[Asterisk-Users] disabling ALERTING message
Hi all, Is there a way of disabling ALERTING message on a PRI channel? I have a problem .* is sending ALERTING message to the Nortel telco switch of my local provider BEFORE it dial the number it has to .if the number is busy or invalid there is no way we can tell this to the switch because it has already been told that the phone is ringing I am using asterisk Asterisk CVS-04/06/04-10:46:21 with T410 PRI card connected to a Nortel switch Thanks
[Asterisk-Users] Re:disabling ALERTING message
I cant use this t410 card to send calls to my telcos switch bse the network manager is complaining about this message being sent to their switch saying that the call is going on even before the number is dialed Thanks
[Asterisk-Users] Problems with PRI with T410 messages
Hi all, I have a box running asterisk with T410 connected to a Nortel DMS 100 switch and another box running SER with grandstream phones on it So if there is a call from the pstn it goes from the Nortel to the asterisk and then to the SER box and finally to the phones.if the phone is busy or the number is invalid the * box will first send an ALERT message to the Nortel and say the call is going on and the phone is ringing (which is not the case )and after it will send a RELEASE message saying that the line is busy or the # is invalid .is there any way * can send a progress message instead of the alerting message until it gets the correct message from SER? Thanks Habiyakare Aimable Phone Services TERRACOM Broadband [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Thursday, June 17, 2004 10:56 AM To: [EMAIL PROTECTED] Subject: Asterisk-Users digest, Vol 1 #4181 - 12 msgs Send Asterisk-Users mailing list submissions to [EMAIL PROTECTED] To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than Re: Contents of Asterisk-Users digest... Today's Topics: 1. RE: Soekris Engineering net4801 (Senad Jordanovic) 2. Accepting SIP calls from unregistered gateways (Axel) 3. Re: pri with TE410P not working (Austria) (Peter Svensson) 4. Re: ZAPHFC - only for * 0.7.2? (Holger Schurig) 5. Calling the firefly network? (Martijn van Oosterhout) 6. RE: IAX2 no compatible codecs (Jason Penton) 7. Re: pri with TE410P not working (Austria) (Wolfgang Pichler) 8. Re: embedded Asterisk (Klaus-Peter Junghanns) 9. Re: pri with TE410P not working (Austria) (Michael Bielicki) 10. RE: Cost of IP Phones, or Isn't It Just Software? (Andy Powell) 11. Re: pri with TE410P not working (Austria) (Peter Svensson) --__--__-- Message: 1 From: Senad Jordanovic [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Soekris Engineering net4801 Date: Thu, 17 Jun 2004 08:34:01 +0100 Reply-To: [EMAIL PROTECTED] John Bittner wrote: Hi, I have it working great. I have debian running on it with music on hold disabled. I setup 10 cisco 7960 phones and tested the 4801 with calls on all 10 phones at the same time through voicepulse with no issues. I ran top with all the phones running and I was only up to 45% cpu. Seems to run ok but I am still in the testing phase. Great... Have you tried to connect a X100P or TDM400P to it? --__--__-- Message: 2 From: Axel [EMAIL PROTECTED] To: [EMAIL PROTECTED] Date: Thu, 17 Jun 2004 03:43:12 -0400 Subject: [Asterisk-Users] Accepting SIP calls from unregistered gateways Reply-To: [EMAIL PROTECTED] This is a multi-part message in MIME format. --=_NextPart_000_0351_01C4541D.36B45830 Content-Type: text/plain; charset=iso-8859-1 Content-Transfer-Encoding: quoted-printable Hi, Is there a way to accept SIP calls from unregistered gateways? autocreatpeer=3Dyes seems to disable checking credentials but the = originating gateway is still required to register itself with a username = and password (which can be anything since it won't check it). I like to be able to receive the call from any gateway without them = having to register even, just like a Cisco gateway that you can = terminate a call from clients who are not registered. Is such thing = possible with Asterisk? Best regards, Axel --=_NextPart_000_0351_01C4541D.36B45830 Content-Type: text/html; charset=iso-8859-1 Content-Transfer-Encoding: quoted-printable !DOCTYPE HTML PUBLIC -//W3C//DTD HTML 4.0 Transitional//EN HTMLHEAD META http-equiv=3DContent-Type content=3Dtext/html; = charset=3Diso-8859-1 META content=3DMSHTML 6.00.2800.1400 name=3DGENERATOR STYLE/STYLE /HEAD BODY bgColor=3D#ff DIVFONT face=3DArial size=3D2Hi,/FONT/DIV DIVFONT face=3DArial size=3D2Is there a way to accept SIP calls from = unregistered gateways?/FONT/DIV DIVFONT face=3DArial size=3D2autocreatpeer=3Dyes seems to disable = checking=20 credentials but the originating gateway is still required to register = itself=20 with a username and password (which can be anything since it won't check = it)./FONT/DIV DIVFONT face=3DArial size=3D2I like to be able to receive the call = from any=20 gateway without them having to register even, just like a Cisco gateway = that you=20 can terminate a call from clients who are not registered.nbsp; Is such = thing=20 possible with Asterisk?/FONT/DIV DIVFONT face=3DArial size=3D2/FONTnbsp;/DIV DIVFONT face=3DArial size=3D2Best regards,/FONT/DIV DIVnbsp;/DIV DIVFONT face=3DArial size=3D2AxelBR/FONT/DIV/BODY/HTML --=_NextPart_000_0351_01C4541D.36B45830-- --__--__-- Message: 3 Date: Thu, 17 Jun 2004
[Asterisk-Users] Problems with PRI with T410 messages
