[asterisk-users] Update callee num or name at caller display
Hi, A calls B and B has it's phone forwarded to C. So the call rings at C. Is there any way to inform A about that forwarding? Best way would be to update the called name so A has B forwarded to C in his display. Any chance to get this? I tried Set(REDIRECTING(to-name)=...). This sends a SIP/2.0 181 Call is being forwarded to the calling phone, but with no information about the new callee name. Regards, Gunnar -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Update callee num or name at caller display
On 01/16/2012 12:55 PM, Olivier wrote: I've never done it myself yet but I think I would look after COLP function (1.8 and above). Asterisk 1.8 and later will do this automatically; if the phone can display the redirection information, it will get displayed (not all phones can do so). Thanks for the feedback. Any documentation abount COLP? On voip-info.org there is noting. The redirection is done in Asterisk dialplan, so I have to tell phone A about the forwarding. exten = B,1,Dial(SIP/C) So I need a dialplan function or something else to send an update to phone A. Regards, Gunnar -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Update callee num or name at caller display
Hello Eric, See http://www.voip-info.org/wiki/view/Asterisk+func+CONNECTEDLINE pay special attention to the sendrpid note. That does not work. CONNECTEDLINE is for answered calls. A calls B. B has a forward to C in Asterisk dialplan. A want's to notice the forwarding _before_ C answers. Cause A only want to speak to B. Sorry if that was not clear before. Regards, Gunnar -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Update callee num or name at caller display
Are both A and B extensions of the same Asterisk system or is A an inbound caller ? Both are snom phones at the same Asterisk (1.8.8). Regards, Gunnar -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] After Monitor() files disappear
I'm moving over to asterisk-dev. Seems to be a bug. Greetings, Gunnar Schaller ___ -- 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] After Monitor() files disappear
Hello list, Using Asterisk 1.2.29 I use the Monitor() application. In extensions.conf I have set MONITOR_EXEC to my script (for mixing files together and convert to mp3) and I set TOUCH_MONITOR on every new channel which has to be recorded. But sometimes I'm missing the recording files. I had a look to the Asterisk-Log and saw those lines: Feb 10 15:18:57 NOTICE[16772] res_monitor.c: monitor executing /root/recordings //var/spool/asterisk/monitor/X_20090210-151258-in.wav //var/spool/asterisk/monitor/X_20090210-151258-out.wav //var/spool/asterisk/monitor/X_20090210-151258.wav X is the phone-number. Exactly these recordings with two slashes at the beginning of the parameters (//var/spool...) I'm missing. My script does not delete them (for testing I cleared my script so it does nothing). While Monitor is running there are files in /var/spool/asterisk/monitor. Asterisk definitly records the call. Seems the files disappear after the channel is hung up. It must be an Asterisk problem. Anyone out there with an idea or a hint? I knew Asterisk 1.2 is out of date. But I can't believe that it is an Asterisk problem and no one else had the same problem. Thanks, Gunnar Schaller ___ -- 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] Monitor and SIP transfers (SIP REFER)
The problem in this particular case is that the actual monitor object is on A's channel. When A is no longer involved in the call, the monitor is gone, and so the call cannot be recorded further. One possible solution is to run the Monitor application on B's channel instead. This can be done by using the M option in the Dial application. The M option allows you to run a macro on the *called* channel's party when he answers. If you start the Monitor application from this macro, you should find that things will work as you expect. Note that the issue you linked was about MixMonitor, not Monitor. They are completely different beasts when it comes to how they operate. In fact, MixMonitor recordings can be set to survive a transfer if you are using Asterisk 1.4.23 and make use of the AUDIOHOOK_INHERIT function. For more information on its use, you can issue the command core show function AUDIOHOOK_INHERIT from the Asterisk CLI. Mark Michelson Thank you very much for your explanation! Gunnar Schaller ___ -- 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] Monitor and SIP transfers (SIP REFER)
Hello list, I need to record all calls. So I'm using application Monitor. Works good until someone transfers a callee to another internal extension. Example: A calls B A set B on hold A calls C A transfers B to C with SIP transfer (SIP REFER - with phone funktions and not Asterisk attended transfer). I found http://bugs.digium.com/view.php?id=0013538 . corruptor asked about this problem, but it seems there is no solution. Now I want to know how anyone deals with this problem. How to record those transfered calls? Any solution with manager commands or some source-code hacking (enabling Monitor for all calls so no Monitor is needed in dialplan). I'm working with Snom phones here - so there is the possibility to work with action url's. Thank you, Gunnar ___ -- 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] Gemeinschaft released
Hello, Do I need any Asterisk Patches? I had a look to the source code, specially the provisioning system. Not very readable, no classes and many code lines commented. The cluster capability is very interesting. Regards, Gunnar Schaller Hi, Just wanted to let you know that we have just made our GPL toolkit Gemeinschaft available to the public. (Finally.) Mostly German for now - about half of the strings in the language strings file have been translated to English. I'm a software developer, not a marketing guy, so ... svn co https://svn.amooma.de/gemeinschaft/trunk gemeinschaft-trunk German readers: see http://www.amooma.de/gemeinschaft/ Regards, Philipp Kempgen ___ -- 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] Call forward at telco
Friday, August 10, 2007, 3:12:39 AM, you wrote:
This doesn't work?
