Re: [Asterisk-Users] IAX vs SIP
Thanks, this is exactly what I was looking for. I tried experimenting with different codecs myself, and GSM seems to be the only one that works... neither iLBC or Speex went thru. I'm using XLite v1.x Asterisk 0.5.0, wonder if it's a softphone's problem? I have got X-Lite to work with G.711 and GSM only, I have never been able to get it to work with iLBC or Speex.. I use iLBC over my IAX trunk and it works fine so I can only guess that there is some compatibility problem between X-Lite and Asterisk.. Later -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Skinny
Cool. I know for a fact my lab setup has it commented out, so with a small tweak I'll be doing real testing. IANAL, but SmartNET likely won't cover the skinny license. Even with Call Manager you have to buy a license for every phone you deploy. The license varies from model to model, but typically costs as much as the cheaper SIP phones that are now available. Lastly, what would be the best way to provide feedback on the channel? Thanks, Dan -Original Message- From: Jeremy McNamara [mailto:[EMAIL PROTECTED] Sent: Sunday, September 21, 2003 7:49 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Skinny At the present time you have to have a VALID ip address in bindaddr for Skinny to work. If bindaddr is either 0.0.0.0 or simply commented out all packets requiring the IP address contain 127.0.0.1. I forgot their nick, but someone in IRC recommended we make Asterisk be smart enough not to pick that interface, but I'm not sure of that is the problem or not. I simply have not had the time to investigate it whatsoever. Also, I have been advised against providing specific material resources to make Skinny work with Asterisk, especially the binary firmware images. So, your on your own for the firmware. I recommend a CCM v3.1 or higher as Cisco was nice enough to give us an XML based config file. Someone needs to do some research to see if having a SmartNet on the phone would be sufficient enough to be able to LEGALLY use the Skinny firmware. And, yes I do have my 7910 working with Asterisk. Which, btw, was graciously donated by a generous Asterisk user (he knows who he is), Granted only with ulaw and none of the soft buttons actually do anything, but I can and do make calls with it all day long. Hopefully in the next week or two I can find some time to spend on fixing all of the various annoying issues with chan_skinny or please feel free to contribute disclaimed patches. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ISDN BRI hardware
You can try AVM FRITZ with chan_capi from kapejod. srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de YO Internet Information Enviado el: lunes, 22 de septiembre de 2003 0:03 Para: [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] ISDN BRI hardware We sell: AVM B1 for development environment Eicon Diva Server BRI card for live system (on-board echo canceller) Tan www.telappliant.com - Original Message - From: Mark Hagler [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, September 21, 2003 10:43 PM Subject: [Asterisk-Users] ISDN BRI hardware Hi, Anybody have lots of experience with PCI ISDN cards and Asterisk? I'm thinking of getting a BRI in my house to deliver more advanced signaling to my PBX (yes, I'm a geek :-)) but I've never played with isdn4linux. Is there any particular BRI card that works better with Asterisk than any other? Also, can the BRI interface cards participate in conference, etc., since they aren't a Zaptel interface? Thanks, M ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN BRI hardware
Hi, Anybody have lots of experience with PCI ISDN cards and Asterisk? I'm thinking of getting a BRI in my house to deliver more advanced signaling to my PBX (yes, I'm a geek :-)) but I've never played with isdn4linux. Is there any particular BRI card that works better with Asterisk than any other? My suggestion (unless you have lots of money) is to get hold of an AVM fritz PCI card, then use Chan_Capi instead of I4L.. I use this setup and have found minimal echo problems and quite good performance all round... The card cost me £3 off ebay.. you really can't beat that.. :) Also, can the BRI interface cards participate in conference, etc., since they aren't a Zaptel interface? I haven't used conferencing but I believe you can load the ztdummy emulator to get it working.. Later.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN BRI hardware
-= On Sun, 21 Sep 2003 14:43:47 -0700, Mark Hagler [EMAIL PROTECTED] said: Hi, Anybody have lots of experience with PCI ISDN cards and Asterisk? I'm thinking of getting a BRI in my house to deliver more advanced signaling to my PBX (yes, I'm a geek :-)) but I've never played with isdn4linux. I tried to do the same here, and it looks like there are quite a few flavors of ISDN. The least-supported in Linux seems to be National ISDN (NI-1, NI-2), what I get here in North America. I'm guessing you're in the same boat, right? Is there any particular BRI card that works better with Asterisk than any other? So far, I've tried Dynalink, HFC-S, and Fritz! AVM PCI.. none has worked correctly with Linux/Asterisk, and I suspect it's an i4l problem with National ISDN. The incoming Caller-ID is always zero, even though I can see the text right there in the SETUP messages. So far, I haven't been able to complete a call. Others have had better luck with the CAPI drivers instead of i4l, but most CAPI drivers are binary only and support only EuroISDN. Klaus-Peter Junghanns suggested the Eicon Diva Server BRI active card as having Linux CAPI drivers with NI-1 support. Those cards are not cheap.. $520 at TheNerds.net. The Passive BRI cards are dirt cheap, under $5 on eBay, so I'm still motivated to make them work. Also, can the BRI interface cards participate in conference, etc., since they aren't a Zaptel interface? I could be wrong, but I read one reference that said you can have conferences so long as there is one Zaptel card in the machine somewhere (for timing?). You can use other cards for your trunk. I'm sure someone can jump in and correct me. In any case, it sounds like we're trying to do similar things. The new low prices on ISDN BRI makes them very attractive here. I especially like getting the CallerID data before the phone rings, and being able to bring up a call in milliseconds. Asterisk is a great match to ISDN BRI, since we are only allowed one phone handset per SPID -- two total. Deploying more phones throughout the house/office really begs for a small PBX like Asterisk and some SIP phones. That's the vision, anyway. -Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP NAT QUESTIONS
Hi, Is there anyway to use xlite though a nat I have a xlite - nat- asterisk. * is on a public IP. When I do this, I get an error on the asterisk server because it is trying to use the dirty ip of the computer running xlite. All of the settings in xlite seem to have no effect! Have you added nat=yet in your sip.conf? -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] G.729A + Cisco AS5300
Hello, I have 5 digium's g.729 codecs and succesfully register with asterisk, I have incomming call from my cisco AS5300 to Asterisk through IP. But Asterisk always use g711 ulaw instead of g.729. When I disable all other codecs other than g.729 in both cisco and asterisk, calls get dropped once connected. The codec list show onmy cisco AS5300for g.729 are: g729r8 g729br8 I suspect thatdigium's g.729 is not compatible with these codec found on cisco AS5300. Am I correct? Any advice will be helpful Foong
Re: [Asterisk-Users] G.729A + Cisco AS5300
the G.729 from digium are the G.729A type. Greetings, Tj -- Tjardick van der Kraan Tel +32 4 34 40 522Fax +32 4 34 40 525GSM +32 497 45 27 36 IAXtel: 1 700 344 0522FWD: 26322IPtel: 91331 Belgium - Original Message - From: Chee Foong To: [EMAIL PROTECTED] Sent: Monday, September 22, 2003 9:10 AM Subject: [Asterisk-Users] G.729A + Cisco AS5300 Hello, I have 5 digium's g.729 codecs and succesfully register with asterisk, I have incomming call from my cisco AS5300 to Asterisk through IP. But Asterisk always use g711 ulaw instead of g.729. When I disable all other codecs other than g.729 in both cisco and asterisk, calls get dropped once connected. The codec list show onmy cisco AS5300for g.729 are: g729r8 g729br8 I suspect thatdigium's g.729 is not compatible with these codec found on cisco AS5300. Am I correct? Any advice will be helpful Foong
[Asterisk-Users] Warnung: File dsp.c, Line 1198 ???
Hi, I have a problem with asterisk-0.5.0 which I don't understand. The monitor says when making a call: *CLI -- Executing Dial(SIP/roger-c456, Modem/ttyI0:BYEXTENSION|60|tTm) in new stack -- Called ttyI0:1234567890 WARNING[196621]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 2 frames WARNING[196621]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 2 frames and the last 2 lines are repeated approx a 100 times every second. Regardless which one of my 2 snom phones is used. I have a snom 100 and a snom 200 attached via SIP. With the snom 200 I have nevertheless good sound quality, but with the snom 100, I have a very bad sound. Sound seems to be interrupted several 10 times every second for a very short time. I already tried different codecs - hard to say, if there were any differences. Any ideas, what went wrong? Roger. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G.729A + Cisco AS5300
IC, does that means they are not compatible?. Funny thing is, call make from asterisk to AS5300is fine using codec G.729. But call from AS5300 to asterisk result in incompatible codec. This is very strange. Foong - Original Message - From: Tjardick van der Kraan To: [EMAIL PROTECTED] Sent: Monday, September 22, 2003 3:50 PM Subject: Re: [Asterisk-Users] G.729A + Cisco AS5300 the G.729 from digium are the G.729A type. Greetings, Tj -- Tjardick van der Kraan Tel +32 4 34 40 522Fax +32 4 34 40 525GSM +32 497 45 27 36 IAXtel: 1 700 344 0522FWD: 26322IPtel: 91331 Belgium - Original Message - From: Chee Foong To: [EMAIL PROTECTED] Sent: Monday, September 22, 2003 9:10 AM Subject: [Asterisk-Users] G.729A + Cisco AS5300 Hello, I have 5 digium's g.729 codecs and succesfully register with asterisk, I have incomming call from my cisco AS5300 to Asterisk through IP. But Asterisk always use g711 ulaw instead of g.729. When I disable all other codecs other than g.729 in both cisco and asterisk, calls get dropped once connected. The codec list show onmy cisco AS5300for g.729 are: g729r8 g729br8 I suspect thatdigium's g.729 is not compatible with these codec found on cisco AS5300. Am I correct? Any advice will be helpful Foong
Re: [Asterisk-Users] h.323 - success
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Monday 22 September 2003 04:02, Jeremy McNamara wrote: You have to enable ring indications exten = whatever,1,Dial,CAPI/22545070:b98013356|300|Tr That doesn't work when you use H323 directly. As in Dial(H323/ip$12.34.56.78|120|r) ... Works fine with OH323 though. - -- Regards, Tais M. Hansen ComX Networks Tel: +45-70257474 Fax: +45-70257374 -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.2 (GNU/Linux) iD8DBQE/bq4/2TEAILET3McRAmUkAJ97wvMuj+O5D1E3Uu9PBZ5G4bIt+QCgiFAA Ufbc9xNFKXOExxoXia0Qits= =iRbu -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G.729A + Cisco AS5300
