calling from a H323 EP to * I have connection but no audio in both sides. Dundi: - if I load dundi module I receive an failed load module - undefined symbol: uncompress I had to remove the -lz at the end of makefile pbx_dundi.so because I had an error at compilation: gcc -shared -Xlinker -x -o pbx_dundi.so pbx_dundi.o dundi_parser.o -lz /usr/bin/ld: cannot find -lz If anyone has an idea. I have those errors since the upgrade from 11/02/04 Before, an 09/25/04 CVS everything went fine. -- Daniel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Video conferencing Meet Me Bounty bumped
Bounty bumped to $US1,000 11/04/04 I've been able to secure some pledges from 2 people outside of the asterisk community that should this feature be made available that they will contribute $300 and $200 respectively. This bounty now stands at $US1,000. So far I haven't received any interest from any developers to work on this, please contact me dean at collins.net.pr should you be interested. Cheers, Dean ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CVS-HEAD-11/03/04-14:09:34 ALERT_INFO Doesn't Get Passed
I saw a previous post about this but I can't find it, CVS-HEAD-11/03/04-14:09:34 does not pass ALERT_INFO to the phones. It used to work but has now stopped. I'm not a coder so I can't look through the code but someone mentioned ALERT_INFO does not exist in app_dial if I remember correctly. Anyone know anything about this? -- MBM ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Perl AGIs TCP Sockets
Hello everybody Do you remember I sent a case to the list about a digit 1 phantom I received when I call the method get_data or stream_file? Fine. I realized that It does not happend when I omit a subrutine I my code where I open a TCP client socket by IO::Socket. I think It is because Asterisk send data to AGI by STDIN, like my socket; and somehow the 1 I get come from the TCP Socket. Am I right? If so... why does It happend after I properly close my socket? Please give me a hand! Thanks in advace. Víctor ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Eicon Diva Server 4BRI
Hi Damon, I have the Eicon Diva Server BRI card working fine on my box. What is important (if you want to use kernel 2.6) is that you need to use at least kernel 2.6.9 because a lot of CAPI/Eicon fixes where added. Divactrl is needed to load the firmware on the card: http://isdn4linux.org/~armin/divas/divactrl_2.1.tar.gz The firmware for the card can be found here: ftp://ftp.isdn4linux.org/pub/isdn4linux/utils/eicon/firmware/ Regards, Patrick Is there a significant advantage (or requirement) to using kernel 2.6 with Eicon / chan_capi / * ? All of my experimentation and use up to now has been redhat 9. What 2.6 distro are you using? fedora core 2? Have there been things that DO NOT work with kernel 2.6 in your experience? ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] supposable timing problem with TE100P
On Thu, 4 Nov 2004, Kurt Bauer wrote: Hi list, every now and then I get the following message in my * logs: chan_zap.c:7379 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 As this is only a notice and voice worked quite well, despite the messages, I didn't bother. _But_, I wanted to try spandsp/fax (latest version) lately and whenever I send a fax to * the pages get chopped, which according to Steve Underwood points at a timing problem, either in * or Hardware. Is your timing source set correctly? If you are connecting to the pstn the pstn connection should be the primary timing source. What does the missed interrupts counter in cat cat /proc/zaptel/1 say? Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 Missing MeetMe App?
On Thu, 2004-11-04 at 08:56 -0500, david winter wrote: All, I was going to mess around with meetme on my cvs install of asterisk, but the app_meetme does not seem to be there. I did a normal make clean;make install. ran asterisk, did a 'show applications' and its not there. i have extensions.conf and meetme.conf setup correctly, when i try to do MeetMe(1000|p) it says there is no MeetMe application registered? Is MeetMe no 'registered' already? Please learn to start a NEW thread. Don't hit reply and delete the previous content. Your mail reader actually tattles on you by telling us you responded tp Paul Rodan's message with subject RE: [Asterisk-Users] Unable to write frame to channel: Success - MeetMeproblem. You also didn't mention what devices you have connected to your asterisk machine. As app_meetme.c was in the CVS tree 2 days ago and I haven't seen anything cross the -cvs mailing list with regards to it being removed, I am left with the impression that you don't have any Zap devices nor the code checked out and compiled. App Meetme requires zaptel drivers as it uses specialized code from the zaptel drivers to actually mix the channels together and also as a timer to make sure samples are going out at regular intervals to keep all portions of the conference chop free. Please refer to the wiki or the mailing list via google for such things as the zap rtc interface or the zap dummy interface to provide you with the proper timers for meetme. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Perl AGIs TCP Sockets
On Thu, 2004-11-04 at 11:15 -0400, Victor Cartes wrote: Hello everybody Do you remember I sent a case to the list about a digit 1 phantom I received when I call the method get_data or stream_file? Fine. I realized that It does not happend when I omit a subrutine I my code where I open a TCP client socket by IO::Socket. I think It is because Asterisk send data to AGI by STDIN, like my socket; and somehow the 1 I get come from the TCP Socket. Am I right? If so... why does It happend after I properly close my socket? Well you seem to have narrowed your problem down to something between you and perl, not asterisk. Without the code, it would be difficult to determine the problem. It may well be a problem with selected input, if you selected the socket and then closed it, you would need to select stdin again. Without code samples, it is just a guessing game. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 79XX - Using built-in 3way conference
We recently switched to 729. I wouldn't expect that to cause built-in conferencing to stop working. Matthew - Original Message - From: Kevin P. Fleming [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Wednesday, November 03, 2004 5:28 PM Subject: Re: [Asterisk-Users] Cisco 79XX - Using built-in 3way conference Matthew Boehm wrote: This works fine if caller 1 and 2 are both other phones in the office or caller 1 is a phone in office and 2 is PSTN. Just doesn't work when both PSTN. Worked before.. What codecs are involved? If you are using low-bandwidth codecs for PSTN connections, the Cisco phones will refuse to conference two low-bandwidth calls together. They may work with a ulaw and a low-bandwidth call, but I haven't tested that. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Multiline (4 or 8) sip phone
All, What is a good multiline sip phone for an operator? Model and and manufacturer. I presume the multiline phone looks like 4 or 8 independent SIP phones and asterisk would handle that by a call queue. Then the operator just does her normal routine answering calls etc... Thanks for the suggestion. Jerry ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] X100P Analog PBX - not RING and not answer
Hi, I connect a X100p in a Analog PBX extension. If I want to call a analog extension (e.g.: using a softphone), the asterisk pick up the extension and dial perfectly. If I call the extension where where the X100Pp is connected (inside the company), the asterisk doesn't answer the call. I do: modprobe wcfxo debug=1 opermode=1 This message repeats lot of times: Setting hook state to 0 (0a) But in syslog, I don't see the RING! and NO RING! messages. This is the outputs of the ztmonitor and audacity: http://www.ad2.com.br/asterisk/teste.wav.bz2 http://www.ad2.com.br/asterisk/teste.png.bz2 And the state of Zap channels is always offhook: *CLI zap show channel 1 Channel: 1 (...) InAlarm: 0 Signalling Type: FXS Kewlstart (...) Actual Hookstate: Offhook If I disconnect the cable of the board, the hookstate changes to onhook. I made a lot of tests with tx and rx gain w/o success. Thanks for any help. Alexandre Arruda Paes ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] supposable timing problem with TE100P
--On Thursday, November 04, 2004 03:19:56 PM +0100 Peter Svensson [EMAIL PROTECTED] wrote: On Thu, 4 Nov 2004, Kurt Bauer wrote: Hi list, every now and then I get the following message in my * logs: chan_zap.c:7379 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 As this is only a notice and voice worked quite well, despite the messages, I didn't bother. _But_, I wanted to try spandsp/fax (latest version) lately and whenever I send a fax to * the pages get chopped, which according to Steve Underwood points at a timing problem, either in * or Hardware. Is your timing source set correctly? If you are connecting to the pstn the pstn connection should be the primary timing source. connection is to a Ericsson MD110 wich is set as network, * is set as CPE. What does the missed interrupts counter in cat cat /proc/zaptel/1 say? cat /proc/zaptel/1 Span 1: WCT1/0 Digium Wildcard E100P E1/PRA Card 0 HDB3/CCS 1 WCT1/0/1 Clear (In use) 2 WCT1/0/2 Clear (In use) 3 WCT1/0/3 Clear (In use) 4 WCT1/0/4 Clear (In use) 5 WCT1/0/5 Clear (In use) 6 WCT1/0/6 Clear (In use) 7 WCT1/0/7 Clear (In use) 8 WCT1/0/8 Clear (In use) 9 WCT1/0/9 Clear (In use) 10 WCT1/0/10 Clear (In use) 11 WCT1/0/11 Clear (In use) 12 WCT1/0/12 Clear (In use) 13 WCT1/0/13 Clear (In use) 14 WCT1/0/14 Clear (In use) 15 WCT1/0/15 Clear (In use) 16 WCT1/0/16 HDLCFCS (In use) 17 WCT1/0/17 Clear (In use) 18 WCT1/0/18 Clear (In use) 19 WCT1/0/19 Clear (In use) 20 WCT1/0/20 Clear (In use) 21 WCT1/0/21 Clear (In use) 22 WCT1/0/22 Clear (In use) 23 WCT1/0/23 Clear (In use) 24 WCT1/0/24 Clear (In use) 25 WCT1/0/25 Clear (In use) 26 WCT1/0/26 Clear (In use) 27 WCT1/0/27 Clear (In use) 28 WCT1/0/28 Clear (In use) 29 WCT1/0/29 Clear (In use) 30 WCT1/0/30 Clear (In use) 31 WCT1/0/31 Clear (In use) br, Kurt ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 IP Conference Station
Hi, Somebody have any idea how I can config a CISCO IP CONFERENCE STATION Model 7935 that work with * . Thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.0.2, Zaptel 1.0.2, Linux 2.6.9 on a PCEngines WRAP\Soekris net4801 in Compact Flash
Brian Wilkins wrote: Sounds cool, but I heard a rumor that CF Cards don't like too many rewrites or they start losing data. Brian, It boots read-only and uses ramdisks for RW stuff. -- Kristian Kielhofner ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] real-time-clock asterisk/meetme/ztdummy in 2.6.9 UML
Hi *, [this goes to [EMAIL PROTECTED] and [EMAIL PROTECTED] i try to setup an VoIP conferencing server within a UML using asterisk and it's 'MeetMe conference bridge'. I have several UMLs running other services, but my asterisk know-how is poor (as you will see ;-). Question #1: did anybody successfully implement an asterisk MeetMe-conference inside a 2.6-UML? It needs a special zaptel timer. As there is no special hardware, i have to use a ztdummy/2.6 or zaprtc: http://www.voip-info.org/tiki-index.php?page=Asterisk%20timer Question #2: is there any chance that either ztdummy or zaprtc will work inside 2.6-UML? I tried ztdummy. After frickeling the Makefile, it compiled against the UML source tree, it loads within the UML on top of zaptel.ko [EMAIL PROTECTED] asterisk]# lsmod Module Size Used by ztdummy 2628 0 zaptel227940 1 ztdummy (no other modules loaded). Asterisk works (e.g. i can hear the conf- onlyperson message when i enter the conference room). But the audio of the conference itself is extremly snatchy. That _may_ be a config problem, but a very similar configuration on a non-UML works, and un- loading the ztdummy.ko on that system results in the same snatchy audio, so i suspect the ztdummy.ko or some device configuration is broken in my UML. At the same time, i observe this http://seclists.org/lists/linux-kernel/2004/Oct/8108.html problem within my UML, too: # tail -f /var/log/messages tail: cannot read realtime clock: Unknown error 516 and tail exits with code 1. Question #3: is there a problem with the RTC in 2.6.9? My host is 2.6.8.1 + skas3-2.6.7-v5 without module support: CONFIG_SMP=y (host is a P4 with HT) CONFIG_SCHED_SMT=y CONFIG_HPET_TIMER=y CONFIG_HPET_EMULATE_RTC=y CONFIG_RTC=y CONFIG_PREEMPT=y My guest is 2.6.9, with module support. Besides the zaptel modules no feature is compiles as a module but static. CONFIG_UML_REAL_TIME_CLOCK=y Question #4: Shall i try different SMP/HT/RTC settings on host or guest? Regards, /nils. host and guest are FC2. I use the asterisk-1.0 from ATrpms http://atrpms.net/dist/fc2/asterisk/ and seft-built zaptel-1.0.2 tarball user-mode-linux: http://user-mode-linux.sourceforge.net/ http://www.user-mode-linux.org/~blaisorblade/ asterisk: http://www.voip-info.org/tiki-index.php?page=Asterisk http://asterisk.org/ meetme: http://www.voip-info.org/wiki-Asterisk+cmd+MeetMe -- there is no sig. -- there is no sig. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Perl AGIs TCP Sockets
[EMAIL PROTECTED] wrote: Without code samples, it is just a guessing game. Maybe there's a bug where the TCP stuff is writing to fd(0), which IIRC is STDIN Guessing is fun! -- Andreas SikkemaRits tele.com Scheepmakersstraat 11 3011 VH Rotterdam t: +31 (0)10 2245544f: +31 (0)10 2245540 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 1.0.2-CVS RPM update
I've uploaded new packages to fix a couple of problems in the previous releases. Again, these are for FC1 and are located in the standard place: ftp://ftp.linuxsys.com/pub/releases/FC1/asterisk-v1.0/ Regards, -- Andrew McRory - President/CTO Linux Systems Engineers, Inc. - http://www.linuxsys.com Located in beautiful Tallahassee, Florida Office 850-224-5737 Office 850-575-7213 Mobile 850-294-7567 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 as sip proxy registrar
Hello, --- Anand S. Katti [EMAIL PROTECTED] wrote: register = [EMAIL PROTECTED]/1000 register = [EMAIL PROTECTED]/1001 register = [EMAIL PROTECTED]/1002 Remove these register lines. You dont need them as your UACs are trying to register with Asterisk. [sourabha] username=1001 The context name and username should be the same. Try making these changes and it should be up. Regards, Girish __ Do you Yahoo!? Check out the new Yahoo! Front Page. www.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hardware Support
Quick Question that I hope someone can answer. Will Asterisk work with basic PCI FaxModems instead of those expensive cards listed on the hardware page? -- Mike Shultz [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Dynamic DNS causes problems
On Thu, 2004-11-04 at 02:58, Larry Hendrickson wrote:
Hi all,
This is my first post to this list, so I apologize if it is a newbie
question. I did quite a bit of reading and a number of Google searches
for answers and found people with dynamic DNS problems, but not the
same one.