exten = _*21*X.,1,Dial(Zap/1/*21*${EXTEN:4})
Then you can dial
*21*destination#
then just push 'send' on your SIP phone and the system will dial it out
for you... ??
this is working for me in .nl, and from what I know .nl and
.de are very simular.
I did a test in Germany, but it is for a customer in Switzerland. And
in Switzerland it doesn't work.
I patched libpri with some backports of the Asterisk function
ZapSendKeypadFacility from Asterisk 1.4 now (Asterisk 1.2 here with
Bristuff). Seems to work, but needs more testing. Would be fine to
have something similar as this function, but for unanswered channels
in Asterisk 1.6.
Regards,
Gunnar
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Re: [asterisk-users] Call forward at telco
Hello Gordon,
Thursday, August 9, 2007, 4:39:44 PM, you wrote:
This doesn't work?
exten = _*21*X.,1,Dial(Zap/1/*21*${EXTEN:4})
Then you can dial
*21*destination#
No that doesn't work. You can't dial this number. You have to send
special facility keypads to telco switch. Normal dialing would
signalling as called number, not as facility keypads.
pri debug span with called number (5 here):
4 Protocol Discriminator: Q.931 (8) len=8
4 Call Ref: len= 1 (reference 4/0x4) (Originator)
4 Message type: INFORMATION (123)
4 [70 02 81 35]
4 Called Number (len= 4) [ Ext: 1 TON: Unknown Number Type (0) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1) '5' ]
4 -- Processing IE 112 (cs0, Called Party Number)
pri debug span with facility keypad (a * in this case):
4 Protocol Discriminator: Q.931 (8) len=7
4 Call Ref: len= 1 (reference 1/0x1) (Originator)
4 Message type: INFORMATION (123)
4 [2c 01 2a]
4 Keypad Facility (len= 1) [ *DÏÈN ]
4 -- Processing IE 44 (cs0, Keypad Facility)
ZapSendKeypadFacility in Asterisk 1.4 does this IN a call. But I do
not have a call. I have to pick up the line and send the information.
After # at the and a voice is telling me service activated or try
again.
Gunnar
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Re: [asterisk-users] Call forward at telco
Hello Gordon,
Thursday, August 9, 2007, 4:39:44 PM, you wrote:
This doesn't work?
exten = _*21*X.,1,Dial(Zap/1/*21*${EXTEN:4})
Then you can dial
*21*destination#
No that doesn't work. You can't dial this number. You have to send
special facility keypads to telco switch. Normal dialing would
signalling as called number, not as facility keypads.
pri debug span with called number (5 here):
4 Protocol Discriminator: Q.931 (8) len=8
4 Call Ref: len= 1 (reference 4/0x4) (Originator)
4 Message type: INFORMATION (123)
4 [70 02 81 35]
4 Called Number (len= 4) [ Ext: 1 TON: Unknown Number Type (0) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1) '5' ]
4 -- Processing IE 112 (cs0, Called Party Number)
pri debug span with facility keypad (a * in this case):
4 Protocol Discriminator: Q.931 (8) len=7
4 Call Ref: len= 1 (reference 1/0x1) (Originator)
4 Message type: INFORMATION (123)
4 [2c 01 2a]
4 Keypad Facility (len= 1) [ *DÏÈN ]
4 -- Processing IE 44 (cs0, Keypad Facility)
ZapSendKeypadFacility in Asterisk 1.4 does this IN a call. But I do
not have a call. I have to pick up the line and send the information.