Are you using SIP or H323? If SIP, what are the allow= and disallow= lines in your sip.conf? On Mon, 2003-09-22 at 03:08, Chee Foong wrote: IC, does that means they are not compatible?. Funny thing is, call make from asterisk to AS5300 is fine using codec G.729. But call from AS5300 to asterisk result in incompatible codec. This is very strange. Foong - Original Message - From: Tjardick van der Kraan To: [EMAIL PROTECTED] Sent: Monday, September 22, 2003 3:50 PM Subject: Re: [Asterisk-Users] G.729A + Cisco AS5300 the G.729 from digium are the G.729A type. Greetings, Tj -- Tjardick van der Kraan Tel +32 4 34 40 522 Fax +32 4 34 40 525 GSM +32 497 45 27 36 IAXtel: 1 700 344 0522 FWD: 26322 IPtel: 91331 Belgium - Original Message - From: Chee Foong To: [EMAIL PROTECTED] Sent: Monday, September 22, 2003 9:10 AM Subject: [Asterisk-Users] G.729A + Cisco AS5300 Hello, I have 5 digium's g.729 codecs and succesfully register with asterisk, I have incomming call from my cisco AS5300 to Asterisk through IP. But Asterisk always use g711 ulaw instead of g.729. When I disable all other codecs other than g.729 in both cisco and asterisk, calls get dropped once connected. The codec list show on my cisco AS5300 for g.729 are: g729r8 g729br8 I suspect that digium's g.729 is not compatible with these codec found on cisco AS5300. Am I correct? Any advice will be helpful Foong __ This message has been 'sanitized'. This means that potentially dangerous content has been rewritten or removed. The following log describes which actions were taken. Sanitizer (start=1064217921): Part (pos=3455): SanitizeFile (filename=unnamed.txt, mimetype=text/plain): Match (names=unnamed.txt, rule=1): ScanFile (file=/tmp/att-3f6ead42-MII-unnamed.txt): Scan succeeded, file is clean. Enforced policy: unknown Match (names=unnamed.txt, rule=3): Enforced policy: accept Added 1 bytes of scratch space. Total modifications so far: 1 Part (pos=5049): SanitizeFile (filename=unnamed.html, mimetype=text/html): Match (names=unnamed.html, rule=1): ScanFile (file=/tmp/att-3f6ead42-PQH-unnamed.html): Scan succeeded, file is clean. Enforced policy: unknown Match (names=unnamed.html, rule=3): Enforced policy: accept Added 1 bytes of scratch space. Note: Styles and layers give attackers many tools to fool the user and common browsers interpret Javascript code found within style definitions. References: - http://www.securityfocus.com/bid/630 - http://archives.indenial.com/hypermail/bugtraq/2001/January2001/0512.html Rewrote HTML tag: _style_0 _/STYLE_ as: _DANGEROUS_style_0 _/STYLE_ Rewrote HTML tag: _DIV_ as: _p__DANGEROUS_DIV_ Rewrote HTML tag: _/DIV_ as: _/p__DANGEROUS_DIV_ Rewrote HTML tag: _DIV_ as: _p__DANGEROUS_DIV_ Rewrote HTML tag: _/DIV_ as: _/p__DANGEROUS_DIV_ Rewrote HTML tag: _DIV_ as: _p__DANGEROUS_DIV_ Rewrote HTML tag: _/DIV_ as: _/p__DANGEROUS_DIV_ Rewrote HTML tag: _DIV_ as: _p__DANGEROUS_DIV_ Rewrote HTML tag: _/DIV_ as: _/p__DANGEROUS_DIV_ Rewrote HTML tag: _DIV_ as: _p__DANGEROUS_DIV_ Rewrote HTML tag: _/DIV_ as: _/p__DANGEROUS_DIV_ Rewrote HTML tag: _DIV_ as:
Re: [Asterisk-Users] h.323 - success
I have found that mixing the Dial() format with | can cause problems. Does Dial(H323/ip$12.34.56.78,120,r) work as expected? On Mon, 2003-09-22 at 03:09, Tais M. Hansen wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Monday 22 September 2003 04:02, Jeremy McNamara wrote: You have to enable ring indications exten = whatever,1,Dial,CAPI/22545070:b98013356|300|Tr That doesn't work when you use H323 directly. As in Dial(H323/ip$12.34.56.78|120|r) ... Works fine with OH323 though. - -- Regards, Tais M. Hansen ComX Networks Tel: +45-70257474 Fax: +45-70257374 -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.2 (GNU/Linux) iD8DBQE/bq4/2TEAILET3McRAmUkAJ97wvMuj+O5D1E3Uu9PBZ5G4bIt+QCgiFAA Ufbc9xNFKXOExxoXia0Qits= =iRbu -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Sample configs and more: http://www.fnords.org/~eric/asterisk/ BTEL Consulting +1-850-484-4535 x2111 (Pensacola) +1-504-595-3916 x2111 (New Orleans) +1-877-677-9643 x2111 (Toll Free) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Registration NOTIFY EVENT
Hi all, when I try register my netergy SIP Phone with *, I can't do it due to the next message: 1 headers, 2 lines Reliably Transmitting: NOTIFY sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.207:5060;branch=z9hG4bK3438300a From: asterisk sip:[EMAIL PROTECTED];tag=as34fa433f To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: application/simple-message-summary Content-Length: 37 Messages-Waiting: yes Voicemail: 1/2 (no NAT) to 192.168.0.155:5060 Sip read: SIP/2.0 405 Method Not Allowed Call-ID: [EMAIL PROTECTED] From: asterisksip:[EMAIL PROTECTED];tag=as34fa433f To: sip:[EMAIL PROTECTED] CSeq: 102 NOTIFY Via: SIP/2.0/UDP 192.168.0.207:5060;branch=z9hG4bK3438300a Allow: REFER,UPDATE,INFO,MESSAGE,OPTIONS,PRACK Content-Length: 0 NOTIFY meesage is nos supported by asterisk? Anyone can help me? Thanks in advance, srsergio ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] h.323 - success
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Monday 22 September 2003 10:16, Eric Wieling wrote: I have found that mixing the Dial() format with | can cause problems. Does Dial(H323/ip$12.34.56.78,120,r) work as expected? Doesn't change anything. Here's a better explanation of the problem. Using chan_h323, it doesn't matter which tech I choose to dial. It doesn't make the ringing sound on the h323 endpoint. I.e. h323 ep - chan_h323 asterisk 1 chan_iax2 - chan_iax2 asterisk 2 If 'asterisk 1' has a extension like this: exten = 1234,1,Dial(IAX2/user:[EMAIL PROTECTED]/[EMAIL PROTECTED],30,r) And 'asterisk 2': exten = 1234,1,Wait(10) exten = 1234,2,Answer() exten ... Dialing 1234 on the h323 endpoint would send the call to 'asterisk 2' but during those 10 seconds wait on 'asterisk 2', there's no indication on my h323 endpoint that it's actually ringing. Using chan_oh323 instead of the native h323, the problem magically disappears. - -- Regards, Tais M. Hansen ComX Networks Tel: +45-70257474 Fax: +45-70257374 -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.2 (GNU/Linux) iD8DBQE/brqt2TEAILET3McRAkEwAJ9Qa23Gmet470GBhU7NHQm6gXgWsQCfTbO6 mXEHmNtd7xgiQ4B8LrDuANY= =VKte -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G.729A + Cisco AS5300
Hello, I am using H.323 with chan_h323. Here is my config in h323.conf: allow=g729 if I set allow=ulaw, G7.11 alway get used. Therefore I disallow it. I want to use G.729. G.711 is too heavy for my network Any with AS5300 manage to get the digium's g.729 working Foong - Original Message - From: Eric Wieling [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, September 22, 2003 4:10 PM Subject: Re: [Asterisk-Users] G.729A + Cisco AS5300 Are you using SIP or H323? If SIP, what are the allow= and disallow= lines in your sip.conf? On Mon, 2003-09-22 at 03:08, Chee Foong wrote: IC, does that means they are not compatible?. Funny thing is, call make from asterisk to AS5300 is fine using codec G.729. But call from AS5300 to asterisk result in incompatible codec. This is very strange. Foong - Original Message - From: Tjardick van der Kraan To: [EMAIL PROTECTED] Sent: Monday, September 22, 2003 3:50 PM Subject: Re: [Asterisk-Users] G.729A + Cisco AS5300 the G.729 from digium are the G.729A type. Greetings, Tj -- Tjardick van der Kraan Tel +32 4 34 40 522 Fax +32 4 34 40 525 GSM +32 497 45 27 36 IAXtel: 1 700 344 0522 FWD: 26322 IPtel: 91331 Belgium - Original Message - From: Chee Foong To: [EMAIL PROTECTED] Sent: Monday, September 22, 2003 9:10 AM Subject: [Asterisk-Users] G.729A + Cisco AS5300 Hello, I have 5 digium's g.729 codecs and succesfully register with asterisk, I have incomming call from my cisco AS5300 to Asterisk through IP. But Asterisk always use g711 ulaw instead of g.729. When I disable all other codecs other than g.729 in both cisco and asterisk, calls get dropped once connected. The codec list show on my cisco AS5300 for g.729 are: g729r8 g729br8 I suspect that digium's g.729 is not compatible with these codec found on cisco AS5300. Am I correct? Any advice will be helpful Foong __ This message has been 'sanitized'. This means that potentially dangerous content has been rewritten or removed. The following log describes which actions were taken. Sanitizer (start=1064217921): Part (pos=3455): SanitizeFile (filename=unnamed.txt, mimetype=text/plain): Match (names=unnamed.txt, rule=1): ScanFile (file=/tmp/att-3f6ead42-MII-unnamed.txt): Scan succeeded, file is clean. Enforced policy: unknown Match (names=unnamed.txt, rule=3): Enforced policy: accept Added 1 bytes of scratch space. Total modifications so far: 1 Part (pos=5049): SanitizeFile (filename=unnamed.html, mimetype=text/html): Match (names=unnamed.html, rule=1): ScanFile (file=/tmp/att-3f6ead42-PQH-unnamed.html): Scan succeeded, file is clean. Enforced policy: unknown Match (names=unnamed.html, rule=3): Enforced policy: accept Added 1 bytes of scratch space. Note: Styles and layers give attackers many tools to fool the user and common browsers interpret Javascript code found within style definitions. References: - http://www.securityfocus.com/bid/630 - http://archives.indenial.com/hypermail/bugtraq/2001/January2001/0512.html Rewrote HTML tag: _style_0 _/STYLE_ as: _DANGEROUS_style_0 _/STYLE_ Rewrote HTML tag: _DIV_ as: _p__DANGEROUS_DIV_ Rewrote HTML tag: _/DIV_ as: _/p__DANGEROUS_DIV_ Rewrote HTML tag: _DIV_ as: _p__DANGEROUS_DIV_ Rewrote HTML tag: _/DIV_ as: _/p__DANGEROUS_DIV_ Rewrote HTML tag: _DIV_ as: _p__DANGEROUS_DIV_ Rewrote HTML tag: _/DIV_ as: _/p__DANGEROUS_DIV_ Rewrote HTML tag: _DIV_ as: _p__DANGEROUS_DIV_ Rewrote HTML tag: _/DIV_ as: _/p__DANGEROUS_DIV_ Rewrote HTML tag: _DIV_ as: _p__DANGEROUS_DIV_ Rewrote HTML tag: _/DIV_ as: _/p__DANGEROUS_DIV_ Rewrote HTML tag: _DIV_ as: _p__DANGEROUS_DIV_ Rewrote HTML tag: _/DIV_ as: _/p__DANGEROUS_DIV_ Rewrote HTML tag: _DIV_ as: _p__DANGEROUS_DIV_ Rewrote HTML
Re: [Asterisk-Users] ISDN BRI hardware
Also, can the BRI interface cards participate in conference, etc., since they aren't a Zaptel interface? I haven't used conferencing but I believe you can load the ztdummy emulator to get it working.. I am successfully using the zaptelrtc from http://www.junghanns.net/asterisk/ It uses the linux RTC interface, thus you must NOT have RTC compiled into your kernel! Also tried ztdummy.o but that needs usb-uhci.o for timing - I have uhci.o loaded. The hfcdummy.o driver didn't work with my hardware (AVM FRITZ!PCI), I guess it's for the Cologne Chipset based cards... Ciao, Alex ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] e100p and E-bit alarm indication
Hi Lele,
that did the work.
now my telco is happy too.
many thanks
Konrad
Lele Forzani wrote:
We connected an * box with an e100p to an E1/PRI from a telco here in Italy.
After we had it working perfectly the telco told us that, despite the circuit
appeared to work fine, and we could place calls on it, they had an E-bit2
alarm indication constantly on that caused the circuit to be flagged as
faulty every time.
(The E-bit indication, is an alarm sent back from us to the telco, telling
them we are getting CRC-errored data from them. It should be incrementing the
Far-End SES on their side)
Since the circuit appeared to work fine, calls went through, and the crc
counters on our side was zero, it was impossible we were getting that many
errors and something must have been wrong with our handling of the E-bit
signal.
I've come across the DS21554 framer documentation and i've seen that it has a
flag for enabling the E-bit generation in the TCR2 register and that the
wct1xxp.c wasn't setting it.
So i tried this small patch and the telco is perfectly happy with it, now, the
E-bit error has disappeared.
Since there had been a thread in May (started by Konrad Gorsky) about weird
far end CRC errors i'm posting in the hope to help somebody.
Note that i do not have a clue on what this does to the *T1* framer. I do not
have the specs for it!
