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 -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Aimable Sent: 17 June 2004 10:29 To: [EMAIL PROTECTED] Subject: [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"> <HTML><HEAD> <META http-equiv=3DContent-Type content=3D"text/html; = charset=3Diso-8859-1"> <META content=3D"MSHTML 6.00.2800.1400" name=3DGENERATOR> <STYLE></STYLE> </HEAD> <BODY bgColor=3D#ffffff> <DIV><FONT face=3DArial size=3D2>Hi,</FONT></DIV> <DIV><FONT face=3DArial size=3D2>Is there a way to accept SIP calls from = unregistered gateways?</FONT></DIV> <DIV><FONT face=3DArial size=3D2>autocreatpeer=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> <DIV><FONT face=3DArial size=3D2>I 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. Is such = thing=20 possible with Asterisk?</FONT></DIV> <DIV><FONT face=3DArial size=3D2></FONT> </DIV> <DIV><FONT face=3DArial size=3D2>Best regards,</FONT></DIV> <DIV> </DIV> <DIV><FONT face=3DArial size=3D2>Axel<BR></FONT></DIV></BODY></HTML> ------=_NextPart_000_0351_01C4541D.36B45830-- --__--__-- Message: 3 Date: Thu, 17 Jun 2004 09:43:37 +0200 (CEST) From: Peter Svensson <[EMAIL PROTECTED]> To: Asterisk-Users Mailinglist <[EMAIL PROTECTED]> Subject: Re: [Asterisk-Users] pri with TE410P not working (Austria) Reply-To: [EMAIL PROTECTED] On Thu, 17 Jun 2004, Wolfgang Pichler wrote: > ... on the card i can see the two leds pulsing red (i think thats the > yellow alaram - or i am wrong) ? Are you sure it is not a red alarm? That would indicate a loss of link. I think you can check with the command zttool. Are you sure the cables are correct? Have you set the jumpers on the card to E1 and not left them on T1? I think the leds should turn green when the card senses a correct carrier and framing on the lines. Peter -- Peter Svensson ! Pgp key available by finger, fingerprint: <[EMAIL PROTECTED]> ! 8A E9 20 98 C1 FF 43 E3 07 FD B9 0A 80 72 70 AF ------------------------------------------------------------------------ Remember, Luke, your source will be with you... always... --__--__-- Message: 4 From: Holger Schurig <[EMAIL PROTECTED]> To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] ZAPHFC - only for * 0.7.2? Date: Thu, 17 Jun 2004 09:59:33 +0200 Reply-To: [EMAIL PROTECTED] > I've got Zaphfc working running Asterisk v. 0.7.2 > > Then I have tried with Asterisk V. 1.0 and the latest from CVS - with > no succes. Has anybody got zaphfc working with newer version than > 0.7.2 zaphfc is in bri-stuff from www.junghanns.net --- or in a patched version at http://capi4linux.thepenguin.de/download/asterisk/. I downloaded the latter and let the ./download.sh and ./compile.sh scripts run normally. Then I install zaptel.o and zaphfc.o to /lib/modules/<kernelversion>/misc and do the usual mambo in /etc/modules to run ztcfg after loading zaphfc and to load zaptel before zaphfc: pre-install zaphfc /sbin/modprobe zaptel post-install zaphfc /sbin/ztcfg -v Now I go to a different directory and do a CVS checkout of Asterisk head. Just before compiling, I replace channels/chan_zap.c with bri-stuff-0.0.2a-pp/asterisk/channels/chan.zap.c. I then change the lines of the form static ast_mutex_t usecnt_lock = AST_MUTEX_INITIALIZER; into AST_MUTEX_DEFINE_STATIC(usecnt_lock); and compile & install. And voila, now I have an Asterisk from (almost) CVS HEAD working with zaphfc. The real solution would have been to apply all the patches from bri-stuff*/libpri.patch to libpri in CVS. After looking at how much has been changed and considering that I don't have a clue about q.921 and q.931 I decided to not doing it that way :-) Also, I'd thing it would be better if KaPeJot put's his software into some CVS so that more than one person can add changes and keep things up-to-date. Greetings, Holger --__--__-- Message: 5 Date: Thu, 17 Jun 2004 18:12:10 +1000 From: Martijn van Oosterhout <[EMAIL PROTECTED]> To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Calling the firefly network? Reply-To: [EMAIL PROTECTED] Is there a way to register with or call the firefly network from an Asterisk server. It would be pretty cool if you could gateway calls onto it. Have a nice day, -- Martijn van Oosterhout --__--__-- Message: 6 From: "Jason Penton" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Subject: RE: [Asterisk-Users] IAX2 no compatible codecs Date: Thu, 17 Jun 2004 