After # at the and a voice is telling me service activated or try
again.
Gunnar
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[asterisk-users] Call forward at telco
Hello, I want to enable call forwarding at my telco. In Germany you can press *21*destination# and all calls will be redirected to the destination without interaction with any equipment on my side. How to dial this with Asterisk and Zap-Channels? It can not be send as called number, it has to be send as keypad facility. Anyone here with some hints? The application ZapSendKeypadFacility in Asterisk 1.4 only supports answered channels if I read it correctly. But my channel is not answered before sending *21*destination# (I get a voice telling me the call forwarding is activated). Thanks, Gunnar ___ --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] Where to find t38modem
Same site, just a few lines later: ... you could run Asterisk and Hylafax with T38modem (by www.openh323.org) on the same box and terminate T.38 calls ... Gunnar Hello, From http://www.voip-info.org/wiki/view/Asterisk+fax you can read: *Update Jul 2007:* For a T.38 gateway you can use Asterisk 1.4's T.38pass-through support in combination with the new OPAL (Open Phone Abstraction Library) - using t38modem (currently CVS) which now supports SIP (and not just H.323) to terminate T.38 calls. You can also use OPAL and chan_woomera to do essentially the same. Where can you find this t38modem stuff ? Google replies things that doesn't seem to match (T.38-SIP termination). 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
[asterisk-users] Problem with Remote-Hold/MusicOnHold
Hello, I have a problem with MoH at attended transfers. - Mobile A dials into Asterisk - Asterisk dials another Mobile B - Mobile B presses *1 for attended transfer and for example 20 to dial extension 20 - Asterisk sends Remote hold message to Mobile A, so the carrier of Mobile A starts playing it's own music-on-hold - Mobile B hang up, so Mobile A should be connected to extension 20. But Asterisk doesn't send Remote retrieve, so the mosic-on-hold of the carrier doesn't stop. You can hear the person on Mobile A, but the person on Mobile A only gets music. Anybody with the same issue? It happens on a bristuffed Asterisk 1.2.19. Is there a way to stop sending Remote hold or a way to send Remote retrieve? See here the the cli output with pri debug span 4 at starting attended transfer: 4 Protocol Discriminator: Q.931 (8) len=7 4 Call Ref: len= 1 (reference 214/0xD6) (Terminator) 4 Message type: NOTIFY (110) 4 [27 01 f9] 4 Notification indicator (len= 3): Ext: 1 Remote hold (121) -- Started music on hold, class 'default', on channel 'Zap/10-1' -- Playing 'pbx-transfer' (language 'de') Best regards, Gunnar Schaller ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with Remote-Hold/MusicOnHold
Hello, I have a problem with MoH at attended transfers. - Mobile A dials into Asterisk - Asterisk dials another Mobile B - Mobile B presses *1 for attended transfer and for example 20 to dial extension 20 - Asterisk sends Remote hold message to Mobile A, so the carrier of Mobile A starts playing it's own music-on-hold - Mobile B hang up, so Mobile A should be connected to extension 20. But Asterisk doesn't send Remote retrieve, so the mosic-on-hold of the carrier doesn't stop. You can hear the person on Mobile A, but the person on Mobile A only gets music. Anybody with the same issue? It happens on a bristuffed Asterisk 1.2.19. Is there a way to stop sending Remote hold or a way to send Remote retrieve? See here the the cli output with pri debug span 4 at starting attended transfer: 4 Protocol Discriminator: Q.931 (8) len=7 4 Call Ref: len= 1 (reference 214/0xD6) (Terminator) 4 Message type: NOTIFY (110) 4 [27 01 f9] 4 Notification indicator (len= 3): Ext: 1 Remote hold (121) -- Started music on hold, class 'default', on channel 'Zap/10-1' -- Playing 'pbx-transfer' (language 'de') Best regards, Gunnar Schaller ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR on transfers of called ZAP channel
Hello Steve,
Thank you for the detailed answer. The first solution you mentioned
seems to be good enough for me. So I have to wait for Asterisk 1.6.
