Re: [asterisk-users] [OpenSIPS-Users] Asterisk is not designed for University with largeuser base?
Hi Yehavi, As Alex said, it depends of what exactly you want to implement. You just have to evaluate your target service and to properly understand what each piece of software is appropriate for and what it has to offer. First of all, you have 2 complementary classes of software : you have softswitches and and you have PBX - you have large capacity softswitches for 100K subscribers with no media support (like OpenSER/OpenSIPS) and you have PBX-like software with advanced and complex media capabilities. (like Asterisk). There is no sigle software to give the magic complete solution - so, far the combination of the two types (opensips + asterisk) proved to be a good solution to covers all needs and all requirements of a complex solution. But again, it is up to what you are looking for (as voip platform) Regards, Bogdan Yehavi Bourvine wrote: Hello, After a long time we had a meeting with our university's management and got a green light to have a proof of concept with open source telephony. Now I have to select the right software to experiment with... Up to now I thought of going with OpenSER for the masses and Asterisk for voicemail and other media related things. However, from reading around it seems like FreeSwitch can give me the benefits of both packages. Anyone has an experience with it? Thanks, __Yehavi: ___ Users mailing list us...@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ -- 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
[asterisk-users] opensips and asterisk canreinvite
Hi,
I'm using opensips as the registrar server for my users.
I am redirecting calls going out to pstn to my asterisk server.
call flow is basically:
ua -- opensips server -- * server -- sip gateway provider
if (uri=~sip:00[0-...@sip\.myserver\.com) {
xlog(L_INFO, Call to PSTN\n);
#strip(2);
#prefix(011);
rewritehostport(20.21.22.23:6050); --- IP and Port of * Server
route(1);
exit;
}
call routing works properly, but i would like for the rtp not to go thru
asterisk, i'm using the canreinvite option, but when i try to make a
call, rtp debug still sees rtp passing thru the asterisk.
Sent RTP packet to 12.34.56.78:16410 (type 18, seq 063687, ts 035408,
len 20)
Got RTP packet from 87.65.43.21:21376 (type 18, seq 000310, ts 074400,
len 30)
Sent RTP packet to 12.34.56.78:16410 (type 18, seq 063688, ts 035568,
len 20)
Got RTP packet from 87.65.43.21:21376 (type 18, seq 000311, ts 074640,
len 30)
12.34.56.78 public IP of the UA, 87.65.43.21 IP of the SIP gw provider.
note: opensips and asterisk are on the same box.
i apologize in advance as i'm not sure if i'm sending it on the correct
list.
regards,
nhadie
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Re: [asterisk-users] async agi question
Hi Moy, thanks a lot for your fix, but I'm afraid it doesn't work. I looked your patch over and I realize the code never passes by neither of the two lines you added with returnstatus = AGI_RESULT_HANGUP. Even, it seems the execution doesn't pass by res_agi.c at all, or at least, it doesn't pass over any ast_log(LOG_DEBUG,... lines like the ones your last patch has above the returnstatus fix. Could be the execution is flowing down by an if - else - break without an ast_log(LOG_DEBUG,... line? In that case, would the returnstatus = AGI_RESULT_HANGUP be added to any places more? Below is the output log for the redirect while playing a file. As you can see, there isn't any res_agi.c output on it: [Apr 13 11:20:09] DEBUG[5804]: manager.c:2108 process_message: Manager received command 'Redirect' [Apr 13 11:20:09] DEBUG[5804]: channel.c:1378 ast_softhangup_nolock: Soft-Hanging up channel 'SIP/501-08287a00' [Apr 13 11:20:09] DEBUG[5815]: channel.c:1793 ast_settimeout: Scheduling timer at 0 sample intervals [Apr 13 11:20:09] DEBUG[5815]: pbx.c:2448 __ast_pbx_run: Extension 801, priority 0 returned normally even though call was hung up [Apr 13 11:20:09] DEBUG[5815]: channel.c:1378 ast_softhangup_nolock: Soft-Hanging up channel 'SIP/501-08287a00' [Apr 13 11:20:09] DEBUG[5815]: channel.c:1477 ast_hangup: Hanging up channel 'SIP/501-08287a00' [Apr 13 11:20:09] DEBUG[5815]: chan_sip.c:3485 sip_hangup: Hangup call SIP/501-08287a00, SIP callid 2dbe6797392cde921fb7db0b16e81...@10.0.5.20) However, if the redirect is done without playing a file, the execution does pass by res_agi.c: [Apr 13 12:03:57] DEBUG[2688]: manager.c:2108 process_message: Manager received command 'Redirect' [Apr 13 12:03:57] DEBUG[2688]: channel.c:1378 ast_softhangup_nolock: Soft-Hanging up channel 'SIP/501-08279028' [Apr 13 12:03:57] DEBUG[2755]: res_agi.c:464 launch_asyncagi: No frame read on channel SIP/501-08279028, going out ... [Apr 13 12:03:57] DEBUG[2755]: pbx.c:2427 __ast_pbx_run: Spawn extension (sip_sercom,500,0) exited non-zero on 'SIP/501-08279028' [Apr 13 12:03:57] == Spawn extension (sip_sercom, 500, 0) exited non-zero on 'SIP/501-08279028' By the way, there's another thing puzzling me: Due you said this AsyncAGI patch was done for asterisk 1.6 and not for asterisk 1.4, and Henrik Westerbeg said it had worked for it as well, (please see: http://lists.digium.com/pipermail/asterisk-users/2008-December/223009.html) then I looked over the last releases at http://bugs.digium.com/bug_view_advanced_page.php?bug_id=11282 for that AsyncAGI patch and I was able to see neither of them have the returnstatus = AGI_RESULT_HANGUP either, however, ¡they work! (as Henrik said). As you can see, I'm a bit confusing about this subject. I would thank you If you can give any guidelines about it in order to be able to investigate deeper and move forward. Thank you very much for your help Jose M Arias -- Moises Silva wrote : It's a bug in the Async AGI feature. I have created a new patch http://www.moythreads.com/asterisk-1.4.18-async-agi.patch Please test it and let me know if it works for you, -- This message was sent on behalf of cyr2...@gmail.com at openSubscriber.com http://www.opensubscriber.com/message/asterisk-users@lists.digium.com/11874935.html ___ -- 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
[asterisk-users] Sending Re-Invite with Dialplan application?
Hi, I have a requirement where an IVR application on asterisk has to play a audio file in g729 and when a digit is pressed, the call should switch to another codec (say ulaw). So, What can I do in the extensions.conf to trigger a re-negotiation of codec? I used exten = 55xx,n,Set(SIP_CODEC=ulaw) but, I suppose this affects the next call and not the current one. Please help ASAP Thanks, Sai Check out the all-new Messenger 9.0! Go to http://in.messenger.yahoo.com/___ -- 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
[asterisk-users] Send Re-invite from Dialplan application?
Hi, I have a requirement where an IVR application on asterisk has to play a audio file in g729 and when a digit is pressed, the call should switch to another codec (say ulaw). So, What can I do in the extensions.conf to trigger a re-negotiation of codec? I used exten = 55xx,n,Set(SIP_CODEC=ulaw) but, I suppose this affects the next call and not the current one. Please help ASAP Thanks, Sai Did you know? You can CHAT without downloading messenger. Go to http://in.webmessenger.yahoo.com/___ -- 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
[asterisk-users] Clock problem with TE122
Hello, I am using TE122 between an ericsson MD110 and asterisk server. I set on ericsson side as master PRI NET and Asterisk is PRI CPE Even i connect jumper on external clock side(on TE122).. or even my configuration is as span = 1,1,0,ccs,hdb3 i still see my TE122 as Internally Clocked on dahdi_tools.. I want my PRI card to get clocking signal from the ericsson side.. How can i do it? Thanks ___ -- 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] FAX reliability
Hi Steve, Steve Underwood wrote: In chan_dahdi.c there is now code that extends the buffering inside dadhi when a FAX is detected, and puts the buffering back to normal at the end. This isn't really a cure - its more of a bandaid. However, I expect it has the desired effect if they have put it into the trunk code. You need to enable this feature in chan_dahdi.conf. Very interestingly, many people who have problems sending from app_txfax or app_fax have no problem sending from iaxmodem + HylaFAX. It seems Asterisk can feed a zaptel/dahdi channel much more reliably when passing a signal through, than when generating on in an app. So are the chan_dahdi.c developments mentioned of any practical value to iaxmodem users? Thanks, Lee. ___ -- 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] FAX reliability
Hi Lee, Lee Howard wrote: Hi Steve, Steve Underwood wrote: In chan_dahdi.c there is now code that extends the buffering inside dadhi when a FAX is detected, and puts the buffering back to normal at the end. This isn't really a cure - its more of a bandaid. However, I expect it has the desired effect if they have put it into the trunk code. You need to enable this feature in chan_dahdi.conf. Very interestingly, many people who have problems sending from app_txfax or app_fax have no problem sending from iaxmodem + HylaFAX. It seems Asterisk can feed a zaptel/dahdi channel much more reliably when passing a signal through, than when generating on in an app. So are the chan_dahdi.c developments mentioned of any practical value to iaxmodem users? That's a good question, and I have no idea about the answer. Some people who have had problems sending from app_txfax say iaxmodem + HylaFAX works OK on the same machine. This seems strange, as you might expect a problem in scheduling I/O would affect passthrough as well as applications. What I don't know is whether they just have a lot less trouble with iaxmodem, or they have no trouble at all. Do you get reports from people who say receive is stable, but transmit is not? Steve ___ -- 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] Followme for multiple persons?
You could use this on-call script to go until you got an acceptance
exten = s,501,Set(ONECELL=${DB(Cell/One)})
exten = s,n,Set(TWOCELL=${DB(Cell/Two)})
exten = s,n,Set(THREECELL=${DB(Cell/Three)})
exten = s,n,Macro(calleng,${TWOCELL},1)
exten = s,n,Macro(calleng,${ONECELL},2)
exten = s,n,Macro(calleng,${THREECELL},3)
exten = s,n,VoiceMail(1...@default)
exten = s,n,Background(vm-goodbye)
exten = s,n,Hangup()
[macro-calleng]
exten = s,1,Background(please-wait-connect-oncall-eng)
exten = s,n,Background(number)
exten = s,n,SayDigits(${ARG2})
exten = s,n,Dial(Zap/g1/${DELAY}${ARG1},30)
You load the values of the numbers to call in the Asterisk DB and the call
goes in 30 second chunks until someone answers or voicemail Is reached.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of JD
Sent: Friday, April 10, 2009 5:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Followme for multiple persons?
I've got a challenge (or clarification request if I am mistaken) for the
group.
I have a non-profit customer on asterisk 1.4 that has multiple
volunteers that work from home. The volunteers are willing to take calls
to help out the organization.
So, a formal queue is out. They don't want their home phones or cell
phones to blindly send them callers. They want to take calls when/if
they happen to be free to take a call at that particular moment. Plus,
the queue function can't handle the roll to voicemail problem that all
cell phones have. Plus, they won't have the discipline to log-in/log-out.
Fine, I thought, I'll just use the followme function in Asterisk 1.4. It
rings four numbers at once. It asks the friendly screening question,
allowing a volunteer to press 1 to take the call. Or, they hang up and
perhaps someone else will take it. (Or, if nobody does, it goes to
voicemail.)
Fine and dandy. Or so I thought. The problem is that followme is
designed to assume that it is only going to reach exactly one person.
