Re: [asterisk-users] Strange problem Solved
Sorry, i had a mistake in my dialplan - Original Message - From: Accursio Avona To: asterisk-users@lists.digium.com Sent: Monday, March 10, 2008 6:42 PM Subject: [asterisk-users] Strange problem Hi All, i'm experiencing a strange problem on sip channel. Sometime appens that the sip client ring as if it recieves 3 calls at the same time from the same number, even if thre is only a single call. I'm experiencing that both on the softphone sjphone and on the sip phone Grandstream GXP2000 an on two asterisk box one asterisk v 1.0.7 the second asterisk 1.2.16 I have not idea where to start for debug this Someone can help me? thank's in advance Accursio -- ___ -- 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 -- No virus found in this incoming message. Checked by AVG. Version: 7.5.518 / Virus Database: 269.21.7/1322 - Release Date: 09/03/2008 12.17 ___ -- 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] Strange problem
Hi All, i'm experiencing a strange problem on sip channel. Sometime appens that the sip client ring as if it recieves 3 calls at the same time from the same number, even if thre is only a single call. I'm experiencing that both on the softphone sjphone and on the sip phone Grandstream GXP2000 an on two asterisk box one asterisk v 1.0.7 the second asterisk 1.2.16 I have not idea where to start for debug this Someone can help me? thank's in advance Accursio___ -- 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] ata hook-flash
Hi all, someone ca suggest me an ata device that can send an hook-flash to fxo port from voip? tank's in advance Regards Accursio Avona ___ --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] ata hook-flash
Rich Adamson ha scritto: Accursio Avona wrote: Hi all, someone ca suggest me an ata device that can send an hook-flash to fxo port from voip? The sipura spa3000 can do it. The user (on the fxs port) must double-flash to make the pstn (fxo) port flash. I don't know of a way for a sip device (or any other non-fxs port device) to cause a pstn (fxo) flash. I have to connect an anolg pbx to a remote asterisk, i thought to connect the pbx to the fxo port of an ata device and the ata to asterisk throught the wan. Calls arriving to the pbx can be forwrded to sip phone connected to asterisk. If the called part wants to transfer the call to an extension of the pbx he have to send an hook-flash throuth the ata's fxo port. Now i guess: is an ata the rigth device to solve this problem (is there an ata that can send an hook-flash to fxo port from sip) or i really need a local asterisk with an fxo card that connect to the remote asetrisk? Any help or suggestion is welcome Thank's Regards Accursio Avona ___ --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] meetme and dtmf
Imran Ahmed wrote:
may or may not work, try at your own risk:
1) Use a sip soft phone and set the dtmf mode = inband.
2) In asterisk set the dtmf mode for that soft phone to be rfc2833 or
info. (this is done so that asterisk ignores the inband dtmf on the
sip channel).
3) Design your dialplan such that asterisk should not depend on dtmf
from the sip call.
ex:
exten xxx, 1, dial(zap/g/client_number) //on answer directed to conference room
exten xxx, 2, dial(zap/g/ivr_number) //on answer directed to conference room.
exten xxx, 3, meetme(conference room)
Thank you very much.
I tried sjphone setting clinet and asterisk as above and it seems to
work. I will test it better in the next hours.
I had a look at meetme.c and i found a portion of code that manage dtmf
if ((f-frametype == AST_FRAME_DTMF) (confflags
CONFFLAG_EXIT_CONTEXT)) {
..
..
-
I think this part manage the case of meetme application is called with
p, X or s option,
but maybe also (i'm not sure, i had not the time to study well enough
the source, and over all i'm not a so good c programmer)
that this part of code prevents asterisk to broadcast the sound to other
channels when it is not inband.
Sorry if my bad english make me not very clear.
Anyway, thank you very much to all for your help.
Accursio Avona
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Re: [Asterisk-Users] meetme and dtmf
Kevin P. Fleming wrote: but maybe also (i'm not sure, i had not the time to study well enough the source, and over all i'm not a so good c programmer) that this part of code prevents asterisk to broadcast the sound to other channels when it is not inband. MeetMe is not designed to pass DTMF through between the parties in any case. It may happen if you use inband DTMF and don't have Asterisk actually paying attention to DTMF for any reason, but it's not intended to work that way. This means that if i'd like to use iax2 protocol (i need to integrate, into a propietary crm, calling features though asterisk, and i thougth to use iaxclient dll) i can't pass DTMF through between the parties? If so is it possible to modify meetme.c to avoid this behaviour? or i must use sip protocol. Thank's Accursio Avona ___ --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] meetme and dtmf
Imran Ahmed wrote:
On 2/1/06, Kevin P. Fleming [EMAIL PROTECTED] wrote:
Imran Ahmed wrote:
Even though no IAX client supports inband dtmf, An IAX client can send
inband dtmf which would have corrected your problem.