I just recently set up Asterisk as a pbx system for my home using
Broadvoice. I would first like to say thank-you for an incredibly
effective program (it took me a while to get used to it, but now I am
quite impressed).
The only remaining problem comes from the fact that I have an ADSL
connection at my home and the pppoe changes my IP address every once in
a while. I have set my sip.conf 'host=' command up with a dyndns
hostname and everything works when I start *. The moment the IP address
changes, though, incoming calls continue to work but outgoing calls give
me a maximum retries exceeded error, and my Dial command exits after
about 3 seconds with NOANSWER. I have confirmed that this is caused
by the change in IP address by restarting *, successfully making a call,
rebooting my DSL router to have it get a new IP address, waiting for
dyndns to register the change (and confirming that this is correct), and
then immediately making another call that has this problem. When the IP
address doesn't change, I have confirmed that outgoing calls work after
several days. A 'service asterisk restart' always gets the outgoing
calls working again, but it seems odd that I would have to restart * so
regularly.
I am assuming that either * or the remote computer is either doing the
DNS lookup once and caching the IP address or there is a socket that is
opened and kept open through the IP address change and not reconnected
afterward (shouldn't the socket on both ends figure out that the
connection is no longer good and reconnect?).
As an ugly hack, I am tempted to have a cron job check for changes in
the ip address and restart *. The problem with this (or one of them) is
that I have to somehow make sure that it isn't in the middle of a phone
call when it does this. Is there a more elegant way of doing this?
I'm not sure about more elegant, but...
Have your cron job issue an asterisk -rx 'restart when convenient'
command instead of a hard restart. That will wait until there are no
active channels to restart. Also, issuing a 'sip reload' instead of
restarting * is probably sufficient to re-register with Broadvoice.
-Seth
Am I doing something wrong?
(And although getting a static IP would be the most elegant solution,
that really isn't an option now.)
Thank-you in advance,
Larry
-- sip.conf
[general]
externip=MY_HOSTNAME.dyndns.org
bindaddr = 0.0.0.0
port=5060
localnet=192.168.0.0/255.255.255.0
disallow=all
allow=gsm
allow=slinear
allow=ulaw
allow=alaw
context=incoming
dtmfmode=inband
register = MY_NAME:MY_PASSWD@sip.broadvoice.com
tos=0x18
srvlookup=yes
nat=no
[Broadvoice]
type=peer
username=MY_NAME
fromuser=MY_NAME
secret=MY_PASSWD
host=147.135.8.129
context=sip
fromdomain=sip.broadvoice.com
canreinvite=no
dtmfmode=inband
nat=no
[broadvoice-incoming]
type=peer
dtmfmode=inband
host=147.135.8.128
context=incoming
qualify=yes
canreinvite=no
disallow=all
allow=gsm
allow=slinear
allow=ulaw
allow=alaw
insecure=yes
nat=no
[broadvoice-incoming2]
type=peer
dtmfmode=inband
host=147.135.0.128
context=incoming
qualify=yes
canreinvite=no
disallow=all
allow=gsm
allow=slinear
allow=ulaw
allow=alaw
insecure=yes
nat=no
--- the extension I am calling on
[trunkld]
;
; US long distance context accessed through trunk
;
;Pattern match US long distance calls
exten = _NX,1,Dial(SIP/[EMAIL PROTECTED],20)
exten = _NX,2,Goto(error-${DIALSTATUS},1)
exten = _NX,3,Congestion
exten = _NX,102,Busy
--
Seth Remington
SaberLogic, LLC
661-B Weber Drive
Wadsworth, Ohio 44281
Phone: (330)335-6442
Fax: (330)336-8559
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RE: [Asterisk-Users] Hardware Support
Quick Question that I hope someone can answer. Will Asterisk work with basic PCI FaxModems instead of those expensive cards listed on the hardware page?--Mike Shultz[EMAIL PROTECTED] Answer is here http://www.voip-info.org/wiki-Asterisk+Hardware Answers to the next several questions you will have are here http://www.voip-info.org/tiki-index.php?page=Asterisk This is just one of the several sites dedicated to answering basic questions There areinexpensive intel chipset modems that will act like the Digium single port FXO card - read the hardware section. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Hardware Support
Hi Mike, Yes it may work with some modems depending on which chipset they use (in fact there are some advertised as supporting asterisk) however a number of us choose to buy from Digium in order to show our support for how much effort they put in by developing Asterisk. Cheers, Dean From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Shultz Sent: Thursday, November 04, 2004 10:11 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Hardware Support Quick Question that I hope someone can answer. Will Asterisk work with basic PCI FaxModems instead of those expensive cards listed on the hardware page? -- Mike Shultz [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Multi-line analog phones with Asterisk?
I am interested in implementing Asterisk and someday hope to have it replace my 8 x 24 Nortel switch. However, I was told by a Telcom friend that my multi line phones (Nortel 7208s) may not work with Asterisk. This is a huge concern because in my business we are constantly jumping back from one line to another (putting people on hold and grabbing another line and going back and forth etc.) Is it possible with Asterisk to (or are there analog phones that allow ) access multiple lines with the press of a button, so that if someone says Richard line 4 is for you, I can easily put my caller on hold and grab line 4? Any help would be very much appreciated. Richard __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Eicon Diva Server 4BRI
The reason I went with kernel 2.6.9 is that it contains all the needed CAPI support while a 2.4 kernel needs to be patched (check melware.de). I am using Fedora Core 2 with the latest kernel from rawhide (2.6.9-1.640 iirc). So no requirements other than not wanting to patch the kernel. An advantage imho is that my Asterisk box doesn't have a Digium card for timing and on 2.6 ztdummy will work without some special USB chipset. Finally afaik 2.6 performs better than 2.4 and I have found nothing that doesn't work. If you want to go for Fedora Core maybe it is an idea to wait 4 or 5 more days because then Fedora Core 3 will be released. Regards, Patrick Thank you for the info, I prefer tested over new, so I might try Fedora Core 2 on the ISDN test box. I will wait for others to try Fedora Core 3 :) Damon ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Hardware Support
for voice they may or may not work, but the audio quality is bound to be bad... unless you're really lucky :) On Nov 4, 2004, at 16:10, Mike Shultz wrote: Quick Question that I hope someone can answer. Will Asterisk work with basic PCI FaxModems instead of those expensive cards listed on the hardware page? -- Mike Shultz [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] Hardware Support
On Thu, 2004-11-04 at 10:10 -0500, Mike Shultz wrote: Quick Question that I hope someone can answer. Will Asterisk work with basic PCI FaxModems instead of those expensive cards listed on the hardware page? Did you bother checking the wiki or using google to check past messages? Or did you just feel like bothering more than 8k people to do your homework for you? It is in the archive with all the included screaming and yelling about how you should support Digium who gives you the software. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Hardware Support
On Thu, 2004-11-04 at 10:10, Mike Shultz wrote: Quick Question that I hope someone can answer. Will Asterisk work with basic PCI FaxModems instead of those expensive cards listed on the hardware page? No. The wcfxo driver only works with a very specific Intel/Ambient chipset. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] real-time-clock asterisk/meetme/ztdummy in 2.6.9 UML
You might have better luck with app_conference (see the wiki) under UML.. It is probably a little more tolerant of loose timing and scheduling. -SteveK nils toedtmann wrote: Hi *, [this goes to [EMAIL PROTECTED] and [EMAIL PROTECTED] i try to setup an VoIP conferencing server within a UML using asterisk and it's 'MeetMe conference bridge'. I have several UMLs running other services, but my asterisk know-how is poor (as you will see ;-). Question #1: did anybody successfully implement an asterisk MeetMe-conference inside a 2.6-UML? It needs a special zaptel timer. As there is no special hardware, i have to use a ztdummy/2.6 or zaprtc: http://www.voip-info.org/tiki-index.php?page=Asterisk%20timer Question #2: is there any chance that either ztdummy or zaprtc will work inside 2.6-UML? I tried ztdummy. After frickeling the Makefile, it compiled against the UML source tree, it loads within the UML on top of zaptel.ko [EMAIL PROTECTED] asterisk]# lsmod Module Size Used by ztdummy 2628 0 zaptel227940 1 ztdummy (no other modules loaded). Asterisk works (e.g. i can hear the conf- onlyperson message when i enter the conference room). But the audio of the conference itself is extremly snatchy. That _may_ be a config problem, but a very similar configuration on a non-UML works, and un- loading the ztdummy.ko on that system results in the same snatchy audio, so i suspect the ztdummy.ko or some device configuration is broken in my UML. At the same time, i observe this http://seclists.org/lists/linux-kernel/2004/Oct/8108.html problem within my UML, too: # tail -f /var/log/messages tail: cannot read realtime clock: Unknown error 516 and tail exits with code 1. Question #3: is there a problem with the RTC in 2.6.9? My host is 2.6.8.1 + skas3-2.6.7-v5 without module support: CONFIG_SMP=y (host is a P4 with HT) CONFIG_SCHED_SMT=y CONFIG_HPET_TIMER=y CONFIG_HPET_EMULATE_RTC=y CONFIG_RTC=y CONFIG_PREEMPT=y My guest is 2.6.9, with module support. Besides the zaptel modules no feature is compiles as a module but static. CONFIG_UML_REAL_TIME_CLOCK=y Question #4: Shall i try different SMP/HT/RTC settings on host or guest? Regards, /nils. host and guest are FC2. I use the asterisk-1.0 from ATrpms http://atrpms.net/dist/fc2/asterisk/ and seft-built zaptel-1.0.2 tarball user-mode-linux: http://user-mode-linux.sourceforge.net/ http://www.user-mode-linux.org/~blaisorblade/ asterisk: http://www.voip-info.org/tiki-index.php?page=Asterisk http://asterisk.org/ meetme: http://www.voip-info.org/wiki-Asterisk+cmd+MeetMe ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 7910 - Success?