So, if a phone answers and they press 2 to reject the call: bam,
asterisk stops trying the other three phone numbers.
I am currently trying to educate the volunteers to refrain from
pressing 2 but that is prone to problems. I'd rather that there not be
a reject function at all. Or, making it so that pressing 2 doesn't
really reject the call, it just hangs it up. I could change the audio,
and remap 2 to 9 and hope nobody presses it, but that seems like an
accident waiting to happen.
Does anyone have suggestions?
John
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[asterisk-users] Agents on asterisk
Hi! I have a question about agents in asterisk. In first place, agent login to asterisk (from the telephone) The question is: Can an agent take a break (using a function *(some number)) from the phone? Thanks to all Regards -- Ing Francisco Roqué 3Tech SRL Plaza Paso Nº92, EP B Buenos Aires, Argentina. www.3tech.com.ar ___ -- 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] Followme for multiple persons?
JD wrote:
I've got a challenge (or clarification request if I am mistaken) for the
group.
I have a non-profit customer on asterisk 1.4 that has multiple
volunteers that work from home. The volunteers are willing to take calls
to help out the organization.
So, a formal queue is out. They don't want their home phones or cell
phones to blindly send them callers. They want to take calls when/if
they happen to be free to take a call at that particular moment. Plus,
the queue function can't handle the roll to voicemail problem that all
cell phones have. Plus, they won't have the discipline to log-in/log-out.
Fine, I thought, I'll just use the followme function in Asterisk 1.4. It
rings four numbers at once. It asks the friendly screening question,
allowing a volunteer to press 1 to take the call. Or, they hang up and
perhaps someone else will take it. (Or, if nobody does, it goes to
voicemail.)
Fine and dandy. Or so I thought. The problem is that followme is
designed to assume that it is only going to reach exactly one person.
So, if a phone answers and they press 2 to reject the call: bam,
asterisk stops trying the other three phone numbers.
I am currently trying to educate the volunteers to refrain from
pressing 2 but that is prone to problems. I'd rather that there not be
a reject function at all. Or, making it so that pressing 2 doesn't
really reject the call, it just hangs it up. I could change the audio,
and remap 2 to 9 and hope nobody presses it, but that seems like an
accident waiting to happen.
Does anyone have suggestions?
John
I think you can make the following code mod to have the next in dial plan
step not do anything. If someone has the time, this would probably be a decent
option to add to the application for future versions to make this behavior
optional via an application option parameter.
[r...@btwtechshowdemoc apps]# svn diff app_followme.c
Index: app_followme.c
===
--- app_followme.c (revision 188040)
+++ app_followme.c (working copy)
@@ -724,9 +724,9 @@
if
(!strcmp(tmpuser-yn, tpargs-nextindp)) {
if
(option_debug)
ast_log(LOG_DEBUG, Next in dial plan step requested.\n);
- *status = 1;
+ // *status = 1;
ast_frfree(f);
- return NULL;
+ // return NULL;
}
}
--
--
Bird's The Word Technologies, Inc.
http://www.btwtech.com/
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Re: [asterisk-users] FAX reliability
Steve Underwood wrote: Hi Lee, Lee Howard wrote: Hi Steve, Steve Underwood wrote: In chan_dahdi.c there is now code that extends the buffering inside dadhi when a FAX is detected, and puts the buffering back to normal at the end. This isn't really a cure - its more of a bandaid. However, I expect it has the desired effect if they have put it into the trunk code. You need to enable this feature in chan_dahdi.conf. Very interestingly, many people who have problems sending from app_txfax or app_fax have no problem sending from iaxmodem + HylaFAX. It seems Asterisk can feed a zaptel/dahdi channel much more reliably when passing a signal through, than when generating on in an app. So are the chan_dahdi.c developments mentioned of any practical value to iaxmodem users? That's a good question, and I have no idea about the answer. Some people who have had problems sending from app_txfax say iaxmodem + HylaFAX works OK on the same machine. This seems strange, as you might expect a problem in scheduling I/O would affect passthrough as well as applications. What I don't know is whether they just have a lot less trouble with iaxmodem, or they have no trouble at all. Do you get reports from people who say receive is stable, but transmit is not? There have been a few reports of that situation, but all of them that I recall had to do with people who were trying to use VoIP for fax (HylaFAX and iaxmodem tend to be much more tolerant of the audio cut-outs caused by jitter than are other receivers). Otherwise, no, I don't have any of those situations that I can point out. I suppose that I'd need to make some recordings to say for certain whether audio cut-outs were occurring. However, what would really be nice would be to see some comments come from whatever developer at Digium made those chan_dahdi.c modifications. Thanks, Lee. ___ -- 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] Agents on asterisk
Thanks James. I read it. But the cmd PauseQueueMember must be executed from an extension. As soon as the agent logged in, asterisk does not recognize the dtmf. There can be an alternative solution? Regards Francisco james.coll...@xtratelecom.es wrote: You can use the PauseQueueMember command. http://www.voip-info.org/wiki/view/Asterisk+cmd+PauseQueueMember Saludos, -James ROQUÉ, Francisco Emiliano wrote: Hi! I have a question about agents in asterisk. In first place, agent login to asterisk (from the telephone) The question is: Can an agent take a break (using a function *(some number)) from the phone? Thanks to all Regards ___ -- 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] Followme for multiple persons?
BJ Weschke wrote:
JD wrote:
I've got a challenge (or clarification request if I am mistaken) for the
group.
I have a non-profit customer on asterisk 1.4 that has multiple
volunteers that work from home. The volunteers are willing to take calls
to help out the organization.
So, a formal queue is out. They don't want their home phones or cell
phones to blindly send them callers. They want to take calls when/if
they happen to be free to take a call at that particular moment. Plus,
the queue function can't handle the roll to voicemail problem that all
cell phones have. Plus, they won't have the discipline to log-in/log-out.
Fine, I thought, I'll just use the followme function in Asterisk 1.4. It
rings four numbers at once. It asks the friendly screening question,
allowing a volunteer to press 1 to take the call. Or, they hang up and
perhaps someone else will take it. (Or, if nobody does, it goes to
voicemail.)
Fine and dandy. Or so I thought. The problem is that followme is
designed to assume that it is only going to reach exactly one person.
So, if a phone answers and they press 2 to reject the call: bam,
asterisk stops trying the other three phone numbers.
I am currently trying to educate the volunteers to refrain from
pressing 2 but that is prone to problems. I'd rather that there not be
a reject function at all. Or, making it so that pressing 2 doesn't
really reject the call, it just hangs it up. I could change the audio,
and remap 2 to 9 and hope nobody presses it, but that seems like an
accident waiting to happen.
Does anyone have suggestions?
John
I think you can make the following code mod to have the next in dial plan
step not do anything. If someone has the time, this would probably be a
decent option to add to the application for future versions to make this
behavior optional via an application option parameter.
[r...@btwtechshowdemoc apps]# svn diff app_followme.c
Index: app_followme.c
===
--- app_followme.c (revision 188040)
+++ app_followme.c (working copy)
@@ -724,9 +724,9 @@
if
(!strcmp(tmpuser-yn, tpargs-nextindp)) {
if
(option_debug)
ast_log(LOG_DEBUG, Next in dial plan step requested.\n);
- *status = 1;
+ // *status =
1;
ast_frfree(f);
- return NULL;
+ // return
NULL;
}
}
Thanks for the code. Excellent idea.
In this particular case, I can't use it as-is because I run many virtual
PBXs on the same machine. I don't want to change the behavior for all
followme, just this one customer. So, your suggestion of making it an
application option would be spot on.
I am a programmer, but I've never done open source stuff before. Is
there a quick readable 1-2-3 guide on submitting code changes to the
community?
John
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Re: [asterisk-users] Followme for multiple persons?
On Apr 13, 2009, at 11:23 AM, JD wrote:
BJ Weschke wrote:
JD wrote:
I've got a challenge (or clarification request if I am mistaken)
for the
group.
I have a non-profit customer on asterisk 1.4 that has multiple
volunteers that work from home. The volunteers are willing to take
calls
to help out the organization.
So, a formal queue is out. They don't want their home phones or cell
phones to blindly send them callers. They want to take calls when/if
they happen to be free to take a call at that particular moment.
Plus,
the queue function can't handle the roll to voicemail problem
that all
cell phones have. Plus, they won't have the discipline to log-in/
log-out.
Fine, I thought, I'll just use the followme function in Asterisk
1.4. It
rings four numbers at once. It asks the friendly screening question,
allowing a volunteer to press 1 to take the call. Or, they hang up
and
perhaps someone else will take it. (Or, if nobody does, it goes to
voicemail.)
Fine and dandy. Or so I thought. The problem is that followme is
designed to assume that it is only going to reach exactly one
person.
So, if a phone answers and they press 2 to reject the call: bam,
asterisk stops trying the other three phone numbers.
I am currently trying to educate the volunteers to refrain from
pressing 2 but that is prone to problems. I'd rather that there
not be
a reject function at all. Or, making it so that pressing 2 doesn't
really reject the call, it just hangs it up. I could change the
audio,
and remap 2 to 9 and hope nobody presses it, but that seems like an
accident waiting to happen.
Does anyone have suggestions?
John
I think you can make the following code mod to have the next in
dial plan step not do anything. If someone has the time, this would
probably be a decent option to add to the application for future
versions to make this behavior optional via an application option
parameter.
[r...@btwtechshowdemoc apps]# svn diff app_followme.c
Index: app_followme.c
===
--- app_followme.c (revision 188040)
+++ app_followme.c (working copy)
@@ -724,9 +724,9 @@
if (!
strcmp(tmpuser-yn, tpargs-nextindp)) {
if
(option_debug)
ast_log
(LOG_DEBUG, Next in dial plan step requested.\n);
-
*status = 1;
+ //
*status = 1;
ast_frfree(f);
-
return NULL;
+ //
return NULL;
}
}
Thanks for the code. Excellent idea.
In this particular case, I can't use it as-is because I run many
virtual
PBXs on the same machine. I don't want to change the behavior for all
followme, just this one customer. So, your suggestion of making it an
application option would be spot on.
I am a programmer, but I've never done open source stuff before. Is
there a quick readable 1-2-3 guide on submitting code changes to the
community?
John
John -
There's a quick summary here:
http://www.asterisk.org/developers/bug-guidelines
The even quicker summary is:
1) Write your code, test it. :-)
2) Join the bug tracker, electronically approve the contributor's
agreement.
3) Follow the bug through.
We're looking forward to seeing your contributions! I'd also suggest
spending some time on the IRC channels (#asterisk-dev) if you have any
questions that you think could be resolved in real-time interaction
with other developers.
JT
---
John Todd email:jt...@digium.com
Digium, Inc. | Asterisk Open Source Community Director
445 Jan Davis Drive NW - Huntsville AL 35806 - USA
direct: +1-256-428-6083 http://www.digium.com/
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Re: [asterisk-users] Agents on asterisk
You could make the agent busy with this kind of logic exten = 2000,1,Answer exten = 2000,2,SetMusicOnHold(default) exten = 2000,n,WaitMusicOnHold(300) exten = 2000,n,Background(vm-goodbye) exten = 2000,n,Hangup This would let the agent play MOH back to his/her self for 5 minutes and tie up the extension. If they come back more quickly, just hang up and be available again. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ROQUÉ, Francisco Emiliano Sent: Monday, April 13, 2009 10:27 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Agents on asterisk Thanks James. I read it. But the cmd PauseQueueMember must be executed from an extension. As soon as the agent logged in, asterisk does not recognize the dtmf. There can be an alternative solution? Regards Francisco james.coll...@xtratelecom.es wrote: You can use the PauseQueueMember command. http://www.voip-info.org/wiki/view/Asterisk+cmd+PauseQueueMember Saludos, -James ROQUÉ, Francisco Emiliano wrote: Hi! I have a question about agents in asterisk. In first place, agent login to asterisk (from the telephone) The question is: Can an agent take a break (using a function *(some number)) from the phone? Thanks to all Regards ___ -- 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 ___ -- 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] Is there documentation explaining res_config_curl?