That's bad, but I have to wait.
My hope was a way with Asterisk 1.2 (or 1.4) and CDR-functions like
ForkCDR or with some local channels. I worked on a mobile-integration
solution. In Switzerland you can have Team Unlimited, that's a
mobile option. Free calls from fixed company lines to the employees
mobile. It's like DECT phones, but not limited to DECT base stations
:-)
But without correct CDR I have to review the whole plan.
Thanks for your help,
Gunnar
Monday, June 11, 2007, 5:11:18 PM, you wrote:
Gunnar--
CDR generation that covers transfers is an umimplemented feature in
Asterisk, in any version.
I have been working on a solution, but unfortunately, my solution is
radical enough that I dare not apply it to 1.2 or even 1.4. It will most
likely break every working implementation of billing that has been built
on Asterisk by end users/developers. Unpleasant visions of angry mobs of
developers armed with baseball bats, who want nothing more than to drag
me out of my home and share their pain and frustration over my
fixes. you get the idea.
Actually, I have TWO solutions! One, is to modify the current CDR
engine, the other is to provide an entirely different solution that is
single-event driven, kinda along the lines of manager events, but more
streamlined for CDR billing purposes.
The first solution somewhat reorganizes CDRS by no longer posting them
to the backend db's when a hangup occurs. Rather, it will post them when
a bridge between channels is finished, or ends. Since a Local
channel acts as a sort of bridge, I think I will have to do the same
thing there. I'm in the middle of it now. I spent/wasted a good amount
of time generating extra CDR's that would describe time in different
parts of a transfer, but as I traveled further down that road, I see
that this will only make things unnecessarily complex. So, I'm not going
to do it. What this means is that a CDR will get generated for each
chunk of a conversation involved in a transfer, but these pieces will
not tell you much about how the chunks relate to each other. The channel
originating the conversation will be the source, and the channel
originally connected to will be the destination. Time spent in 3-way
conferences, music on hold, etc. etc. will most likely not be available.
My theory is that, in most cases, it won't matter. All you REALLY want
to know is who to bill, and for how much time. If a transfer occurs, it
involves someone internally dialing another party. This second
conversation, will generate another CDR, and the guy who dialed it
will be assigned that call, even if he hung up before the call was
answered (blind xfer). For example, picture this: a switch in Modesto
gets a call from Sacramento, and extension 151 gets this call, and dials
Shanghai, and blind transfers the Sacramento call to Shanghai, and then
Sacramento and Shanghai talk for an hour. Two CDR's will be generated.
One will cover the incoming call from Sacramento, and will be little
over an hour. The other CDR that will come out will say 151 dialed
Shanghai and talked an hour. That's it.
The second solution, the event-based one, will generate an event record
for each significant event in the life of each channel. So, START
events when a channel is born; ANSWER events when someone answers a
call; END events when somebody hangs up. There will also be Park,
and Transfer, and MOH, and 3-WAY, Conference-Join, and several
others. Just enough information will be included with each event to
thread together billable sequences. Along with each event record will be
the time the event happened, and channel info. This approach will be
very much more fine-grained, and allow you to do fancy things like
figure out that Sacramento was the only person talking to Shanghai, and
allow you to bill the call to the guy/gal in Sacramento. Trouble with
this approach is that threading together the event records is a
non-trivial operation! But I hope to provide some tools that will make
this easier to do.
So, the bad news is: you will not see any solutions for this problem, in
1.2, or 1.4. the CDR fix (first solution) will most likely end up in
1.6, the event-based solution will probably not be available until 1.8
or 1.10; we shall see.
murf
On Mon, 2007-06-11 at 14:26 +0200, Gunnar Schaller wrote:
Hello list,
I have a problem with called ZAP channels making an attended-transfer
or blind-xfer. Signalling at the phones is ok, but the CDR of Asterisk
is wrong.