I know that the 7910 only works with Skinny. We have a possible client that wants to bring 80 lines to us off his current provider. All 80 of his phones are Cisco 7910s. Does anyone run 7910s with chan_sccp or the like and find that it works good? Thanks, Matthew ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Hardware Support
Yes look at Ebay for x100P compatible cards On Thu, 04 Nov 2004 10:10:50 -0500, Mike Shultz [EMAIL PROTECTED] wrote: Quick Question that I hope someone can answer. Will Asterisk work with basic PCI FaxModems instead of those expensive cards listed on the hardware page? ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Hardware Support
On November 4, 2004 10:10 am, Mike Shultz wrote: Quick Question that I hope someone can answer. Will Asterisk work with basic PCI FaxModems instead of those expensive cards listed on the hardware page? Quick answer: No. Read up on the Wiki about this -- you can likely get it to work but you won't have support from Digium nor the majority of the community. If you know what you're doing, go for it. If not, make use of the support Digium is selling you with the card. -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Multiline (4 or 8) sip phone
What is a good multiline sip phone for an operator? Model and and manufacturer. The list that I came up with for multiple line/presence SIP phones is: Snom 190, 200 (5 lines) Snom 220 (Expandable number of lines) Cisco 7940 (2 lines) Cisco 7960 (6 lines) Polycom IP 500 (3 lines) Polycom IP 600 (6 lines) ipDialog SipTone (2 lines) Zultys 4x4, 4x5 (4 lines) I'm pretty sure there are others (QTelNet, etc.) but these are the only ones I got a chance to test. After testing the ones above, I found the Polycoms to have the best sound on handset and speakerphone. The Polycoms support fewer codecs, though (ulaw, alaw, and G.729 I think). They are also reasonably priced compared to the Cisco models. I had to search for this info for a LONG time. Eventually, I'll get around to posting it on the Wiki. And yes, the multiple lines are done as multiple SIP registrations with Asterisk. Thanks! Noah ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] supposable timing problem with TE100P
On Thu, 4 Nov 2004, Kurt Bauer wrote: Is your timing source set correctly? If you are connecting to the pstn the pstn connection should be the primary timing source. connection is to a Ericsson MD110 wich is set as network, * is set as CPE. Have you set the span as the timing source? (second number in the span line in zaptel.conf). What does the missed interrupts counter in cat cat /proc/zaptel/1 say? cat /proc/zaptel/1 Span 1: WCT1/0 Digium Wildcard E100P E1/PRA Card 0 HDB3/CCS 1 WCT1/0/1 Clear (In use) 2 WCT1/0/2 Clear (In use) 3 WCT1/0/3 Clear (In use) 4 WCT1/0/4 Clear (In use) Is there no IRQ miss counter for the E100P card? Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Here's a tough question
HI all, I have a question and I cant seem to find the answer anywhere. Is there a way to limit the amount of digits dialed? For example I have a * box set up for the department of corrections for prisoners to call home. It has the Digium 2 FXO/ 2 FXS card in it. I have two Lines brought in to the fxo ports and 2 standard 2500 analog sets for the prisoners to use to dial out. Everything seems to work great. We use * to record all the calls. After the prisoner dials the original number and their PIN, I do not want them to be able to send anymore DTMF tones. The PIN number is not processed by *. It is processed by the LECs switch. Thanks in advance. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Multi-line analog phones with Asterisk?
On Thu, Nov 04, 2004 at 07:18:38AM -0800, Richard Reina said: I am interested in implementing Asterisk and someday hope to have it replace my 8 x 24 Nortel switch. However, I was told by a Telcom friend that my multi line phones (Nortel 7208s) may not work with Asterisk. This is a huge concern because in my business we are constantly jumping back from one line to another (putting people on hold and grabbing another line and going back and forth etc.) Is it possible with Asterisk to (or are there analog phones that allow ) access multiple lines with the press of a button, so that if someone says Richard line 4 is for you, I can easily put my caller on hold and grab line 4? The Nortel 7208 is a proprietary digital phone that only works with nortel equipment, so no, you won't be able to use those phones specifically. However, there are other multi-line phones such as the polycom IP 500 / 600 or the Cisco IP phones that will work just fine with asterisk. I'm guessing, but it sounds like you have the line buttons on your phones mapped to actual phone company lines. This is called a key system type setup. Asterisk is a PBX. You MAY be able to make it function like a key system, but it would be a royal pain. http://experts.about.com/q/2419/1801187.htm With phones that have multiple line appearances such as the polycom or cisco phones mentioned above, you can juggle anywhere from 2 to 6 calls for YOU specifically at once. So if you have 8 people in your office and use phones with 6 line appearances, you could theoretically collectivly juggle 48 calls (how insane would that be?? :-) Anyway, Asterisk has all the features and capabilities of the big boys - well beyond your current norstar system. Rather than someone yelling over that you have a call on line 6, they would just transfer it to your phone, or park the call and yell over that you have a call parked on extension 706 or whatever, which you can go grab, or they send it to voicemail. Flexability here is just about unlimited. Check out: http://www.voip-info.org/wiki-PBX+features http://www.voip-info.org/wiki-Asterisk+PBX+functions http://www.millenigence.com/articles/asterisk-non-technical-review.pdf ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX -- SIP DTMF
This may be a no brainer for some of you out there, simply put it seems that we have a problem passing DTMF from IAX to SIP. The digits cannot be heard coming from the IAX side nor do they seem to register in Asterisk. This seems to happen with any Codec we use so that part has been ruled out. Ideas? ty ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Limit DTMF tones
I sent this to the list earlier but I never saw the post show up, I apologize if this is a repeat post. ___ HI all, I have a question and I cant seem to find the answer anywhere. Is there a way to limit the amount of digits dialed? For example I have a * box set up for the department of corrections for prisoners to call home. It has the Digium 2 FXO/ 2 FXS card in it. I have two Lines brought in to the fxo ports and 2 standard 2500 analog sets for the prisoners to use to dial out. Everything seems to work great. We use * to record all the calls. After the prisoner dials the original number and their PIN, I do not want them to be able to send anymore DTMF tones. The PIN number is not processed by *. It is processed by the LECs switch. Thanks in advance ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] res_features.so Segmentation fault
Sorry, problem solved, it's my mistake.. - Original Message - From: Serge To: [EMAIL PROTECTED] Sent: Thursday, November 04, 2004 9:38 AM Subject: [Asterisk-Users] res_features.so Segmentation fault Have anyone some idea ?Asterisk - latest cvs,RedHat9==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_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'Segmentation faultThanks.Serge. ___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Multiline (4 or 8) sip phone
The list that I came up with for multiple line/presence SIP phones is: Snom 190, 200 (5 lines) Snom 220 (Expandable number of lines) Cisco 7940 (2 lines) Cisco 7960 (6 lines) Polycom IP 500 (3 lines) Polycom IP 600 (6 lines) ipDialog SipTone (2 lines) Zultys 4x4, 4x5 (4 lines) aastra 480i (4 lines SIP,MGCP) http://www.sayson.com/product/voip_phone.htm ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] H323 ISDN
I'm assuming nobody has experience with running ISDN / BRI over H.323... -Original Message- From: Huddleston, Robert [mailto:[EMAIL PROTECTED] Sent: Wednesday, November 03, 2004 8:22 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] H323 ISDN All I need some help here!!! I've configured our * to use OH323 Channel driver as we have a Lucent VoIP box that would only work with OH323 channel driver. I'm trying to run ISDN over H323 - the Lucent VoIP box supports the ISDN over H323 - but it appears that during registration of the line I get all sorts of troubles.. Has anyone been able to use ISDN over H323 on *... I figured that if I could get ISDN BRI working over H323 then I could have multiple call appearances and do some other cool stuff with DID etc... ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Newbie question: forwarding call from PSTN to VoIP
Hi, That's problaby a easy question to solve but I couldn't figure out how to do what I need. My PSTN line is connected to a phone and a FXO card. What I need is when someone calls me, and I don't answer in 3 or 4 rings, * makes a VoIP call to my office, where I'll pickup that call. Or I want to configure that, for example, if I know that nobody will be at home, I want to set that the phone immediately is forwarded to my office. Configuration examples? Where I configure this? Please? Thank you very much. Cheers, Luciano Macedo Rodrigues Opensoft - Porto Alegre/RS ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Limit DTMF tones
On November 4, 2004 10:56 am, Henry Devito wrote:
I have a question and I can't seem to find the answer anywhere. Is there a
way to limit the amount of digits dialed? For example I have a * box set
up for the department of corrections for prisoners to call home. It has
the Digium 2 FXO/ 2 FXS card in it. I have two Lines brought in to the fxo
ports and 2 standard 2500 analog sets for the prisoners to use to dial out.
Everything seems to work great. We use * to record all the calls. After
the prisoner dials the original number and their PIN, I do not want them
to be able to send anymore DTMF tones. The PIN number is not processed by
*. It is processed by the LEC's switch. Thanks in advance
I just did something similar:
[recordcall]
exten = s,1,SetVar(CALLFILENAME=/tmp/calls/${CALLERIDNUM}-${TIMESTAMP})
exten = s,2,Monitor(wav,${CALLFILENAME},m)
exten = s,3,Playback(agent-pass)
exten = s,4,DISA(/path/to/asterisk/passwd/file)
and then just make sure that the extension you dump them in to does not allow
them to dial anything ohter than what you want (i.e. 7-digit #s or whatnot)
I don't know of any way to suppress DTMF if that's what you're talking about.
-A.
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[Asterisk-Users] Stop AGI proccess after user hang-up
Does anybody know how to stop the AGI process after the user Hang-Up? 'Cause it stills running if the transaction is ended by the user. Thanks Víctor ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] res_features.so Segmentation fault
On Thu, 2004-11-04 at 16:57 +0100, Serge wrote: Sorry, problem solved, it's my mistake.. Please share more information. Asterisk shouldn't segfault on simple user problems. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_capi patch : fax support
Hello,
For those of you who have a CAPI card with an on-board DSP (like some Eicon
Diva Server), this patch allows you to receive faxes.
