Re: [asterisk-users] Top Posting
On 01/17/2011 10:31 AM, Mark Murawski wrote: On 01/16/2011 10:28 PM, Mark Murawski wrote: We obviously have all our own opinions about being on top or bottom. And it boils down to personal preference obviously. And it looks like I top posted, heh. I just usually hit reply and start typing, the default is top. I guess I go both ways. :P Hi, I thought this kind of discussion didn't exists in asterisk list. I guess most tech list members will argue on top-vs-bottom subject at some point :) anton -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fast busy out?
On 09/04/2010 08:40 PM, Thomas Perron wrote: why does this not work? i simply want to hear the recorded message exten = s,1,Answer() ;exten = s,n,Record(zipcodegutter1.gsm) ;zcg1 exten = s,n,Playback(zipcodegutter1) exten = s,n,Dial(SIP/c01s/159,120,A,(demo-thanks)) hi, try to put exten = s,n,Playback(silence/1) after Answer(), before your actual playback anton -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] channel stay up when extension unreachable
On 08/24/2010 01:13 AM, Anton Raharja wrote: === electricity down in 801's room and 801 became unreachable: [Aug 20 14:46:45] NOTICE[8052] chan_sip.c: Peer '801' is now UNREACHABLE! Last qualify: 7 === after 25 minutes power restored and 801 re-registered. 801 continue testing, dialed several other destinations, also dialed *43 several times. He didn't noticed any suspicious log and didn't bother to check it coz 801 worked, calls were made and seems to be completed normally. [Aug 20 15:13:04] VERBOSE[8052] logger.c: -- Registered SIP '801' at xxx.xxx.xxx.xxx port 1806 [Aug 20 15:13:04] VERBOSE[8052] logger.c: -- Saved useragent X-Lite release 1104o stamp 56125 for peer 801 [Aug 20 15:13:04] NOTICE[8052] chan_sip.c: Peer '801' is now Reachable. (17ms / 2000ms) === tonight, in our server, I noticed that I have a channel associated with 801 elapsed for at least 81 hours after a core show channels, while theres no way 801 still available or making that call Hi, problem solved, rtptimeout option fix this. anton -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] channel stay up when extension unreachable
Hi, We are using asterisk 1.4.34, ubuntu 10.4, below is suspicious activity recorded in our full log. Could you help us to explain what had happened. Thanks. === my friend, 801, from his room did a test by dialing echo test in freepbx, *43: [Aug 20 14:42:46] VERBOSE[14427] logger.c: -- Executing [...@from-internal:1] Answer(SIP/801-03f5, ) in new stack [Aug 20 14:42:46] VERBOSE[14427] logger.c: -- Executing [...@from-internal:2] Wait(SIP/801-03f5, 1) in new stack [Aug 20 14:42:47] VERBOSE[14427] logger.c: -- Executing [...@from-internal:3] Playback(SIP/801-03f5, demo-echotest) in new stack [Aug 20 14:42:47] VERBOSE[14427] logger.c: -- SIP/801-03f5 Playing 'demo-echotest' (language 'en') [Aug 20 14:43:07] VERBOSE[14427] logger.c: -- Executing [...@from-internal:4] Echo(SIP/801-03f5, ) in new stack === electricity down in 801's room and 801 became unreachable: [Aug 20 14:46:45] NOTICE[8052] chan_sip.c: Peer '801' is now UNREACHABLE! Last qualify: 7 === after 25 minutes power restored and 801 re-registered. 