10:22:10 +0200 Reply-To: [EMAIL PROTECTED] Hi Adam Thanks - Here are the two attempts: This is the first one where * dials firefly via the dialplan (which works fine): Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 00001ms SCall: 00004 DCall: 00000 [146.231.125.65:4569] VERSION : 2 CALLED NUMBER : s CALLING NUMBER : 7001 CALLING NAME : Alfredo+Terzoli LANGUAGE : en FORMAT : 4 CAPABILITY : 2147483647 ADSICPE : 2 DATE TIME : 147935435 Now the following output is when I use the manager ORIGINATE command: Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 00001ms SCall: 00004 DCall: 00000 [146.231.125.65:4569] VERSION : 2 CALLED NUMBER : s LANGUAGE : en FORMAT : 64 CAPABILITY : 2147483647 ADSICPE : 0 DATE TIME : 147935484 Jun 17 10:07:57 WARNING[180236]: chan_iax2.c:4534 socket_read: Call rejected by 146.231.125.65: No compatible Codecs I can see the inconsitency with the FORMAT header of the two setup messages. According to the IAX protocol spec. The FORMAT (0x4) represents G.711 U-LAW, which is exactly what the resulting call uses. However, the funny thing is that the protocol spec has no entry for FORMAT(0x64) in the second message - an undefined format. The quesiton is how the * manager API causes * to inititiate an IAX call with this FORMAT type (0x64)??????? An how we can fix it ???????. Any ideas, anyone Thanks again Adam for the help Cheers Jason > -----Original Message----- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Adam Hart > Sent: 17 June 2004 09:19 AM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] IAX2 no compatible codecs > > iax2 debug is your friend, looks at the capibilities asterisk > is sending > in it's NEW message > > Jason Penton wrote: > > > Hi Adam > > > > Done all that but still the same problem. > > > > Do you have any other ideas? > > > > Cheers > > Jason > > > > > >>-----Original Message----- > >>From: [EMAIL PROTECTED] > >>[mailto:[EMAIL PROTECTED] On Behalf Of > Adam Hart > >>Sent: 17 June 2004 08:29 AM > >>To: [EMAIL PROTECTED] > >>Subject: Re: [Asterisk-Users] IAX2 no compatible codecs > >> > >>check under your network settings that you have all the > >>codecs selected > >>and obviously type IAX > >> > >>Jason Penton wrote: > >> > >>>Hi All > >>> > >>>I have a strange problem using IAX2. When placing a call to > >> > >>my IAX clients > >> > >>>(firefly) via the Asterisk dialplan all works great. > >> > >>However trying to > >> > >>>initiate a call via the manager interface to the IAX client > >> > >>using the > >> > >>>following command results in an error: > >>> > >>>Action: Originate > >>>Channel: IAX2/7000 > >>>Extension: 7000 > >>>Context: local > >>>Priority: 1 > >>>ActionID: 1 > >>> > >>>The error I get in the CLI is "Jun 17 08:18:36 WARNING[180236]: > >>>chan_iax2.c:4534 socket_read: Call rejected by #IP: No > >> > >>compatible Codecs" > >> > >>>Does anyone have any ideas. > >>> > >>>Thanks in advance > >>>Jason > >>> > >>>_______________________________________________ > >>>Asterisk-Users mailing list [EMAIL PROTECTED] > >>>http://lists.digium.com/mailman/listinfo/asterisk-users > >>>To UNSUBSCRIBE or update options visit: > >>> http://lists.digium.com/mailman/listinfo/asterisk-users > >>> > >> > >>_______________________________________________ > >>Asterisk-Users mailing list > >>[EMAIL PROTECTED] > >>http://lists.digium.com/mailman/listinfo/asterisk-users > >>To UNSUBSCRIBE or update options visit: > >> http://lists.digium.com/mailman/listinfo/asterisk-users > >> > >> > > > > > > _______________________________________________ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > _______________________________________________ > 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: 7 Subject: Re: [Asterisk-Users] pri with TE410P not working (Austria) From: Wolfgang Pichler <[EMAIL PROTECTED]> To: Asterisk-Users Mailinglist <[EMAIL PROTECTED]> Date: Thu, 17 Jun 2004 10:28:09 +0200 Reply-To: [EMAIL PROTECTED] Am Do, den 17.06.2004 schrieb Peter Svensson um 9:43: > On Thu, 17 Jun 2004, Wolfgang Pichler wrote: > > > ... on the card i can see the two leds pulsing red (i think thats the > > yellow alaram - or i am wrong) ? > > Are you sure it is not a red alarm? That would indicate a loss of link. > I think you can check with the command zttool. you are right - its a red alarm - zttool says "Red Alarm/Not Open" > > Are you sure the cables are correct? > Have