Now what is the normal behavior and how can I set it so that * behaves normally? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Thursday, June 17, 2004 2:06 PM To: [EMAIL PROTECTED] Subject: Asterisk-Users digest, Vol 1 #4186 - 11 msgs Send Asterisk-Users mailing list submissions to [EMAIL PROTECTED] To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than Re: Contents of Asterisk-Users digest... Today's Topics: 1. Re[2]: [Asterisk-Users] HFC ISDN card with bristuff from junghanns.n et? (Alessio Focardi) 2. RE: LDAP synchronization script (Stefan de Konink) 3. Re: Problems with PRI with T410 messages (CW_ASN) 4. RE: Re[2]: [Asterisk-Users] HFC ISDN card with bristuff from jung hanns.n et? (Robinson Tim-W10277) 5. RE: LDAP synchronization script (David Hajek) 6. Zapata.conf Signaling for Bulgaria (PSTN: Siemens PABX) (Miroslav Nachev) 7. Re: embedded Asterisk (listas iPfone) 8. Re[4]: [Asterisk-Users] HFC ISDN card with bristuff from jung hanns.n et? (Alessio Focardi) 9. SFTP (Dean Collins) 10. Re: embedded Asterisk (Stefan de Konink) --__--__-- Message: 1 Date: Thu, 17 Jun 2004 13:18:51 +0200 From: Alessio Focardi [EMAIL PROTECTED] To: Robinson Tim-W10277 [EMAIL PROTECTED], [EMAIL PROTECTED] Subject: Re[2]: [Asterisk-Users] HFC ISDN card with bristuff from junghanns.n et? Reply-To: [EMAIL PROTECTED] Hello Robinson, Thursday, June 17, 2004, 12:42:21 PM, you wrote: RTW Please can you explain in more details as to what your RTW problem is? I have 2 cards working in one PC, but have had RTW problems with Dell motherboards. voice is out of sync, it syncs for some second if I run something over another console, like, for instance a find / then slips away again. I suspect an Irq problem, what do you think ? What kind of problems have you found with dell's ? Tnx for the help ! -- Best regards, Alessiomailto:[EMAIL PROTECTED] --__--__-- Message: 2 Date: Thu, 17 Jun 2004 13:12:25 +0200 (CEST) From: Stefan de Konink [EMAIL PROTECTED] To: David Hajek [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] LDAP synchronization script Reply-To: [EMAIL PROTECTED] I'm planning to incorporate this (native and dynamic) LDAP for my own system on short term. Do you have any LDAP design in mind? Stefan On Thu, 17 Jun 2004, Jeremy Jones wrote: David Hajek Sent: Thursday, June 17, 2004 2:41 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] LDAP synchronization script Hello, I understand there's no possibility to have asterisk configuration (sipusers, extensions, voicemail) in LDAP right now. I'm thinking about put the (sipusers, extensions, voicemail) info in LDAP and then run some synchronization script on the asterisk server which will build up appropriate configuration files and reload asterisk. I'm sure this script is already around. Can some share one with me/us? Not aware of any scripts like that, but... you could use the odbc support in asterisk in conjunction with some slick odbc-ldap connectivity. Jeremy Jones ___ 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 --__--__-- Message: 3 From: CW_ASN [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Problems with PRI with T410 messages Date: Thu, 17 Jun 2004 08:13:03 -0300 Reply-To: [EMAIL PROTECTED] This is a problem I pointed out to Digium a while back, but I am not sure Markster understood the issue and I didn't really have the time to follow it up. It does need fixing though, as it is a major drawback in the current architecture. Rgds Tim Hi all, I have a box running asterisk with T410 connected to a Nortel DMS 100 switch and another box running SER with grandstream phones on it So if there is a call from the pstn it goes from the Nortel to the asterisk and then to the SER box and finally to the phones.if the phone is busy or the number is invalid the * box will first send an ALERT message to the Nortel and say the call is going on and the phone is ringing (which is not the case )and after it will send a RELEASE message saying that the line is busy or the # is invalid .is there any way * can send a progress message instead of the alerting message until it gets the correct message from SER? Thanks Habiyakare Aimable Call Proceeding can be sent only by transit network, not by the local switch or pbx. AFAIK, * behavior for this scenario
[Asterisk-Users] Asterisk PRI calls to SER problem
Hi all, I need help. I have a Linux box with SER as a proxy server with ip phones attached on it , and another linux box with Asterisk and T410 card connect to an E1 line .Whenever there is a call from PSTN it is passed to Asterisk and then to SER box and then to the phone .every time an invalid number dialed from PSTN to SIP phones connected to SER asterisk says that the call is progressing while it is not the case and send an alerting message to the Nortel DMS switch attached to it. Is there any way I can remove that alerting message and send the collect message to the switch? I think that the reason is that * is not directly connected to the phones it is calling my setup is like this. SIP phonesSER---AsteriskPSTN(PRI connected to NORTEL DES 100 switch) I would like to find a way of informing Asterisk that the call is progressing or something like that, not ringing until it gets the correct message from SER . I am using Asterisk CVS-04/06/04-10:46:21 on Red Hat 9 and Sip Express Router version 12 on Red Hat 9. I tried to use PRI_causes and r extension in extension.conf but still the problem is there. Any idea on how I can solve this problem?