801 continue testing, dialed several other destinations, also dialed *43 several times. He didn't noticed any suspicious log and didn't bother to check it coz 801 worked, calls were made and seems to be completed normally. [Aug 20 15:13:04] VERBOSE[8052] logger.c: -- Registered SIP '801' at xxx.xxx.xxx.xxx port 1806 [Aug 20 15:13:04] VERBOSE[8052] logger.c: -- Saved useragent X-Lite release 1104o stamp 56125 for peer 801 [Aug 20 15:13:04] NOTICE[8052] chan_sip.c: Peer '801' is now Reachable. (17ms / 2000ms) === tonight, in our server, I noticed that I have a channel associated with 801 elapsed for at least 81 hours after a core show channels, while theres no way 801 still available or making that call === later on after soft hangup: [Aug 23 23:52:01] VERBOSE[14427] logger.c: == Spawn extension (from-internal, *43, 4) exited non-zero on 'SIP/801-03f5' [Aug 23 23:52:01] VERBOSE[14427] logger.c: -- Executing [...@from-internal:1] Macro(SIP/801-03f5, hangupcall) in new stack [Aug 23 23:52:01] VERBOSE[14427] logger.c: -- Executing [...@macro-hangupcall:1] ResetCDR(SIP/801-03f5, w) in new stack [Aug 23 23:52:01] DEBUG[14427] app_macro.c: Executed application: ResetCDR [Aug 23 23:52:01] VERBOSE[14427] logger.c: -- Executing [...@macro-hangupcall:2] NoCDR(SIP/801-03f5, ) in new stack [Aug 23 23:52:01] DEBUG[14427] app_macro.c: Executed application: NoCDR [Aug 23 23:52:01] VERBOSE[14427] logger.c: -- Executing [...@macro-hangupcall:3] GotoIf(SIP/801-03f5, 1?skiprg) in new stack [Aug 23 23:52:01] VERBOSE[14427] logger.c: -- Goto (macro-hangupcall,s,6) [Aug 23 23:52:01] DEBUG[14427] app_macro.c: Executed application: GotoIf [Aug 23 23:52:01] VERBOSE[14427] logger.c: -- Executing [...@macro-hangupcall:6] GotoIf(SIP/801-03f5, 1?skipblkvm) in new stack [Aug 23 23:52:01] VERBOSE[14427] logger.c: -- Goto (macro-hangupcall,s,9) [Aug 23 23:52:01] DEBUG[14427] app_macro.c: Executed application: GotoIf [Aug 23 23:52:01] VERBOSE[14427] logger.c: -- Executing [...@macro-hangupcall:9] GotoIf(SIP/801-03f5, 1?theend) in new stack [Aug 23 23:52:01] VERBOSE[14427] logger.c: -- Goto (macro-hangupcall,s,11) [Aug 23 23:52:01] DEBUG[14427] app_macro.c: Executed application: GotoIf [Aug 23 23:52:01] VERBOSE[14427] logger.c: -- Executing [...@macro-hangupcall:11] Hangup(SIP/801-03f5, ) in new stack [Aug 23 23:52:01] VERBOSE[14427] logger.c: == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/801-03f5' in macro 'hangupcall' [Aug 23 23:52:01] VERBOSE[14427] logger.c: == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/801-03f5' anton -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sending sms from Asterisk server
On 08/19/2010 08:21 AM, Tiago Geada wrote:
I would rather use .call files. So easy to produce a text file...
On 18 August 2010 21:02, Steve Edwards asterisk.org
http://asterisk.org@sedwards.com http://sedwards.com wrote:
Un-top-posting...
On 08/17/2010 09:00 AM, Tino wrote:
I would like to send sms to some external phone numbers from
my asterisk server. Is it possible to send sms via softphones
like X-Lite ? . Any tips regarding this will be helpful
On Wed, Aug 18, 2010 at 3:13 AM, Johann Hoehn
johann.ho...@ecommerce.com
mailto:johann.ho...@ecommerce.com wrote:
This is easy to do by using email to SMS gateways. A list of
them is on wikipedia
(http://en.wikipedia.org/wiki/List_of_SMS_gateways). For the
Asterisk side, you have an extension that sends the email. I
personally use an AGI script for this part, but you could use
a System() call as well.