If you want to answer a channel in fax mode, use capiAnswerFax() instead of
Answer()
If you use Answer(), you will be in voice mode. If the hardware DSP detects
a fax tone, you can switch from voice to fax mode by calling
capiAnswerFax().
Example of use :
line number 123, play something, if a fax tone is detected, handle it
line number 124, answer directly in fax mode
[incoming]
exten = 123,1,Answer()
exten = 123,2,BackGround(jpop)
exten = 124,1,Goto(handle_fax,s,1)
exten = fax,1,Goto(handle_fax,s,1)
[handle_fax]
exten = s,1,capiAnswerFax(/tmp/${UNIQUEID})
exten = s,2,Hangup()
exten = h,1,deadagi,fax.php // Run sfftobmp and mail it.
The output of capiAnswerFax is a SFF file. Use sfftobmp to convert it.
With a Diva Server, theses features are allowed : fax up to 33600, high
resolution. Color Fax /JPEG Compression is disabled (I can't test it).
You can download the patch at :
http://www.mlkj.net/asterisk/chan_capi-0.3.5-patch.tar.bz2
A fix for a dead lock issue is also included (Oct 22 18:06:00
WARNING[11275]: channel.c:472 ast_channel_walk_locked: Avoided initial
deadlock for 'CAPI[contr1/173720007]/7', 10 retries!)
--
Carl
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[Asterisk-Users] avm fritz box fon
Hi List I recently got one of those boxes (without wlan). Works as ata device with my local asterisk. Just tested the basic stuff like call the box and make call from the box. It uses sip with alaw/ulaw/g726 codecs. Runs on linux, kernel 2.4.17, mipsel. HardWare: -wan (UR2/annexB) -ethernet (10/100) -usb (usb-net) -3 capicontrolers (avm-fcclassic) 1 for fxo, 2 for fxs -16MB flash with squashfs -4MB memory... SoftWare: -busybox -telnetd -webserver Connections: -wan (Modem on board) -lan -usb -fxo (isdn or anlog) -2fxs Nice box maybe one day someone replaces the avm-software with asterisk! It looks like almost the same arch as the linksys-router. I will shortly set up some detailed readme about the inside. -- Tho/\/\as ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Best Linux base for small Asterisk server?
I'm in the process of building up a small (1x1) test Asterisk box based on a 1Ghz VIA C3 Mini-ITX box with one PCI slot (and a FX100P). Anyone have suggestions as to the best Linux distribution (or kernel) to base the system on? I'll just have one FXO/POTS line and then a Grandstream Budgetone 101 IP phone; this is more for playing with IVR functionality than anything else. Thanks. Bill ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 noise on ADSL line.
Following on from the message below I have discovered that the X100P causes the SNR on my ADSL line to drop even with the Asterisk box **switched off** and the power unplugged... This seems very strange.. Why should a card in a switched off PC cause noise on a line meaning that it drops out and has to reconnect quite often.. Anyone got any other ideas to try and stop it messing up my internet connection cos its causing havoc with my VoIP calls coming in and going out over the ADSL line.. Later.. WipeOut wrote: Hi, This may be one for the broadband guru's out there.. I have a single analog line coming into the house.. This line is for my ADSL and home phone.. My Asterisk box uses an X100P card to connect to the analog line.. I have a microfilter on the line etc.. The rest of my phone system works inbound and outbound calls via a VoIP provider over the ADSL line.. The problem I am having is that the X100P seems to introduce a lot of noise on the line when it its connected to the phone socket on the microfilter and this causes the ADSL quality to drop quite badly.. When the X100P is not connected I have a signal to noise ratio of 29dB downstream and 30dB upstream (this stays the same when I connect an analog phone) when I connect the X100P the SNR drops to 12dB downstream and 30dB upstream.. At 12dB I get a large number of CRC errors and errored seconds on the ADSL connection.. Anyone got any ideas why the X100P would cause this kind of deterioration? Only thing I can think of is possibly something to do with ring detection or that its acting on some of the frequencies that are being used by the ADSL.. Thanks for any thoughts.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ValetParking
Does anyone that the source for app_valetparking.c Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] chan_capi patch : fax support
On Thu, 2004-11-04 at 17:31, Carl Sempla wrote: Hello, For those of you who have a CAPI card with an on-board DSP (like some Eicon Diva Server), this patch allows you to receive faxes. [snip] A fix for a dead lock issue is also included (Oct 22 18:06:00 WARNING[11275]: channel.c:472 ast_channel_walk_locked: Avoided initial deadlock for 'CAPI[contr1/173720007]/7', 10 retries!) Hi Carl, Thanks for the patch. I'll apply it later today and give it a whirl. Maybe it is an idea to send the patch also to kapejod who wrote chan_capi. His site/contact details can be found at www.junghanns.net Regards, Patrick ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Best Linux base for small Asterisk server?
On November 4, 2004 11:39 am, Bill Bradford wrote: Anyone have suggestions as to the best Linux distribution (or kernel) to base the system on? I'm a fan of a trimmed-down slackware but there are smaller distros yet. My entire * setup fits into less than 500MB and that's without really cutting in there and trimming things out. -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] system errors
Im getting the following error when I do an strace -p # for asterisk . Its taking alot of resources. ioctl(0, SNDCTL_TMR_TIMEBASE or TCGETS, 0xb290) = -1 EIO (Input/output error) write(1, voip1*CLI , 11) = -1 EIO (Input/output error) I checked the WIKI and the list and found nothing. Kyle ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] system errors
On Thu, 2004-11-04 at 10:11 -0700, kyle Hagan wrote: Im getting the following error when I do an strace -p # for asterisk . Its taking alot of resources. ioctl(0, SNDCTL_TMR_TIMEBASE or TCGETS, 0xb290) = -1 EIO (Input/output error) write(1, voip1*CLI , 11) = -1 EIO (Input/output error) I checked the WIKI and the list and found nothing. strace takes a lot of resources and depending on the driver for the display, you can really suck a lot of CPU just trying to keep up with the output. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Multi-line analog phones with Asterisk?
Thank you very much for your thoghtful and thorough response. I guess I don't wan't to set up * to behave like a key system, thank godness, I just want to be able to juggle calls which it sould like Asterisk can do fine. Just to clarify though, can the polycom IP 500 / 600 work on analog lines? Thanks Again, Richard --- Walt Reed [EMAIL PROTECTED] wrote: On Thu, Nov 04, 2004 at 07:18:38AM -0800, Richard Reina said: I am interested in implementing Asterisk and someday hope to have it replace my 8 x 24 Nortel switch. However, I was told by a Telcom friend that my multi line phones (Nortel 7208s) may not work with Asterisk. This is a huge concern because in my business we are constantly jumping back from one line to another (putting people on hold and grabbing another line and going back and forth etc.) Is it possible with Asterisk to (or are there analog phones that allow ) access multiple lines with the press of a button, so that if someone says Richard line 4 is for you, I can easily put my caller on hold and grab line 4? The Nortel 7208 is a proprietary digital phone that only works with nortel equipment, so no, you won't be able to use those phones specifically. However, there are other multi-line phones such as the polycom IP 500 / 600 or the Cisco IP phones that will work just fine with asterisk. I'm guessing, but it sounds like you have the line buttons on your phones mapped to actual phone company lines. This is called a key system type setup. Asterisk is a PBX. You MAY be able to make it function like a key system, but it would be a royal pain. http://experts.about.com/q/2419/1801187.htm With phones that have multiple line appearances such as the polycom or cisco phones mentioned above, you can juggle anywhere from 2 to 6 calls for YOU specifically at once. So if you have 8 people in your office and use phones with 6 line appearances, you could theoretically collectivly juggle 48 calls (how insane would that be?? :-) Anyway, Asterisk has all the features and capabilities of the big boys - well beyond your current norstar system. Rather than someone yelling over that you have a call on line 6, they would just transfer it to your phone, or park the call and yell over that you have a call parked on extension 706 or whatever, which you can go grab, or they send it to voicemail. Flexability here is just about unlimited. Check out: http://www.voip-info.org/wiki-PBX+features http://www.voip-info.org/wiki-Asterisk+PBX+functions http://www.millenigence.com/articles/asterisk-non-technical-review.pdf ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? Check out the new Yahoo! Front Page. www.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] system errors
Im not running strace constantly, I just run it to see whats going on. And that error it coming in on the Asterisk PID at a rate of about 10 per second, from what I can see. Kyle Steven Critchfield wrote: On Thu, 2004-11-04 at 10:11 -0700, kyle Hagan wrote: Im getting the following error when I do an strace -p # for asterisk . Its taking alot of resources. ioctl(0, SNDCTL_TMR_TIMEBASE or TCGETS, 0xb290) = -1 EIO (Input/output error) write(1, voip1*CLI , 11) = -1 EIO (Input/output error) I checked the WIKI and the list and found nothing. strace takes a lot of resources and depending on the driver for the display, you can really suck a lot of CPU just trying to keep up with the output. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] system errors
Im not running strace constantly, I just run it to see whats going on. And that error it coming in on the Asterisk PID at a rate of about 10 per second, from what I can see. here is my TOP: Cpu(s): 86.3% user, 13.7% system, 0.0% nice, 0.0% idle Mem:904204k total, 862688k used,41516k free,0k buffers Swap: 1004020k total,0k used, 1004020k free, 578036k cached PID USER PR NI VIRT RES SHR S %CPU %MEMTIME+ COMMAND 10744 root 20 0 1400 1400 1096 R 50.0 0.2 699:46.68 asterisk 23199 root 19 0 1408 1408 1096 R 49.7 0.2 536:18.27 asterisk 13838 root 9 0 7404 7332 3644 R 0.3 0.8 0:00.14 asterisk its showing 50% CPU x 2. when I strace -p 10744 or strace -p 23199 thats when I get the error. Kyle Steven Critchfield wrote: On Thu, 2004-11-04 at 10:11 -0700, kyle Hagan wrote: Im getting the following error when I do an strace -p # for asterisk . Its taking alot of resources. ioctl(0, SNDCTL_TMR_TIMEBASE or TCGETS, 0xb290) = -1 EIO (Input/output error) write(1, voip1*CLI , 11) = -1 EIO (Input/output error) I checked the WIKI and the list and found nothing. strace takes a lot of resources and depending on the driver for the display, you can really suck a lot of CPU just trying to keep up with the output. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Limit DTMF tones
The issue is the inmates have figured out a way to dial long distance
numbers by calling different private phone numbers and using that companies
DISA to complete calls. So in order to stop that I have to suppress dtmf
after so many digits are dialed. Any idea's?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Kohlsmith
Sent: Thursday, November 04, 2004 10:13 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Limit DTMF tones
On November 4, 2004 10:56 am, Henry Devito wrote:
I have a question and I can't seem to find the answer anywhere. Is there
a
way to limit the amount of digits dialed? For example I have a * box set
up for the department of corrections for prisoners to call home. It has
the Digium 2 FXO/ 2 FXS card in it. I have two Lines brought in to the
fxo
ports and 2 standard 2500 analog sets for the prisoners to use to dial
out.