On Sunday 12 April 2009 02:34:10 am Julian Lyndon-Smith wrote: Eric Chamberlain wrote: On Apr 11, 2009, at 12:53 AM, Julian Lyndon-Smith wrote: Eric Chamberlain wrote: [snip] Thank you, that bug does have useful information. We are working on moving from res_config_odbc to res_config_curl, so all asterisk requests go through our django backend, rather than django and asterisk sharing database tables. We had a buggy odbc driver (a 3rd party closed one) - we went from 2-3 crashes per day to zero in the last year, running nearly 3M config_curl requests per month now ;) It's, like, wow man ! As an additional note, please see contrib/scripts/dbsep.cgi, which I wrote as a reference implementation for the CGI backend of res_config_curl. It implements several additional methods to what JMLS is using (basically, for all the methods in trunk), so it may be useful in that regard. -- Tilghman ___ -- 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] Is there documentation explaining res_config_curl?
On Apr 13, 2009, at 8:52 AM, Tilghman Lesher wrote: On Sunday 12 April 2009 02:34:10 am Julian Lyndon-Smith wrote: Eric Chamberlain wrote: On Apr 11, 2009, at 12:53 AM, Julian Lyndon-Smith wrote: Eric Chamberlain wrote: [snip] Thank you, that bug does have useful information. We are working on moving from res_config_odbc to res_config_curl, so all asterisk requests go through our django backend, rather than django and asterisk sharing database tables. We had a buggy odbc driver (a 3rd party closed one) - we went from 2-3 crashes per day to zero in the last year, running nearly 3M config_curl requests per month now ;) It's, like, wow man ! As an additional note, please see contrib/scripts/dbsep.cgi, which I wrote as a reference implementation for the CGI backend of res_config_curl. It implements several additional methods to what JMLS is using (basically, for all the methods in trunk), so it may be useful in that regard. dbsep.cgi looks very helpful, thanks. Mapping the require function looks to be a challenge. Is there any way to add additional information to the res_config_curl POST request? We need to authenticate each https request and we'd rather not put the username and password in the path info to keep the password out of the server logs. ___ -- 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
[asterisk-users] Asterisk-beginner : cannot make phonecalls using Asterisk
Hi there, this is the first time that I'm building an Asterisk-server. I have compiled Asterisk together with Zaptel on an CentOS 5.3-system. Zaptel is for later, when configuring the POTS-line. Now first internal communication with SIP. Thought it would go easier... I have 2 Grandstream IP-phones : BT-201 and GXP-1200. These are my settings : sip.conf : [r...@asterisk asterisk]# cat sip.conf [general] bindport=5060 bindaddr = 0.0.0.0 [BT201] type=friend context=intern host=192.168.4.210 secret=testpaswoord [GXP1200] type=friend context=intern host=192.168.4.211 secret=testpaswoord extensions.conf : [r...@asterisk asterisk]# cat extensions.conf [intern] exten = 210,1,Dial(SIP/BT201) exten = 211,1,Dial(SIP/GXP1200) Asterisk CLI shows me : asterisk*CLI sip reload Reloading SIP == Parsing '/etc/asterisk/sip.conf': Found == Parsing '/etc/asterisk/users.conf': Found == Parsing '/etc/asterisk/sip_notify.conf': Found asterisk*CLI sip show peers Name/username HostDyn Nat ACL Port Status GXP1200192.168.4.211 5060 Unmonitored BT201 192.168.4.210 5060 Unmonitored 2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline] asterisk*CLI dialplan show intern [ Context 'intern' created by 'pbx_config' ] '210' = 1. Dial(SIP/BT201) [pbx_config] '211' = 1. Dial(SIP/GXP1200) [pbx_config] I pick up the phone of the BT201 and dial 211... nothing happens. I pick up the phone of the GXP1200 and dial 210... nothing happens. I would love to have your feedback on this. Where could this problem be situated ? I notice (on the Asterisk CLI) that my SIP-phones do not register. They have a fixed IP and there account information is set via the web interface. Greetingz, Jonas. ___ -- 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-beginner : cannot make phonecalls using Asterisk
jonas kellens wrote: Hi there, this is the first time that I'm building an Asterisk-server. I have compiled Asterisk together with Zaptel on an CentOS 5.3-system. Zaptel is for later, when configuring the POTS-line. Now first internal communication with SIP. Thought it would go easier... I have 2 Grandstream IP-phones : BT-201 and GXP-1200. These are my settings : sip.conf : /[r...@asterisk asterisk]# cat sip.conf/ /[general]/ /bindport=5060/ /bindaddr = 0.0.0.0/ /[BT201]/ /type=friend/ /context=intern/ /host=192.168.4.210/ /secret=testpaswoord/ /[GXP1200]/ /type=friend/ /context=intern/ /host=192.168.4.211/ /secret=testpaswoord/ extensions.conf : /[r...@asterisk asterisk]# cat extensions.conf/ /[intern]/ /exten = 210,1,Dial(SIP/BT201)/ /exten = 211,1,Dial(SIP/GXP1200)/ Asterisk CLI shows me : /asterisk*CLI sip reload/ /Reloading SIP/ / == Parsing '/etc/asterisk/sip.conf': Found/ / == Parsing '/etc/asterisk/users.conf': Found/ / == Parsing '/etc/asterisk/sip_notify.conf': Found/ /asterisk*CLI sip show peers/ /Name/username HostDyn Nat ACL Port Status / /GXP1200192.168.4.211 5060 Unmonitored / /BT201 192.168.4.210 5060 Unmonitored / /2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline]/ /asterisk*CLI dialplan show intern/ /[ Context 'intern' created by 'pbx_config' ]/ / '210' = 1. Dial(SIP/BT201) [pbx_config]/ / '211' = 1. Dial(SIP/GXP1200) [pbx_config]/ I pick up the phone of the BT201 and dial 211... nothing happens. I pick up the phone of the GXP1200 and dial 210... nothing happens. I would love to have your feedback on this. Where could this problem be situated ? I notice (on the Asterisk CLI) that my SIP-phones do not register. They have a fixed IP and there account information is set via the web interface. Greetingz, Jonas. ___ -- 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 I had the same issue. I set the hosts to dynamic and and explicitly set their IP's via a dhcp server using their MAC addresses. The phones registered and all is well. Regards, Mike ___ -- 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-beginner : cannot make phonecalls using Asterisk
bindaddr = 0.0.0.0 I would set this to the ethernet interface IP address, I believe this may be your issue. Registration is only for receiving calls, if you are not seeing information on the dial, then the phone is not talking to the server. I would make sure of the settings in the web-interface as well. Tony Plack ___ -- 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-beginner : cannot make phonecalls usingAsterisk
What do you see when you run asterisk –r and dial 210 or 211 from one of the phones James Shigley Monroe Telephone Answering Service 409-981-9213 Infinity 5.5,UC 4.02.3803, Blink 3.0.104 Ecreator:2.21, eResponse 1.1.7 Webportal,WebApps, CONFIDENTIALITY NOTICE: This email, including any attachments, contains information which may be confidential or privileged. The information is intended to be for the use of the individual or entity named above. If you are not the intended recipient, be aware that any disclosure, copying, distribution or use of the contents of this information is prohibited. If you have received this email in error, please notify the sender immediately by reply to sender only message and destroy all electronic and hard copies of the communication, including attachments. Common sense is the collection of prejudices acquired by age eighteen. -- Albert Einstein Once you can accept the universe as matter expanding into nothing that is something,wearing stripes with plaid comes easy. -- Albert Einstein I know a little of everything, but a lot of nothing From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jonas kellens Sent: Monday, April 13, 2009 11:19 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk-beginner : cannot make phonecalls usingAsterisk Hi there, this is the first time that I'm building an Asterisk-server. I have compiled Asterisk together with Zaptel on an CentOS 5.3-system. Zaptel is for later, when configuring the POTS-line. Now first internal communication with SIP. Thought it would go easier... I have 2 Grandstream IP-phones : BT-201 and GXP-1200. These are my settings : sip.conf : [r...@asterisk asterisk]# cat sip.conf [general] bindport=5060 bindaddr = 0.0.0.0 [BT201] type=friend context=intern host=192.168.4.210 secret=testpaswoord [GXP1200] type=friend context=intern host=192.168.4.211 secret=testpaswoord extensions.conf : [r...@asterisk asterisk]# cat extensions.conf [intern] exten = 210,1,Dial(SIP/BT201) exten = 211,1,Dial(SIP/GXP1200) Asterisk CLI shows me : asterisk*CLI sip reload Reloading SIP == Parsing '/etc/asterisk/sip.conf': Found == Parsing '/etc/asterisk/users.conf': Found == Parsing '/etc/asterisk/sip_notify.conf': Found asterisk*CLI sip show peers Name/username HostDyn Nat ACL Port Status GXP1200192.168.4.211 5060 Unmonitored BT201 192.168.4.210 5060 Unmonitored 2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline] asterisk*CLI dialplan show intern [ Context 'intern' created by 'pbx_config' ] '210' = 1. Dial(SIP/BT201)[pbx_config] '211' = 1. Dial(SIP/GXP1200) [pbx_config] I pick up the phone of the BT201 and dial 211... nothing happens. I pick up the phone of the GXP1200 and dial 210... nothing happens. I would love to have your feedback on this. Where could this problem be situated ? I notice (on the Asterisk CLI) that my SIP-phones do not register. They have a fixed IP and there account information is set via the web interface. Greetingz, Jonas. ___ -- 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] ***SPAM*** Re: Asterisk-beginner : cannot make phonecalls using Asterisk
In your sip.conf or sip_nat.conf (for elastix) set the variables: externhost= externip= domain= externrefresh= localnet= Regards Francisco Anthony Plack wrote: bindaddr = 0.0.0.0 I would set this to the ethernet interface IP address, I believe this may be your issue. Registration is only for receiving calls, if you are not seeing information on the dial, then the phone is not talking to the server. I would make sure of the settings in the web-interface as well. Tony Plack ___ -- 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 ___ -- 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-beginner : cannot make phonecalls using Asterisk