At the moment there is a bristuffed Asterisk 1.2.18 running with
bristuff-0.3.0-PRE-1y-g. Here is my dialplan, I simplified it a bit:
[default]
exten = 0123456789,1,Macro(dialpstn,${EXTEN})
[macro-dialpstn]
exten = s,1,Set(TRANSFER_CONTEXT=transfer)
exten = s,2,Set(FORWARD_CONTEXT=transfer)
exten = s,3
[asterisk-users] CDR on transfers of called ZAP channel
Hello list,
I have a problem with called ZAP channels making an attended-transfer
or blind-xfer. Signalling at the phones is ok, but the CDR of Asterisk
is wrong.
At the moment there is a bristuffed Asterisk 1.2.18 running with
bristuff-0.3.0-PRE-1y-g. Here is my dialplan, I simplified it a bit:
[default]
exten = 0123456789,1,Macro(dialpstn,${EXTEN})
[macro-dialpstn]
exten = s,1,Set(TRANSFER_CONTEXT=transfer)
exten = s,2,Set(FORWARD_CONTEXT=transfer)
exten = s,3,Set(CIDNUMORIG=${CALLERID(num)}) ; save callerid-num
exten = s,4,Set(CALLERID(num)=98765) ; dial out using msn 98765
exten = s,5,Dial(Zap/g1/${ARG1}|30|t)
exten = h,1,Set(CALLERID(num)=${CIDNUMORIG}) restore callerid-num for
CDR
[transfer]
exten = _X.,1,Set(CALLERID(all)=External 0123456789)
exten = _X.,1,Dial(SIP/${EXTEN})
So I call 0123456789 with SIP phone 10. The callee dials *1 20 for
attended transfer and SIP phone 20 (I have *1 for attended transfer in
features.conf). The called SIP-phone shows the caller-information I
set in context transfer. But the CDR is wrong, it has 98765 in MySQL
field src. So it seems I can't overwrite the ${CALLERID(num)} for cdr,
but for the called channel.
Anybody who can explain that? Or any solution for called Zap channels
making an attended transfer?
--
Best regards,
Gunnar Schaller
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Re[2]: [asterisk-users] Re: Verizon Interconnection
Hello, Switzerland is good :-) My company in Switzerland has a contract with Verizon for wholesale in pstn and we are thinking about SIP. Can you please tell me your experience with their SIP product? Regards, Gunnar Schaller Wednesday, June 6, 2007, 2:55:54 PM, you wrote: HI, yes we are interconnected with Verizon in SIP, but we are in europe (Switzerland) so I don't know If it is the same process in USA ... Laurent 2007/6/6, Matt [EMAIL PROTECTED]: So absolutely no one here was interconnected with Verizon? I am going to shoot this over to asterisk-biz, also, in hopes someone may have missed it that is on the biz list. The question again is: Has anyone on this list connected with Verizon's SIP product? We are currently undergoing interop testing with Verizon, and honestly, it seems like the most convoluted process. I'd be interested in talking with someone else who has gone through this and run a few things past you. ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [asterisk-users] openvz resources
Hello, Can you tell more about Xen? I would like to install Debian Etch with Xen and use A Digium 4-port E1 in a guest domain. Is it possible? I read of much problems with cards in a guest domain. I have Xen running with DNS-server/ Web-server guests, also a VoIP only Asterisk, but a telephony card is missing in a guest. Gunnar Schaller Saturday, April 14, 2007, 1:01:07 AM, you wrote: No relevant experience with OpenVZ, but plenty with Xen if you would find that interesting. Bryan Johns Partner Shelton | Johns Office: 678.248.2637 FindMe: 678.229.1809 http://www.sheltonjohns.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dell Poweredge SC430 and Digium cards compatability enquiry
Hello Matthew, It depends on the chipset on the mainboard. I had problems with a SC1420, the only way to solve it was to get a new server (without Intel chipset). So don't try a chipset which is listed on the Digium compatibility site. Wednesday, September 6, 2006, 8:55:58 AM, you wrote: We're looking at using a number of Dell Poweredge SC430 servers as Asterisk hosts in our smaller overseas offices with Digium cards in to provide local breakout over the pre-existing analogue or digital phone lines (One office uses ISDN2 the others analogue) I note that the SC420 is listed as incompatible but the SC430 appears to be a slightly different beast in terms of chipset, the 430 has the newer E7230 as opposed to the E7221 - does this make a difference to compatibility? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Softphone in German
Hello, X-Lite in German: http://www.globalipphones.com/xlite Gunnar ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] E1 signalling pridialplan