No, it won't. No IAX2 client will start a DSP to listen for inband DTMF,
because IAX2 is defined to always send out-of-band DTMF.
At best, if the receiving IAX2 system is just passing the audio along to
another protocol that does support inband DTMF, then sending it in the
audio stream would work. If the application receiving the DTMF is on the
other IAX2 end, though (like MeetMe in this case), then it will never
'see' the DTMF, because Asterisk will not look in the audio stream for DTMF.
I agree, but the other ends of the conference were zap channels in
this case, at least that is what I figured by the first email.
Maybe if a paint better my scenario it would help the discussion.
Step 1: A IAX client make a call executing the following command
Dial(ZAP/g1/${EXTEN})
If aswered this call is tranfered to a conference room.
Step 2: The IAX client make a second call executing again
Dial(ZAP/g1/${EXTEN})
an IVR answer this call and the IAX client have to send some
DTMF stil now everything works very well.
At this point call is transfered to the previous conference room
and The IAX client reach the conference too.
Step 3 The Iax client heve to send some other DTMF to the IVR.
NOW THE IVR DOES NOT HEAR DTMF SENDED BY THE IAX CLIENT, EVEN IF
IT CAN HEAR DTMF SENDED BY THE FIRST ZAP CHANNEL.
Hoping to be clear enough
thank yuo very much for any help or suggestion.
Accursio Avona
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Re: [Asterisk-Users] meetme and dtmf
Kevin P. Fleming wrote:
Accursio Avona wrote:
Step 2: The IAX client make a second call executing again
Dial(ZAP/g1/${EXTEN})
an IVR answer this call and the IAX client have to send some
DTMF stil now everything works very well.
At this point call is transfered to the previous conference
room and The IAX client reach the conference too.
Step 3 The Iax client heve to send some other DTMF to the IVR.
How is the IVR still involved if the call has been transferred into a
conference room?
The IVR records the conversation between the other partecipant to the
conference and wait '#' to stop recording and a '1' to save the file.
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Re: [Asterisk-Users] meetme and dtmf
Imran Ahmed wrote: Here is my problem, at this point the IVR doesn't hear the dtmf sended by the iax client, even if it can hear the dtmf sended by the first zap channel. I donot know if IaxComm has inband dtmf mode available, if so enable it and see if it works. Someone can suggest me a Iax softphone with inband dtmf mode available ?? Thank's in advance Regards Accursio Avona ___ --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] meetme and dtmf
Francesco Peeters (Asterisk) wrote: On Wed, February 1, 2006 12:07, Accursio Avona said: Imran Ahmed wrote: Here is my problem, at this point the IVR doesn't hear the dtmf sended by the iax client, even if it can hear the dtmf sended by the first zap channel. I donot know if IaxComm has inband dtmf mode available, if so enable it and see if it works. Someone can suggest me a Iax softphone with inband dtmf mode available ?? Thank's in advance AFAIK there's no DTMF option in IAX2... IAX always sends DTMF inline, eliminating the confusion often found with SIP. http://www.voip-info.org/wiki-IAX If so, wy the IVR does not hear the dtmf sended by the iax client and it hear that one sendee by the zap channel? Could it be a meetme problem? and if so what can i do? Thank yuo very much for any help. Accursio Avona ___ --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] meetme and dtmf
Francesco Peeters (Asterisk) wrote: Are you sure it *is* sending DTMF in the first place? (Just trying to find a logical place to start here...) I do not use meetme, but when I use idefisk, my (*) server recognizes the DTMF. Have you tried whether the IAXCOMM DTMF is recognized OUTSIDE meetme? Yes it is, outside meetme everything works fine. Thank's for any help or suggestion. Accursio Avona ___ --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] meetme and dtmf
Hi all, I'm experiencing a problem with meetme i can't resolve. This is my scenario: A iax client, say IaxComm, make a call through a zap channel. When it answers it is tranfered to a conference room. Then the iax client make a second call though a second zap channel, at the other side there is an IVR. Iax client send some dtmf to the IVR then it transfers the IVR to the previos conference room. At this point iax client joins to the conference and talking to the first zap channel need to send dtmf to the IVR. Here is my problem, at this point the IVR doesn't hear the dtmf sended by the iax client, even if it can hear the dtmf sended by the first zap channel. Is there someone that can help me? any suggestion i welcome. Best Regards Accursio Avona ___ --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] Meetme Question