Mike, thank you for your reply. However I do not have the option of a DHCP-server. On the network where Asterisk needs to be implemented all is configured statically, so also the IP-phones need to be statically assigned an IP-address. Surely this can not be thé problem... Greetingz, Jonas. On Mon, 2009-04-13 at 12:28 -0400, Michael van der Stoop wrote: jonas kellens wrote: Hi there, this is the first time that I'm building an Asterisk-server. I have compiled Asterisk together with Zaptel on an CentOS 5.3-system. Zaptel is for later, when configuring the POTS-line. Now first internal communication with SIP. Thought it would go easier... I have 2 Grandstream IP-phones : BT-201 and GXP-1200. These are my settings : sip.conf : /[r...@asterisk asterisk]# cat sip.conf/ /[general]/ /bindport=5060/ /bindaddr = 0.0.0.0/ /[BT201]/ /type=friend/ /context=intern/ /host=192.168.4.210/ /secret=testpaswoord/ /[GXP1200]/ /type=friend/ /context=intern/ /host=192.168.4.211/ /secret=testpaswoord/ extensions.conf : /[r...@asterisk asterisk]# cat extensions.conf/ /[intern]/ /exten = 210,1,Dial(SIP/BT201)/ /exten = 211,1,Dial(SIP/GXP1200)/ Asterisk CLI shows me : /asterisk*CLI sip reload/ /Reloading SIP/ / == Parsing '/etc/asterisk/sip.conf': Found/ / == Parsing '/etc/asterisk/users.conf': Found/ / == Parsing '/etc/asterisk/sip_notify.conf': Found/ /asterisk*CLI sip show peers/ /Name/username HostDyn Nat ACL Port Status / /GXP1200192.168.4.211 5060 Unmonitored / /BT201 192.168.4.210 5060 Unmonitored / /2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline]/ /asterisk*CLI dialplan show intern/ /[ Context 'intern' created by 'pbx_config' ]/ / '210' = 1. Dial(SIP/BT201) [pbx_config]/ / '211' = 1. Dial(SIP/GXP1200) [pbx_config]/ I pick up the phone of the BT201 and dial 211... nothing happens. I pick up the phone of the GXP1200 and dial 210... nothing happens. I would love to have your feedback on this. Where could this problem be situated ? I notice (on the Asterisk CLI) that my SIP-phones do not register. They have a fixed IP and there account information is set via the web interface. Greetingz, Jonas. ___ -- 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 I had the same issue. I set the hosts to dynamic and and explicitly set their IP's via a dhcp server using their MAC addresses. The phones registered and all is well. Regards, Mike ___ -- 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 ___ -- 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-beginner : cannot make phonecallsusingAsterisk
Do you have include=intern in the default context? If no, * will come back with can't find peer 210 (or 211). From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jonas kellens Sent: Monday, April 13, 2009 11:19 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk-beginner : cannot make phonecalls usingAsterisk Hi there, this is the first time that I'm building an Asterisk-server. I have compiled Asterisk together with Zaptel on an CentOS 5.3-system. Zaptel is for later, when configuring the POTS-line. Now first internal communication with SIP. Thought it would go easier... I have 2 Grandstream IP-phones : BT-201 and GXP-1200. These are my settings : sip.conf : [r...@asterisk asterisk]# cat sip.conf [general] bindport=5060 bindaddr = 0.0.0.0 [BT201] type=friend context=intern host=192.168.4.210 secret=testpaswoord [GXP1200] type=friend context=intern host=192.168.4.211 secret=testpaswoord extensions.conf : [r...@asterisk asterisk]# cat extensions.conf [intern] exten = 210,1,Dial(SIP/BT201) exten = 211,1,Dial(SIP/GXP1200) Asterisk CLI shows me : asterisk*CLI sip reload Reloading SIP == Parsing '/etc/asterisk/sip.conf': Found == Parsing '/etc/asterisk/users.conf': Found == Parsing '/etc/asterisk/sip_notify.conf': Found asterisk*CLI sip show peers Name/username HostDyn Nat ACL Port Status GXP1200192.168.4.211 5060 Unmonitored BT201 192.168.4.210 5060 Unmonitored 2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline] asterisk*CLI dialplan show intern [ Context 'intern' created by 'pbx_config' ] '210' = 1. Dial(SIP/BT201) [pbx_config] '211' = 1. Dial(SIP/GXP1200) [pbx_config] I pick up the phone of the BT201 and dial 211... nothing happens. I pick up the phone of the GXP1200 and dial 210... nothing happens. I would love to have your feedback on this. Where could this problem be situated ? I notice (on the Asterisk CLI) that my SIP-phones do not register. They have a fixed IP and there account information is set via the web interface. Greetingz, Jonas. ___ -- 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-beginner : cannot make phonecalls using Asterisk
On Mon, 13 Apr 2009, Anthony Plack wrote: bindaddr = 0.0.0.0 I would set this to the ethernet interface IP address, I believe this may be your issue. Binding to 0.0.0.0 means listen to all IP addresses on the box. It is not the issue. Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- 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-beginner : cannot make phonecalls using Asterisk
Tony Plack, this is the result form Asterisk CLI : [r...@asterisk asterisk]# /usr/sbin/asterisk -vr Asterisk 1.4.24, Copyright (C) 1999 - 2008 Digium, Inc. and others. Created by Mark Spencer marks...@digium.com Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. = == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Found Connected to Asterisk 1.4.24 currently running on asterisk (pid = 3895) Verbosity is at least 5 asterisk*CLI dialplan reload Dialplan reloaded. == Parsing '/etc/asterisk/extensions.conf': Found -- Registered extension context 'intern' -- Added extension '210' priority 1 to intern -- Added extension '211' priority 1 to intern == Parsing '/etc/asterisk/users.conf': Found asterisk*CLI sip reload Reloading SIP == Parsing '/etc/asterisk/sip.conf': Found == Parsing '/etc/asterisk/users.conf': Found == SIP Listening on 192.168.4.248:5060 == Using SIP TOS: none == Parsing '/etc/asterisk/sip_notify.conf': Found So I've changed the bindaddr... Still no change I'm afraid... Thanks for your reply ! Please help me a bit further cause this a work I'm doing as thesis. Jonas. On Mon, 2009-04-13 at 11:34 -0500, Anthony Plack wrote: bindaddr = 0.0.0.0 I would set this to the ethernet interface IP address, I believe this may be your issue. Registration is only for receiving calls, if you are not seeing information on the dial, then the phone is not talking to the server. I would make sure of the settings in the web-interface as well. Tony Plack ___ -- 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-beginner : cannot make phonecalls using Asterisk
Alright again, what do you see on the CLI when you make a call to 210/211? James Shigley From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jonas kellens Sent: Monday, April 13, 2009 12:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk-beginner : cannot make phonecalls using Asterisk Tony Plack, this is the result form Asterisk CLI : [r...@asterisk asterisk]# /usr/sbin/asterisk -vr Asterisk 1.4.24, Copyright (C) 1999 - 2008 Digium, Inc. and others. Created by Mark Spencer marks...@digium.com Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. = == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Found Connected to Asterisk 1.4.24 currently running on asterisk (pid = 3895) Verbosity is at least 5 asterisk*CLI dialplan reload Dialplan reloaded. == Parsing '/etc/asterisk/extensions.conf': Found -- Registered extension context 'intern' -- Added extension '210' priority 1 to intern -- Added extension '211' priority 1 to intern == Parsing '/etc/asterisk/users.conf': Found asterisk*CLI sip reload Reloading SIP == Parsing '/etc/asterisk/sip.conf': Found == Parsing '/etc/asterisk/users.conf': Found == SIP Listening on 192.168.4.248:5060 == Using SIP TOS: none == Parsing '/etc/asterisk/sip_notify.conf': Found So I've changed the bindaddr... Still no change I'm afraid... Thanks for your reply ! Please help me a bit further cause this a work I'm doing as thesis. Jonas. On Mon, 2009-04-13 at 11:34 -0500, Anthony Plack wrote: bindaddr = 0.0.0.0 I would set this to the ethernet interface IP address, I believe this may be your issue. Registration is only for receiving calls, if you are not seeing information on the dial, then the phone is not talking to the server. I would make sure of the settings in the web-interface as well. Tony Plack ___ -- 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
[asterisk-users] Asterisk-beginner : cannot make phonecalls using Asterisk
James, when I run Asterisk -vr and I enter 210 on one phone to call the other, nothing is displayed on the CommandLine... I know this is not right, just don't know what is wrong. I really need someone to guide me a bit... [r...@asterisk asterisk]# /usr/sbin/asterisk -vr Asterisk 1.4.24, Copyright (C) 1999 - 2008 Digium, Inc. and others. Created by Mark Spencer marks...@digium.com Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. = == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Found Connected to Asterisk 1.4.24 currently running on asterisk (pid = 3895) Verbosity is at least 5 asterisk*CLI sip reload Reloading SIP == Parsing '/etc/asterisk/sip.conf': Found == Parsing '/etc/asterisk/users.conf': Found == Parsing '/etc/asterisk/sip_notify.conf': Found asterisk*CLI sip show peers Name/username HostDyn Nat ACL Port Status GXP1200/GXP1200192.168.4.211 5060 Unmonitored BT201/BT201192.168.4.210 5060 Unmonitored 2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline] Thanks for your reply ! Jonas. On Mon, 2009-04-13 at 11:59 -0500, James A. Shigley wrote: What do you see when you run asterisk –r and dial 210 or 211 from one of the phones James Shigley ___ -- 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-beginner : cannot make phonecalls using Asterisk
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: April-13-09 1:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk-beginner : cannot make phonecalls using Asterisk On Mon, 13 Apr 2009, Anthony Plack wrote: bindaddr = 0.0.0.0 I would set this to the ethernet interface IP address, I believe this may be your issue. Binding to 0.0.0.0 means listen to all IP addresses on the box. It is not the issue. Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 Same problem here, have 5 nics, One for outside, One for inside One for virtual vz's One for openvpn So if you listen on all , you open your box out.. (bad..) If you don't.. then you can't choose both the outside, and openvpn ___ -- 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] Is there documentation explaining res_config_curl?