bye
lele
--- zaptel/wct1xxp.c2003-09-12 10:12:01.0 +0200
+++ zaptel-i/wct1xxp.c 2003-09-11 19:24:53.0 +0200
@@ -411,13 +411,14 @@
int alreadyrunning = wc-span.flags ZT_FLAG_RUNNING;
long flags;
char *crcing = ;
- unsigned char ccr1, tcr1;
+ unsigned char ccr1, tcr1, tcr2;
spin_lock_irqsave(wc-lock, flags);
/* Build up config */
ccr1 = 0;
tcr1 = 8;
+ tcr2 = 0;
if (wc-span.lineconfig ZT_CONFIG_CCS) {
coding = CCS; /* Receive CCS */
ccr1 |= 8;
@@ -433,9 +434,11 @@
}
if (wc-span.lineconfig ZT_CONFIG_CRC4) {
ccr1 |= 0x11;
+ tcr2 |= 0x02; // xxx Enable E-bit alarm
crcing = with CRC4;
}
__t1_set_reg(wc, 0x12, tcr1);
+ __t1_set_reg(wc, 0x13, tcr2);
__t1_set_reg(wc, 0x14, ccr1);
__t1_set_reg(wc, 0x18, 0x20); /* 120 Ohm */
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Re: [Asterisk-Users] G.729A + Cisco AS5300
add a disallow=all above the allow=g729 line. On Mon, 2003-09-22 at 04:28, Chee Foong wrote: Hello, I am using H.323 with chan_h323. Here is my config in h323.conf: allow=g729 if I set allow=ulaw, G7.11 alway get used. Therefore I disallow it. I want to use G.729. G.711 is too heavy for my network Any with AS5300 manage to get the digium's g.729 working Foong - Original Message - From: Eric Wieling [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, September 22, 2003 4:10 PM Subject: Re: [Asterisk-Users] G.729A + Cisco AS5300 Are you using SIP or H323? If SIP, what are the allow= and disallow= lines in your sip.conf? On Mon, 2003-09-22 at 03:08, Chee Foong wrote: IC, does that means they are not compatible?. Funny thing is, call make from asterisk to AS5300 is fine using codec G.729. But call from AS5300 to asterisk result in incompatible codec. This is very strange. Foong - Original Message - From: Tjardick van der Kraan To: [EMAIL PROTECTED] Sent: Monday, September 22, 2003 3:50 PM Subject: Re: [Asterisk-Users] G.729A + Cisco AS5300 the G.729 from digium are the G.729A type. Greetings, Tj -- Tjardick van der Kraan Tel +32 4 34 40 522 Fax +32 4 34 40 525 GSM +32 497 45 27 36 IAXtel: 1 700 344 0522 FWD: 26322 IPtel: 91331 Belgium - Original Message - From: Chee Foong To: [EMAIL PROTECTED] Sent: Monday, September 22, 2003 9:10 AM Subject: [Asterisk-Users] G.729A + Cisco AS5300 Hello, I have 5 digium's g.729 codecs and succesfully register with asterisk, I have incomming call from my cisco AS5300 to Asterisk through IP. But Asterisk always use g711 ulaw instead of g.729. When I disable all other codecs other than g.729 in both cisco and asterisk, calls get dropped once connected. The codec list show on my cisco AS5300 for g.729 are: g729r8 g729br8 I suspect that digium's g.729 is not compatible with these codec found on cisco AS5300. Am I correct? Any advice will be helpful Foong __ This message has been 'sanitized'. This means that potentially dangerous content has been rewritten or removed. The following log describes which actions were taken. Sanitizer (start=1064217921): Part (pos=3455): SanitizeFile (filename=unnamed.txt, mimetype=text/plain): Match (names=unnamed.txt, rule=1): ScanFile (file=/tmp/att-3f6ead42-MII-unnamed.txt): Scan succeeded, file is clean. Enforced policy: unknown Match (names=unnamed.txt, rule=3): Enforced policy: accept Added 1 bytes of scratch space. Total modifications so far: 1 Part (pos=5049): SanitizeFile (filename=unnamed.html, mimetype=text/html): Match (names=unnamed.html, rule=1): ScanFile (file=/tmp/att-3f6ead42-PQH-unnamed.html): Scan succeeded, file is clean. Enforced policy: unknown Match (names=unnamed.html, rule=3): Enforced policy: accept Added 1 bytes of scratch space. Note: Styles and layers give attackers many tools to fool the user and common browsers interpret Javascript code found within style definitions. References: - http://www.securityfocus.com/bid/630 - http://archives.indenial.com/hypermail/bugtraq/2001/January2001/0512.html Rewrote HTML tag: _style_0 _/STYLE_ as: _DANGEROUS_style_0 _/STYLE_ Rewrote HTML tag: _DIV_ as: _p__DANGEROUS_DIV_ Rewrote HTML tag: _/DIV_ as: _/p__DANGEROUS_DIV_ Rewrote HTML tag: _DIV_ as: _p__DANGEROUS_DIV_ Rewrote HTML tag: _/DIV_ as: _/p__DANGEROUS_DIV_ Rewrote HTML tag: _DIV_ as: _p__DANGEROUS_DIV_ Rewrote HTML tag: _/DIV_ as: _/p__DANGEROUS_DIV_ Rewrote HTML tag: _DIV_ as: _p__DANGEROUS_DIV_ Rewrote HTML tag: _/DIV_ as: _/p__DANGEROUS_DIV_ Rewrote HTML tag: _DIV_ as: _p__DANGEROUS_DIV_ Rewrote HTML tag:
Re: [Asterisk-Users] G.729A + Cisco AS5300
hello, I have tried that but get disconnected once asterisk answer the call. Got the following error 1:02.899 H225 Answer:813ae50 h323.cxx(4167) H323 CreateLogicalChannel - unknown data type Guess it's the difference btw g.729 on AS5300 and g.729 on asterisk. Cisco AS5300 has G.729 and G.729 Annex-B while digium's is G.729 Annex-A. Still wondering why calling from asterisk to AS5300 works using the digium codec since they are different. Thanks Foong - Original Message - From: Eric Wieling [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, September 22, 2003 5:30 PM Subject: Re: [Asterisk-Users] G.729A + Cisco AS5300 add a disallow=all above the allow=g729 line. On Mon, 2003-09-22 at 04:28, Chee Foong wrote: Hello, I am using H.323 with chan_h323. Here is my config in h323.conf: allow=g729 if I set allow=ulaw, G7.11 alway get used. Therefore I disallow it. I want to use G.729. G.711 is too heavy for my network Any with AS5300 manage to get the digium's g.729 working Foong - Original Message - From: Eric Wieling [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, September 22, 2003 4:10 PM Subject: Re: [Asterisk-Users] G.729A + Cisco AS5300 Are you using SIP or H323? If SIP, what are the allow= and disallow= lines in your sip.conf? On Mon, 2003-09-22 at 03:08, Chee Foong wrote: IC, does that means they are not compatible?. Funny thing is, call make from asterisk to AS5300 is fine using codec G.729. But call from AS5300 to asterisk result in incompatible codec. This is very strange. Foong - Original Message - From: Tjardick van der Kraan To: [EMAIL PROTECTED] Sent: Monday, September 22, 2003 3:50 PM Subject: Re: [Asterisk-Users] G.729A + Cisco AS5300 the G.729 from digium are the G.729A type. Greetings, Tj -- Tjardick van der Kraan Tel +32 4 34 40 522 Fax +32 4 34 40 525 GSM +32 497 45 27 36 IAXtel: 1 700 344 0522 FWD: 26322 IPtel: 91331 Belgium - Original Message - From: Chee Foong To: [EMAIL PROTECTED] Sent: Monday, September 22, 2003 9:10 AM Subject: [Asterisk-Users] G.729A + Cisco AS5300 Hello, I have 5 digium's g.729 codecs and succesfully register with asterisk, I have incomming call from my cisco AS5300 to Asterisk through IP. But Asterisk always use g711 ulaw instead of g.729. When I disable all other codecs other than g.729 in both cisco and asterisk, calls get dropped once connected. The codec list show on my cisco AS5300 for g.729 are: g729r8 g729br8 I suspect that digium's g.729 is not compatible with these codec found on cisco AS5300. Am I correct? Any advice will be helpful Foong __ This message has been 'sanitized'. This means that potentially dangerous content has been rewritten or removed. The following log describes which actions were taken. Sanitizer (start=1064217921): Part (pos=3455): SanitizeFile (filename=unnamed.txt, mimetype=text/plain): Match (names=unnamed.txt, rule=1): ScanFile (file=/tmp/att-3f6ead42-MII-unnamed.txt): Scan succeeded, file is clean. Enforced policy: unknown Match (names=unnamed.txt, rule=3): Enforced policy: accept Added 1 bytes of scratch space. Total modifications so far: 1 Part (pos=5049): SanitizeFile (filename=unnamed.html, mimetype=text/html): Match (names=unnamed.html, rule=1): ScanFile (file=/tmp/att-3f6ead42-PQH-unnamed.html): Scan succeeded, file is clean. Enforced policy: unknown Match (names=unnamed.html, rule=3): Enforced policy: accept Added 1 bytes of scratch space. Note: Styles and layers give attackers many tools to fool the user and common browsers interpret Javascript code found within style definitions. References: - http://www.securityfocus.com/bid/630 -
Re: [Asterisk-Users] G.729A + Cisco AS5300
I doubt that it's a codec problem. Maybe chan_h323 doesnt' support G729. JerJer would know. On Mon, 2003-09-22 at 04:55, Chee Foong wrote: hello, I have tried that but get disconnected once asterisk answer the call. Got the following error 1:02.899 H225 Answer:813ae50 h323.cxx(4167) H323 CreateLogicalChannel - unknown data type Guess it's the difference btw g.729 on AS5300 and g.729 on asterisk. Cisco AS5300 has G.729 and G.729 Annex-B while digium's is G.729 Annex-A. Still wondering why calling from asterisk to AS5300 works using the digium codec since they are different. Thanks Foong - Original Message - From: Eric Wieling [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, September 22, 2003 5:30 PM Subject: Re: [Asterisk-Users] G.729A + Cisco AS5300 add a disallow=all above the allow=g729 line. On Mon, 2003-09-22 at 04:28, Chee Foong wrote: Hello, I am using H.323 with chan_h323. Here is my config in h323.conf: allow=g729 if I set allow=ulaw, G7.11 alway get used. Therefore I disallow it. I want to use G.729. G.711 is too heavy for my network Any with AS5300 manage to get the digium's g.729 working Foong - Original Message - From: Eric Wieling [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, September 22, 2003 4:10 PM Subject: Re: [Asterisk-Users] G.729A + Cisco AS5300 Are you using SIP or H323? If SIP, what are the allow= and disallow= lines in your sip.conf? On Mon, 2003-09-22 at 03:08, Chee Foong wrote: IC, does that means they are not compatible?. Funny thing is, call make from asterisk to AS5300 is fine using codec G.729. But call from AS5300 to asterisk result in incompatible codec. This is very strange. Foong - Original Message - From: Tjardick van der Kraan To: [EMAIL PROTECTED] Sent: Monday, September 22, 2003 3:50 PM Subject: Re: [Asterisk-Users] G.729A + Cisco AS5300 the G.729 from digium are the G.729A type. Greetings, Tj -- Tjardick van der Kraan Tel +32 4 34 40 522 Fax +32 4 34 40 525 GSM +32 497 45 27 36 IAXtel: 1 700 344 0522 FWD: 26322 IPtel: 91331 Belgium - Original Message - From: Chee Foong To: [EMAIL PROTECTED] Sent: Monday, September 22, 2003 9:10 AM Subject: [Asterisk-Users] G.729A + Cisco AS5300 Hello, I have 5 digium's g.729 codecs and succesfully register with asterisk, I have incomming call from my cisco AS5300 to Asterisk through IP. But Asterisk always use g711 ulaw instead of g.729. When I disable all other codecs other than g.729 in both cisco and asterisk, calls get dropped once connected. The codec list show on my cisco AS5300 for g.729 are: g729r8 g729br8 I suspect that digium's g.729 is not compatible with these codec found on cisco AS5300. Am I correct? Any advice will be helpful Foong __ This message has been 'sanitized'. This means that potentially dangerous content has been rewritten or removed. The following log describes which actions were taken. Sanitizer (start=1064217921): Part (pos=3455): SanitizeFile (filename=unnamed.txt, mimetype=text/plain): Match (names=unnamed.txt, rule=1): ScanFile (file=/tmp/att-3f6ead42-MII-unnamed.txt): Scan succeeded, file is clean. Enforced policy: unknown Match (names=unnamed.txt, rule=3): Enforced policy: accept Added 1 bytes of scratch space. Total modifications so far: 1 Part (pos=5049): SanitizeFile (filename=unnamed.html, mimetype=text/html): Match (names=unnamed.html, rule=1): ScanFile (file=/tmp/att-3f6ead42-PQH-unnamed.html): Scan succeeded, file is clean. Enforced policy: unknown Match (names=unnamed.html, rule=3): Enforced
[Asterisk-Users] Chan_h323 config
Hello, Camparing chan_h323 config with chan_oh323 config, In the codec section chan_oh323 allow me to specify frame value. Is there a equivalent in chan_h323? Or if not, what is the default frame value if I use G.729(digium). Foong
[Asterisk-Users] Setting up MySQL CDR??
Hi, I am running Redhat, I loaded the mysql and mysql-devel RPM's and then recompiled *.. I thought it would be that simple but it looks like I have missed something becasue it doesn't look like the module has been complied.. What did I leave out? -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G.729A + Cisco AS5300
I had a similar problem a while ago, The g729 negotiation with chan_h323 might cause problems sometimes with compatibility between g729a and g729b. While g729a and b are perfectly compatible, the as5300 might have problems recognizing g729b as g729. (I had to allow g729a,b and ab on my hardware to make it work with chan_h323, g729.) If you can't, a workaround is to use chan_oh323 instead of chan_h323. (you can allow different g729 codecs in that config if i recall correctly.) Or kindly ask the almighty JerJer to add the option for you in chan_h323. Joachim. At 04:57 22/09/2003 -0500, you wrote: I doubt that it's a codec problem. Maybe chan_h323 doesnt' support G729. JerJer would know. On Mon, 2003-09-22 at 04:55, Chee Foong wrote: hello, I have tried that but get disconnected once asterisk answer the call. Got the following error 1:02.899 H225 Answer:813ae50 h323.cxx(4167) H323 CreateLogicalChannel - unknown data type Guess it's the difference btw g.729 on AS5300 and g.729 on asterisk. Cisco AS5300 has G.729 and G.729 Annex-B while digium's is G.729 Annex-A. Still wondering why calling from asterisk to AS5300 works using the digium codec since they are different. Thanks Foong - Original Message - From: Eric Wieling [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, September 22, 2003 5:30 PM Subject: Re: [Asterisk-Users] G.729A + Cisco AS5300 add a disallow=all above the allow=g729 line. On Mon, 2003-09-22 at 04:28, Chee Foong wrote: Hello, I am using H.323 with chan_h323. Here is my config in h323.conf: allow=g729 if I set allow=ulaw, G7.11 alway get used. Therefore I disallow it. I want to use G.729. G.711 is too heavy for my network Any with AS5300 manage to get the digium's g.729 working Foong - Original Message - From: Eric Wieling [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, September 22, 2003 4:10 PM Subject: Re: [Asterisk-Users] G.729A + Cisco AS5300 Are you using SIP or H323? If SIP, what are the allow= and disallow= lines in your sip.conf? On Mon, 2003-09-22 at 03:08, Chee Foong wrote: IC, does that means they are not compatible?. Funny thing is, call make from asterisk to AS5300 is fine using codec G.729. But call from AS5300 to asterisk result in incompatible codec. This is very strange. Foong - Original Message - From: Tjardick van der Kraan To: [EMAIL PROTECTED] Sent: Monday, September 22, 2003 3:50 PM Subject: Re: [Asterisk-Users] G.729A + Cisco AS5300 the G.729 from digium are the G.729A type. Greetings, Tj -- Tjardick van der Kraan Tel +32 4 34 40 522 Fax +32 4 34 40 525 GSM +32 497 45 27 36 IAXtel: 1 700 344 0522 FWD: 26322 IPtel: 91331 Belgium - Original Message - From: Chee Foong To: [EMAIL PROTECTED] Sent: Monday, September 22, 2003 9:10 AM Subject: [Asterisk-Users] G.729A + Cisco AS5300 Hello, I have 5 digium's g.729 codecs and succesfully register with asterisk, I have incomming call from my cisco AS5300 to Asterisk through IP. But Asterisk always use g711 ulaw instead of g.729. When I disable all other codecs other than g.729 in both cisco and asterisk, calls get dropped once connected. The codec list show on my cisco AS5300 for g.729 are: g729r8 g729br8 I suspect that digium's g.729 is not compatible with these codec found on cisco AS5300. Am I correct? Any advice will be helpful Foong __ This message has been 'sanitized'. This means that potentially dangerous content has been rewritten or removed. The following log describes which actions were taken. Sanitizer (start=1064217921): Part (pos=3455): SanitizeFile (filename=unnamed.txt, mimetype=text/plain): Match (names=unnamed.txt, rule=1): ScanFile (file=/tmp/att-3f6ead42-MII-unnamed.txt): Scan succeeded,
[Asterisk-Users] Meetme Admin menu
Hello, Is there a asterisk developer guide/source code doc or something like that? I want to see if I can implement the admin menu function for meetme. Foong
Re: [Asterisk-Users] Budget Hotel PBX
-- Original Message --
From: Bill Schultz [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]
Date: Fri, 19 Sep 2003 17:28:18 -0800
I'm considering using asterisk to replace an existing PBX in a 40 room hotel and
would appreciate any comments, corrections or insight before I begin.