Hi, Thank you very much for your suggestion this was what i nedded. Best Regards Accursio Avona The question is, how can i indicate the marked user? A quick search of the archives reveals: Example: meetme.conf conf = 1000 extensions.conf ; ** Normal users enter the conference here ** exten = 4823,1,SetMusicOnHold(random) exten = 4823,2,Meetme(|Msciw) exten = 4823,3,Hangup() ; ** Extension to mark conference users* exten = 4824,1,Authenticate(12345) exten = 4824,2,Meetme(|Asci) exten = 4824,3,Hangup() Users using extension 4823 and entering conference 1000 will listen to hold music until the marked users enters. Users using extension 4824 and entering a password of 12345 will be able to select conference 1000 as the marked user. Doug ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Meetme Question
Hi all, I'd like to use the w option of the meetme application. >From tiki i read: 'w' wait until the marked user enters the conference All other connected users will hear MusicOnHold until the marked user enters. The question is, how can i indicate the "marked user"? thank's in advance best regards Accursio Avona ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Meetme Question
Hi all, I'd like to use the w option of the meetme application. From tiki i read: 'w' — wait until the marked user enters the conference * All other connected users will hear MusicOnHold until the marked user enters. The question is, how can i indicate the marked user? thank's in advance best regards Accursio Avona ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco ATA186 + Dell 1600n printer-fax
Hi All, Is there someone who have used a Dell 1600n as fax machine? Any information or suggestion is welcome. Thank's in advance Best Regards Accursio Avona ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium Wildcard TE110P IRQ problem
This worked for me:
before compile bristuff edit the file
wcte1xxp.c
near line 1526 initialize the array pci_device_id t1xxp_pci_tbl[]
this way:
static struct pci_device_id t1xxp_pci_tbl[] = {
{ 0xe159, 0x0001, 0x79fe, PCI_ANY_ID, 0, 0, (unsigned long)
Digium Wildcard TE110P T1/E1 Board },
{ 0xe159, 0x0001, 0x797e, PCI_ANY_ID, 0, 0, (unsigned long)
Digium Wildcard TE110P T1/E1 Board },
{ 0xe159, 0x0001, 0x79de, PCI_ANY_ID, 0, 0, (unsigned long)
Digium Wildcard TE110P T1/E1 Board },
{ 0xe159, 0x0001, 0x795e, PCI_ANY_ID, 0, 0, (unsigned long)
Digium Wildcard TE110P T1/E1 Board },
{ 0xe159, 0x0001, 0x79be, PCI_ANY_ID, 0, 0, (unsigned long)
Digium Wildcard TE110P T1/E1 Board },
{ 0xe159, 0x0001, 0x793e, PCI_ANY_ID, 0, 0, (unsigned long)
Digium Wildcard TE110P T1/E1 Board },
{ 0xe159, 0x0001, 0x791e, PCI_ANY_ID, 0, 0, (unsigned long)
Digium Wildcard TE110P T1/E1 Board },
{ 0xe159, 0x0001, 0x799e, PCI_ANY_ID, 0, 0, (unsigned long)
Digium Wildcard TE110P T1/E1 Board },
{ 0 }
};
hope i helped you.
Regards
Accursio
[EMAIL PROTECTED] wrote:
Folks:
I'm trying to reach for help, I have a Digium Wildcard TE110P, on an E1,
the problem is every time the server reboots sometimes, and the system
does not recognize the card. I have on the Bios disabled the PNP OS and
the irq's are assigned manually, IRQ 9 for slot 3 (where I have installed
the card) bus still have the same problem.
Is there a way to fix or hardcode the irq for the card?
Thanks in advance for your help
JR
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Re: [Asterisk-Users] asterisk box after an analogic pbx
Many, Many Thanks.
This was exactly what I needed.
This way I solved the problem.
Best Regards
Accursio Avona
Julian J. M. wrote:
exten = _X.,1,Dial(Zap/1/0www${EXTEN})
That doesn't wait for dialtone, just dial 0, sleep for 1,5sec, and
dial the number.
Julian.
On 7/5/05, Accursio Avona [EMAIL PROTECTED] wrote:
Hi all,
I'm newbe with asterisk and i'm facing with this problem that i'm not
able to solve.
I've to put an asterisk box after an analogic pbx wich require a 0 digit
to give the dialtone.
So when a client ask asterisk to dial an extension it should
1) send the 0 digit
2) wait for the dialtone
3) dial the extension the client send.
How can i obtain this result?
Thank's in advance
Best regards
Accursio Avona.
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[Asterisk-Users] asterisk box after an analogic pbx
Hi all, I'm newbe with asterisk and i'm facing with this problem that i'm not able to solve. I've to put an asterisk box after an analogic pbx wich require a 0 digit to give the dialtone. So when a client ask asterisk to dial an extension it should 1) send the 0 digit 2) wait for the dialtone 3) dial the extension the client send. How can i obtain this result? Thank's in advance Best regards Accursio Avona. ___ 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