Hello, I have a little problem with signalling. An E100p is connected to an Alcatel PBX, wich has an E1 to the outside. Located in Germany. zapata.conf: switchtype=euroisdn pridialplan=local prilocaldialplan=local overlapdial=yes signalling=pri_cpe With asterisk 1.0.2 I can call from a SIP phone to a phone connected to the Alcatel and the SIP number is correctly displayed at the caller. Let's say SIP phone is 12345, then 12345 is displayed. Asterisk cvs head 2004.12.16.23.00.00 seems to ignore my settings in zapata.conf, the phone connected to the Alcatel displays 0012345. Can anybody give me a hint with the 00's? I think pridialplan and prilocaldialplan are set correctly. Another question: When a SIP phone calls to the pstn, how to set the correct callerid? I tried SetCIDNum (012345) and SetCIDNum(12345), but it did't work. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicemail and ast_data
Hi, I have a problem with Voicemail. Asterisk 1.0.1 patched with ast_data connected to a mysql-server, mailboxes are in a mysql database. When I call to VoicemailMain to hear my messages it don't tell me the time the message was left. Only Message 1 and then the message. For testing I defined a mailbox in voicemail.conf. Hearing messages from this account in VoicemailMain tells me the time the message was left. For example Message one, received thuesday . and then the message. There are no special options I set. Can anybody give me a hint? Gunnar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail and ast_data
Thanks a lot, that was the option I needed! Gunnar Gunnar, Add the voicemail options you want to use for that mailbox to the users table as text with multiple option=value pairs separated by a '|' (pipe) character just as it was in the voicemail.conf file. I believe the envelope=yes option is what you want to set. Karl Putz Forte Communications -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Gunnar Schaller Sent: Thursday, October 21, 2004 10:41 AM To: Asterisk Users Mailing List Subject: [Asterisk-Users] Voicemail and ast_data Hi, I have a problem with Voicemail. Asterisk 1.0.1 patched with ast_data connected to a mysql-server, mailboxes are in a mysql database. When I call to VoicemailMain to hear my messages it don't tell me the time the message was left. Only Message 1 and then the message. For testing I defined a mailbox in voicemail.conf. Hearing messages from this account in VoicemailMain tells me the time the message was left. For example Message one, received thuesday . and then the message. There are no special options I set. Can anybody give me a hint? Gunnar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: ast_data and dialplan in mysql
+++ Gunnar Schaller [14/10/04 22:40 +0200]: I have also ast_data and extensions in mysql, works fine for me. Your crash might be something other, I dont't think it's ast_data. Do you have more infos to the crash? Logfiles? Gunnar what kind of log files can i provide here. I mean i'm not so familiar with asterisk log files. btw do u use the latest asterisk cvs version or some specific version I mean /var/log/asterisk, in some circumstances the log-files there might help you. My asterisk-server runs with version 1.01, because chan_capi didn't compile with latest cvs (the day I tried it). Normally I have latest cvs... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ast_data and dialplan in mysql
I have also ast_data and extensions in mysql, works fine for me. Your crash might be something other, I dont't think it's ast_data. Do you have more infos to the crash? Logfiles? Gunnar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] conf from pgsql database
Hai there, Is it possible to save all the configuration of the asterisk on the pgsql database? or just the cdr record? best regards, Freddy Setiawan ~SimpleWare Solusion~ Read the wiki: http://www.voip-info.org/tiki-index.php?page=Asterisk+configuration+from+database And this: http://svn.asteriskdocs.org/res_data/ Gunnar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail
J Which voicemail is current and latest? J Voicemail J or J Voicemail2 I think Voicemail ist the latest. Greetings Gunnar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ast_data, mysql, md5 hashes for passwords
I have asterisk (CVS 7.06.2004) with newest ast_data. My sip-users are in a MySQL database. Is there a possibility to store the passwords (column secret) as md5 hashes? In that way that asterisk takes the password from a client which want to log in, do a md5 hash and compare that with the one in the database? I don't want them plain text in the database because of security reasons. At the moment I plan a web-gui for sip-users to register and making changes on their accounts themself. It's for my university. If there is no way with the current source-code, can anybody give me hints where to do it? Thanks, Gunnar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] german localization for mailbox available?