Only 8 PSTN connections are initially required but since the guests need dial-up
We setup a similar system using Asterisk. We are working with something very similar
so I know it will work. We are faxing and using modem on our system without problems.
internet access in the rooms it has to be Frac-T1 as opposed to using FXO ports on
a channel bank.
IP phones are not an option strictly because of price. The analog phones must
have FSK message waiting lights instead of the cheaper voltage type since asterisk
doesn't support that.
So, a TE410P {or 400} and two Zhone 24FXS channel banks will be used.
I have 8 Zhone 24FXS unit which are pretty much junk for this operation. We got some
great Atran 750 to replace them. If you value your time and money the Zhone are good
for testing and learning but for real world they just don't cut it!
I couldn't google up any info on what mobo but I'd like to start with a 450mhz since
I
have one laying around with 64bit slots but if that's marginal I could get a dual
Athlon server board or whatever.
We actually setup the system on a Dual Xeon Dell system and had to move it to a plain
single processor P4 system. It seems there is allot of noise from the Dual processor
system. We need to redo this and start the test over again. But the Plain Intel base
MB with the P4 and having only 256mg RAM is more power then what we really need.
I'd also greatly appreciate knowing if anyone out there is actually using asterisk in
a
similar hotel application today.
If you want you can email me directly. We have worked with Hotel systems for over 10
years now! And we are starting to move to the Asterisks now! It seems to be the best
bang for the money.
TIA
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[Asterisk-Users] SIP stage
Title: Mensaje
Hi,
I
would like to configure a stage for SIP phones. This stage would be the
next:
two
netergy SIP phones connected to Asterisk through chan_sip.
one
X100P or AVM FRITZ to outside lines.
I
think that sip.conf would be the next:
;;
SIP Configuration for Asterisk;[general]port =
5060
; Port to bind tobindaddr =
192.168.0.207
; Address to bind tocontext =
outgoing
; Default for incoming
callsdisallow=allallow=alawmaxexpirey=3600
; Max length of incoming registration we
allowdefaultexpirey=120
; Default length of incoming/outoing registration
[704]type=friendusername=704;secret=704host=192.168.0.154dtmfmode=rfc2833mailbox=704callerid=704context=outgoingreinvite=yescanreinvite=yesqualify=yesnat=-1
[705]type=friendusername=705;secret=705host=192.168.0.155;defaultip=192.168.0.5dtmfmode=rfc2833mailbox=705callerid=705context=outgoingreinvite=yescanreinvite=yesqualify=yesnat=-1
And my
extensions.conf would be the next:
[outgoing]
exten=i,1,Playback(invalid)exten=t,1,Hungup()
exten=_7XX,1,Goto(SIP|${EXTEN}|1)exten=_X,1,ChanIsAvail(CAPI/951014943CAPI/951014944)exten=_X,2,SubString,CANAL=${AVAILCHAN}|12|9exten=_X,3,Dial(CAPI/@${CANAL}:B${EXTEN}|17)
[SIP]
exten=704,1,Dial(SIP/704|tTm)exten=705,1,Dial(SIP/705|tTm)
are
these files correct?
Why
hwen I try call from one phone to other only rings once and then
hungup?
Any
idea,
thanks,
srsergio
Re: [Asterisk-Users] how many production systems are there?
-- Original Message -- From: Steve Totaro [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] Date: Sat, 20 Sep 2003 12:29:54 -0700 i am just curious how many * systems are in the real world with more than one user. do you run a certain version? you dont update CVS do you? any admins running a system of over twenty? over fifty? over one-hundred? I am an admin of a real world Asterisk system. About 96 anlog devices as in dial up modem lines, Faxes and just plain phone extensions. We are now starting to bridge our off site offices with this system. It's very good but hard to configure at the beginning. It it had better documentation this would be a system far ahead of the Nortel's out there. But since it does not it's hard to configure and maintain! But once you learn the system it starts to get easier. i deal with 3com and nec systems all day (i am cerified in the 3com nbx advanced network telephony, elite voice mail, CCNA, A+, nec ipk, and soon to be asterisk school of Northern Virginia (once its a mainstay and reliable). they are five nines but totally proprietary and extremely expensive. i think the nec runs dos and i know the 3com nbx run vxworks. the way i see it, i am sitting on a goldmine. i wish i could code but i would certainly pay the asterisk/digium squad to be my exclusive distributor. i will even pay to come visit your operations and host you at mine. no kidding at least let me come meet you guys and your operation. there is a company that sells and supports NEC systems. the name is telco, they even have a voice mail that fits the NEC (in skin and out) they completely support the product. i would like to position my company at this point. obviously you have cornered the open source market. time to take it further. total customer support,,, sure, in less than fifteen minutes in a real CDR and IVR, my system is down takes the ssh cake. otherwise i will respond but give me twenty four to ninety-six hours to fix your problem. besides playing with this thing, has anyone deployed it? what were your results, or ongoing results? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Setting up MySQL CDR??
Hi, I am running Redhat, I loaded the mysql and mysql-devel RPM's and then recompiled *.. I thought it would be that simple but it looks like I have missed something becasue it doesn't look like the module has been complied.. What did I leave out? Ok been doing some testing and I get the following when trying to build the cdr modules.. [EMAIL PROTECTED] cdr]# make ../mkdep -fPIC -I/usr/include/mysql `ls *.c` cc -fPIC -I/usr/include/mysql -c -o cdr_csv.o cdr_csv.c cc -shared -Xlinker -x -o cdr_csv.so cdr_csv.o cc -fPIC -I/usr/include/mysql -c -o cdr_mysql.o cdr_mysql.c cc -shared -Xlinker -x -o cdr_mysql.so cdr_mysql.o -lmysqlclient -lz -L/usr/lib/mysql /usr/bin/ld: cannot find -lz collect2: ld returned 1 exit status make: *** [cdr_mysql.so] Error 1 rm cdr_csv.o What is -lz ?? Is it looking for an application called z ?? Later.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] X100P and PCI-X slots
Does the X100P card work in PCI-X (3.3v) slots or will it in the future. Thanks Chad ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Setting up MySQL CDR??
The zlib compression library. zlib-devel or so and zlib would probably be the packages you are looking for. That did it.. Thanks.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Switch between calls without initiating a threeway converstaion
I was just wondering if there was a way that you could have two calls on one line and switch between the two without initiating a threeway conversation? I would imagine that Flash is the way to do this, but when I Flash twice, a 3-way call is initiated. If I turn threeway off, then I can't transfer. Also, is it possible to hang up one of the calls, and then continue talking to the second call? -- Thank you for your time __ Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software http://sitebuilder.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G.729A + Cisco AS5300
It does support it but you have to uncomment -DWANT-G729 in h323/Makefile On Mon, 22 Sep 2003, Eric Wieling wrote: I doubt that it's a codec problem. Maybe chan_h323 doesnt' support G729. JerJer would know. On Mon, 2003-09-22 at 04:55, Chee Foong wrote: hello, I have tried that but get disconnected once asterisk answer the call. Got the following error 1:02.899 H225 Answer:813ae50 h323.cxx(4167) H323 CreateLogicalChannel - unknown data type Guess it's the difference btw g.729 on AS5300 and g.729 on asterisk. Cisco AS5300 has G.729 and G.729 Annex-B while digium's is G.729 Annex-A. Still wondering why calling from asterisk to AS5300 works using the digium codec since they are different. Thanks Foong - Original Message - From: Eric Wieling [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, September 22, 2003 5:30 PM Subject: Re: [Asterisk-Users] G.729A + Cisco AS5300 add a disallow=all above the allow=g729 line. On Mon, 2003-09-22 at 04:28, Chee Foong wrote: Hello, I am using H.323 with chan_h323. Here is my config in h323.conf: allow=g729 if I set allow=ulaw, G7.11 alway get used. Therefore I disallow it. I want to use G.729. G.711 is too heavy for my network Any with AS5300 manage to get the digium's g.729 working Foong - Original Message - From: Eric Wieling [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, September 22, 2003 4:10 PM Subject: Re: [Asterisk-Users] G.729A + Cisco AS5300 Are you using SIP or H323? If SIP, what are the allow= and disallow= lines in your sip.conf? On Mon, 2003-09-22 at 03:08, Chee Foong wrote: IC, does that means they are not compatible?. Funny thing is, call make from asterisk to AS5300 is fine using codec G.729. But call from AS5300 to asterisk result in incompatible codec. This is very strange. Foong - Original Message - From: Tjardick van der Kraan To: [EMAIL PROTECTED] Sent: Monday, September 22, 2003 3:50 PM Subject: Re: [Asterisk-Users] G.729A + Cisco AS5300 the G.729 from digium are the G.729A type. Greetings, Tj -- Tjardick van der Kraan Tel +32 4 34 40 522 Fax +32 4 34 40 525 GSM +32 497 45 27 36 IAXtel: 1 700 344 0522 FWD: 26322 IPtel: 91331 Belgium - Original Message - From: Chee Foong To: [EMAIL PROTECTED] Sent: Monday, September 22, 2003 9:10 AM Subject: [Asterisk-Users] G.729A + Cisco AS5300 Hello, I have 5 digium's g.729 codecs and succesfully register with asterisk, I have incomming call from my cisco AS5300 to Asterisk through IP. But Asterisk always use g711 ulaw instead of g.729. When I disable all other codecs other than g.729 in both cisco and asterisk, calls get dropped once connected. The codec list show on my cisco AS5300 for g.729 are: g729r8 g729br8 I suspect that digium's g.729 is not compatible with these codec found on cisco AS5300. Am I correct? Any advice will be helpful Foong __ This message has been 'sanitized'. This means that potentially dangerous content has been rewritten or removed. The following log describes which actions were taken. Sanitizer (start=1064217921): Part (pos=3455): SanitizeFile (filename=unnamed.txt, mimetype=text/plain): Match (names=unnamed.txt, rule=1): ScanFile (file=/tmp/att-3f6ead42-MII-unnamed.txt): Scan succeeded, file is clean. Enforced policy: unknown Match (names=unnamed.txt, rule=3): Enforced policy: accept Added 1 bytes of scratch space. Total modifications so far: 1 Part (pos=5049): SanitizeFile
RE: [Asterisk-Users] False RING (incoming call) on Digium X101P FXO
On Saturday, September 20, 2003 10:37 PM, Mark Spencer [SMTP:[EMAIL PROTECTED] wrote: I was surprised to see that it's 240 volts (peak-to-peak)! Egad.. no wonder it shocks fingertips. 20Hz (50ms cycle), 2 second long clean sine waveform. I was just surprised to see twice as much voltage as expected. 240 sounds like a lot. Are you sure you were doing DC measurement? Normal is +/- 90 - 120, generally at least 48VRMS or about 70V. The local phone company will, while the phone is on hook, provide a 48 to 52 dc voltage at the central office. To ring a phone a 50 v ac signal is placed on top of this ~50 v dc. Don Pobanz Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Skinny
Dan Austin wrote: Cool. I know for a fact my lab setup has it commented out, so with a small tweak I'll be doing real testing. IANAL, but SmartNET likely won't cover the skinny license. Even with Call Manager you have to buy a license for every phone you deploy. The license varies from model to model, but typically costs as much as the cheaper SIP phones that are now available. Yep, I agree completely. Lastly, what would be the best way to provide feedback on the channel? http://bugs.digium.com/ Thanks, Dan -Original Message- From: Jeremy McNamara [mailto:[EMAIL PROTECTED] Sent: Sunday, September 21, 2003 7:49 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Skinny At the present time you have to have a VALID ip address in bindaddr for Skinny to work. If bindaddr is either 0.0.0.0 or simply commented out all packets requiring the IP address contain 127.0.0.1. I forgot their nick, but someone in IRC recommended we make Asterisk be smart enough not to pick that interface, but I'm not sure of that is the problem or not. I simply have not had the time to investigate it whatsoever. Also, I have been advised against providing specific material resources to make Skinny work with Asterisk, especially the binary firmware images. So, your on your own for the firmware. I recommend a CCM v3.1 or higher as Cisco was nice enough to give us an XML based config file. Someone needs to do some research to see if having a SmartNet on the phone would be sufficient enough to be able to LEGALLY use the Skinny firmware. And, yes I do have my 7910 working with Asterisk. Which, btw, was graciously donated by a generous Asterisk user (he knows who he is), Granted only with ulaw and none of the soft buttons actually do anything, but I can and do make calls with it all day long. Hopefully in the next week or two I can find some time to spend on fixing all of the various annoying issues with chan_skinny or please feel free to contribute disclaimed patches. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Chan_capi account code..