Everything seems to work great. We use * to record all the calls. After
the prisoner dials the original number and their PIN, I do not want them
to be able to send anymore DTMF tones. The PIN number is not processed by
*. It is processed by the LEC's switch. Thanks in advance
I just did something similar:
[recordcall]
exten = s,1,SetVar(CALLFILENAME=/tmp/calls/${CALLERIDNUM}-${TIMESTAMP})
exten = s,2,Monitor(wav,${CALLFILENAME},m)
exten = s,3,Playback(agent-pass)
exten = s,4,DISA(/path/to/asterisk/passwd/file)
and then just make sure that the extension you dump them in to does not
allow
them to dial anything ohter than what you want (i.e. 7-digit #s or whatnot)
I don't know of any way to suppress DTMF if that's what you're talking
about.
-A.
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RE: [Asterisk-Users] Best Linux base for small Asterisk server?
I'm using RH9 and Mandrake 10, because they were easy to install. I heard good things about Gentoo when properly built and tweaked, but it requires some effort. Search the list archives for previous discussions. -Original Message- From: Bill Bradford [mailto:[EMAIL PROTECTED] Sent: Thursday, November 04, 2004 10:40 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Best Linux base for small Asterisk server? I'm in the process of building up a small (1x1) test Asterisk box based on a 1Ghz VIA C3 Mini-ITX box with one PCI slot (and a FX100P). Anyone have suggestions as to the best Linux distribution (or kernel) to base the system on? I'll just have one FXO/POTS line and then a Grandstream Budgetone 101 IP phone; this is more for playing with IVR functionality than anything else. Thanks. Bill ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/aster isk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Stop AGI proccess after user hang-up
You have to use DeadAGI instead of AGI to call your script, for example: exten = 77,1,Answer exten = 77,2,DeadAGI(astcc.agi) exten = 77,3,Hangup Regards.. Nahuel Ramos. On Thu, 4 Nov 2004 13:14:56 -0400, Victor Cartes [EMAIL PROTECTED] wrote: Does anybody know how to stop the AGI process after the user Hang-Up? 'Cause it stills running if the transaction is ended by the user. Thanks Víctor ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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: anyone using pointone?
Sorry for the OT message but I'm very curious to see if anyone on this list uses pointone for long distance sip call termination? We've been having an off and on problem with them saying they do not support sip message with a fqdn in the from field.. which to me appears to be a breakage of the sip rfc.. and to top it off all our other calls process through them just fine expect to a current problem area code out in California.. I feel they are giving us a very generic white-washed answer and do not wish to actually provide good customer service.. opinions, comments, or cuss words? begin:vcard fn:Matt Hess n:Hess;Matt org:LiveWireNet adr;dom:;;4577 Pecos St;Denver;CO;80211 email;internet:[EMAIL PROTECTED] title:Senior Network Engineer tel;work:303-458-5667 tel;fax:303-458-5725 x-mozilla-html:FALSE url:http://www.livewirenet.com/ version:2.1 end:vcard ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Limit DTMF tones
On Thu, 2004-11-04 at 11:30 -0600, Henry Devito wrote:
The issue is the inmates have figured out a way to dial long distance
numbers by calling different private phone numbers and using that companies
DISA to complete calls. So in order to stop that I have to suppress dtmf
after so many digits are dialed. Any idea's?
look at the code for # transfer, Asterisk is finding the DTMF while a
call is in progress. You could probably do something along the way of
checking the call timer and once it exceeds a certain point, all DTMFs
are just ignored. It doesn't seem like it would be difficult, but I
haven't looked at the code, nor thought about how that could be merged
back into the codebase as a configurable option.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Kohlsmith
Sent: Thursday, November 04, 2004 10:13 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Limit DTMF tones
On November 4, 2004 10:56 am, Henry Devito wrote:
I have a question and I can't seem to find the answer anywhere. Is there
a
way to limit the amount of digits dialed? For example I have a * box set
up for the department of corrections for prisoners to call home. It has
the Digium 2 FXO/ 2 FXS card in it. I have two Lines brought in to the
fxo
ports and 2 standard 2500 analog sets for the prisoners to use to dial
out.
Everything seems to work great. We use * to record all the calls. After
the prisoner dials the original number and their PIN, I do not want them
to be able to send anymore DTMF tones. The PIN number is not processed by
*. It is processed by the LEC's switch. Thanks in advance
I just did something similar:
[recordcall]
exten = s,1,SetVar(CALLFILENAME=/tmp/calls/${CALLERIDNUM}-${TIMESTAMP})
exten = s,2,Monitor(wav,${CALLFILENAME},m)
exten = s,3,Playback(agent-pass)
exten = s,4,DISA(/path/to/asterisk/passwd/file)
and then just make sure that the extension you dump them in to does not
allow
them to dial anything ohter than what you want (i.e. 7-digit #s or whatnot)
I don't know of any way to suppress DTMF if that's what you're talking
about.
-A.
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--
Steven Critchfield [EMAIL PROTECTED]
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Re: [Asterisk-Users] Best Linux base for small Asterisk server?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi I'm using Fedora RC1 without problems (kernel 2.4.22) Anyone here have tried Whitebox Respin 1 with asterisk ? Maybe nest week i'll install a small Asterisk Server on it. Asterisk + Apache only. Bill Bradford wrote: | I'm in the process of building up a small (1x1) test Asterisk box | based on a 1Ghz VIA C3 Mini-ITX box with one PCI slot (and a | FX100P). | | Anyone have suggestions as to the best Linux distribution (or | kernel) to base the system on? | | I'll just have one FXO/POTS line and then a Grandstream Budgetone | 101 IP phone; this is more for playing with IVR functionality than | anything else. | | Thanks. | | Bill ___ Asterisk-Users | mailing list [EMAIL PROTECTED] | http://lists.digium.com/mailman/listinfo/asterisk-users To | UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | - -- Manuel João S. Costa Amaro [EMAIL PROTECTED] ICQ: 57398499 MSN: [EMAIL PROTECTED] As únicas pessoas que aprecio são os loucos: os que são loucos para viver, loucos para falar, loucos para se salvar, desejosos de tudo ao mesmo tempo; os que nunca bocejam nem dizem lugares-comuns, mas que ardem, se inflamam e brilham como fabulosos fogos-de-artifício. (Jack Kerouac) -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQFBimtIJUm/Bor63CERAsV+AJ9ns8COTviXq1wuq1ulRbAr017HywCgmQES EvLwOwbb64aZoNs0Lsg/PrY= =7olh -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Limit DTMF tones
On November 4, 2004 12:30 pm, Henry Devito wrote: The issue is the inmates have figured out a way to dial long distance numbers by calling different private phone numbers and using that companies DISA to complete calls. So in order to stop that I have to suppress dtmf after so many digits are dialed. Any idea's? Without hacking the source... no. -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip clients not longer registering
Karl Brose wrote: The REGISTER requests that your SIP UAs are sending as listed are not requests to *register*, but request to *unregister* The contacts are '*' and expirations are '0' Granted that Asterisk doesn't do registrations correctly, but it does need a proper registration request with a contact and an Expires value 0 to enter something in its location database. David Filion wrote: Hi, We have been using Asterisk since version 0.9x with little or no problems. However, for an unknow reasons, our sip clients can nolonger register. We updated to Asterisk 1.0.2 hoping that would solve the problem, but no luck. Here is the entry from sip.conf for one of our clients: [10012200] host=dynamic nat=yes type=friend [EMAIL PROTECTED] username=10012200 secret= context=1001 port=5060 quality=1000 dtmfmode=rfc2833 canreinvite=no callerid=Muffin Man 1222333 disallow=all allow=g729 The settings have been check in the sip client (a gs 486) and they match. Below is a couple of sip sessions from when the user device attempts to register: sip*CLI Sip read: REGISTER sip:111.222.333.444 SIP/2.0 Via: SIP/2.0/UDP 555.666.777.888;branch=z9hG4bKd4dd151b65533090 From: David Filion sip:[EMAIL PROTECTED];tag=706ef4f92086984a To: sip:[EMAIL PROTECTED] Contact: * Call-ID: [EMAIL PROTECTED] CSeq: 100 REGISTER Expires: 0 User-Agent: Grandstream HT486 1.0.5.16 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Length: 0 12 headers, 0 lines Using latest request as basis request Sending to 555.666.777.888 : 5060 (non-NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 555.666.777.888;branch=z9hG4bKd4dd151b65533090;received=555.666.777.888;rport=1024 From: David Filion sip:[EMAIL PROTECTED];tag=706ef4f92086984a To: sip:[EMAIL PROTECTED];tag=as3055bbba Call-ID: [EMAIL PROTECTED] CSeq: 100 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 555.666.777.888:1024 Transmitting (NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 555.666.777.888;branch=z9hG4bKd4dd151b65533090;received=555.666.777.888;rport=1024 From: David Filion sip:[EMAIL PROTECTED];tag=706ef4f92086984a To: sip:[EMAIL PROTECTED];tag=as3055bbba Call-ID: [EMAIL PROTECTED] CSeq: 100 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] WWW-Authenticate: Digest realm=asterisk, nonce=626079d1 Content-Length: 0 to 555.666.777.888:1024 Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms sip*CLI Sip read: REGISTER sip:111.222.333.444 SIP/2.0 Via: SIP/2.0/UDP 555.666.777.888;branch=z9hG4bKd4dd151b65533090 From: David Filion sip:[EMAIL PROTECTED];tag=706ef4f92086984a To: sip:[EMAIL PROTECTED] Contact: * Call-ID: [EMAIL PROTECTED] CSeq: 100 REGISTER Expires: 0 User-Agent: Grandstream HT486 1.0.5.16 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Length: 0 12 headers, 0 lines Using latest request as basis request Sending to 555.666.777.888 : 5060 (NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 555.666.777.888;branch=z9hG4bKd4dd151b65533090;received=555.666.777.888;rport=1024 From: David Filion sip:[EMAIL PROTECTED];tag=706ef4f92086984a To: sip:[EMAIL PROTECTED];tag=as3055bbba Call-ID: [EMAIL PROTECTED] CSeq: 100 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 555.666.777.888:1024 Transmitting (NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 555.666.777.888;branch=z9hG4bKd4dd151b65533090;received=555.666.777.888;rport=1024 From: David Filion sip:[EMAIL PROTECTED];tag=706ef4f92086984a To: sip:[EMAIL PROTECTED];tag=as3055bbba Call-ID: [EMAIL PROTECTED] CSeq: 100 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] WWW-Authenticate: Digest realm=asterisk, nonce=626079d1 Content-Length: 0 to 555.666.777.888:1024 Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms sip*CLI Sip read: REGISTER sip:111.222.333.444 SIP/2.0 Via: SIP/2.0/UDP 555.666.777.888;branch=z9hG4bKd4dd151b65533090 From: David Filion sip:[EMAIL PROTECTED];tag=706ef4f92086984a To: sip:[EMAIL PROTECTED] Contact: * Call-ID: [EMAIL PROTECTED] CSeq: 100 REGISTER Expires: 0 User-Agent: Grandstream HT486 1.0.5.16 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Length: 0 12 headers, 0 lines Using latest request as basis request Sending to 555.666.777.888 : 5060 (NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 555.666.777.888;branch=z9hG4bKd4dd151b65533090;received=555.666.777.888;rport=1024 From: David Filion sip:[EMAIL PROTECTED];tag=706ef4f92086984a To: sip:[EMAIL PROTECTED];tag=as3055bbba Call-ID: [EMAIL PROTECTED] CSeq: 100 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact:
Re: [Asterisk-Users] Best Linux base for small Asterisk server?