Using system() is almost always a hack -- and not the good kind :)
On Wed, 18 Aug 2010, Tino wrote:
Thanks for your advice in this matter. But i am not sure how
to pass the numbers to be sent sms in the dialplan.
You have a choice: you can pass them as channel variables or as
command line options. I use both, frequently in the same program.
Unfortunately, I can't clearly articulate why I use one over the
other. If the variable is something that exists for the life of
the call like ${CLIENT-ID} I tend to access it as a channel
variable. If it's something that modifies the behavior of the AGI
(--debug or --verbose) I always pass it as a command line option
and use getopt_long()
First, you need to pick a language. If this is a SOHOish hobby
project, it doesn't matter -- pick a language you are comfortable
with.
If this is a high volume, performance critical project -- I'd vote
for c.
Once you've decided on a language, search out an established AGI
library and learn a bit about the protocol. It's very simple but
not always obvious. The 3 biggest stumbling blocks that trip up
programmers are:
1) You have to read the AGI environment before anything else.
2) It's a request followed by a response. If you don't read the
response, bad things will happen.
3) It's STDIN/STDOUT based. If you try to debug by writing
variables or messages using echo/printf/puts/etc, bad things will
happen.
Check out voip-info.org http://voip-info.org for more
information on AGI.
Hi,
how do you get the text to send?
text that is sent from X-Lite for example.
thx,
anton
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] E1 not synchronized
Hello, We're using OpenVox D410 to connect Asterisk 1.4.21.2 box to E1 line from local telco operator. Once in a while we experienced lines always busy. I'm not sure but reported as sometime on making outgoing calls only, sometime both outgoing and incoming. Rebooting (not just restarting asterisk) solved the problem, lines become available and everything works as previous. Engineers at local telco operator said that between their equipment and our box is not synchronized. They gave us a clue how to solve it: CRC4=ON SENDING DIGIT=FULL CENTRAL EWSD=NET.3 Any information appreciated. Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk with IM
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Mochamad Susantok wrote: Hi all, Howto configure asterisk 1.2.13 (debian-base) with support Instant Messaging, especially using client Xlite v.3. Thanks Hello, Im using my patched chan_sip.c for that. http://www.voiprakyat.or.id/download/server/asterisk/sip-messaging/1.2.13/ anton -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (MingW32) iD8DBQFFffXU5ByPs8h3tvwRAtvrAJ4+otMwOEdohO6acrLgdPPuBPuZRwCgv3Up IPheq/tk8dV5eCmK7hVbJro= =vrNg -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous registry
in other SIP proxy server, this can be done easily, i mean its default 1 or more phone could be registered at 1 number (12345) and resulting same effect as u ask SER (SIP Express Router, http://iptel.org/ser) can deal with this SER is a friend to asterisk, i think :), you can accept calls with SER and pass it to asterisk to process complex dialplan but if this feature implemented in asterisk alone, it would be nice *** REPLY SEPARATOR *** On 11/07/2004 at 6:00 Kannaiyan Natesan wrote: Paul, The question is very simple. When I call a SIP user, the phone should ring in more than one extentions. Also more than one phone should be able to register with asterisk. Right now it is not the case. The last phone which register will be receiving the calls. This type of situations might be needed in call centres. Called 12345 |---(12345) Ringing |---(12345) Ringing |---(12345) Ringing So you don't need to disturb asterisk when you add more devices to it to receive calls. Such facility is not available in asterisk at this moment. I hope this helps. Since I feel this is a great feature, I will topup up to $100/- -.Kannaiyan http://www.goods2world.com -- Your Only VoIP - Original Message - From: Paul Mahler [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, July 11, 2004 5:44 AM Subject: RE: [Asterisk-Users] New Asterisk bounty: SIP simultaneous registry I'm not sure I understand what you are trying to do. You have an administrative assistant and several other staff. You want the administrator to be able to take calls directed to the staff