On Monday 13 April 2009 11:14:14 am Eric Chamberlain wrote: On Apr 13, 2009, at 8:52 AM, Tilghman Lesher wrote: On Sunday 12 April 2009 02:34:10 am Julian Lyndon-Smith wrote: Eric Chamberlain wrote: On Apr 11, 2009, at 12:53 AM, Julian Lyndon-Smith wrote: Eric Chamberlain wrote: [snip] Thank you, that bug does have useful information. We are working on moving from res_config_odbc to res_config_curl, so all asterisk requests go through our django backend, rather than django and asterisk sharing database tables. We had a buggy odbc driver (a 3rd party closed one) - we went from 2-3 crashes per day to zero in the last year, running nearly 3M config_curl requests per month now ;) It's, like, wow man ! As an additional note, please see contrib/scripts/dbsep.cgi, which I wrote as a reference implementation for the CGI backend of res_config_curl. It implements several additional methods to what JMLS is using (basically, for all the methods in trunk), so it may be useful in that regard. dbsep.cgi looks very helpful, thanks. Mapping the require function looks to be a challenge. Is there any way to add additional information to the res_config_curl POST request? We need to authenticate each https request and we'd rather not put the username and password in the path info to keep the password out of the server logs. Not to the POST, no, but in 1.6.2 and higher, you can use the CURLOPT function in the [globals] section of extensions.conf to set an HTTP header containing the authentication username and password: [globals] CURLOPT(userpwd)=username:password Note that you may need to preload pbx_config.so and func_curl.so prior to res_config_curl.so in order for these settings to be in place for the realtime load at startup. -- Tilghman ___ -- 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
[asterisk-users] Asterisk-beginner : cannot make phonecallsusingAsterisk
Danny, this is from the Asterisk CLI : asterisk*CLI dialplan reload Dialplan reloaded. == Parsing '/etc/asterisk/extensions.conf': Found -- Registered extension context 'default' -- Including context 'intern' in context 'default' -- Registered extension context 'intern' -- Added extension '210' priority 1 to intern -- Added extension '211' priority 1 to intern == Parsing '/etc/asterisk/users.conf': Found asterisk*CLI dialplan show default [ Context 'default' created by 'pbx_config' ] Include ='intern' [pbx_config] -= 0 extensions (0 priorities) in 1 context. =- Asterisk does not show me can't find peer 210 (or 211)... There is no output on the CLI... Thanks for your reply ! Jonas. On Mon, 2009-04-13 at 11:58 -0500, Danny Nicholas wrote: Do you have include=intern in the default context? If no, * will come back with can’t find peer 210 (or 211). ___ -- 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
[asterisk-users] MySQL queries
I'm running some mysql queries on the standard sql logging of calls, and am interested if anyone has any they'd like to share to get good statistics. I'm interested in # of calls per day, based on DST. Number of Calls per day based on SRC, avg duration of calls, etc.. Thanks. Jeremy Mann Director of IT Texas Health Management Group Direct Line: 817-310-4956 Main Line: 817-310-4999 Helpdesk: 817-310-4999 x3 Fax: 817-310-4990 Email: jm...@txhmg.com This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- 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
[asterisk-users] Asterisk-beginner : cannot make phonecalls using Asterisk
I pick up the phone, and dial 211 on the BT201. This is the Asterisk CLI : Connected to Asterisk 1.4.24 currently running on asterisk (pid = 3895) Verbosity is at least 5 asterisk*CLI Nothing is displayed... it stays that way... Jonas. On Mon, 2009-04-13 at 11:59 -0500, James A. Shigley wrote: What do you see when you run asterisk –r and dial 210 or 211 from one of the phones James Shigley __ ___ -- 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
[asterisk-users] Asterisk-beginner : cannot make phonecalls using Asterisk
These are the settings on my BT201 (GXP1200 is the same interface) : Account Name:(e.g., MyCompany) SIP Server:(e.g., sip.mycompany.com, or IP address) Outbound Proxy:(e.g., proxy.myprovider.com, or IP address) SIP User ID:(the user part of an SIP address) -- I put here the same as username=BT201 Authenticate ID:(can be same or different from SIP UserID) Authenticate Password:(not displayed for security protection) -- I've put here the same is secret=testpaswoord Name:(optional, e.g., John Doe) sip.conf : [r...@asterisk asterisk]# cat sip.conf [general] bindport=5060 bindaddr = 192.168.4.248 [BT201] type=friend context=intern host=192.168.4.210 username=BT201 secret=testpaswoord [GXP1200] type=friend context=intern host=192.168.4.211 username=GXP1200 secret=testpaswoord ___ -- 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-beginner : cannot make phonecalls using Asterisk
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 jonas kellens wrote: I pick up the phone, and dial 211 on the BT201. This is the Asterisk CLI : /Connected to Asterisk 1.4.24 currently running on asterisk (pid = 3895)/ /Verbosity is at least 5/ /asterisk*CLI / Nothing is displayed... it stays that way... Jonas. Is there a Send button on that phone? It sounds to me as though the phone is still waiting for more digits. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFJ437BCFu3bIiwtTARAs73AJ9spwpr7ULu6VyimPPoDIPnzFK6JQCbBEDO bQ0m2dROkUEkdtwCHtbHTBI= =4Zmk -END PGP SIGNATURE- ___ -- 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-beginner : cannot make phonecallsusingAsterisk
Danny Nicholas wrote: Do you have include=intern in the default context? If no, * will come back with can't find peer 210 (or 211). *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *jonas kellens *Sent:* Monday, April 13, 2009 11:19 AM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] Asterisk-beginner : cannot make phonecalls usingAsterisk Hi there, this is the first time that I'm building an Asterisk-server. I have compiled Asterisk together with Zaptel on an CentOS 5.3-system. Zaptel is for later, when configuring the POTS-line. Now first internal communication with SIP. Thought it would go easier... I have 2 Grandstream IP-phones : BT-201 and GXP-1200. These are my settings : sip.conf : /[r...@asterisk asterisk]# cat sip.conf/ /[general]/ /bindport=5060/ /bindaddr = 0.0.0.0/ /[BT201]/ /type=friend/ /context=intern/ /host=192.168.4.210/ /secret=testpaswoord/ /[GXP1200]/ /type=friend/ /context=intern/ /host=192.168.4.211/ /secret=testpaswoord/ extensions.conf : /[r...@asterisk asterisk]# cat extensions.conf/ /[intern]/ /exten = 210,1,Dial(SIP/BT201)/ /exten = 211,1,Dial(SIP/GXP1200)/ Asterisk CLI shows me : /asterisk*CLI sip reload/ /Reloading SIP/ / == Parsing '/etc/asterisk/sip.conf': Found/ / == Parsing '/etc/asterisk/users.conf': Found/ / == Parsing '/etc/asterisk/sip_notify.conf': Found/ /asterisk*CLI sip show peers/ /Name/username HostDyn Nat ACL Port Status / /GXP1200192.168.4.211 5060 Unmonitored / /BT201 192.168.4.210 5060 Unmonitored / /2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline]/ /asterisk*CLI dialplan show intern/ /[ Context 'intern' created by 'pbx_config' ]/ / '210' = 1. Dial(SIP/BT201) [pbx_config]/ / '211' = 1. Dial(SIP/GXP1200) [pbx_config]/ I pick up the phone of the BT201 and dial 211... nothing happens. I pick up the phone of the GXP1200 and dial 210... nothing happens. I would love to have your feedback on this. Where could this problem be situated ? I notice (on the Asterisk CLI) that my SIP-phones do not register. They have a fixed IP and there account information is set via the web interface. Greetingz, Jonas. ___ -- 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 This is not the case since both of his phones are configured to come in to the intern context by default. In the real world, if you intern context had access to outside calls and you included it in the default context and happened to allow guest access, then anybody coming in to your box could make outbound calls. ___ -- 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-beginner : cannot make phonecalls using Asterisk
Barry, there is a 'send' button but pushing it before or after dialing '211' does not really change anything... I get no dial tone, no ring tone on the other phone and no output on the Asterisk CLI... I thought this would go easier... Don't know what is going on here. I followed the book Asterisk, the future of telephony... Thanks for your reply ! Greetingz, Jonas. On Mon, 2009-04-13 at 14:04 -0400, Barry L. Kline wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 jonas kellens wrote: I pick up the phone, and dial 211 on the BT201. This is the Asterisk CLI : /Connected to Asterisk 1.4.24 currently running on asterisk (pid = 3895)/ /Verbosity is at least 5/ /asterisk*CLI / Nothing is displayed... it stays that way... Jonas. Is there a Send button on that phone? It sounds to me as though the phone is still waiting for more digits. Barry ___ -- 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-beginner : cannot make phonecalls using Asterisk
jonas kellens wrote: Hi there, I notice (on the Asterisk CLI) that my SIP-phones do not register. They have a fixed IP and there account information is If your phones don't register, then your not going to be able to make a call. The Grandstream phones have a web interface (At least if memory serves correctly) and you'll need to tell the phones: server ip address username/extension to use password of that extension. Once they've registered, things will get a lot easier. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation 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
[asterisk-users] Asterisk-beginner : cannot make phonecalls using Asterisk (update)
Hey there again ! I've changed some things now : 1) IP-phones get there IP from a DHCP 2) sip-accounts simplified : [r...@asterisk asterisk]# cat sip.conf [general] context=default port=5060 bindaddr=0.0.0.0 srvlookup=yes disallow=all allow=ulaw [210] type=friend context=intern host=dynamic [211] type=friend context=intern host=dynamic 3) dial plan simplified : [r...@asterisk asterisk]# cat extensions.conf [globals] [default] include = intern [intern] exten = 210,1,Dial(SIP/210) exten = 211,1,Dial(SIP/211) The IP-phones are set as DHCP-client... I reloaded everything on the Asterisk CLI. I put off the power of the IP-phones and then put them back on. I still see no register-message on the CLI. This really should happen now, as they are defined host=dynamic ! What can be going wrong here... Tell me, I'm not writing a wrong sip.conf or extensions.conf, do I ? I will now hang my portable on the switch and monitor the network with wireshark to see if the phones send a SIP-register to the Asterisk-server... In the mean time... every feedback on this is very welcome, thanks. Jonas. ___ -- 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-beginner : cannot make phonecalls using Asterisk
Hi On Mon, Apr 13, 2009 at 06:18:58PM +0200, jonas kellens wrote: I pick up the phone of the BT201 and dial 211... nothing happens. I pick up the phone of the GXP1200 and dial 210... nothing happens. I would love to have your feedback on this. Where could this problem be situated ? Your basic mistake at troubleshooting this is trying to test two things at the same time. Let's test them separately. 1. A call from Asterisk to the phones: In the Asterisk CLI: originate SIP/BT201 application playback demo-instruct And the other one: originate SIP/GXP1200 application playback demo-instruct Alternatively, use the echo-test aplication: originate SIP/BT201 application echo 2. Next, test calling from the phones to Asterisk. Add those two extensions to [intern] exten = 250,1,Answer exten = 250,n,Playback(demo-instruct) exten = 250,n,Hangup exten = 251,1,Answer exten = 251,1,Echo exten = 251,1,Hangup Make sure you reload for that to take effect, and then try dialing 250 or 251. Another useful tools: 'sip debug'. It tends to generate a very noisy output that is normally not readable for mere mortals. However it does indicate that something is happening. If you call from a remote SIP phone and there's nothing on the SIP debug, the problem is probably with the settings of the phone, as it is not getting to you. Last and not least: a sanity check as you see nothing: what is the output of: 'logger show channels' ? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- 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-beginner : cannot make phonecalls using Asterisk (update)