Hi, Where or wgat is the correct way to set the accountcode= setting when using chan_capi? Do I do it in capi.conf or extensions.conf with a setvar? -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G.729A + Cisco AS5300
Eric Wieling wrote: I doubt that it's a codec problem. Maybe chan_h323 doesnt' support G729. JerJer would know. I babysit systems that terminate hundreds of thousands of G.729 based H.323 calls per day using chan_h323 and As5300. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] how to dial a h323 destination ?
Hi all,
i have big problems to make a h323 call over the gatekeeper from my
provider.
The provider demanded following account data:
H323 ID:XXX-XXX-XX-X
DetinationNumer: XXX
I have configured the oh323.conf following:
gatekeeper=XX.XX.XXX.XXX
alias=XXX-XXX-XX-X
Isx the alias equal to the h323id ?
And how i have to make a call with the dial app ?
I have following config:
exten = _01099X.,1,Dial,OH323/${EXTEN:7}
exten = _01099X.,2,Hangup
I thought it would be enough when i give the destination number if i
registered at the gk, isn't it ?
Or is a ip and something like a userbname necessary ? And if how can i dial
so?
Can somebody help please ?
Thanks,
Thomas.
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Re: [Asterisk-Users] Meetme Admin menu
Its fairly simple.. meetme isn't that big you can find where the hooks are its commented in the code. bkw On Mon, 22 Sep 2003, Chee Foong wrote: Hello, Is there a asterisk developer guide/source code doc or something like that? I want to see if I can implement the admin menu function for meetme. Foong ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Switch between calls without initiating a threeway converstaion
I was just wondering if there was a way that you could have two calls on one line and switch between the two without initiating a threeway conversation? I would imagine that Flash is the way to do this, but when I Flash twice, a 3-way call is initiated. It's a tricky one isn't it.. There were changes to the way this worked in the UK a couple of years back which made it easier for people to use at the cost of lost functionality. It used to be that you made a call, flashed, made another call. If you flashed again you got dialtone and could press 1 to end the current call and revert to the other one, 2 to toggle between the two calls, or 3 to join them in a 3 way call. The same held true for call waiting - hear the beep and press flash then 0 to reject the call and turn off call waiting for the rest of the call, press 1 to end the current call and answer the new incoming call, or press 2 to switch between them. In both these scenarious you could do flash then 2 as many times as you wanted. In a call waiting scenario it gets confusing if the inbound call hungup just before you hit flash. The exchange knows the call's gone away, and thinks you want to do a consultative or three way call. It gives you dial tone inviting dialing of a number. Meanwhile, you're thinking you're in command mode and issue a single digit to toggle to the new call. The exchange waits for more digits, it thinks you're dialling a phone number.. confusion ensues! To stop the madness, they made it work more like it does in North America where it's all about the flash hook and little else. Back to your problem then.. hmm.. You could probably hack the code round a bit to give you the functionality you want for Zap attached devices, but I wonder how it all works with external boxes like ATA-186s.. it sees the flash hook and starts issuing SIP messages for 3 way calling and stuff? I bow to the greater knowledge on the list there - maybe someone can shed some light? Also, is it possible to hang up one of the calls, and then continue talking to the second call? Yes, this should work - the held call should recall back to you, making your phone ring. Cheers Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G.729A + Cisco AS5300
zoa wrote: I had a similar problem a while ago, The g729 negotiation with chan_h323 might cause problems sometimes with compatibility between g729a and g729b. While g729a and b are perfectly compatible, the as5300 might have problems recognizing g729b as g729. (I had to allow g729a,b and ab on my hardware to make it work with chan_h323, g729.) If you can't, a workaround is to use chan_oh323 instead of chan_h323. (you can allow different g729 codecs in that config if i recall correctly.) Or kindly ask the almighty JerJer to add the option for you in chan_h323. I don't see any mention of this on http://bugs.digium.com and a patch would certainly speed the process up. My plate is over-flowing as it is already, I simply cannot stuff any more on it right now. Sorry, Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MY Sql CDR
On Sunday 21 September 2003 11:30 pm, Uriel Carrasquilla wrote: I like it. I am thinking of putting this query in a C++ but I am a bit concern on 1) scalability 2) delays in setting up the calls shoud I be concerned? The query: mysql select sum(billsec) from cdr where calldate '2003-09-01 00:00:00' and '2' in (src,dst); This gets into the meat of database management, so it's a little far afield from this list, but I'll try to explain as best I can. It really depends upon the design of the database table and the indexes you have on various fields. For my referenced query, you'll want to have indexes on the fields in the WHERE clause (i.e. src, dst, calldate) if you get beyond about 10,000 records or so. Also, make sure that you define your table with type CHAR, not type VARCHAR, and have no text or blob fields. This will ensure that the table will be fixed width, which can speed up database access. And, you'll want to have an archival policy, whereby old records are purged from the active table. This keeps your table and index sizes down. If the database resides on the same machine as the Asterisk installation, use the Unix socket method of connecting to the database, as this removes the overhead of TCP and could produce as much as a 15% overall speed improvement. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] THIS IS STRANGE
Hello everyone, I have posted once a message that I had problem with Asterisk, ATA and X-Lite. The problem was: When I called from ATA to X-Lie it did not want to work. The connectuion apper in astersik but I could not hear anything. Right now I updated asterisk from CVS and I still have this problem but... When I call directly to X-Lite from ATA it doesn't work but when I call to X-lie with queue from ATA is works... It is really strange. -- Bart
[Asterisk-Users] connecting to ICH
Hi All, I need an example of sip.conf connection with ICH My connection don´t works Thanks! Miklos ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] connecting to ICH
I need an example of sip.conf connection with ICH
/etc/asterisk/sip.conf:
[iconnect]
type=friend
secret=1234
username=12345678
host=sipauth.deltathree.com
dtmfmode=inband
/etc/asterisk/extension.conf
exten = _61XX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],45)
Hope this helps. If not, what error messages are you getting?
Paul
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[Asterisk-Users] 3 fritz-cards pci
hello, got anybody succesfully setup asterisk with three avm fritz pci cards - using the howto described in http://www.quiss.org/caiviar/Two-Fritzcards-HOWTO i already have asterisk working with 2 cards, by when i add third card and compile driver ( see capiinit debug below ) asterisk freeze on capi initialization does anybody know how to solve this ? regards Marian -- Using /lib/modules/2.4.20/kernel/drivers/isdn/avmb1/capiutil.o Using /lib/modules/2.4.20/kernel/drivers/isdn/avmb1/kernelcapi.o Using /lib/modules/2.4.20/kernel/drivers/isdn/avmb1/capiutil.o Using /lib/modules/2.4.20/kernel/drivers/isdn/avmb1/kernelcapi.o Using /lib/modules/2.4.20/kernel/drivers/isdn/avmb1/capifs.o Using /lib/modules/2.4.20/kernel/drivers/isdn/avmb1/capi.o Using /lib/modules/2.4.20/kernel/drivers/isdn/avmb1/fcpci.o Warning: loading fcpci will taint the kernel: non-GPL license - Proprietary See http://www.tux.org/lkml/#export-tainted for information about tainted modules Using /lib/modules/2.4.20/kernel/drivers/isdn/avmb1/f2pci.o Warning: loading f2pci will taint the kernel: non-GPL license - Proprietary See http://www.tux.org/lkml/#export-tainted for information about tainted modules Using /lib/modules/2.4.20/kernel/drivers/isdn/avmb1/f3pci.o Warning: loading f3pci will taint the kernel: non-GPL license - Proprietary See http://www.tux.org/lkml/#export-tainted for information about tainted modules fcpci - -(0)- - - - f2pci - -(0)- - - - f3pci - -(0)- - - - 1 fcpci running fritz-pciA1 3.10-02 0xdc00 5 2 f2pci running fritz2-pci A1 3.10-02 0xe000 12 3 f3pci running fritz3-pci A1 3.10-02 0xe400 10 -- SUNTEQ s. r. o. Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/ A mind is like a parachute... it only works when it's open. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MS Outlook
If you are using Microsoft Outlook and you are reading this message you need to make 500% sure you are not propagating virii. I posted our support (at) nufone d0t net addy on this mailing list last night and have never posted it in an unprotected fashion like that anywhere else. So far today we have received over a hundred email virii to that address. I suggest you upgrade to a more secure email client that doesn't enable java, javascript or ActiveX. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ISDN and Whisper
Hi, Im new to PBX, and now IP-PBX. I found out about asterisk and need to find see if it is suitable. Firstly I live in Australia (any ACA rules I need to know before purchasing hardware?) Before I purchase hardware, can asterisk do the following, - 3 businesses want me to setup a small call centre to answer there calls. - Each business has a number that needs to be answered with there business name. Can a whisper be played to the operator to say, "Call from Company A" depending on what extension was dialled. (i.e.. Each company would have an extension.) - Can I use Telstra ISDN with a card like AVM Fritz? Is it reliable? I think it is the cheap card. - For the operators is there a compatible USB phone or headset (windows based)? From what I have read so far Asterisk supports everything else I need. Regards Matt
[Asterisk-Users] Also CR Spam filters
[EMAIL PROTECTED] needs to fix their spam filter. Please stop using it or learn to configure it. + 1 Sep 22 AntiSpam UOL (6828) RE:Re: [Asterisk-Users] MS Outlook Thanks, Brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MS Outlook
On Mon, 2003-09-22 at 13:42, Brian West wrote: I second that... I have received a load of virii from people on this list.. Received: from torch.junct.com (sootbox.junct.com [65.168.64.10]) by www.bkw.org (8.11.6/8.11.6) with ESMTP id h8MIcEJ06998 for [EMAIL PROTECTED]; Mon, 22 Sep 2003 13:38:14 -0500 Received: from wdxmvur (unknown [207.41.124.63]) by torch.junct.com (Postfix) with SMTP id 461DF4159; Mon, 22 Sep 2003 13:37:08 -0500 (CDT) For those of you that have no reason whatsoever to receive windows executables, here is a procmail rule that matches the beginning of a windows executable no matter what it is named. # Base 64 encoded windows executable :0B: *TVqQAAME//8AALgAQAAA You can use this and deliver the mail wherever you want to. This works on the last Sobig, klez and the Swen virus so far. This is what I had in my virii folder to test it against. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicemailmain2 user docs?
Has anyone browsed through the source code and made a list of menu option for VoiceMailMain2? Or know of some user documentation hiding in Internet land some place? If not there well be soon. Ho hum. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is there any MFC-R2 implementation for asterisk?
Any news on this regard? If this is not implemented yet, what alternatives do we have? A channel bank? - Original Message - From: Paulo Mannheimer [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, September 11, 2003 10:23 AM Subject: RE: [Asterisk-Users] Is there any MFC-R2 implementation for asterisk? Me too. I sent Steve an email about this, but didn't get a reply. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of LQ (Asterisk) Sent: September 11, 2003 10:19 AM To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Subject: [Asterisk-Users] Is there any MFC-R2 implementation for asterisk? The last thing that I read about it was: Steve Underwood [EMAIL PROTECTED] wrote on Sep 3: Is EM designed to work with the E1 driver code? I think probably not. I had to fix some things to get proper access to the CAS signaling bits when I implemented MFC/R2... So, apparently he implemented it. I was trying to contact Steve, but he isn't answering me. Does anybody have any news about it? Regards, Pablo. -Original Message- From: Herry Sitepu [mailto:[EMAIL PROTECTED] Posted At: Thursday, September 11, 2003 5:07 Posted To: Asterisk Conversation: [Asterisk-Users] Is there any MFC-R2 implementation for asterisk? Subject: [Asterisk-Users] Is there any MFC-R2 implementation for asterisk? Hi guys, Is there anyone has implemented MFC-R2 for astrisk? Regards Herry Sitepu ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] re: Anyone looking for IP Phones?