extensions? If I have the requirement right, you could accomplish this by ringing the staff extension and the admin extension at the same time. The Dial command allows you to ring multiple extensions simultaneously. If you want to be able to more easily recognize what extension the traffic if for, you can add additional extensions to the 7960. For example, if you have two staff the admin monitors, add two additional extensions to the 7960. The admin can tell who is being called by the extension that rings. Paul Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Daniel Jimenez Sent: Saturday, July 10, 2004 3:05 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous registry http://voip-info.org/tiki-index.php?page=Asterisk+bounty+SIP+s imultaneous+registry Updated, Allow a SIP device to register more than once so a single extension may exist in multiple locations. Upped total to $75. Daniel... Daniel Jimenez wrote: http://voip-info.org/tiki-index.php?page=Asterisk+bounty+SIP+simultane ous+registry From the WIKI: Contributions Manager: Daniel Jimenez (cuban) Bounty: $50 USD Date opened: July 10, 2004 Contributors: cuban ($50) Detail Yes, Yes I know you could do all sorts of fun with the dialplan to produce a similar effect, but I still would like to be able to do this. Plus it's easy money :). I have some users with a 7960 who are administrative assistants who monitor calls for 3 or 4 other people. It'd be nice to have two line instances for them, and one for the person(s) whom they assist. Contact me: djimenez at pobox.com if you're interested in making this happen. -- Daniel Jimenez djimenez[at]pobox[dot]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 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 http://sleepless.ngoprek.org VoIP Rakyat: (0921) 20006 ___ 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] xlite calls not approved
asterisk 0.9.1 with regular sip.conf and extensions.conf sjPhone able to register and make calls xlite said logged in but when i start to call/dial it said calls not approved n i dont see anything while my asterisk sip debug enabled can anyone give me a clue whats happening? http://sleepless.ngoprek.org VoIP Rakyat: (0921) 20006 ___ 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] xlite calls not approved
ok, this is my sip.conf xlite cant calls, sjPhone can i wish sjPhone dont hav that popup thing :) [general] port = 5060 bindaddr = 0.0.0.0 context = intern tos=lowdelay videosupport=yes disallow=all allow=gsm allow=ulaw allow=alaw register = sleepless:pwd:[EMAIL PROTECTED] [voiprakyat.net] type=peer context=intern username=sleepless secret=pwd host=voiprakyat.net nat=yes canreinvite=no [1234] type=friend context=intern username=1234 secret=pwd host=dynamic nat=yes canreinvite=yes [5678] type=friend context=intern username=5678 secret=pwd host=dynamic nat=yes canreinvite=yes *** REPLY SEPARATOR *** On 09/07/2004 at 14:29 Jay Milk wrote: Show us your sip.conf -- probably a config issue -Original Message- From: R. Anton Raharja [mailto:[EMAIL PROTECTED] Sent: Friday, July 09, 2004 1:16 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] xlite calls not approved asterisk 0.9.1 with regular sip.conf and extensions.conf sjPhone able to register and make calls xlite said logged in but when i start to call/dial it said calls not approved n i dont see anything while my asterisk sip debug enabled can anyone give me a clue whats happening? http://sleepless.ngoprek.org VoIP Rakyat: (0921) 20006 ___ 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] xlite calls not approved
now tht i guess your dtmf problem fixed too, mind to tell us (or me) wht u've done to fix xlite call not approved problem? *** REPLY SEPARATOR *** On 09/07/2004 at 13:15 CHS wrote: ok, I've finally got it working. I can get to the demo extension '1000' and I hear the voice, etc.. only one problem, I can't seem to hit any of the demo extensions (like 2 for more detailed info, etc..) http://sleepless.ngoprek.org VoIP Rakyat: (0921) 20006 ___ 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] voicetronix n asterisk
im sure this is the perfect place to ask bout asterisk http://www.voicetronix.com/openpbx.htm + asterisk is this a good solution for VoIP on private network (6 office, each has their own existing PBX) good means relatively cheap, stable n reliable thx ___ 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