Hello, Do you know this site? http://www.karl.aegee.org/asterisk.nsf/HT/sound-de Gunnar hi, i just wanted to ask if there is a german localization for the audio files of the mailbox available on the net. regards frank sautter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] Syntax for 2 ISDN Cards
When I try to make 3 simultaneous connections from SIP to ISDN the first and second one works, but on the third connection this happens: -- Executing Dial(SIP/gunnar-26ea, CAPI/7501:7986:bBYEXTENSION) in new stack chan_capi.c:1147 capi_request: didn't find capi device with outgoing msn = 7501. you should check your config! app_dial.c:673 dial_exec: Unable to create channel of type 'CAPI' == Everyone is busy at this time -- Executing Dial(SIP/gunnar-26ea, CAPI/7502:7986:bBYEXTENSION) in new stack chan_capi.c:1147 capi_request: didn't find capi device with outgoing msn = 7502. you should check your config! app_dial.c:673 dial_exec: Unable to create channel of type 'CAPI' == Everyone is busy at this time Do you have 4 b-channels? (2 Lines with 2 channels) According to your email you are in germany, there you need a 2nd NTBA. Well, I could be wrong at all, just my thoughts. Yes there are 4 b-channels and yes it's germany :o) There are 2 lines with 2 b-channels each. My Asterisk operates at a internal telephone system. As I wrote I can do 2 simultaneous connections, in this case capi info shows that contr1 has no free channels. Gunnar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] Syntax for 2 ISDN Cards
Hello,
Thanks for your capi.conf! It works great! I made the changes,
restarted Asterisk and made 3 calls with success.
Thanks again,
Gunnar Schaller
Hi Gunnar,
here is our capi.conf for two controllers on two different ISDN lines
;
; CAPI config
;
;
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
[interfaces]
msn=041120
incomingmsn=*
controller=1
softdtmf=0
context=default
devices=2
msn=041682
incomingmsn=*
controller=2
softdtmf=0
context=assistenza
devices=2
believe it or not, but you can see in the chan_capi source code,
the creation of the lines are activated by parsing the line devices=
so it seems that MUST be the last line of every interface parameters.
With this capi.conf and two passive AVM controllers (one PCI, une USB)
with hacked drivers, we do have random problems, when we have many calls,
our server hangs and we must reboot. Actually we are trying to understand if
problems are on chan_capi and this capi.conf or on then AVM hack.
Please let me know if this syntax works for you.
Bye.
Francesco Sibilla
- Original Message -
From: Gunnar Schaller [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, June 01, 2004 11:34 PM
Subject: [Asterisk-Users] Syntax for 2 ISDN Cards
Hi there,
I searched in mailinglist and in web, but no answer to my problem...
Only this post with no answers:
http://lists.digium.com/pipermail/asterisk-users/2004-March/038994.html
I'm using CVS Asterisk (05/17/04) with chan_capi 0.3.1. (multiple
controller support). In my Asterisk-box there are 2 Fritzcards
(module for second card compiled with changes on sourcecode found in
the web). capi info shows:
Contr1: 2 B channels total, 2 B channels free.
Contr2: 2 B channels total, 2 B channels free.
Here a snipplet of my capi.conf:
[interfaces]
msn=7501,7502
incomingmsn=*
controller=1,2
devices=2,2
Is that correct? I also tried devices=4.