João Amaro wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi I'm using Fedora RC1 without problems (kernel 2.4.22) Anyone here have tried Whitebox Respin 1 with asterisk ? Maybe nest week i'll install a small Asterisk Server on it. Asterisk + Apache only. Bill Bradford wrote: | I'm in the process of building up a small (1x1) test Asterisk box | based on a 1Ghz VIA C3 Mini-ITX box with one PCI slot (and a | FX100P). | | Anyone have suggestions as to the best Linux distribution (or | kernel) to base the system on? | | I'll just have one FXO/POTS line and then a Grandstream Budgetone | 101 IP phone; this is more for playing with IVR functionality than | anything else. | | Thanks. | | Bill ___ Asterisk-Users | mailing list [EMAIL PROTECTED] | http://lists.digium.com/mailman/listinfo/asterisk-users To | UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | - -- Manuel João S. Costa Amaro [EMAIL PROTECTED] ICQ: 57398499 MSN: [EMAIL PROTECTED] As únicas pessoas que aprecio são os loucos: os que são loucos para viver, loucos para falar, loucos para se salvar, desejosos de tudo ao mesmo tempo; os que nunca bocejam nem dizem lugares-comuns, mas que ardem, se inflamam e brilham como fabulosos fogos-de-artifício. (Jack Kerouac) -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQFBimtIJUm/Bor63CERAsV+AJ9ns8COTviXq1wuq1ulRbAr017HywCgmQES EvLwOwbb64aZoNs0Lsg/PrY= =7olh -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Wasn't there a long thread about this just last week? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ATCC - Astcc-Admin.cgi File
Does anyone in the list have a fully functional ASTCC and would like to share their CGI, AGI and CONF files/Scripts and database installation that is customized for: 1) Accepting user input for a Pin for authentication 2) Recognizes the caller id for authentication 3) Has a better GUI to manage the cards and users 4) PHPMysqlAdmin installation for managing the database in exchange for a brand new 2 Port version of eezeephone(Netweb-302) (would this offer be deemed a commercial discussion by the list?? :) ) Please contact off the list Seshu Kanuri 732-213-2422 [EMAIL PROTECTED] [EMAIL PROTECTED] NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 ISDN HFC-S card (Biilion) instead of Fritz Capi ?
Hi I am trying the fine iso at http://www.asterisk.de.ms/ but are having problems with Capi probably due to having to old Fritz PCI card. Trying with both non version marked version and version marked V 2.0. I get following error when booting Astrisk on Debian: Oct 31 02:26:10 asterisk kernel: kcapi: capi20 attached Oct 31 02:26:10 asterisk kernel: capi20: Rev 1.1.4.2: started up with major 68 ( middleware+capifs) Oct 31 02:26:10 asterisk kernel: fcpci: AVM FRITZ!Card PCI driver, revision 0.2 Oct 31 02:26:10 asterisk kernel: fcpci: (fcpci built on Jun 3 2004 at 12:02:52) Oct 31 02:26:10 asterisk kernel: fcpci: Loading... Oct 31 02:26:10 asterisk kernel: fcpci: Driver 'fcpci' attached to stack Oct 31 02:26:10 asterisk kernel: kcapi: driver fcpci attached Oct 31 02:26:10 asterisk kernel: fcpci: Auto-attaching... Oct 31 02:26:10 asterisk kernel: PCI: Found IRQ 11 for device 00:0c.0 Oct 31 02:26:10 asterisk kernel: fcpci: Error: Invalid parameters (base=0xe800, irq=11) Oct 31 02:26:10 asterisk kernel: fcpci: Not loaded. Oct 31 02:26:10 asterisk kernel: kcapi: driver fcpci detached Can anbody on the list give a clue on this problem ? Or can you please direct me to a guide on how to install driver for HFC-S based ISDN card on Asterisk/Debian ? Thank you ! HB Norway ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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: Hardware Support
Thanks to everyone who answered my stupid question. I did not see the wiki nor an answer to my question on their main site. My main experience is with a 3Com NBX 25 which is small and really simple so I never had to learn much except the dial plan, which is why most of this is still foreign to me. I have one more question. Are there special interface cards for IP Phone/Network connectivity or is this a job for a basic NIC? I've been reading through the wiki on most of the protocols but it doesn't state what media they run on. I would prefer to run this over the current data network I already have as I would not need to run new cables. I'd rather not use analog phones at all either so I'm not looking for one of those S100I media converters. Any suggestions would be appreciated. -- Mike Shultz [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Grandstream BT100 - Failed to write frame
Hi everyone, I'm having problems with Playback() on a Grandstream Budge Tone-100. Every time Playback is used I get the following messages: WARNING[229388]: file.c:550 ast_readaudio_callback: Failed to write frame == Spawn extension (from-sip, , 1) exited non-zero on 'SIP/2002-559c' I tried pretty much every codec on the phone with no succes. I'm using: Asterisk CVS-D2004.10.25.17.09.17-11/04/04-12:25:54 GrandStream Firmware: Program--1.0.4.65 sip.conf entry for the BT-100 Phone: [BT100] type=friend context=from-sip username=2002 fromuser=2002 secret=123456 host=dynamic nat=no The grandstream phone and asterisk are on the same lan. Thanks a lot, Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] ATCC - Astcc-Admin.cgi File
Sounds more like a requirement for custom development since I'm sure your needs will vary from some others that are also using astcc as a starting point for their prepaid cards -- William ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Stop AGI proccess after user hang-up
On 11/4/2004, Victor Cartes [EMAIL PROTECTED] wrote: Does anybody know how to stop the AGI process after the user Hang-Up? 'Cause it stills running if the transaction is ended by the user. I'm unsure (haven't had a need yet to use AGI), but perhaps you could use the DeadAgi application in your dialplan to execute another app that kills the first one. Do a show application DeadAgi at the CLI. Other than that you might have to build the logic within your AGI itself, basically terminating it after say X seconds of no activity when the caller's supposed to be doing something. Flynn ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Best Linux base for small Asterisk server?
I'm using * in a prepaid environment with Whitebox Respin with all the bells and whistles loaded. I made the move when Redhat lost their mind and discontinued support of 9.0. The asterisk is still in test and development mode (not a lot of traffic yet) but it seems to work okay. I am, however running an older version of asterisk CVS from May 2004, as versions after that gave me fits with some of the apps I depend on (playinterruptibletones for instance). I doubt the problems I encountered were the result of the O/S, more likely they are the result of improvements? to the asterisk CVS that conflict with some of the apps I'm using. Installing, compiling and running * on Whitebox was straight forward and trouble free. - Original Message - From: João Amaro [EMAIL PROTECTED] To: Bill Bradford [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Thursday, November 04, 2004 11:47 AM Subject: Re: [Asterisk-Users] Best Linux base for small Asterisk server? -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi I'm using Fedora RC1 without problems (kernel 2.4.22) Anyone here have tried Whitebox Respin 1 with asterisk ? Maybe nest week i'll install a small Asterisk Server on it. Asterisk + Apache only. Bill Bradford wrote: | I'm in the process of building up a small (1x1) test Asterisk box | based on a 1Ghz VIA C3 Mini-ITX box with one PCI slot (and a | FX100P). | | Anyone have suggestions as to the best Linux distribution (or | kernel) to base the system on? | | I'll just have one FXO/POTS line and then a Grandstream Budgetone | 101 IP phone; this is more for playing with IVR functionality than | anything else. | | Thanks. | | Bill ___ Asterisk-Users | mailing list [EMAIL PROTECTED] | http://lists.digium.com/mailman/listinfo/asterisk-users To | UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | - -- Manuel João S. Costa Amaro [EMAIL PROTECTED] ICQ: 57398499 MSN: [EMAIL PROTECTED] As únicas pessoas que aprecio são os loucos: os que são loucos para viver, loucos para falar, loucos para se salvar, desejosos de tudo ao mesmo tempo; os que nunca bocejam nem dizem lugares-comuns, mas que ardem, se inflamam e brilham como fabulosos fogos-de-artifício. (Jack Kerouac) -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQFBimtIJUm/Bor63CERAsV+AJ9ns8COTviXq1wuq1ulRbAr017HywCgmQES EvLwOwbb64aZoNs0Lsg/PrY= =7olh -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 question: forwarding call from PSTN to VoIP
On Thu, 4 Nov 2004, Luciano Macedo Rodrigues wrote:
That's problaby a easy question to solve but I couldn't figure out how to do
what I need.
My PSTN line is connected to a phone and a FXO card. What I need is when
someone calls me, and I don't answer in 3 or 4 rings, * makes a VoIP call to
my office, where I'll pickup that call. Or I want to configure that, for
example, if I know that nobody will be at home, I want to set that the phone
immediately is forwarded to my office.
Configuration examples? Where I configure this? Please?
It would be configured in extensions.conf. Maybe a timeout on your Dial()
will help accomplish what you're after. For example:
exten = 100,1,Dial(ZAPexten,10)
exten = 100,2,Dial(SIP/youroffice,10)
would ring the zap extension for 10 seconds, then try the SIP extension
for 10 seconds, and then would drop off to somewhere (you might want to
route to voicemail, or play an automated greeting, or simply hang up the
call).
Use the help facility in the CLI ('show application dial') to find out
more.
Greg
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Re: [Asterisk-Users] Multi-line analog phones with Asterisk?
On Nov 4, 2004, at 9:23 AM, Richard Reina wrote: Thank you very much for your thoghtful and thorough response. I guess I don't wan't to set up * to behave like a key system, thank godness, I just want to be able to juggle calls which it sould like Asterisk can do fine. Just to clarify though, can the polycom IP 500 / 600 work on analog lines? It depends on what you mean by analog lines. If you mean will it work with calls that come into Asterisk over analog phone lines then the answer is yes--the phones don't know or care how the phone call made its way into Asterisk. If you mean can I plug the Polycom IP500/600 into a regular analog phone jack then the answer is no--they plug into an Ethernet jack, not an analog phone jack. Scott ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Passing callerID info to a forwarded line
Hi everyone, I have a question regarding the use of callerID and call forwarding. When I forward any of my Zap extensions in the office to an outside line, such as a cell phone, the callerID info shows up as originating from that office phone, rather than from whoever actually originated the call into that office phone. Does anyone have an idea of how to pass the callerID info of the originating caller to the forwarded phone? Thanks, Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] ATCC - Astcc-Admin.cgi File
Not necessarily. The need is a generic calling card app - take user Input or recognize the ANI, Allow the calls The users and pins are stored in Mysql database In order to make the database easy to manage - as the users and pins are stored in Mysql database, PHPMysqlAdmin (which is a generic GNU install) is a much more preferred way for me to manage the user data, as that way I can upload and download tonnes of Pins and users in one go. Also PHPMysqlAdmin helps the Admin and make it is easy to drop them quickly. Reporting is another need for needing to use PHPMysqlAdmin, so that we can give a frontend to the user to check their calls and CDRs, in a few minutes of effort. Seshu Kanuri -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of William Suffill Sent: Thursday, November 04, 2004 1:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] ATCC - Astcc-Admin.cgi File Sounds more like a requirement for custom development since I'm sure your needs will vary from some others that are also using astcc as a starting point for their prepaid cards -- William ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Multi-line analog phones with Asterisk?