On 13 Apr 2009, at 20:52, jonas kellens wrote: Hey there again ! If you are new to all this wouldn't going with some pre-made dialplan be useful? Go for something like FreePBX ___ -- 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-beginner : cannot make phonecalls using Asterisk (update)
On Mon, 13 Apr 2009, jonas kellens wrote: 1) IP-phones get there IP from a DHCP The source of the address is not the issue. I still see no register-message on the CLI. This really should happen now, as they are defined host=dynamic ! I suspect you have not [correctly] configured the phones to register to the Asterisk server. I will now hang my portable on the switch and monitor the network with wireshark to see if the phones send a SIP-register to the Asterisk-server... sudo netstat -a -n -p | grep 5060 will show you if Asterisk is actually listening. It should look something like: udp 0 0 0.0.0.0:5060 0.0.0.0:* 3283/asterisk sudo tcpdump port 5060 will show you if the phones are talking to the box. It should look something like: 13:11:30.432163 IP spa841.example.com.sip asterisk.example.com.sip: UDP, length 431 13:11:30.432443 IP asterisk.example.com.sip spa841.example.com.sip: UDP, length 398 13:11:30.432520 IP asterisk.example.com.sip spa841.example.com.sip: UDP, length 460 13:11:30.451350 IP spa841.example.com.sip asterisk.example.com.sip: UDP, length 578 13:11:30.451525 IP asterisk.example.com.sip spa841.example.com.sip: UDP, length 398 13:11:30.460889 IP asterisk.example.com.sip spa841.example.com.sip: UDP, length 481 13:11:30.461231 IP asterisk.example.com.sip spa841.example.com.sip: UDP, length 476 13:11:30.461541 IP asterisk.example.com.sip spa841.example.com.sip: UDP, length 540 13:11:30.474515 IP spa841.example.com.sip asterisk.example.com.sip: UDP, length 383 13:11:30.497854 IP spa841.example.com.sip asterisk.example.com.sip: UDP, length 319 sip debug at the Asterisk console will show the messages as the are received and responded to by Asterisk. It should look something like: -- SIP read from 192.168.0.19:5060: SIP/2.0 200 OK To: sip:spa...@192.168.0.19:5060;tag=d732d5ba46660f68i0 From: asterisk sip:aster...@192.168.0.1;tag=as51d58666 Call-ID: 7e81b5850a48114430b5bd505bfd3...@192.168.0.1 CSeq: 102 NOTIFY Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK25449e4a Server: Sipura/SPA841-3.1.4(a) Content-Length: 0 --- (8 headers 0 lines) --- Destroying call '7e81b5850a48114430b5bd505bfd3...@192.168.0.1' 12 headers, 0 lines Reliably Transmitting (no NAT) to 192.168.0.19:5060: OPTIONS sip:spa...@192.168.0.19:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK6d5660c5 From: asterisk sip:aster...@192.168.0.1;tag=as079a9a44 To: sip:spa...@192.168.0.19:5060 Contact: sip:aster...@192.168.0.1 Call-ID: 16bb21000690e22e53bff2f90b43d...@192.168.0.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 13 Apr 2009 20:18:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- 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-beginner : cannot make phonecalls using Asterisk (update)
jonas kellens escribió: Hey there again ! Hey, just my two cents: I've changed some things now : 1) IP-phones get there IP from a DHCP 2) sip-accounts simplified : /[r...@asterisk asterisk]# cat sip.conf/ /[general]/ /context=default/ /port=5060/ /bindaddr=0.0.0.0/ /srvlookup=yes/ /disallow=all/ /allow=ulaw/ /[210]/ /type=friend/ /context=intern/ /host=dynamic/ /[211]/ /type=friend/ /context=intern/ /host=dynamic/ Mi first cent: This is oversimplified. I think you need to put the username=2XX here too. You can check the configuration with the asterisk CLI commands sip show users, sip show peers, sip show user XXX and sip show peer XXX 3) dial plan simplified : /[r...@asterisk asterisk]# cat extensions.conf/ /[globals]/ /[default]/ /include = intern/ /[intern]/ /exten = 210,1,Dial(SIP/210)/ /exten = 211,1,Dial(SIP/211)/ The IP-phones are set as DHCP-client... I reloaded everything on the Asterisk CLI. I put off the power of the IP-phones and then put them back on. My second cent: check again if your phones are configured to register, and recheck your network configuration. Something like 255.255.255.255 on your netmask will make the communication impossible. I still see no register-message on the CLI. This really should happen now, as they are defined host=dynamic ! What can be going wrong here... Tell me, I'm not writing a wrong sip.conf or extensions.conf, do I ? I will now hang my portable on the switch and monitor the network with wireshark to see if the phones send a SIP-register to the Asterisk-server... In the mean time... every feedback on this is very welcome, thanks. You're welcome. Jonas. ___ -- 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 -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center PBX: (+57 1)6500800 ext. 1201 Fax: (+57 1)6500816 Móvil: (+57)3138873587 ___ -- 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-beginner : cannot make phonecalls using Asterisk
Hi Tzafrir, yet with the first test, things get wrong : asterisk*CLI logger show channels Channel Type StatusConfiguration --- --- /var/log/asterisk/messages File Enabled- Warning Notice Error Console Enabled- Warning Notice Error asterisk*CLI asterisk*CLI originate SIP/210 application playback demo-instruct [Apr 13 22:33:23] NOTICE[3481]: channel.c:3033 __ast_request_and_dial: Unable to request channel SIP/210 asterisk*CLI Instead of naming the phone BT201, I've named it after its internal telephone number. For clearity for myself :-). But when I dial the IP-phone from the CLI, I get the output of above... Thank for your reply ! Jonas. On Mon, 2009-04-13 at 23:11 +0300, Tzafrir Cohen wrote: Hi On Mon, Apr 13, 2009 at 06:18:58PM +0200, jonas kellens wrote: I pick up the phone of the BT201 and dial 211... nothing happens. I pick up the phone of the GXP1200 and dial 210... nothing happens. I would love to have your feedback on this. Where could this problem be situated ? Your basic mistake at troubleshooting this is trying to test two things at the same time. Let's test them separately. 1. A call from Asterisk to the phones: In the Asterisk CLI: originate SIP/BT201 application playback demo-instruct And the other one: originate SIP/GXP1200 application playback demo-instruct Alternatively, use the echo-test aplication: originate SIP/BT201 application echo 2. Next, test calling from the phones to Asterisk. Add those two extensions to [intern] exten = 250,1,Answer exten = 250,n,Playback(demo-instruct) exten = 250,n,Hangup exten = 251,1,Answer exten = 251,1,Echo exten = 251,1,Hangup Make sure you reload for that to take effect, and then try dialing 250 or 251. Another useful tools: 'sip debug'. It tends to generate a very noisy output that is normally not readable for mere mortals. However it does indicate that something is happening. If you call from a remote SIP phone and there's nothing on the SIP debug, the problem is probably with the settings of the phone, as it is not getting to you. Last and not least: a sanity check as you see nothing: what is the output of: 'logger show channels' ? attachment: stock_smiley-1.png___ -- 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-beginner : cannot make phonecalls using Asterisk (update)
1) IP-phones get there IP from a DHCP The source of the address is not the issue. I still see no register-message on the CLI. This really should happen now, as they are defined host=dynamic ! I suspect you have not [correctly] configured the phones to register to the Asterisk server. You may want to pay attention to what he is saying. If you read through teh "Asterisk - The Future of Telephony - 2nd Edition" then read page 83 where it says.. "The host option is used to define where the clients exists on the network when Asterisk needs to send a call to it." In your case it appears the phone (not Asterisk) is the one having a problem sending a call. Have you visisted the phone's web configuration page and entered the IP address of the Asterisk server? If you have not then you need to plug in the Asterisk server IP address, authorization userid (usually the extension) and the authorization password (defined by 'secret='). There was another post I would second regarding FreePBX. My first 'installation' of Asterisk was Digium's Switchvox server delivered to my company's office (followed quickly by their 'free' version on my home network). My second installation was FreePBX for a friend's small company which allowed me to get my feet wet in a production environment. I firmly believe that starting off with source compiled version of Asterisk would have been more than I could chew and probably ruined my appetite for this fascinating platform. FreePBX (if you read the configuration files carefully) will allow you to get a functional installation working more quickly. After you have a chance to see how it works (and how it doesn't) then you can graduate to a custom installation of Asterisk. Glad to see you're suffering through this though. It's a great learning experience once it works. ___ -- 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-beginner : cannot make phonecalls using Asterisk
On Mon, Apr 13, 2009 at 10:39:49PM +0200, jonas kellens wrote: Hi Tzafrir, yet with the first test, things get wrong : asterisk*CLI logger show channels Channel Type StatusConfiguration --- --- /var/log/asterisk/messages File Enabled- Warning Notice Error Console Enabled- Warning Notice Error asterisk*CLI asterisk*CLI originate SIP/210 application playback demo-instruct [Apr 13 22:33:23] NOTICE[3481]: channel.c:3033 __ast_request_and_dial: Unable to request channel SIP/210 asterisk*CLI Instead of naming the phone BT201, I've named it after its internal telephone number. For clearity for myself :-). But when I dial the IP-phone from the CLI, I get the output of above... At that point, what is the output of: sip show peers A more verbose output would help -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- 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-beginner : cannot make phonecalls using Asterisk (update)
[r...@asterisk asterisk]# netstat -a -n -p | grep 5060 udp0 0 0.0.0.0:50600.0.0.0:* 3047/asterisk [r...@asterisk asterisk]# /usr/sbin/tcpdump port 5060 tcpdump: verbose output suppressed, use -v or -vv for full protocol decode listening on eth1, link-type EN10MB (Ethernet), capture size 96 bytes 23:04:59.522498 IP 192.168.4.114.sip 192.168.4.248.sip: SIP, length: 530 23:05:01.233460 IP 192.168.4.112.sip 192.168.4.248.sip: SIP, length: 540 23:05:23.521076 IP 192.168.4.114.sip 192.168.4.248.sip: SIP, length: 530 23:05:24.520486 IP 192.168.4.114.sip 192.168.4.248.sip: SIP, length: 530 23:05:25.232068 IP 192.168.4.112.sip 192.168.4.248.sip: SIP, length: 540 23:05:26.231229 IP 192.168.4.112.sip 192.168.4.248.sip: SIP, length: 540 23:05:26.520308 IP 192.168.4.114.sip 192.168.4.248.sip: SIP, length: 530 23:05:28.231050 IP 192.168.4.112.sip 192.168.4.248.sip: SIP, length: 540 23:05:30.519957 IP 192.168.4.114.sip 192.168.4.248.sip: SIP, length: 530 23:05:32.230693 IP 192.168.4.112.sip 192.168.4.248.sip: SIP, length: 540 23:05:34.521843 IP 192.168.4.114.sip 192.168.4.248.sip: SIP, length: 925 23:05:34.530587 IP 192.168.4.114.sip 192.168.4.248.sip: SIP, length: 530 23:05:35.519255 IP 192.168.4.114.sip 192.168.4.248.sip: SIP, length: 925 23:05:36.230336 IP 192.168.4.112.sip 192.168.4.248.sip: SIP, length: 540 23:05:37.519077 IP 192.168.4.114.sip 192.168.4.248.sip: SIP, length: 925 23:05:41.518720 IP 192.168.4.114.sip 192.168.4.248.sip: SIP, length: 925 Connected to Asterisk 1.4.24 currently running on asterisk (pid = 3047) Verbosity is at least 3 asterisk*CLI sip debug SIP Debugging re-enabled asterisk*CLI and it stays that way... Greetingz, Jonas. On Mon, 2009-04-13 at 13:21 -0700, Steve Edwards wrote: On Mon, 13 Apr 2009, jonas kellens wrote: 1) IP-phones get there IP from a DHCP The source of the address is not the issue. I still see no register-message on the CLI. This really should happen now, as they are defined host=dynamic ! I suspect you have not [correctly] configured the phones to register to the Asterisk server. I will now hang my portable on the switch and monitor the network with wireshark to see if the phones send a SIP-register to the Asterisk-server... sudo netstat -a -n -p | grep 5060 will show you if Asterisk is actually listening. It should look something like: udp 0 0 0.0.0.0:5060 0.0.0.0:* 3283/asterisk sudo tcpdump port 5060 will show you if the phones are talking to the box. It should look something like: 13:11:30.432163 IP spa841.example.com.sip asterisk.example.com.sip: UDP, length 431 13:11:30.432443 IP asterisk.example.com.sip spa841.example.com.sip: UDP, length 398 13:11:30.432520 IP asterisk.example.com.sip spa841.example.com.sip: UDP, length 460 13:11:30.451350 IP spa841.example.com.sip asterisk.example.com.sip: UDP, length 578 13:11:30.451525 IP asterisk.example.com.sip spa841.example.com.sip: UDP, length 398 13:11:30.460889 IP asterisk.example.com.sip spa841.example.com.sip: UDP, length 481 13:11:30.461231 IP asterisk.example.com.sip spa841.example.com.sip: UDP, length 476 13:11:30.461541 IP asterisk.example.com.sip spa841.example.com.sip: UDP, length 540 13:11:30.474515 IP spa841.example.com.sip asterisk.example.com.sip: UDP, length 383 13:11:30.497854 IP spa841.example.com.sip asterisk.example.com.sip: UDP, length 319 sip debug at the Asterisk console will show the messages as the are received and responded to by Asterisk. It should look something like: -- SIP read from 192.168.0.19:5060: SIP/2.0 200 OK To: sip:spa...