[Asterisk-Users] RE:Asterisk PRI calls to SER problem
I have checked my SER configs and for cpb numbers validation I don't know
what it means .Can anyone who does help me?
Thanks
the reason is that you have a bug in your config files, most probably on SER
which sends provisional response instead of an error response to * which in
turn translates that to alerting on isdn. Verify your configs on SER and
make sure you send an error back to * when the sip phone is unavailbale. You
might also want to validate your cpb numbers on * so that if the number is
invalid you send back a release with invalid number format back to the
switch instead of forwarding the call to SER.
BR
Dawid
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Aimable
Sent: Friday, June 11, 2004 12:05 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Asterisk PRI calls to SER problem
Hi all,
I need help. I have a Linux box with SER as a proxy server with ip phones
attached on it , and another linux box with Asterisk and T410 card connect
to an E1 line .Whenever there is a call from PSTN it is passed to Asterisk
and then to SER box and then to the phone .every time an invalid number
dialed from PSTN to SIP phones connected to SER asterisk says
that the call is progressing while it is not the case and send an alerting
message to the Nortel DMS switch attached to it. Is there any way I can
remove that alerting message and send the collect message to the switch? I
think that the reason is that * is not directly connected to the phones it
is calling
my setup is like this.
SIP
phonesSER---AsteriskPSTN(PRI
connected to NORTEL DES 100 switch)
I would like to find a way of
informing Asterisk that the call is progressing or something like that,
not ringing until it gets the correct message from SER .
I am using Asterisk CVS-04/06/04-10:46:21 on Red Hat 9 and Sip Express
Router version 12 on Red Hat 9.
I tried to use PRI_causes and r extension in extension.conf but still
the problem is there.
Any idea on how I can solve this problem?
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reason is that you have a bug in your config files, most probably on SER =
which=20
sends provisional response instead of an error response to * which in =
turn=20
translates that to alerting on isdn. Verify your configs on SER and make =
sure=20
you send an error back to * when the sip phone is unavailbale. You might =
also=20
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you send=20
backnbsp;a release withnbsp;invalid number format back to the switch =
instead=20
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DIVSPAN class=3D640503710-11062004FONT face=3DArial color=3D#ff =
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[Asterisk-Users] Asterisk PRI messages
Hi all, I have decided to send this e-mail because you are the developer of Asterisk . We are developing a phone system using Asterisk as the VOIP gateway with 1 t410 PRI card and Sip Express Router as the proxy server but we have a problem. Our phone system setup like this: SIP phonesSER---AsteriskPSTN(PRI connected to NORTEL DES 100 switch) transfer the call to Sip Express router then to the phone. So when there is a call from the pstn through asterisk and the phone is busy or the number is invalid ,asterisk tells the switch that the call is going on and the phone is ringing while it is not the case. I would like to find a way of informing Asterisk that the call is progressing or something like that, not ringing until it gets the correct message from SER . I am using Asterisk CVS-03/22/04-15:45:54 on Red Hat 9 and Sip Express Router version 12 on Red Hat 9. I tried to use PRI_causes but still the problem is there. Any idea on how I can solve this problem? Thanks
[Asterisk-Users] FW: Problem with Asterisk PRI forwarding to SER
From: Habiyakare Aimable [mailto:[EMAIL PROTECTED] Sent: Monday, June 07, 2004 11:49 AM To: '[EMAIL PROTECTED]'; 'gt'; '[EMAIL PROTECTED]' Subject: Problem with Asterisk PRI forwarding to SER Hi all, I have a problem. We have a phone system setup like this: SIP phones--SER-Asterisk--PSTN(PRI connected to NORTEL DES 100 switch) So when there is a call from the pstn through asterisk and the phone is busy or the number is invalid ,asterisk tells the switch that the call is going and the phone is ringing while it is not the case. I cant find a way of informing Asterisk that the call is progressing or something like that . I am using Asterisk CVS-03/22/04-15:45:54 on Red Hat 9 and Sip Express Router version 12 on Red Hat 9 I tried to use PRI_causes but still the problem is there. Any idea on how I can solve this problem? Thanks