Thank you very much for that clarification. --- Scott Laird [EMAIL PROTECTED] wrote: On Nov 4, 2004, at 9:23 AM, Richard Reina wrote: Thank you very much for your thoghtful and thorough response. I guess I don't wan't to set up * to behave like a key system, thank godness, I just want to be able to juggle calls which it sould like Asterisk can do fine. Just to clarify though, can the polycom IP 500 / 600 work on analog lines? It depends on what you mean by analog lines. If you mean will it work with calls that come into Asterisk over analog phone lines then the answer is yes--the phones don't know or care how the phone call made its way into Asterisk. If you mean can I plug the Polycom IP500/600 into a regular analog phone jack then the answer is no--they plug into an Ethernet jack, not an analog phone jack. Scott ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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: Hardware Support
On Thu, 2004-11-04 at 13:15 -0500, Mike Shultz wrote: Thanks to everyone who answered my stupid question. I did not see the wiki nor an answer to my question on their main site. My main experience is with a 3Com NBX 25 which is small and really simple so I never had to learn much except the dial plan, which is why most of this is still foreign to me. I have one more question. Are there special interface cards for IP Phone/Network connectivity or is this a job for a basic NIC? I've been reading through the wiki on most of the protocols but it doesn't state what media they run on. I would prefer to run this over the current data network I already have as I would not need to run new cables. I'd rather not use analog phones at all either so I'm not looking for one of those S100I media converters. Any suggestions would be appreciated. At least based on your stated background this one isn't stupid. You are correct in your assumption though that the VoIP part is just packets for the NIC to handle. There is no special hardware for it. Although when you get to some of the conference portions of asterisk, you will need a timing source that is provided either by the PSTN hardware Digium sells or via psuedo devices such as the ZapRTC or ztdummy interface. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Stop AGI proccess after user hang-up
On Thu, 2004-11-04 at 13:14 -0400, Victor Cartes wrote: Does anybody know how to stop the AGI process after the user Hang-Up? 'Cause it stills running if the transaction is ended by the user. You must handle t his in your AGI application. If you start getting back broken reads from the STDIN filehandle, the other side of the pipe has gone away and you need to clean up and exit. Also most AGI commands will let you know when asterisk has detected a hangup or other failure condition and you should handle those appropriately. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Passing callerID info to a forwarded line
On Thu, 4 Nov 2004, Chris Goodwin wrote: I have a question regarding the use of callerID and call forwarding. When I forward any of my Zap extensions in the office to an outside line, such as a cell phone, the callerID info shows up as originating from that office phone, rather than from whoever actually originated the call into that office phone. Does anyone have an idea of how to pass the callerID info of the originating caller to the forwarded phone? Area you using a PRI line, or what? If a PRI, you need your provider to allow you to set the outgoing CallerID to whatever you'd like, instead of just one of your own numbers. If BRI, Analog, etc, I don't think there is a way to set your own CallerID. | nate carlson | [EMAIL PROTECTED] | http://www.natecarlson.com | | depriving some poor village of its idiot since 1981| ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Is it possible to use IAXY device to make 56K modem calls
ALL, is it possible to plug a standard analog 56K modem into my iaxy device and make a modem call out? 9600 baud call would be fine actually. I just want to make a call out with my iAXy device and eliminate my PSTN line. THanks, Jerry ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Limit DTMF tones
On 11/4/2004, Henry Devito [EMAIL PROTECTED] wrote:
The issue is the inmates have figured out a way to dial long distance
numbers by calling different private phone numbers and using that companies
DISA to complete calls. So in order to stop that I have to suppress dtmf
after so many digits are dialed. Any idea's?
My, prisoners are getting devious :)
Anyways, you'd only be able to do this by hacking the code, as others
have pointed out. What you want is in res_features.c (if I understand
the code correctly), in the function called ast_bridge_call. On *
release 1.0.1 it's somewhere on line 520:
if (f (f-frametype == AST_FRAME_DTMF)) {
if (who == peer)
ast_write(chan, f);
else
ast_write(peer, f);
}
So you'd have to hack it by disabling commenting out that section. I
think this bit of code is only executed once the two legs of the call
are bridged, so it probably wouldn't affect anything else.
I also think that if you were at some point required to be able to send
DTMF after the initial dial pattern, you could programmatically via the
dialplan use the D option in the Dial application to send dtmf
digits.
Hope you do test this out before putting it live ;) Free advice, so
don't knock me out if it breaks something else!!
Flynn
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[Asterisk-Users] Grandstream BT100 - Does not recognize DTMF
One of my customers use Grandstream for ASTCC and it suddenly stopped recognizing DTMF for my ASTCC Application. When ASTCC asks to enter destination number, and when the the digits Are entered, the phone keys does not take any of them. They are dead. Any suggestions Seshu Kanuri NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Leg/Transaction Does Not Exist back
from 172.16.3.13
Date: Thu, 4 Nov 2004 18:55:30 -
MIME-Version: 1.0
Content-type: text/plain; charset=iso-8859-1
Content-Transfer-Encoding: quoted-printable
Hi all,
I hope someone can shed some light on the following: - I came across
a thread with a similiar problem but it didnt fix the problem=2E
I have two windows messenger clients registered with asterisk=2E When I
execute the command 'show sip peers', I can see them online=2E However
they cannot see each other and I am getting the following error:
-- Got SIP response 481 Call Leg/Transaction Does Not Exist back
from 172=2E16=2E3=2E13
-- Got SIP response 481 Call Leg/Transaction Does Not Exist
back from 172=2E16=2E3=2E17
I cannot see anything in 'show sip registry' and I'm wondering if
maybe that has anything got to do with it=2E=20
Anyway I read in a previous post from a person who had similiar
problems that if I change allow=3Dall to diallow=3Dall and allow=3Dulaw in=
the sip=2Econf file, the problem will be fixed=2E I also looked at the
OnLamp config site and followed their instructions there as this
person did=2E
However this is not solving the problem=2E I have included my config
files below and would be extremely grateful if anyone had any ideas
on the problem=2E
sip=2Econf
[general]
port =3D 5060 ; Port to bind to (SIP is 5060)
bindaddr =3D 0=2E0=2E0=2E0 ; Address to bind to (all addresses on machine)=
diallow=3Dall=20
allow=3Dulaw
context =3D from-sip ; Send SIP callers that we don't know about here
;register =3D 2000:[EMAIL PROTECTED]
[2000]
type=3Dfriend ; This device takes and makes calls
username=3D2000 ; Username on device
secret=3Dsuzuki ; Password for device
host=3Ddynamic ; This host is not on the same IP addr every
time
mailbox=3D100 ; Activate the message waiting light if this
; voicemailbox has messages in it
; Windows messenger client1
[2001] ; Duplicate of 2000, except with different auth
data
type=3Dfriend
username=3D2001
secret=3Dbla
host=3Ddynamic
mailbox=3D101
;Windows Messenger Client2
[odriscolla] ; Duplicate of 2000, except with different auth
data
type=3Dfriend
username=3Dodriscolla
secret=3Dpatoldham
host=3Ddynamic
mailbox=3D102
;extension=2Econf
[general]
static=3D3Dyes
writeprotect=3D3Dyes
[bogon-calls]
exten=3D3D =5F=2E,1,Congestion
[from-sip]
exten =3D3D odriscolla,1,Dial(SIP/odriscolla,20)
exten =3D3D odriscolla,2,Voicemail(uodriscolla)
exten =3D3D odriscolla,102,Voicemail(bodriscolla)
exten =3D3D odriscolla,103,Hangup
exten =3D3D 2001,1,Dial(SIP/2001,20)
exten =3D3D 2001,2,Voicemail(u2001)
exten =3D3D 2001,102,Voicemail(b2001)
exten =3D3D 2001,103,Hangup
exten=3D3D 2999,1,VoicemailMain(${CALLERIDNUM})
voicemail=2Econf
; voicemail=2Econf
[general]
format=3D3Dwav
[local]
odriscolla =3D3D 1234,aisling,[EMAIL PROTECTED]
2000 =3D3D 4321,julien,[EMAIL PROTECTED]
Cheers,
Aisling=2E
=20
=
---Legal Disclaimer-=
--
The above electronic mail transmission is confidential and intended only =
for the person to whom it is addressed. Its contents may be protected by =
legal and/or professional privilege. Should it be received by you in erro=
r please contact the sender at the above quoted email address. Any unauth=
orised form of reproduction of this message is strictly prohibited. The I=
nstitute does not guarantee the security of any information electronicall=
y transmitted and is not liable if the information contained in this comm=
unication is not a proper and complete record of the message as transmitt=
ed by the sender nor for any delay in its receipt.
-=
---=
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Re: [Asterisk-Users] Limit DTMF tones
Andrew Kohlsmith wrote: On November 4, 2004 12:30 pm, Henry Devito wrote: The issue is the inmates have figured out a way to dial long distance numbers by calling different private phone numbers and using that companies DISA to complete calls. So in order to stop that I have to suppress dtmf after so many digits are dialed. Any idea's? If you configured a SIP phone to not transmit inband DTMF, would asterisk translate that to inband DTMF when bridged to an inband DTMF only connection, ie your POTS line? Note: Just talking out of my head here, I've not actually tested this... In any case, chan_sip would be much more likely to be hackable to make DTMF quit working. -- Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Remote MWI (I know it's possible)
Hey Folks, I am trying to light a MWI located on a remote SIP phone. In other words, the phones register to one server but the voicemail app lives on a different one. I am guessing it has something to do with passing a user command in the voicemail.conf file. Of course I would also need to clear it... PLEASE let me know if anyone has any ideas. Thanks, Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Grandstream BT100 - Does not recognize DTMF
Maybe they switched to a codec that doesn't support inband DTMF and it isn't configured to use SIP INFO or likewise? -Michael -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kanuri, Seshu (Company IT) Sent: Thursday, November 04, 2004 12:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Grandstream BT100 - Does not recognize DTMF One of my customers use Grandstream for ASTCC and it suddenly stopped recognizing DTMF for my ASTCC Application. When ASTCC asks to enter destination number, and when the the digits Are entered, the phone keys does not take any of them. They are dead. Any suggestions Seshu Kanuri NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] Limit DTMF tones
On Thu, 2004-11-04 at 13:59 -0500, Andrew Thompson wrote: Andrew Kohlsmith wrote: On November 4, 2004 12:30 pm, Henry Devito wrote: The issue is the inmates have figured out a way to dial long distance numbers by calling different private phone numbers and using that companies DISA to complete calls. So in order to stop that I have to suppress dtmf after so many digits are dialed. Any idea's? If you configured a SIP phone to not transmit inband DTMF, would asterisk translate that to inband DTMF when bridged to an inband DTMF only connection, ie your POTS line? Depends on the codec if it would be able to detect and therefore squelch. Note: Just talking out of my head here, I've not actually tested this... In any case, chan_sip would be much more likely to be hackable to make DTMF quit working. As long as asterisk is looking for DTMF, and it is connected, the best place would be in the bridging where it is looking at the frames. As has been posted before, when you are reading the frames as they come in, you could just look at the frame type and decide whether it needed to be sent or acted upon. In this case, acted upon could be dropping it to the floor and replacing it with a silence frame of the proper duration. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Limit DTMF tones
On 11/4/2004, Andrew Thompson [EMAIL PROTECTED] wrote: In any case, chan_sip would be much more likely to be hackable to make DTMF quit working. Possibly, but his working configuration most likely doesn't use SIP (I would presume): It has the Digium 2 FXO/ 2 FXS card in it. I have two Lines brought in to the fxo ports and 2 standard 2500 analog sets for the prisoners to use to dial out. Flynn ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 it possible to use IAXY device to make 56K modem calls
On Thu, 2004-11-04 at 13:50 -0500, Jerry Geis wrote: ALL, is it possible to plug a standard analog 56K modem into my iaxy device and make a modem call out? 9600 baud call would be fine actually. I just want to make a call out with my iAXy device and eliminate my PSTN line. Depends on network quality. Not reliably and not fast. On a clean cable modem with a channel bank and T1 card I could occasionally get 14.4 connections but it would quickly drop to unusable as soon as either a frame was dropped or the jitter increased. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Call Leg/Transaction Does Not Exist back
On 11/4/2004, Ashling O'Driscoll [EMAIL PROTECTED] wrote: [general] port =3D 5060 ; Port to bind to (SIP is 5060) bindaddr =3D 0=2E0=2E0=2E0 ; Address to bind to (all addresses on machine)= diallow=3Dall=20 allow=3Dulaw context =3D from-sip ; Send SIP callers that we don't know about here ;register =3D 2000:[EMAIL PROTECTED] the HTML posting sort of screwed up the content of your email, but if i interpret it correctly it looks like you've got a line that says diallow=all shouldn't that be disallow=all Flynn ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 and PRIORITIES
is it possible after a meetme call to keep going on in the context, like the meet me is priority 2 I want it to hit priority 3 (after the party disconnects)? thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Limit DTMF tones
Flynn wrote: Possibly, but his working configuration most likely doesn't use SIP (I would presume): It has the Digium 2 FXO/ 2 FXS card in it. I have two Lines brought in to the fxo ports and 2 standard 2500 analog sets for the prisoners to use to dial out. Yeah, I saw that, but the replies I'd seen so far were not looking real promising, so I thought I'd throw out another idea. Even if the handsets were ruggedized, a Sipura could sit in between them and asterisk. Critchfield's response about the bridge code seems the place to look, but that's going to require coding and testing. If a SIP adapter could be dropped in and as a side effect of the configuration it broke sending DTMF out, only a few changes to the dialplan would be required to get things back in order. Anyway, it was just an idea, and he did say he was looking for ideas. -- Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Limit DTMF tones
Hi Flynn,
Feel free to contact me offline if you feel this isn't suitable
conversation for online but I read an article about this a few weeks ago
about how you were freezing out other carriers for offering cheaper
calls to inmates than the inflated prices you charged. And I don't
understand how you are legally allowed to do this.