@192.168.0.19:5060;tag=d732d5ba46660f68i0 From: asterisk sip:aster...@192.168.0.1;tag=as51d58666 Call-ID: 7e81b5850a48114430b5bd505bfd3...@192.168.0.1 CSeq: 102 NOTIFY Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK25449e4a Server: Sipura/SPA841-3.1.4(a) Content-Length: 0 --- (8 headers 0 lines) --- Destroying call '7e81b5850a48114430b5bd505bfd3...@192.168.0.1' 12 headers, 0 lines Reliably Transmitting (no NAT) to 192.168.0.19:5060: OPTIONS sip:spa...@192.168.0.19:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK6d5660c5 From: asterisk sip:aster...@192.168.0.1;tag=as079a9a44 To: sip:spa...@192.168.0.19:5060 Contact: sip:aster...@192.168.0.1 Call-ID: 16bb21000690e22e53bff2f90b43d...@192.168.0.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 13 Apr 2009 20:18:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update
[asterisk-users] duration of rfc2833 generated dtmf
Hi. I have a SIP provider which wants RFC2833 for the dtmfmode, however I would like to increase the duration of the tone, its pretty short and some IVR's are unhappy or don't detect it. I did poke around, but it looks like when RFC2833 is used, it actually generates rtp packets of some sort, so I have no idea how to increase that duration. Any assistance would be appreciated. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com ___ -- 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-beginner : cannot make phonecalls using Asterisk (update)
On Mon, 13 Apr 2009, jonas kellens wrote: [r...@asterisk asterisk]# /usr/sbin/tcpdump port 5060 tcpdump: verbose output suppressed, use -v or -vv for full protocol decode listening on eth1, link-type EN10MB (Ethernet), capture size 96 bytes 23:04:59.522498 IP 192.168.4.114.sip 192.168.4.248.sip: SIP, length: 530 23:05:01.233460 IP 192.168.4.112.sip 192.168.4.248.sip: SIP, length: 540 I'm assuming that 192.168.4.112 and 192.168.4.114 are your phones. It looks like they are trying to talk to 192.168.4.248. Any chance your Asterisk server is not at this address? What does sudo ifconfig -a show? Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- 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] duration of rfc2833 generated dtmf
John covici wrote: Hi. I have a SIP provider which wants RFC2833 for the dtmfmode, however I would like to increase the duration of the tone, its pretty short and some IVR's are unhappy or don't detect it. I did poke around, but it looks like when RFC2833 is used, it actually generates rtp packets of some sort, so I have no idea how to increase that duration. Any assistance would be appreciated. If your provider insists on rfc2833, then their servers will be responsible for setting the tone duration sent to PSTN lines. ___ -- 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
[asterisk-users] wanpipe 3.2.7.1 Compiling error
Hi everybody! I'm triying to install a Sangoma A200-R FXO card on a Debian Linux 5 (Lenny), 2.6.26 kernel. To install wanpipe driver I type: WANPIPE_FOLDER# ./Setup install Everything seems to be ok. There are no broken dependencies and the hardware is well detected, even zaptel is recompiled with no errors, but at the time to compile wanpipe I'm getting this error message: -- include/linux/wanrouter.h:344: error: expected specifier-qualifier-list before get_info_t -- Could anyone help me on this. Thanks in advance. -- ._._._._._._._._._._._._._._._._._._._._ D.G.S.C.A U.N.A.M Dirección de Telecomunicaciones Proyectos Especiales e Innovación Tecnológica Giovanni Andrés Nopal Pascual www.voip.unam.mx ___ -- 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] Send Re-invite from Dialplan application?
Y On Mon, Apr 13, 2009 at 6:05 AM, Sai P. Varanasi saiprabhak...@yahoo.com wrote: Hi, I have a requirement where an IVR application on asterisk has to play a audio file in g729 and when a digit is pressed, the call should switch to another codec (say ulaw). So, What can I do in the extensions.conf to trigger a re-negotiation of codec? I used exten = 55xx,n,Set(SIP_CODEC=ulaw) but, I suppose this affects the next call and not the current one. Please help ASAP Thanks, Sai Add more friends to your messenger and enjoy! Invite them now. ___ -- 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 ___ -- 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] Send Re-invite from Dialplan application?
sorry, You can set SIP_CODEC before the call is answered ... most likely as one of the first priorities. It causes the 200 OK to INVITE contain the codec you specify as the first one. I'm not aware of reinviting while in a call other than to switchover to T.38 ... it can be coded in but I'm not sure if it's already there Martin On Mon, Apr 13, 2009 at 5:51 PM, Martin asteriskl...@callthem.info wrote: Y On Mon, Apr 13, 2009 at 6:05 AM, Sai P. Varanasi saiprabhak...@yahoo.com wrote: Hi, I have a requirement where an IVR application on asterisk has to play a audio file in g729 and when a digit is pressed, the call should switch to another codec (say ulaw). So, What can I do in the extensions.conf to trigger a re-negotiation of codec? I used exten = 55xx,n,Set(SIP_CODEC=ulaw) ___ -- 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] Voicemail Greetings Will Not Save
Hi All, -My asterisk will not save voicemail greetings when you call in and record them. -It also will not save voicemail messages after emailing them,even though delete=no. -Folder permissions are fine, no errors in asterisk cli. -If i go into /var/spool/asterisk/voicemail/default/200 and touch unavail.wav, and then call in and record new unavail message, unavail.wav disappears? Can anyone help point me towards any possible info to fix this, i'm stumped and losing hair! I am running FreePBX 2.5.1 Asterisk 1.4.23.1 CentOS 5.3 CLI Output during my attempt to call in and record a greeting: -- Executing [...@from-internal:4] Macro(SIP/200-00fd3150, get- vmcontext|200) in new stack -- Executing [...@macro-get-vmcontext:1] Set40m(SIP/200-00fd3150, VMCONTEXT=default) in new stack -- Executing [...@macro-get-vmcontext:2] GotoIf(SIP/200-00fd3150, 0?200:300) in new stack -- Goto (macro-get-vmcontext,s,300) -- Executing [...@macro-get-vmcontext:300] NoOp(SIP/200-00fd3150, ) in new stack -- Executing [...@from-internal:5] MailboxExists37;40m(SIP/ 200-00fd3150, 2...@default) in new stack -- Executing [...@from-internal:6] GotoIf(SIP/200-00fd3150, 1? mbexist) in new stack -- Goto (from-internal,*97,106) -- Executing [...@from-internal:106] VoiceMailMain(SIP/ 200-00fd3150, 2...@default) in new stack -- SIP/200-00fd3150 Playing 'vm-password' (language 'en') == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/manager_additional.conf': Found == Parsing '/etc/asterisk/manager_custom.conf': Found == Manager 'admin' logged on from 127.0.0.1 == Manager 'admin' logged off from 127.0.0.1 -- SIP/200-00fd3150 Playing 'vm-youhave' (language 'en') -- SIP/200-00fd3150 Playing 'vm-no' (language 'en') -- SIP/200-00fd3150 Playing 'vm-messages' (language 'en') -- SIP/200-00fd3150 Playing 'vm-opts' (language 'en') -- SIP/200-00fd3150 Playing 'vm-options' (language 'en') -- Recording the message -- SIP/200-00fd3150 Playing 'vm-rec-unv' (language 'en') == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/manager_additional.conf': Found == Parsing '/etc/asterisk/manager_custom.conf': Found == Manager 'admin' logged on from 127.0.0.1 == Manager 'admin' logged off from 127.0.0.1 -- SIP/200-00fd3150 Playing 'beep' (language 'en') -- x=0, open writing: /var/spool/asterisk/voicemail/default/200/ unavail.tmp format: wav49, 0x97d788 -- x=1, open writing: /var/spool/asterisk/voicemail/default/200/ unavail.tmp format: gsm, 0x9eecd8 -- x=2, open writing: /var/spool/asterisk/voicemail/default/200/ unavail.tmp format: wav, 0xa4f778 == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/manager_additional.conf': Found == Parsing '/etc/asterisk/manager_custom.conf': Found == Manager 'admin' logged on from 127.0.0.1 == Manager 'admin' logged off from 127.0.0.1 -- User ended message by pressing # -- SIP/200-00fd3150 Playing 'auth-thankyou' (language 'en') -- SIP/200-00fd3150 Playing 'vm-review' (language 'en') == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/manager_additional.conf': Found == Parsing '/etc/asterisk/manager_custom.conf': Found == Manager 'admin' logged on from 127.0.0.1 == Manager 'admin' logged off from 127.0.0.1 -- Saving message as is -- SIP/200-00fd3150 Playing 'vm-msgsaved' (language 'en') == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/manager_additional.conf': Found == Parsing '/etc/asterisk/manager_custom.conf': Found == Manager 'admin' logged on from 127.0.0.1 -- SIP/200-00fd3150 Playing 'vm-options' (language 'en') == Manager 'admin' logged off from 127.0.0.1 == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/manager_additional.conf': Found == Parsing '/etc/asterisk/manager_custom.conf': Found == Manager 'admin' logged on from 127.0.0.1 == Manager 'admin' logged off from 127.0.0.1 == Spawn extension (from-internal, *97, 106) exited non-zero on 'SIP/200-00fd3150' -- Executing [...@from-internal:1] Macro(SIP/200-00fd3150, hangupcall) in new stack -- Executing [...@macro-hangupcall:1] ResetCDR(SIP/200-00fd3150, w) in new stack -- Executing [...@macro-hangupcall:2] NoCDR(SIP/200-00fd3150, ) in new stack -- Executing [...@macro-hangupcall:3] GotoIf(SIP/200-00fd3150, 1? skiprg) in new stack -- Goto (macro-hangupcall,s,6) -- Executing [...@macro-hangupcall:6] GotoIf(SIP/200-00fd3150, 1? skipblkvm) in new stack -- Goto (macro-hangupcall,s,9) -- Executing [...@macro-hangupcall:9] GotoIf(SIP/200-00fd3150, 1? theend) in new stack -- Goto (macro-hangupcall,s,11) -- Executing [...@macro-hangupcall:11] Hangup(SIP/200-00fd3150, ) in new stack == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/200-00fd3150' in macro 'hangupcall' == Spawn
Re: [asterisk-users] wanpipe 3.2.7.1 Compiling error
Which Wanpipe version did you download? 2009/4/13 Giovanni Andrés Nopal Pascual giova...@voip.unam.mx: Hi everybody! I'm triying to install a Sangoma A200-R FXO card on a Debian Linux 5 (Lenny), 2.6.26 kernel. To install wanpipe driver I type: WANPIPE_FOLDER# ./Setup install Everything seems to be ok. There are no broken dependencies and the hardware is well detected, even zaptel is recompiled with no errors, but at the time to compile wanpipe I'm getting this error message: -- include/linux/wanrouter.h:344: error: expected specifier-qualifier-list before ‘get_info_t’ -- Could anyone help me on this. Thanks in advance. -- ._._._._._._._._._._._._._._._._._._._._ D.G.S.C.A U.N.A.M Dirección de Telecomunicaciones Proyectos Especiales e Innovación Tecnológica Giovanni Andrés Nopal Pascual www.voip.unam.mx ___ -- 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 -- I do not agree with what you have to say, but I’ll defend to the death your right to say it. Voltaire ___ -- 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] Voicemail Greetings Will Not Save