My company has approx. 500 Cisco CP-7960G IP Phones that are coming out of service. They were deployed for about 6 months. These include the AC power adapter and station license. We also have some other related equipment. If someone is reading this and is interested, shoot me an email [EMAIL PROTECTED] Thanks Cory Andrews * b2 Technologies ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] failed to load chan_zap
Suddenly after recompiling my 2.4.22 kernel I can no longer load chan_zap: Sep 22 21:25:08 ERROR[16384]: File chan_zap.c, Line 5145 (mkintf): Unable to get span status: Inappropriate ioctl for device Sep 22 21:25:08 ERROR[16384]: File chan_zap.c, Line 6638 (load_module): Unable to register channel '1' Sep 22 21:25:08 WARNING[16384]: File loader.c, Line 301 (ast_load_resource): chan_zap.so: load_module failed, returning -1 Sep 22 21:25:08 WARNING[16384]: File loader.c, Line 396 (load_modules): Loading module chan_zap.so failed! The wcfxo module loaded without warning: Sep 22 21:23:44 zenon kernel: Zapata Telephony Interface Registered on major 196 Sep 22 21:23:44 zenon kernel: Found a Wildcard FXO: Wildcard X101P Sep 22 21:23:44 zenon kernel: Registered tone zone 2 (France) What could be wrong? -- (remember when we spent millions coming up with a pen that would write in zero-G and the Russians just used pencils?) -- limekiller4 on /. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemailmain2 user docs?
James Sizemore wrote: Has anyone browsed through the source code and made a list of menu option for VoiceMailMain2? Or know of some user documentation hiding in Internet land some place? If not there well be soon. Ho hum. Start here http://www.voip-info.org/wiki-Asterisk+cmd+VoiceMailMain2 Also check here http://www.voip-info.org/wiki-Asterisk+config+voicemail.conf ...and if you don't find what you're looking for, please help us add more information to the wiki. I've asked earlier on the mailinglist without getting any answer: - What's the differences between voicemail and voicemail2 ? Thank you! /O ...this time with the correct URL :-) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] failed to load chan_zap
cvs update the zaptel source and make clean install it. Jeremy McNamara Louis-David Mitterrand wrote: Suddenly after recompiling my 2.4.22 kernel I can no longer load chan_zap: Sep 22 21:25:08 ERROR[16384]: File chan_zap.c, Line 5145 (mkintf): Unable to get span status: Inappropriate ioctl for device Sep 22 21:25:08 ERROR[16384]: File chan_zap.c, Line 6638 (load_module): Unable to register channel '1' Sep 22 21:25:08 WARNING[16384]: File loader.c, Line 301 (ast_load_resource): chan_zap.so: load_module failed, returning -1 Sep 22 21:25:08 WARNING[16384]: File loader.c, Line 396 (load_modules): Loading module chan_zap.so failed! The wcfxo module loaded without warning: Sep 22 21:23:44 zenon kernel: Zapata Telephony Interface Registered on major 196 Sep 22 21:23:44 zenon kernel: Found a Wildcard FXO: Wildcard X101P Sep 22 21:23:44 zenon kernel: Registered tone zone 2 (France) What could be wrong? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Voicemailmain2 user docs?
I've asked earlier on the mailinglist without getting any answer: - What's the differences between voicemail and voicemail2 ? From my initial look I thought it was that voicemail2 allowed contexts as well as mailboxes.. so you could do virtual hosting or multicompany working.. Company A has mailbox 1234, as does company B, but they're completely separate or partitioned? Reading the list since then, I think it's also true to say that new features etc are all being added to Voicemail2 and that the voicemail app isn't being worked on any more? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Status of shipdate on the 4 port FX0 card?
Does any-e-one know if the 4 port FX0 cards will be shipping anytime soon? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] app_festival volume problems
I'm using app_festival to speak some text to callers. I'm having two problems with this. The first is with IAX calls (I've not tried others) the first few seconds of the speech is garbled. The second problem I'm having is the the volume of the speech IS VERY LOUD. I tried putting the following in the siteinit.scm but it didn't seem to make any difference. (set! default_after_synth_hooks (list (lambda (utt) (utt.wave.rescale utt .5 t Does anyone have any suggestions? -- Sample configs and more: http://www.fnords.org/~eric/asterisk/ BTEL Consulting +1-850-484-4535 x2111 (Pensacola) +1-504-595-3916 x2111 (New Orleans) +1-877-677-9643 x2111 (Toll Free) This message has been 'sanitized'. This means that potentially dangerous content has been rewritten or removed. The following log describes which actions were taken. Sanitizer (start=1064261367): ParseHeader (): Using Eric Wieling [EMAIL PROTECTED] as reply-to address. Using [EMAIL PROTECTED] as errors address. Got MIME info: _disposition=inline, _encoding=7bit, _type=text/plain, boundary=, charset=iso-8859-1 Finished parsing message header. Forcing message to be multipart/mixed, to facilitate logging. Parsing body as text/plain WrapWithMultipart Writer (pos=726): Set MIME info to: _encoding=8bit, _type=multipart/mixed, boundary=MIMEStream=_0+167520_3733915496221_57676455469 CleanMultipart ParserUnclosedMultipart Part (pos=800): ParseHeader (): Got MIME info: _disposition=inline, _encoding=7bit, _type=text/plain, boundary=, charset=iso-8859-1 Parsing body as text/* CleanUnknown CleanText Added 1 bytes of scratch space. Writer (pos=648): Set MIME info to: _disposition=inline, _encoding=7bit, _type=text/plain, boundary=, charset=iso-8859-1 Total modifications so far: 1 Added 1 bytes of scratch space. Anomy 0.0.0 : Sanitizer.pm $Id: Sanitizer.pm,v 1.79 2003/06/19 19:22:00 bre Exp $
Re: [Asterisk-Users] failed to load chan_zap
On Mon, Sep 22, 2003 at 03:41:43PM -0400, Jeremy McNamara wrote: cvs update the zaptel source and make clean install it. That did it, thanks a lot. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Undocumented variables in chan_sip.c
Trying to read and understand bits and pieces of chan_sip.c I've found these I would like someone to clarify: * srvlookup=yes|no * pedantic * canreinvite=update|yes --update seems new Being curious, especially for srvlookup functionality... /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Anyone looking for IP Phones?
On Mon, Sep 22, 2003 at 03:25:00PM -0400, Sales wrote: My company has approx. 500 Cisco CP-7960G IP Phones that are coming out of service. They were deployed for about 6 months. These include the AC power adapter and station license. We also have some other related equipment. If someone is reading this and is interested, shoot me an email [EMAIL PROTECTED] Man, 500 phones at ~ $450 a pop decommissioned after a mere 6 months? Did they not fulfil your needs? Were you disappointed with them? As I'm thinking about deploying cisco 79xx phones, I'm just curious about your experience. -- Logiciels libres : nourris au code source sans farine animale. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MS Outlook
Actually, MS Outlook by default blocks all executables. I'm not sure why there is so much negativity around the Outlook client. Perhaps we should all go back to the cave and use Pine. -Sean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Critchfield Sent: Monday, September 22, 2003 2:10 PM To: [EMAIL PROTECTED] On Mon, 2003-09-22 at 13:42, Brian West wrote: I second that... I have received a load of virii from people on this list.. Received: from torch.junct.com (sootbox.junct.com [65.168.64.10]) by www.bkw.org (8.11.6/8.11.6) with ESMTP id h8MIcEJ06998 for [EMAIL PROTECTED]; Mon, 22 Sep 2003 13:38:14 -0500 Received: from wdxmvur (unknown [207.41.124.63]) by torch.junct.com (Postfix) with SMTP id 461DF4159; Mon, 22 Sep 2003 13:37:08 -0500 (CDT) For those of you that have no reason whatsoever to receive windows executables, here is a procmail rule that matches the beginning of a windows executable no matter what it is named. # Base 64 encoded windows executable :0B: *TVqQAAME//8AALgAQAA A You can use this and deliver the mail wherever you want to. This works on the last Sobig, klez and the Swen virus so far. This is what I had in my virii folder to test it against. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Anyone looking for IP Phones?
-- Original Message -- From: Louis-David Mitterrand [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] Date: Mon, 22 Sep 2003 22:28:40 +0200 On Mon, Sep 22, 2003 at 03:25:00PM -0400, Sales wrote: My company has approx. 500 Cisco CP-7960G IP Phones that are coming out of service. They were deployed for about 6 months. These include the AC power adapter and station license. We also have some other related equipment. If someone is reading this and is interested, shoot me an email [EMAIL PROTECTED] Man, 500 phones at ~ $450 a pop decommissioned after a mere 6 months? Did they not fulfil your needs? Were you disappointed with them? As I'm thinking about deploying cisco 79xx phones, I'm just curious about your experience. I am very interested in why your no longer using these phones. We have one for testing and so far it's not working. Are they not working? Is the main problem configuration? Any information will be very helpful. Logiciels libres : nourris au code source sans farine animale. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk call waiting X100P - MGCP ata 186
I am running CVS-09/11/03-14:03 on Redhat 9.0 Trying to get call waiting / call waiting callerid working. The setup is: X100P asterisk - MGCP -- analog phone. What changes to I need to make to my mgcp.conf and extensions.conf file to allow answering of Call waiting calls? And how do you answer call waiting calls with the system. I have usecallerid=yes, hidecallerid=no, callwaiting=yes, callwaitingcallerid=yes, threewaycalling=yes, transfer=yes, cancallforward=yes, callreturn=yes, echocalcel=yes and echocancelwhenbridged=yes in the zapta.conf file. How do you use these features? Or where can I find some documentation. I found some information about *0 on the thread but I couldn't find the corresponding config information. Also, is thier a list of functions like *0 and how to use them. Also, on another note. Using this setup to make normal calls I am getting significant echo for the first 15-20 seconds of a call and then it goes away. I assume this is the echo canceller but is their any way to fix this. Thanks for your help. Chad Graham ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] re: Anyone looking for IP Phones?
Are you selling the phones individually? -Original Message- From: Sales [mailto:[EMAIL PROTECTED] Sent: Monday, September 22, 2003 3:25 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] re: Anyone looking for IP Phones? My company has approx. 500 Cisco CP-7960G IP Phones that are coming out of service. They were deployed for about 6 months. These include the AC power adapter and station license. We also have some other related equipment. If someone is reading this and is interested, shoot me an email [EMAIL PROTECTED] Thanks Cory Andrews * b2 Technologies ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MS Outlook
On Mon, 2003-09-22 at 14:30, Sean Heiney wrote: Actually, MS Outlook by default blocks all executables. I'm not sure why there is so much negativity around the Outlook client. Perhaps we should all go back to the cave and use Pine. -Sean Well, that's Outlook 2002 (and maybe with a Service Pack?). Outlook 2000 (which most Outhouse.. er, I mean Outlook users are using) does not block the executables. In fact, they can be run just by previewing the message. Unfortunately, this is not the time or place for the Outlook flame war... Jared Smith ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MS Outlook
What am I if I use mutt...besides virus free? ;) On Mon, Sep 22, 2003 at 04:30:49PM -0400, Sean Heiney wrote: Actually, MS Outlook by default blocks all executables. I'm not sure why there is so much negativity around the Outlook client. Perhaps we should all go back to the cave and use Pine. -Sean ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Anyone looking for IP Phones?
On Mon, 2003-09-22 at 14:33, Ariel Batista wrote: I am very interested in why your no longer using these phones. We have one for testing and so far it's not working. Are they not working? Is the main problem configuration? Any information will be very helpful. I think you're misunderstanding... this company resells used Cisco equipment. I don't think they actually used these phones, they're just reselling them. Jared Smith ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Anyone looking for IP Phones?
On Mon, Sep 22, 2003 at 04:33:44PM -0400, Ariel Batista wrote: I am very interested in why your no longer using these phones. We have one for testing and so far it's not working. Are they not working? Is the main problem configuration? Any information will be very helpful. There web site indicates that they are ligquidators of dot bomb's. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MS Outlook
On Mon, 2003-09-22 at 15:30, Sean Heiney wrote: Actually, MS Outlook by default blocks all executables. I'm not sure why there is so much negativity around the Outlook client. Perhaps we should all go back to the cave and use Pine. I'll assume you don't understand the english words you just wrote well enough to defend yourself. Outlook does not block executables. It receives them via mail like any other mail message. It by default doesn't run executables that are sent as executables. But we all know about the current stupidities of Microsoft in that they look at the mime header to determine if it is safe to use the file(wav, mid, txt, whatever that should be a data file), but then executes the file so that they can use a shortcut to whatever app you defined to run that data file with. The problem being that they package exe files with a mime header for one of those innocuous files and the executable shortcut runs the virus. Not to mention that Outlook is set to by default to display HTML email and that a HTML mail with an embedded link to the data file inside will cause automatic running of the virus. So Outlook is not going to block the attachment from taking up residence on your drive. Outlook has poor security checking, and can be easily tricked into doing evil things. Microsoft recently stated themselves that Windows is not designed to sit on the internet out of the box, but requires a fair amount of hardening. This applies to all their other software as well as it is all tightly integrated. Admit it, Microsoft has been patching crap software for a long time. Linux had an advantage of not caring about market share and trying to do things the right way. Linux also grew up after the internet was around and while it was gaining popularity therefore it has had to grow up in a rough neighborhood and keep itself hardened. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Critchfield Sent: Monday, September 22, 2003 2:10 PM To: [EMAIL PROTECTED] On Mon, 2003-09-22 at 13:42, Brian West wrote: I second that... I have received a load of virii from people on this list.. Received: from torch.junct.com (sootbox.junct.com [65.168.64.10]) by www.bkw.org (8.11.6/8.11.6) with ESMTP id h8MIcEJ06998 for [EMAIL PROTECTED]; Mon, 22 Sep 2003 13:38:14 -0500 Received: from wdxmvur (unknown [207.41.124.63]) by torch.junct.com (Postfix) with SMTP id 461DF4159; Mon, 22 Sep 2003 13:37:08 -0500 (CDT) For those of you that have no reason whatsoever to receive windows executables, here is a procmail rule that matches the beginning of a windows executable no matter what it is named. # Base 64 encoded windows executable :0B: *TVqQAAME//8AALgAQAA A You can use this and deliver the mail wherever you want to. This works on the last Sobig, klez and the Swen virus so far. This is what I had in my virii folder to test it against. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemailmain2 user docs?