you set the jumpers on the card to E1 and not left them on T1? The jumpers are on E1 - the cables should be ok (they are working with other hardware) - and the card is directly connected to a simens ULAF+ STU Desktop (can't really find much information about this device on the net) - which turns off a red led when i load the driver and do a ztcfg. > > I think the leds should turn green when the card senses a correct carrier > and framing on the lines. green is always a wounderful color ;-) so, what else could cause this ? wolfgang --__--__-- Message: 8 Subject: Re: [Asterisk-Users] embedded Asterisk From: Klaus-Peter Junghanns <[EMAIL PROTECTED]> To: [EMAIL PROTECTED] Organization: Junghanns.NET GmbH Date: Thu, 17 Jun 2004 10:11:11 +0200 Reply-To: [EMAIL PROTECTED] Hi, > Actually, you the Geode CPU mentioned below is a 5x86 (486 platform) at > 233 MHz. If you take Pebble (http://www.nycwireless.net/pebble/), which > is a downstripped Debian (< 64 MB) on a readonly ext2 filesystem, you > should be grand. Installing asterisk + some extra stuff will probably > require, that you have at least a 128MB or 256MB flash or so. Dont go for "stripped down but complete" distributions which include a lot of stuff that you dont need, e.g. gcc. Go for a rescue system, like i used the SuSE rescue system (14 mb), then you can add what you need (sshd,...) and compile asterisk on another box and then just copy it. My compressed ramdisk image is 32 mb, including all voice prompts and some mp3s for MOH. > > There are actually quite some board around on that CPU, like Soekris, > pcengines and i think also Mikrotik at prices from 120EUR and up. > I just put together the demo system for Linuxtag: - Via EPIA 5000 (C3-533), EUR 80,- - Morex case with external power supply, EUR 80,- - some old 256 mb SDRAMM - 128 MB USB memory stick, EUR 30,- - 1 quadBRI (could also easily handle an octoBRI, or a PRI card, with the dual riser pci card you can use 2 cards) The C3-533 is an i586 CPU. According to "show translation" it needs 30 ms for transcoding 1 channel from g711 to gsm (and vice versa). So, neglecting any overhead caused by channel handling it could transcode 30 channels to gsm. Linux BIOS has support for the EPIA boards, so you can speed up booting very much and also disable the VGA port (very useful for production deployments....). > I'm running pebble on a pcengines board, just needed to customize the > kernel a bit, haven't been testing asterisk on that yet, but i definatly > will in the sooner future. > > Kind regards, > Martin List-Petersen > martin (at) list (dash) petersen (dot) net best regards Klaus -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Strasse 13a - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ --__--__-- Message: 9 Subject: Re: [Asterisk-Users] pri with TE410P not working (Austria) From: Michael Bielicki <[EMAIL PROTECTED]> To: [EMAIL PROTECTED] Organization: TAAN Consultants Ltd. Date: Thu, 17 Jun 2004 10:32:41 +0200 Reply-To: [EMAIL PROTECTED] What is in your config file ? zaptel.conf ? also, check the crc4 settings and maybe the wire you are using is wrong since some equippment needs crossed wires, some needs straight wires. Crossed would be 1-4 2-5 cheers Michael On Thu, 2004-06-17 at 10:28, Wolfgang Pichler wrote: > Am Do, den 17.06.2004 schrieb Peter Svensson um 9:43: > > On Thu, 17 Jun 2004, Wolfgang Pichler wrote: > > > > > ... on the card i can see the two leds pulsing red (i think thats the > > > yellow alaram - or i am wrong) ? > > > > Are you sure it is not a red alarm? That would indicate a loss of link. > > I think you can check with the command zttool. > you are right - its a red alarm - zttool says "Red Alarm/Not Open" > > > > Are you sure the cables are correct? > > Have you set the jumpers on the card to E1 and not left them on T1? > The jumpers are on E1 - the cables should be ok (they are working with > other hardware) - and the card is directly connected to a simens ULAF+ > STU Desktop (can't really find much information about this device on the > net) - which turns off a red led when i load the driver and do a ztcfg. > > > > I think the leds should turn green when the card senses a correct carrier > > and framing on the lines. > green is always a wounderful color ;-) > > so, what else could cause this ? > > wolfgang > > _______________________________________________ > 