On Monday 13 April 2009 05:54:38 pm Dr. Kenneth Noisewater wrote: Hi All, -My asterisk will not save voicemail greetings when you call in and record them. -It also will not save voicemail messages after emailing them,even though delete=no. -Folder permissions are fine, no errors in asterisk cli. -If i go into /var/spool/asterisk/voicemail/default/200 and touch unavail.wav, and then call in and record new unavail message, unavail.wav disappears? Can anyone help point me towards any possible info to fix this, i'm stumped and losing hair! You wouldn't happen to have built voicemail with ODBC and/or IMAP support, would you? That would make the most sense, as both of those engines remove recordings from the directory after having sucked them into the relevant backend storage device. -- Tilghman ___ -- 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
[asterisk-users] Changing menuselect values from CLI and not TUI
Hi All, I'm in the process of writing an install script and I would like to change some settings for the install process but I don't want the user to go into menuselect and make the changes manually. Is there a way to make the changes to menuselect from the CLI? As an example, selecting the iLBC codec. menuselect codec ilbc on Regards David. ___ -- 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] Voicemail Greetings Will Not Save
I very probably did build them with ODBC or MySQL support. IMAP I don't think so, but where would I look for configs that tell asterisk to use such support? I'm almost positive I compiled it to support database, but I definitely never configured it for use. Or is this something it does automatically and I need to recompile? Thank you very much for your help. On Apr 13, 2009, at 6:14 PM, Tilghman Lesher wrote: On Monday 13 April 2009 05:54:38 pm Dr. Kenneth Noisewater wrote: Hi All, -My asterisk will not save voicemail greetings when you call in and record them. -It also will not save voicemail messages after emailing them,even though delete=no. -Folder permissions are fine, no errors in asterisk cli. -If i go into /var/spool/asterisk/voicemail/default/200 and touch unavail.wav, and then call in and record new unavail message, unavail.wav disappears? Can anyone help point me towards any possible info to fix this, i'm stumped and losing hair! You wouldn't happen to have built voicemail with ODBC and/or IMAP support, would you? That would make the most sense, as both of those engines remove recordings from the directory after having sucked them into the relevant backend storage device. -- Tilghman ___ -- 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 ___ -- 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] Changing menuselect values from CLI and not TUI
Is a drive image out of the question? PaulH David Klaverstyn wrote: Hi All, I’m in the process of writing an install script and I would like to change some settings for the install process but I don’t want the user to go into menuselect and make the changes manually. Is there a way to make the changes to menuselect from the CLI? As an example, selecting the iLBC codec. menuselect codec ilbc on Regards David. ___ -- 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 ___ -- 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
[asterisk-users] dynamic menus in dialplan
I have an application that needs to vary the menu choices available based upon the availability of an external resource at a given time. What I have in mind is a system that can uplink a user to one of many different satellites. Due to the nature of orbital mechanics a satellite may be out of range at any given time. I only want to present a menu of available satellites. I can query an external program for a list of available satellites, but how can I use that list to present menu options for selection? What's the best way of doing this? Does anyone know of similar examples? Thanks, Eric ___ -- 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] wanpipe 3.2.7.1 Compiling error
Duh, read the subject. I suggest to try 3.3.16 beta, given that is probably a kernel version issue. On Mon, Apr 13, 2009 at 7:15 PM, Moises Silva moises.si...@gmail.com wrote: Which Wanpipe version did you download? 2009/4/13 Giovanni Andrés Nopal Pascual giova...@voip.unam.mx: Hi everybody! I'm triying to install a Sangoma A200-R FXO card on a Debian Linux 5 (Lenny), 2.6.26 kernel. To install wanpipe driver I type: WANPIPE_FOLDER# ./Setup install Everything seems to be ok. There are no broken dependencies and the hardware is well detected, even zaptel is recompiled with no errors, but at the time to compile wanpipe I'm getting this error message: -- include/linux/wanrouter.h:344: error: expected specifier-qualifier-list before ‘get_info_t’ -- Could anyone help me on this. Thanks in advance. -- ._._._._._._._._._._._._._._._._._._._._ D.G.S.C.A U.N.A.M Dirección de Telecomunicaciones Proyectos Especiales e Innovación Tecnológica Giovanni Andrés Nopal Pascual www.voip.unam.mx ___ -- 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 -- I do not agree with what you have to say, but I’ll defend to the death your right to say it. Voltaire -- I do not agree with what you have to say, but I’ll defend to the death your right to say it. Voltaire ___ -- 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] dynamic menus in dialplan
On Mon, 13 Apr 2009, Eric Fort wrote: I have an application that needs to vary the menu choices available based upon the availability of an external resource at a given time. What I have in mind is a system that can uplink a user to one of many different satellites. Due to the nature of orbital mechanics a satellite may be out of range at any given time. I only want to present a menu of available satellites. I can query an external program for a list of available satellites, but how can I use that list to present menu options for selection? What's the best way of doing this? Does anyone know of similar examples? If availability is defined purely by time, Asterisk has an IFTIME function available in the dialplan. If availability is defined by a ping, write a daemon to ping the birds that should be in view and update that status in a database. When a call is processed, call an AGI that queries the database. Depending on what your needs are, either have the AGI set a series of channel variables to be processed by the dialplan or have the AGI present the options and accept the caller's choice. I've done both. My current project is a third party verification system. Dozens of channel variables (STEP-x-PROMPT, STEP-x-RETRY-PROMPT, STEP-x-MAX-LENGTH, etc.) are set in an AGI based on the DNIS. The channel variables then control processing in an AEL while loop. Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- 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] dynamic menus in dialplan
hi
you can use AGI or a database internal or external
then if you know all the satellites and are a few you can
if(${SAT1}=1)
playback(SAT1)
if(${SAT2}=1)
playback(SAT2)
.
.
.
or you can use an agi
David
2009/4/14 Eric Fort eric.f...@gmail.com
I have an application that needs to vary the menu choices available based
upon the availability of an external resource at a given time. What I have
in mind is a system that can uplink a user to one of many different
satellites. Due to the nature of orbital mechanics a satellite may be out
of range at any given time. I only want to present a menu of available
satellites. I can query an external program for a list of available
satellites, but how can I use that list to present menu options for
selection? What's the best way of doing this? Does anyone know of similar
examples?
Thanks,
Eric
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Re: [asterisk-users] Asterisk-beginner : cannot make phonecalls using Asterisk (update)
2009/4/14 jonas kellens jonas.kell...@telenet.be [r...@asterisk asterisk]# netstat -a -n -p | grep 5060 udp0 0 0.0.0.0:50600.0.0.0:* 3047/asterisk [r...@asterisk asterisk]# /usr/sbin/tcpdump port 5060 tcpdump: verbose output suppressed, use -v or -vv for full protocol decode listening on eth1, link-type EN10MB (Ethernet), capture size 96 bytes 23:04:59.522498 IP 192.168.4.114.sip 192.168.4.248.sip: SIP, length: 530 23:05:01.233460 IP 192.168.4.112.sip 192.168.4.248.sip: SIP, length: 540 23:05:23.521076 IP 192.168.4.114.sip 192.168.4.248.sip: SIP, length: 530 23:05:24.520486 IP 192.168.4.114.sip 192.168.4.248.sip: SIP, length: 530 23:05:25.232068 IP 192.168.4.112.sip 192.168.4.248.sip: SIP, length: 540 23:05:26.231229 IP 192.168.4.112.sip 192.168.4.248.sip: SIP, length: 540 23:05:26.520308 IP 192.168.4.114.sip 192.168.4.248.sip: SIP, length: 530 23:05:28.231050 IP 192.168.4.112.sip 192.168.4.248.sip: SIP, length: 540 23:05:30.519957 IP 192.168.4.114.sip 192.168.4.248.sip: SIP, length: 530 23:05:32.230693 IP 192.168.4.112.sip 192.168.4.248.sip: SIP, length: 540 23:05:34.521843 IP 192.168.4.114.sip 192.168.4.248.sip: SIP, length: 925 23:05:34.530587 IP 192.168.4.114.sip 192.168.4.248.sip: SIP, length: 530 23:05:35.519255 IP 192.168.4.114.sip 192.168.4.248.sip: SIP, length: 925 23:05:36.230336 IP 192.168.4.112.sip 192.168.4.248.sip: SIP, length: 540 23:05:37.519077 IP 192.168.4.114.sip 192.168.4.248.sip: SIP, length: 925 23:05:41.518720 IP 192.168.4.114.sip 192.168.4.248.sip: SIP, length: 925 Connected to Asterisk 1.4.24 currently running on asterisk (pid = 3047) Verbosity is at least 3 asterisk*CLI sip debug SIP Debugging re-enabled asterisk*CLI and it stays that way... So far, what you've provided as information suggests that asterisk is running, that it is listening on the SIP port, udp 5060, that the phones are sending information to asterisk, the machine asterisk is running on can see the packets in a tcpdump, but, asterisk never sees what the phones are sending going back the lack of anything showing up in the asterisk cli... Make sure you set at least udp port 5060 in your iptables configuration as ACCEPTed... To do it as a one off, you can do 'iptables -I INPUT -i eth1 -p udp -m udp -s 192.168.4.0/24 --dport 5060 -j ACCEPT', though unless you change your configuration this will not survive across a reboot... This wouldn't necessarily be enough to get SIP working, but, the phones should at least register and you should be able to see them at least try to make calls... If you get that far, you'll also need to open a range of ports for RTP... d ___ -- 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] Voicemail Greetings Will Not Save
Hi All, Just wanted to post a follow up in case anyone else has the same issue in the future. I recompiled Asterisk and in the makemenu system there is a Voicemail Build Options, in there there is []ODBC Storage and []IMAP Storage. I had ODBC Storage checked on my last compile, I unchecked it, finished building and it all works now. Apparenlty this does not install the option of using ODBC storage, it commits you to ODBC storage without any additional configuration. Tilghman, thanks, your question is what ultimately led me to my solution. Respectfully, Dr. Kenneth Noisewater, Phd On Apr 13, 2009, at 6:14 PM, Tilghman Lesher wrote: On Monday 13 April 2009 05:54:38 pm Dr. Kenneth Noisewater wrote: Hi All, -My asterisk will not save voicemail greetings when you call in and record them. -It also will not save voicemail messages after emailing them,even though delete=no. -Folder permissions are fine, no errors in asterisk cli. -If i go into /var/spool/asterisk/voicemail/default/200 and touch unavail.wav, and then call in and record new unavail message, unavail.wav disappears? Can anyone help point me towards any possible info to fix this, i'm stumped and losing hair! You wouldn't happen to have built voicemail with ODBC and/or IMAP support, would you? That would make the most sense, as both of those engines remove recordings from the directory after having sucked them into the relevant backend storage device. -- Tilghman ___ -- 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 ___ -- 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] retransmision error con asterisk 1.4.24.1
2009/4/12 Martin asteriskl...@callthem.info: 1) your asterisk box talks to OpenSIPS yes , he talk with opensips 2) in that case OpenSIPS should traverse NAT no , my users are of opensips , asterisk is set mode comedia 3) you should not do nat=yes for that device since Asterisk talks to OpenSIPS (but then it might not matter) Either take OpenSIPS out of the way or configure NAT traversal w/media and it should work Martin before spending to this asterisk version, this scenario works me very well, alone it upgrade the asterisk version and pum the problem regardss.. -- rickygm http://gnuforever.homelinux.com ___ -- 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