It must be profit driven because surely I could call one of my approved
numbers eg sister and then have her pass along the information to a
third party.
Surely if you were looking to solve this 'isolation' issue and were
serious about it you would be tackling this problem another way.
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Flynn
Sent: Thursday, November 04, 2004 1:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Limit DTMF tones
On 11/4/2004, Henry Devito [EMAIL PROTECTED] wrote:
The issue is the inmates have figured out a way to dial long distance
numbers by calling different private phone numbers and using that
companies
DISA to complete calls. So in order to stop that I have to suppress
dtmf
after so many digits are dialed. Any idea's?
My, prisoners are getting devious :)
Anyways, you'd only be able to do this by hacking the code, as others
have pointed out. What you want is in res_features.c (if I understand
the code correctly), in the function called ast_bridge_call. On *
release 1.0.1 it's somewhere on line 520:
if (f (f-frametype == AST_FRAME_DTMF)) {
if (who == peer)
ast_write(chan, f);
else
ast_write(peer, f);
}
So you'd have to hack it by disabling commenting out that section. I
think this bit of code is only executed once the two legs of the call
are bridged, so it probably wouldn't affect anything else.
I also think that if you were at some point required to be able to send
DTMF after the initial dial pattern, you could programmatically via the
dialplan use the D option in the Dial application to send dtmf
digits.
Hope you do test this out before putting it live ;) Free advice, so
don't knock me out if it breaks something else!!
Flynn
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[Asterisk-Users] NAT with Linksys
Hi, I am living a wear thing. I am using my asterisk with all kind of NAT / PAT / NPAT, with multiple ports on the same network address and all works perfect. The problem I have been trying to solve is with a Cisco ATA behind a Linksys NAT. The ATA's register work ok, but when I execute sip show peers on the CLI I get a UNREACHABLE at the status column, and when I try to call it, the asterisk send me to his voicemail because is unreachable. But when the ATAs dial, the call work great. I have other ATAs behind other NATs (f.e. iptables) and they work perfectly. I have tried using the DMZ option on my linksys (putting the ATA private IP) but I stay having the problem. Could anyone give me a guide to search for a solution. I have been googleing but I could not find a solution. Thank you very much. Nahuel Ramos. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Call Leg/Transaction Does Not Exist
Hi, Thanks for the reply. Yes I had left out the 's'(as I had copied from the previous thread) but that is not the problem. I still have the 'call leg transaction does not exist' error. I have included the debug sip messages below if that will help any bit. I read that this error should have something got to do with a sip cancel message, an incorrect invite message or the to header. Since I am not inviting anyone and I dont cancel I dont think they apply. However I also think my 'to' header syntax is okso any ideas? Thanks again, Aisling. Sip read: REGISTER sip:172.16.3.15 SIP/2.0 Via: SIP/2.0/UDP 172.16.3.13:12568 Max-Forwards: 70 From: sip:[EMAIL PROTECTED];tag=4c0cbfbcc5414b0b91cbf6f08d1badcb;epid =f9d3957208 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 71 REGISTER Contact: sip:172.16.3.13:12568;methods=INVITE, MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK, REFER User-Agent: RTC/1.2.4949 (Messenger 5.0.0482) Event: registration Allow-Events: presence Content-Length: 0 12 headers, 0 lines Using latest request as basis request Sending to 172.16.3.13 : 12568 (non-NAT) Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.16.3.13:12568 From: sip:[EMAIL PROTECTED];tag=4c0cbfbcc5414b0b91cbf6f08d1badcb;epid =f9d3957208 To: sip:[EMAIL PROTECTED];tag=as0442c120 Call-ID: [EMAIL PROTECTED] CSeq: 71 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 172.16.3.13:12568 Transmitting (no NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 172.16.3.13:12568 From: sip:[EMAIL PROTECTED];tag=4c0cbfbcc5414b0b91cbf6f08d1badcb;epid =f9d3957208 To: sip:[EMAIL PROTECTED];tag=as0442c120 Call-ID: [EMAIL PROTECTED] CSeq: 71 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] WWW-Authenticate: Digest realm=asterisk, nonce=09fbe581 Content-Length: 0 to 172.16.3.13:12568 Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms Sip read: REGISTER sip:172.16.3.15 SIP/2.0 Via: SIP/2.0/UDP 172.16.3.13:12568 Max-Forwards: 70 From: sip:[EMAIL PROTECTED];tag=4c0cbfbcc5414b0b91cbf6f08d1badcb;epid =f9d3957208 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 72 REGISTER Contact: sip:172.16.3.13:12568;methods=INVITE, MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK, REFER User-Agent: RTC/1.2.4949 (Messenger 5.0.0482) Authorization: Digest username=odriscolla, realm=asterisk, algorithm=md5, uri=sip:172.16.3.15, nonce=09fbe581, response=488e7216327e85c4bc1976050ce81310 Event: registration Allow-Events: presence Content-Length: 0 13 headers, 0 lines Using latest request as basis request Sending to 172.16.3.13 : 12568 (non-NAT) Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.16.3.13:12568 From: sip:[EMAIL PROTECTED];tag=4c0cbfbcc5414b0b91cbf6f08d1badcb;epid =f9d3957208 To: sip:[EMAIL PROTECTED];tag=as0442c120 Call-ID: [EMAIL PROTECTED] CSeq: 72 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 172.16.3.13:12568 Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.3.13:12568 From: sip:[EMAIL PROTECTED];tag=4c0cbfbcc5414b0b91cbf6f08d1badcb;epid =f9d3957208 To: sip:[EMAIL PROTECTED];tag=as0442c120 Call-ID: [EMAIL PROTECTED] CSeq: 72 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 120 Contact: sip:172.16.3.13:12568;expires=120 Date: Thu, 04 Nov 2004 19:31:22 GMT Content-Length: 0 to 172.16.3.13:12568 Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms 11 headers, 2 lines Reliably Transmitting: NOTIFY sip:172.16.3.13:12568 SIP/2.0 Via: SIP/2.0/UDP 172.16.3.15:5060;branch=z9hG4bK2284d025 From: asterisk sip:[EMAIL PROTECTED];tag=as12bc656c To: sip:172.16.3.13:12568 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: application/simple-message-summary Content-Length: 38 Messages-Waiting: no Voicemail: 0/0 (no NAT) to 172.16.3.13:12568 Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms Sip read: SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP 172.16.3.15:5060;branch=z9hG4bK2284d025 From: asterisk sip:[EMAIL PROTECTED];tag=as12bc656c To: sip:172.16.3.13:12568;tag=a05ddee6260049778a66b59fb903130d Call-ID: [EMAIL PROTECTED] CSeq: 102 NOTIFY User-Agent: RTC/1.2 Content-Length: 0 8 headers, 0 lines -- Got SIP response 481 Call Leg/Transaction Does Not Exist back from 172.16.3.13 Destroying call '[EMAIL PROTECTED]' Destroying call '[EMAIL PROTECTED]' Original Message From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Call Leg/Transaction Does Not Exist back Date: Fri, 05 Nov 2004 03:16:13 +0800 On 11/4/2004, Ashling O'Driscoll [EMAIL PROTECTED] wrote: [general] port =3D 5060 ; Port to bind to (SIP is 5060) bindaddr
Re: [Asterisk-Users] Passing callerID info to a forwarded line
On Nov 4, 2004, at 10:28 AM, Chris Goodwin wrote:
Hi everyone,
I have a question regarding the use of callerID and call forwarding.
When I forward any of my Zap extensions in the office to an outside
line, such as a cell phone, the callerID info shows up as originating
from that office phone, rather than from whoever actually originated
the call into that office phone. Does anyone have an idea of how to
pass the callerID info of the originating caller to the forwarded
phone?
First, you need a connection to the PSTN that lets you set your own
caller ID value. POTS lines never have this. PRIs may, depending on
the configuration. Some VoIP providers let you control the caller ID
also; NuFone does, but I'm not sure about others.
Using NuFone, it's really easy--just send the call to them without
resetting the caller ID value that you received on the incoming call.
In my dial plan, I created a macro that calls SetCallerID with my own
phone number if and only if the existing caller ID value is 4 or fewer
digits long. Here's an example:
[macro-condsetcid]
exten = s,1,NoOp
exten = s,2,GotoIf($[${CALLERIDNUM:0:4} = ${CALLERIDNUM}]?3:4)
exten = s,3,SetCallerID(425488)
exten = s,4,NoOp
Then, in my outbound dialing context, I just do 'Macro(condsetcid)'
before doing 'Dial(${NUFONE}/${EXTEN})'. Works perfectly. There are a
couple more examples at
http://scottstuff.net/scott/archives/cat_asterisk.html
Scott
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[Asterisk-Users] Is it possible to use IAXY device to make 56Kmodem calls
steve, Thanks, do you recall what config commands you gave the modem to drop it down and only connect at lower speeds? I'm not a modem guru. Thanks, Jerry On Thu, 2004-11-04 at 13:50 -0500, Jerry Geis wrote: / ALL, // // is it possible to plug a standard analog 56K modem into my // iaxy device and make a modem call out? 9600 baud call would // be fine actually. I just want to make a call out with my iAXy // device and eliminate my PSTN line. / Depends on network quality. Not reliably and not fast. On a clean cable modem with a channel bank and T1 card I could occasionally get 14.4 connections but it would quickly drop to unusable as soon as either a frame was dropped or the jitter increased. -- Steven Critchfield critch at basesys.com http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