I have looked both these URLs over , neither is a User level description of the menu choices. I'm trudging through the code now, the only thing I have found so far that is not listed in the voice mail prompts is that you can press 0 if you have a o extinction in the same contest. I was hoping there were some keys for the users to skip the greetings or a key to goto the VoiceMailMain from VoiceMail. But have not seen any yet. Olle E. Johansson wrote: James Sizemore wrote: Has anyone browsed through the source code and made a list of menu option for VoiceMailMain2? Or know of some user documentation hiding in Internet land some place? If not there well be soon. Ho hum. Start here http://www.voip-info.org/wiki-Asterisk+cmd+VoiceMailMain2 Also check here http://www.voip-info.org/wiki-Asterisk+config+voicemail.conf ...and if you don't find what you're looking for, please help us add more information to the wiki. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MS Outlook
do you like outlook look feel ? fdisk /dev/hda install Linux (redhat/mandrake/slack/debian/gentoo) install evolution - it rocks Il lun, 2003-09-22 alle 22:30, Sean Heiney ha scritto: Actually, MS Outlook by default blocks all executables. I'm not sure why there is so much negativity around the Outlook client. Perhaps we should all go back to the cave and use Pine. -Sean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Critchfield Sent: Monday, September 22, 2003 2:10 PM To: [EMAIL PROTECTED] On Mon, 2003-09-22 at 13:42, Brian West wrote: I second that... I have received a load of virii from people on this list.. Received: from torch.junct.com (sootbox.junct.com [65.168.64.10]) by www.bkw.org (8.11.6/8.11.6) with ESMTP id h8MIcEJ06998 for [EMAIL PROTECTED]; Mon, 22 Sep 2003 13:38:14 -0500 Received: from wdxmvur (unknown [207.41.124.63]) by torch.junct.com (Postfix) with SMTP id 461DF4159; Mon, 22 Sep 2003 13:37:08 -0500 (CDT) For those of you that have no reason whatsoever to receive windows executables, here is a procmail rule that matches the beginning of a windows executable no matter what it is named. # Base 64 encoded windows executable :0B: *TVqQAAME//8AALgAQAA A You can use this and deliver the mail wherever you want to. This works on the last Sobig, klez and the Swen virus so far. This is what I had in my virii folder to test it against. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Pickup problem with cisco 7960 (SIP)
Please help! When I try to place a call pickup from a cisco phone 7960 using *8 the call is picked up but the other phone continues ringing. Is there any problem with call pickup in SIP. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Example weather report AGI by Zip Code using Festival available
I have posted a link to the tarball of my rather simple AGI script that allows a user to input a Zip Code (USA only) via DTMF and have the current weather conditions spoken to them. This is the first release and I'm sure it will have some bugs. It requires a few modules from CPAN and the asterisk-perl AGI interface. It's a very small script. Available at http://www.fnords.org/~eric/asterisk/ --Eric -- Sample configs and more: http://www.fnords.org/~eric/asterisk/ BTEL Consulting +1-850-484-4535 x2111 (Pensacola) +1-504-595-3916 x2111 (New Orleans) +1-877-677-9643 x2111 (Toll Free) This message has been 'sanitized'. This means that potentially dangerous content has been rewritten or removed. The following log describes which actions were taken. Sanitizer (start=1064266495): ParseHeader (): Using Eric Wieling [EMAIL PROTECTED] as reply-to address. Using [EMAIL PROTECTED] as errors address. Got MIME info: _disposition=inline, _encoding=7bit, _type=text/plain, boundary=, charset=iso-8859-1 Finished parsing message header. Forcing message to be multipart/mixed, to facilitate logging. Parsing body as text/plain WrapWithMultipart Writer (pos=761): Set MIME info to: _encoding=8bit, _type=multipart/mixed, boundary=MIMEStream=_0+342_4934350964362533_64804639686 CleanMultipart ParserUnclosedMultipart Part (pos=835): ParseHeader (): Got MIME info: _disposition=inline, _encoding=7bit, _type=text/plain, boundary=, charset=iso-8859-1 Parsing body as text/* CleanUnknown CleanText Added 1 bytes of scratch space. Writer (pos=602): Set MIME info to: _disposition=inline, _encoding=7bit, _type=text/plain, boundary=, charset=iso-8859-1 Total modifications so far: 1 Added 1 bytes of scratch space. Anomy 0.0.0 : Sanitizer.pm $Id: Sanitizer.pm,v 1.79 2003/06/19 19:22:00 bre Exp $
Reply Button was RE: [Asterisk-Users] MS Outlook
Just to add another post to that thread... I'm wondering why people don't use the 'REPLY' button correctly... some (read many) just hit REPLY on any list email to start a new thread... resulting in a messed-up view in thread aware clients... thread views are very good, since can give an overview of the entire forum, but improper REPLY use will put messages into the wrong thread... resulting into ignored messages? Ever wondered what 'New Message' 'Reply' means? I don't want to switch off my threaded view... -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Pickup problem with cisco 7960 (SIP)
On Mon, 2003-09-22 at 15:42, Manuel Marn Garca wrote: Please help! When I try to place a call pickup from a cisco phone 7960 using *8 the call is picked up but the other phone continues ringing. Is there any problem with call pickup in SIP. It's a known problem... I wish someone would hurry up and fix it. Jared Smith ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Speaking of Outlook
Does anybody have a reasonable solution for an Outlook MAPI plugin that works with asterisk? At very least, I would like Asterisk to push incoming call information to the computer, which should then open an Outlook form, launch a web browser, etc. Beyond that, it would be cool to have Outlook initiate outgoing calls. Shouldn't be too difficult, and I know some of you are working along similar lines. Outlook itself is only an example. What I'm looking for is a simple incoming call, launch a browser with the callerid in the query string type app for windows. Thanks, --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Pickup problem with cisco 7960 (SIP)
Here's one thats way out in left field... don't use call pickup! :P Problem solved sorta! bkw On Mon, 22 Sep 2003, Jared Smith wrote: On Mon, 2003-09-22 at 15:42, Manuel Marn Garca wrote: Please help! When I try to place a call pickup from a cisco phone 7960 using *8 the call is picked up but the other phone continues ringing. Is there any problem with call pickup in SIP. It's a known problem... I wish someone would hurry up and fix it. Jared Smith ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Last call: Asterisk BoF in Boston, Tuesday 23rd
Hello - The final schedule for the Asterisk birds-of-a-feather meeting (as an adjunct to the VON conference) in Boston looks like this: Tuesday, September 23rd at 8:15 at VinnyT's of Boston near the Hynes Convention Center. We'll try to get a corner booth in the downstairs room, and look for the guy wearing the blue button-down shirt with VON logos on the breast pocket - that will be me. JT Details: Vinny T's of Boston Back Bay 867 Boylston Street Boston MA 617.262.6699 Directions: From I-90 (Mass Turnpike): Go to the Prudential Center, Copley Square exit. When you exit, it splits; follow the signs to the Prudential Center (left side of split). When you come up out of the tunnel, you will be on Huntington Ave. Take the next 3 right hand turns and you will be on Boylston St. Vinny's is located between Gloucester and Fairfield Streets, directly across from the Prudential Building. From the Central Artery (from the South Shore and the North Shore): Take the Storrow Drive exit from the Artery and follow the signs to the Back Bay (it is a left turn exit). Turn right at the first light and go Gloucester St. Turn left. Gloucester ends at the intersection with Boylston St. Turn left and Vinny's is immediately on the left, directly across from the Prudential Building. From the Orange Line on the T: exit at the Back Bay stop, use the Dartmouth Street Exit. Walk down Dartmouth to Boylston St. Turn left on Boylston St. and go 2.5 blocks (between Fairfield and Gloucester). Vinny's will be across the street from the Prudential Building (which you should be in front of). From the Green Line on the T: Take any OUTBOUND Green train. You can get off at either Copley (which is in front of the Public Library), or Convention Center (which exits on Mass Ave.). From the library, Vinny's is 3 blocks left. From the Mass Ave exit of the T, turn left, walk .5 blocks to Boylston St., cross the bridge and Vinny's is 1.5 blocks from the fire station you will pass. The major cross streets in the Back Bay are alphabetically named: Arlington Berkely Clarendon Dartmouth Exeter Fairfield Gloucester If you get lost, all the streets are one way, so just make a right loop or a left loop to the correct cross street and these will take you back to Boylston. Or if all else fails, look up at the skyline. Find the Prudential Building-we are across the street!!! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G.729A + Cisco AS5300
Hello, Actually call from asterisk to AS5300 works fine with G.729. But not the other way round. I have tried enable all codecs, enable only g.729 on AS5300 but did not manage to get it work May I know what's you setting on both side Jeremy? Thanks for the reply Foong - Original Message - From: Jeremy McNamara [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, September 23, 2003 12:18 AM Subject: Re: [Asterisk-Users] G.729A + Cisco AS5300 Eric Wieling wrote: I doubt that it's a codec problem. Maybe chan_h323 doesnt' support G729. JerJer would know. I babysit systems that terminate hundreds of thousands of G.729 based H.323 calls per day using chan_h323 and As5300. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Recommended OS
Is there a recommended OS that Asterisk should be used with? I have been trying to get Asterisk running on Red Hat 9.0 with little success. Thanks! Michael
RE: [Asterisk-Users] MS Outlook
Go back to your cave. On your way, don't forget to patch sendmail (twice in the last 30 days), OpenSSH, gtkhtml, and pam_smb. Just in the last month. Linux. Security. Made for the Internet. Made for the cave. Regards, Sean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Critchfield Sent: Monday, September 22, 2003 3:58 PM To: [EMAIL PROTECTED] On Mon, 2003-09-22 at 15:30, Sean Heiney wrote: Actually, MS Outlook by default blocks all executables. I'm not sure why there is so much negativity around the Outlook client. Perhaps we should all go back to the cave and use Pine. I'll assume you don't understand the english words you just wrote well enough to defend yourself. Outlook does not block executables. It receives them via mail like any other mail message. It by default doesn't run executables that are sent as executables. But we all know about the current stupidities of Microsoft in that they look at the mime header to determine if it is safe to use the file(wav, mid, txt, whatever that should be a data file), but then executes the file so that they can use a shortcut to whatever app you defined to run that data file with. The problem being that they package exe files with a mime header for one of those innocuous files and the executable shortcut runs the virus. Not to mention that Outlook is set to by default to display HTML email and that a HTML mail with an embedded link to the data file inside will cause automatic running of the virus. So Outlook is not going to block the attachment from taking up residence on your drive. Outlook has poor security checking, and can be easily tricked into doing evil things. Microsoft recently stated themselves that Windows is not designed to sit on the internet out of the box, but requires a fair amount of hardening. This applies to all their other software as well as it is all tightly integrated. Admit it, Microsoft has been patching crap software for a long time. Linux had an advantage of not caring about market share and trying to do things the right way. Linux also grew up after the internet was around and while it was gaining popularity therefore it has had to grow up in a rough neighborhood and keep itself hardened. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Critchfield Sent: Monday, September 22, 2003 2:10 PM To: [EMAIL PROTECTED] On Mon, 2003-09-22 at 13:42, Brian West wrote: I second that... I have received a load of virii from people on this list.. Received: from torch.junct.com (sootbox.junct.com [65.168.64.10]) by www.bkw.org (8.11.6/8.11.6) with ESMTP id h8MIcEJ06998 for [EMAIL PROTECTED]; Mon, 22 Sep 2003 13:38:14 -0500 Received: from wdxmvur (unknown [207.41.124.63]) by torch.junct.com (Postfix) with SMTP id 461DF4159; Mon, 22 Sep 2003 13:37:08 -0500 (CDT) For those of you that have no reason whatsoever to receive windows executables, here is a procmail rule that matches the beginning of a windows executable no matter what it is named. # Base 64 encoded windows executable :0B: *TVqQAAME//8AALgAQ AA A You can use this and deliver the mail wherever you want to. This works on the last Sobig, klez and the Swen virus so far. This is what I had in my virii folder to test it against. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Recommended OS
On Mon, 2003-09-22 at 21:39, Michael A. Miller wrote: Is there a recommended OS that Asterisk should be used with? I have been trying to get Asterisk running on Red Hat 9.0 with little success. You hit 2 of my pet peeves at once. Fist, please understand that HTML has no business in normal email communication. Turn it off or you will start getting ignored(hopefully for you). Linux is a OS. Asterisk runs on Linux with only limited success on some *bsd system. If you search the archives you will find someone who has had some problem or another with just about every distribution. RH has a higher number of support problems, but I'll grant that more newbies pick RH and that contributes to it's problem count. There are quite a few people here that us RH9 though. So if you wish to post some of your problems we can them help you get over some of these bumps in the road. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G.729A + Cisco AS5300
Chee Foong wrote: Hello, Actually call from asterisk to AS5300 works fine with G.729. But not the other way round. I have tried enable all codecs, enable only g.729 on AS5300 but did not manage to get it work May I know what's you setting on both side Jeremy? My systems only do termination: Asterisk---Dial,H3235300PSTN To be clear you are talking about a H.323 incoming call from the As5300 doesn't succeed? Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Recommended OS
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael A. Miller Sent: Monday, September 22, 2003 10:40 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Recommended OS Is there a recommended OS that Asterisk should be used with? I have been trying to get Asterisk running on Red Hat 9.0 with little success. Thanks! Michael I have run Asterisk on two different RedHat 7.3 boxes. I have hundreds of RH7.3 boxes deployed doing a wide variety of tasks and have never had an issue. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Can't get simple config working!