When I try to make 3 simultaneous connections from SIP to ISDN the
first and second one works, but on the third connection this happens:
-- Executing Dial(SIP/gunnar-26ea, CAPI/7501:7986:bBYEXTENSION) in
new stack
chan_capi.c:1147 capi_request: didn't find capi device with outgoing
msn = 7501. you should check your config!
app_dial.c:673 dial_exec: Unable to create channel of type 'CAPI'
== Everyone is busy at this time
-- Executing Dial(SIP/gunnar-26ea, CAPI/7502:7986:bBYEXTENSION) in
new stack
chan_capi.c:1147 capi_request: didn't find capi device with outgoing
msn = 7502. you should check your config!
app_dial.c:673 dial_exec: Unable to create channel of type 'CAPI'
== Everyone is busy at this time
The interesting part of extensions.conf:
exten = _,1,Dial,CAPI/7501:${EXTEN}:bBYEXTENSION
exten = _,102,Dial,CAPI/7502:${EXTEN}:bBYEXTENSION
Can anyone tell me how to use the B-channels of the second Fritzcard?
Gunnar Schaller
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Re[2]: [Asterisk-Users] Syntax for 2 ISDN Cards
On Wed, Jun 02, 2004 at 09:27:14AM +0200, Gunnar Schaller wrote: ...cut chan_capi.c:1147 capi_request: didn't find capi device with outgoing msn = 7502. you should check your config! app_dial.c:673 dial_exec: Unable to create channel of type 'CAPI' == Everyone is busy at this time Do you have 4 b-channels? (2 Lines with 2 channels) According to your email you are in germany, there you need a 2nd NTBA. Well, I could be wrong at all, just my thoughts. Yes there are 4 b-channels and yes it's germany :o) There are 2 lines with 2 b-channels each. My Asterisk operates at a internal telephone system. As I wrote I can do 2 simultaneous connections, in this case capi info shows that contr1 has no free channels. Next questions: Can you see any messages from 2nd line/card via isdnlog? Can you call your * via the other cards? What msn's do the two established calls use? Do you try to access a 3rd call with msn's from first line? Is it right that you have only two msn's in the capi.conf My problem is solved by Francesco Sibilla in this thread. I changed my capi.conf and now it works. Isdnlog didn't work on my machine, it needs hisax but I don't want to load it because I read anywhere to not load it with Asterisk. Anyway, thanks for your help. Gunnar Schaller ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Syntax for 2 ISDN Cards
Hi there,
I searched in mailinglist and in web, but no answer to my problem...
Only this post with no answers:
http://lists.digium.com/pipermail/asterisk-users/2004-March/038994.html
I'm using CVS Asterisk (05/17/04) with chan_capi 0.3.1. (multiple
controller support). In my Asterisk-box there are 2 Fritzcards
(module for second card compiled with changes on sourcecode found in
the web). capi info shows:
Contr1: 2 B channels total, 2 B channels free.
Contr2: 2 B channels total, 2 B channels free.
Here a snipplet of my capi.conf:
[interfaces]
msn=7501,7502
incomingmsn=*
controller=1,2
devices=2,2
Is that correct? I also tried devices=4.
When I try to make 3 simultaneous connections from SIP to ISDN the
first and second one works, but on the third connection this happens:
-- Executing Dial(SIP/gunnar-26ea, CAPI/7501:7986:bBYEXTENSION) in
new stack
chan_capi.c:1147 capi_request: didn't find capi device with outgoing
msn = 7501. you should check your config!
app_dial.c:673 dial_exec: Unable to create channel of type 'CAPI'
== Everyone is busy at this time
-- Executing Dial(SIP/gunnar-26ea, CAPI/7502:7986:bBYEXTENSION) in
new stack
chan_capi.c:1147 capi_request: didn't find capi device with outgoing
msn = 7502. you should check your config!
app_dial.c:673 dial_exec: Unable to create channel of type 'CAPI'
== Everyone is busy at this time
The interesting part of extensions.conf:
exten = _,1,Dial,CAPI/7501:${EXTEN}:bBYEXTENSION
exten = _,102,Dial,CAPI/7502:${EXTEN}:bBYEXTENSION
Can anyone tell me how to use the B-channels of the second Fritzcard?
Gunnar Schaller
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