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: 10 Date: Thu, 17 Jun 2004 10:33:52 +0200 From: "Andy Powell" <[EMAIL PROTECTED]> To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Cost of IP Phones, or Isn't It Just Software? Reply-To: [EMAIL PROTECTED] On 16/06/2004 at 22:53 Jay Milk wrote: >You're correct -- I believe I pointed out in my original post that there >is a $200+ difference between a cordless Cisco with/without software. >And that's plain ridiculous. Plus, the phone alone isn't worth $500 in >hardware -- so we're obviously dealing with GREED here. > >My knee-jerk response to such business tactics always has been to do it >better and cheaper. Six years ago, I was talking to IT personel in >industry "X". There were two established mainframe solutions in that >industry serving 80% of the market, costing $50K-$75K start-up cost per >location, plus $1K+ per seat. Never mind the $10K-$15K monthly >"maintenance" cost. Never mind that everyone had to be able to work a >terminal with a lovely amber on black, text-based "GUI". > <snip for brevity> I think you're missing the point. When you develop hardware or software you need to recoup the cost of development (the period in which you aren't= selling anything, so not making any money). Now Cisco has it's fingers in many pies so they aren't going to suffer to much from that now, but they do have to= fund development. Secondly, Cisco don't really care if their phones are out of your price= range, they are typically sold as part of a solution costing 10's of 1000's or= 100's of 1000's of USD/GBP/EUR and (most probably) with big discounts. Thirdly, If I make a device at a cost of $5 and sell it for $500, some= people will buy it, up to the point where someone builds a similar device and sells it= for $150 ...You have a choice. companies are not charities, they do this to= make money. This is what we call capitalism. I don't want to dig at your business, and this isn't intended to but.. what= you did is look at what was already on offer and it's costs, how it worked etc and= built a cheaper solution. The reason you could do this is because you had the= exposure to the 'system' as was.. i.e. You looked at it and said 'I can do that= cheaper' but without that original system you probably wouldn't have. One final point... There are some companies that have this weird feeling= that anything under a certain amount must be cheap and nasty and not work properly. These= people are fools imho, but they do exist...and they wont buy an cheap phone,= they'll buy an expensive phone, regardless of it's ability... as we've seen recently some= governments will even buy helicopters that can't fly in fog or where it's sandy for= silly money... Now I feel dirty... Andy --__--__-- Message: 11 Date: Thu, 17 Jun 2004 10:38:30 +0200 (CEST) From: Peter Svensson <[EMAIL PROTECTED]> To: Asterisk-Users Mailinglist <[EMAIL PROTECTED]> Subject: Re: [Asterisk-Users] pri with TE410P not working (Austria) Reply-To: [EMAIL PROTECTED] On Thu, 17 Jun 2004, Wolfgang Pichler wrote: > > Are you sure the cables are correct? > > Have you set the jumpers on the card to E1 and not left them on T1? > The jumpers are on E1 - the cables should be ok (they are working with > other hardware) - and the card is directly connected to a simens ULAF+ > STU Desktop (can't really find much information about this device on the > net) - which turns off a red led when i load the driver and do a ztcfg. Then the tx (from TE410P to the Siemens equipment) circuit is ok but the rx may not be. > > I think the leds should turn green when the card senses a correct carrier > > and framing on the lines. > green is always a wounderful color ;-) > > so, what else could cause this ? I'd try to find out if the cable is wired the way the TE410P expects it to be. Do you know the pinout of both ends of the cables? RX (from the TE410P point of view) should be on the pins 1-2 at the TE410P end and TX on 4-5. Peter -- Peter Svensson ! Pgp key available by finger, fingerprint: <[EMAIL PROTECTED]> ! 8A E9 20 98 C1 FF 43 E3 07 FD B9 0A 80 72 70 AF ------------------------------------------------------------------------ Remember, Luke, your source will be with you... always... --__--__-- _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users End of Asterisk-Users Digest _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