Hi all. I'm trying to get a simple configuration working so I can later expand it to something more interesting. I'm using kphone to call an extension on the * server. When I try to connect, I get this error: DEBUG[81926]: File chan_sip.c, Line 3562 (check_user): Setting NAT on RTP to 0 DEBUG[81926]: File chan_sip.c, Line 527 (__sip_ack): Stopping retransmission on '[EMAIL PROTECTED]' of Response 4963: Found DEBUG[81926]: File chan_sip.c, Line 3562 (check_user): Setting NAT on RTP to 0 NOTICE[81926]: File pbx.c, Line 1171 (pbx_extension_helper): Cannot find extension context 'from-sip' DEBUG[81926]: File chan_sip.c, Line 527 (__sip_ack): Stopping retransmission on '[EMAIL PROTECTED]' of Response 4964: Not Found This is my extensions.conf file: [general] [from-sip] exten = 1001,1,Dial(sip/[EMAIL PROTECTED],20) exten = 1001,2,Voicemail(u1001) exten = 1001,102,Voicemail(b1001) exten = 1001,103,Hangup exten = 1002,1,Dial(1002,20) exten = 1002,2,Voicemail(u1002) exten = 1002,102,Voicemail(b1002) exten = 1002,103,Hangup And this is my sip.conf file. [general] port = 5060 bindaddr = 0.0.0.0 allow = all dtmfmode = inband context = from-sip [1001] username = 1001 type = friend context = from-sip mailbox = 1001 host = dynamic secret = liebchen nat = 0 [1002] username = 1002 type = friend context = from-sip mailbox = 1002 host = dynamic secret = liebchen nat = 0 Any ideas? Thanx, Mike. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G.729A + Cisco AS5300
Yes, you are right. H.323 incoming call from the As5300 doesn't succeed. outgoing call to AS5300 works fine like your system. Foong - Original Message - From: Jeremy McNamara [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, September 23, 2003 11:14 AM Subject: Re: [Asterisk-Users] G.729A + Cisco AS5300 Chee Foong wrote: Hello, Actually call from asterisk to AS5300 works fine with G.729. But not the other way round. I have tried enable all codecs, enable only g.729 on AS5300 but did not manage to get it work May I know what's you setting on both side Jeremy? My systems only do termination: Asterisk---Dial,H3235300PSTN To be clear you are talking about a H.323 incoming call from the As5300 doesn't succeed? Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MS Outlook
And we all certainly know that Windows is so secure. I am by no means a Linux or Windows fanatic, they each have their strong spots. And I find this thread a little off topic, totally not related to Asterisk or VoIP/phone systems. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Sean Heiney Sent: Monday, September 22, 2003 10:55 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] MS Outlook Go back to your cave. On your way, don't forget to patch sendmail (twice in the last 30 days), OpenSSH, gtkhtml, and pam_smb. Just in the last month. Linux. Security. Made for the Internet. Made for the cave. Regards, Sean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Critchfield Sent: Monday, September 22, 2003 3:58 PM To: [EMAIL PROTECTED] On Mon, 2003-09-22 at 15:30, Sean Heiney wrote: Actually, MS Outlook by default blocks all executables. I'm not sure why there is so much negativity around the Outlook client. Perhaps we should all go back to the cave and use Pine. I'll assume you don't understand the english words you just wrote well enough to defend yourself. Outlook does not block executables. It receives them via mail like any other mail message. It by default doesn't run executables that are sent as executables. But we all know about the current stupidities of Microsoft in that they look at the mime header to determine if it is safe to use the file(wav, mid, txt, whatever that should be a data file), but then executes the file so that they can use a shortcut to whatever app you defined to run that data file with. The problem being that they package exe files with a mime header for one of those innocuous files and the executable shortcut runs the virus. Not to mention that Outlook is set to by default to display HTML email and that a HTML mail with an embedded link to the data file inside will cause automatic running of the virus. So Outlook is not going to block the attachment from taking up residence on your drive. Outlook has poor security checking, and can be easily tricked into doing evil things. Microsoft recently stated themselves that Windows is not designed to sit on the internet out of the box, but requires a fair amount of hardening. This applies to all their other software as well as it is all tightly integrated. Admit it, Microsoft has been patching crap software for a long time. Linux had an advantage of not caring about market share and trying to do things the right way. Linux also grew up after the internet was around and while it was gaining popularity therefore it has had to grow up in a rough neighborhood and keep itself hardened. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Critchfield Sent: Monday, September 22, 2003 2:10 PM To: [EMAIL PROTECTED] On Mon, 2003-09-22 at 13:42, Brian West wrote: I second that... I have received a load of virii from people on this list.. Received: from torch.junct.com (sootbox.junct.com [65.168.64.10]) by www.bkw.org (8.11.6/8.11.6) with ESMTP id h8MIcEJ06998 for [EMAIL PROTECTED]; Mon, 22 Sep 2003 13:38:14 -0500 Received: from wdxmvur (unknown [207.41.124.63]) by torch.junct.com (Postfix) with SMTP id 461DF4159; Mon, 22 Sep 2003 13:37:08 -0500 (CDT) For those of you that have no reason whatsoever to receive windows executables, here is a procmail rule that matches the beginning of a windows executable no matter what it is named. # Base 64 encoded windows executable :0B: *TVqQAAME//8AALgAQ AA A You can use this and deliver the mail wherever you want to. This works on the last Sobig, klez and the Swen virus so far. This is what I had in my virii folder to test it against. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Recommended OS
On Monday 22 September 2003 11:25 pm, Andrew Joakimsen wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael A. Miller Sent: Monday, September 22, 2003 10:40 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Recommended OS Is there a recommended OS that Asterisk should be used with? I have been trying to get Asterisk running on Red Hat 9.0 with little success. I've been running it on RH9.0 w no problems. Ditto with getting new updates and recompiling. Are you aware of the software requirements? bison cvs gcc kernel-sources libtermcap-devel newt-devel ncurses-devel openssl096b openssl-devel readline41 readline-devel Thanks! Michael I have run Asterisk on two different RedHat 7.3 boxes. I have hundreds of RH7.3 boxes deployed doing a wide variety of tasks and have never had an issue. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Steve __ I hear SCO are closing their doors soon, let's turn their offices into an OpenSource library! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Can't get simple config working!
Try Nat = yes Or Nat = no -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Mike Diehl (Encrypted email prefer red) Sent: Monday, September 22, 2003 11:38 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Can't get simple config working! Hi all. I'm trying to get a simple configuration working so I can later expand it to something more interesting. I'm using kphone to call an extension on the * server. When I try to connect, I get this error: DEBUG[81926]: File chan_sip.c, Line 3562 (check_user): Setting NAT on RTP to 0 DEBUG[81926]: File chan_sip.c, Line 527 (__sip_ack): Stopping retransmission on '[EMAIL PROTECTED]' of Response 4963: Found DEBUG[81926]: File chan_sip.c, Line 3562 (check_user): Setting NAT on RTP to 0 NOTICE[81926]: File pbx.c, Line 1171 (pbx_extension_helper): Cannot find extension context 'from-sip' DEBUG[81926]: File chan_sip.c, Line 527 (__sip_ack): Stopping retransmission on '[EMAIL PROTECTED]' of Response 4964: Not Found This is my extensions.conf file: [general] [from-sip] exten = 1001,1,Dial(sip/[EMAIL PROTECTED],20) exten = 1001,2,Voicemail(u1001) exten = 1001,102,Voicemail(b1001) exten = 1001,103,Hangup exten = 1002,1,Dial(1002,20) exten = 1002,2,Voicemail(u1002) exten = 1002,102,Voicemail(b1002) exten = 1002,103,Hangup And this is my sip.conf file. [general] port = 5060 bindaddr = 0.0.0.0 allow = all dtmfmode = inband context = from-sip [1001] username = 1001 type = friend context = from-sip mailbox = 1001 host = dynamic secret = liebchen nat = 0 [1002] username = 1002 type = friend context = from-sip mailbox = 1002 host = dynamic secret = liebchen nat = 0 Any ideas? Thanx, Mike. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can't get simple config working!
Well, didn't I SAY it was a simple config? grin Thanx, that worked. Mike On Monday 22 September 2003 09:49 pm, Andrew Joakimsen wrote: Try Nat = yes Or Nat = no -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Mike Diehl (Encrypted email prefer red) Sent: Monday, September 22, 2003 11:38 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Can't get simple config working! Hi all. I'm trying to get a simple configuration working so I can later expand it to something more interesting. I'm using kphone to call an extension on the * server. When I try to connect, I get this error: DEBUG[81926]: File chan_sip.c, Line 3562 (check_user): Setting NAT on RTP to 0 DEBUG[81926]: File chan_sip.c, Line 527 (__sip_ack): Stopping retransmission on '[EMAIL PROTECTED]' of Response 4963: Found DEBUG[81926]: File chan_sip.c, Line 3562 (check_user): Setting NAT on RTP to 0 NOTICE[81926]: File pbx.c, Line 1171 (pbx_extension_helper): Cannot find extension context 'from-sip' DEBUG[81926]: File chan_sip.c, Line 527 (__sip_ack): Stopping retransmission on '[EMAIL PROTECTED]' of Response 4964: Not Found This is my extensions.conf file: [general] [from-sip] exten = 1001,1,Dial(sip/[EMAIL PROTECTED],20) exten = 1001,2,Voicemail(u1001) exten = 1001,102,Voicemail(b1001) exten = 1001,103,Hangup exten = 1002,1,Dial(1002,20) exten = 1002,2,Voicemail(u1002) exten = 1002,102,Voicemail(b1002) exten = 1002,103,Hangup And this is my sip.conf file. [general] port = 5060 bindaddr = 0.0.0.0 allow = all dtmfmode = inband context = from-sip [1001] username = 1001 type = friend context = from-sip mailbox = 1001 host = dynamic secret = liebchen nat = 0 [1002] username = 1002 type = friend context = from-sip mailbox = 1002 host = dynamic secret = liebchen nat = 0 Any ideas? Thanx, Mike. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Recommended OS
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Michael A. Miller wrote: | Is there a recommended OS that Asterisk should be used with? I have been | trying to get Asterisk running on Red Hat 9.0 with little success. I have successfully gotten * to compile on Redhat 9.0 and ClarkConnect v1.2 (RH7.3 based) and ClarkConnect v2.0 (RH9.0 based). As long as you have all the packages installed that * needs, then it's simply the make clean ; make install in zaptel, libpri and asterisk. No problems, no confusion, no mess. At least this has been my experience. Just make sure you install the development suite in RH. That should pretty much do it... Leif Madsen. -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.2 (Cygwin) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQE/b8pz6gq3eQ0gpNURAo8RAKC38nnhfyWzbs5ZN7O7ih4D7SENPgCfQI+f rETTnoVp//V9mqGoENUkXxM= =ABpe -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re: [Asterisk-Users] Recommended OS
and I think use those cvs with rh7.3 and apt for RH is works well :-) mack_jpn Tilghman Lesher wrote: On Monday 22 September 2003 22:37, Steve wrote: On Monday 22 September 2003 11:25 pm, Andrew Joakimsen wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael A. Miller Sent: Monday, September 22, 2003 10:40 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Recommended OS Is there a recommended OS that Asterisk should be used with? I have been trying to get Asterisk running on Red Hat 9.0 with little success. I've been running it on RH9.0 w no problems. Ditto with getting new updates and recompiling. Are you aware of the software requirements? bison cvs gcc kernel-sources libtermcap-devel newt-devel ncurses-devel openssl096b openssl-devel readline41 readline-devel Actually, readline should not be necessary anymore. There's now an implementation of readline included in the source (BSD-derived). -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Recommended OS
On Tue, 23 Sep 2003, Michael A. Miller wrote: Is there a recommended OS that Asterisk should be used with? I have been trying to get Asterisk running on Red Hat 9.0 with little success. At this time, Asterisk only seems to run with Various versions of Linux. There are patches for other unix systems, but the code hasn't been cleaned up to use automake/etc yet, but that does seem to be in the works. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
