[Asterisk-Users] Prepaid Calling Card
I am planning to sell prepaid calling cards to my service. The system is already working but I wanna print a good quality prepaid calling cards for it. Anyone would recommend me a good and cheap pre-paid card printing company anywhere in the world? Thanks in advance, Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cepstral TTS Code
Your website is refusing connections at the moment. Or more properly I should say I get Connection refused when I try to access the Cepstral link you posted earlier today to the Asterisk-users list. FYI. Thx. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] help *** newbie
You have not described your hardware configuration. Zapata.conf is in general used for Digium cards. If you have not such card you can just say noload = chan_zap.so in your /etc/asterisk/modules.conf You can obtain more information about asterisk configuration from next sources: a) http://www.digium.com/handbook-draft.pdf b) http://www.voip-info.org/tiki-index.php?page=Asterisk (and other asterisk links) c) http://asterisk.sohoskyway.net/Asterisk_Doc/current/docs-html/book1.html d) and archive of mailing list Good luck, poorman On Thu, 05 Feb 2004 04:29:35 +, FRANCISCO PEREZ-LANDAETA [EMAIL PROTECTED] wrote : This is a multi-part message in MIME format. can anyone help me on this ? i am having problems configuring the asterisk. i have included an attachment because for some reason i could not cut and past from the terminal to my hotmail account. your help is appreciated. thanks, *** please look at the errors francisco _ Check out the new MSN 9 Dial-up #65533; fast reliable Internet access with prime features! http://join.msn.com/?pgmarket=en-uspage=dialup/homeST=1 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] talking clock
On Wed, 4 Feb 2004, John Todd wrote: At 11:50 PM + 2/4/04, Dan Tucny wrote: ; ; Talking clock (123) ; exten = 123,1,SayUnixTime(|GB|HM 'vm-and' S 'digits/seconds') exten = 123,2,Wait(1) exten = 123,3,Goto(1) the seconds sound can be picked up from John Todd's site, http://www.loligo.com/asterisk/ Dan [snip] The file seconds.gsm is also in asterisk-sounds, which along with many other interesting and amusing clips can be pulled from the CVS server just like asterisk, zaptel, etc. Kudos to whomever requested All your base are belong to us and We're off gambling and getting drunk. ;) -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] talking clock
yOn Thu, 5 Feb 2004, Deepakumar JV wrote: Thanks to everyone. I got the talking clock working the way i wanted. thanks again Deepak How about a followup post showing exactly what your extensions.conf entries look like, and what you had to go to get it twekaed to your satisfaction? - Original Message - From: John Todd [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, February 05, 2004 03:20 AM Subject: Re: [Asterisk-Users] talking clock At 11:50 PM + 2/4/04, Dan Tucny wrote: ; ; Talking clock (123) ; exten = 123,1,SayUnixTime(|GB|HM 'vm-and' S 'digits/seconds') exten = 123,2,Wait(1) exten = 123,3,Goto(1) the seconds sound can be picked up from John Todd's site, http://www.loligo.com/asterisk/ Dan [snip] The file seconds.gsm is also in asterisk-sounds, which along with many other interesting and amusing clips can be pulled from the CVS server just like asterisk, zaptel, etc. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Data call transfer
Hi everyone I have TE410P with one E1 link connected to telecom PSTN, and another E1 to my internal legacy PBX. On this PBX I have one extension where my RAS server for both ISDN and analogue calls is located. Can anyone tell me what has to be done to transfer voice call from one E1 to another as voice, and if Asterisk detects that the call is a data call to transfer it further as data? Tomica
AW: [Asterisk-Users] Data call transfer
Hi Tomica,
i had the same problem and here is the solution from Maik Schmitt:
exten = _X.,1,GotoIf,$[${CALLTYPE} = DIGITAL]?50:100
exten = _X.,50,Dial(Zap/g3d/${EXTEN})
exten = _X.,100,Dial(Zap/g3/${EXTEN})
But maybe the dataendpoint would never be reached, and so can try out this:
go to bugs.digium.com and look at bug number 960 at libpri project
Regards,
Thomas.
-Ursprungliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Auftrag von Tomica Crnek
Gesendet: Donnerstag, 5. Februar 2004 10:05
An: [EMAIL PROTECTED]
Betreff: [Asterisk-Users] Data call transfer
Hi everyone
I have TE410P with one E1 link connected to telecom PSTN, and another E1 to
my internal legacy PBX. On this PBX I have one extension where my RAS server
for both ISDN and analogue calls is located.
Can anyone tell me what has to be done to transfer voice call from one E1 to
another as voice, and if Asterisk detects that the call is a data call to
transfer it further as data?
Tomica
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Re: [Asterisk-Users] Minor Registration Problem With Polycom Soun dpoint IP 500
mattf wrote: I have all of my Polycom's set to friend so I know that's not your problem. One day you too will get bitten by the type=friend's EVIL and you will see the light. Trust me, Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Minor Registration Problem With Polycom Soun dpoint IP 500
Could you tell us a little bit how exactly it works? The wiki pages don't say much about type=friend, user, and peer. I tried using type=user but can't seem to register. And what implications are there for using type=friend? David - Original Message - From: Jeremy McNamara [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, February 05, 2004 2:47 AM Subject: Re: [Asterisk-Users] Minor Registration Problem With Polycom Soun dpoint IP 500 mattf wrote: I have all of my Polycom's set to friend so I know that's not your problem. One day you too will get bitten by the type=friend's EVIL and you will see the light. Trust me, Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iax2 jitter buffer help
On Thu, 5 Feb 2004 [EMAIL PROTECTED] wrote: Hi I wonder if anyone has a fix or any advice for the IAX2 jitter buffer. My internet connection here in South Africa has an international ping time of 550ms +- 50 ms. According to the scientific approach I would like to add a 100ms jitter buffer. (nevermind the latency)! I have tried playing with maxjitterbuffer and maxexcessjitterbuffer settings, I also tried from the CLI IAX2 set jitter 700 with all kinds of parameters. Hi Clive, Are you on a Telkom ADSL line? I've found it unusable for VOIP over the last two weeks - simply not enough throughput. Its only a few prioritised ports (eg port 80 - web, 21 - ftp) that have any decent throughput. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] H323 calls via provider
Hello I am trying to use a VOIP provider (PC to PSTN). Is it possible to use asterisk as a client and make calls via a H323 provider? Can anyone guide me how the oh323.conf should be and extension.conf should be. I have a IP, userid and password given by them. I am using www.mywebcalls.com. Has anyone tried using * like this? Regards Deepak
Re: [Asterisk-Users] talking clock
How about a followup post showing exactly what your extensions.conf
entries look like, and what you had to go to get it twekaed to your
satisfaction?
Here is the working extension.conf i came up with
[time]
exten = 5559,1,Answer()
exten = 5559,2,Playback(time)
exten = 5559,3,SayUnixTime(||IM)
exten = 5559,4,SetVar(TIME1=${DATETIME})
exten = 5559,5,SubString,TIME2=${TIME1}|-2|2
exten = 5559,6,Playback(beep)
exten = 5559,7,SayNumber(${TIME2})
exten = 5559,8,Playback(second)
exten = 5559,9,Wait(1)
exten = 5559,10,Goto(time,5559,2)
but then i got to know about the S option in SayUnixTime() from Dan. THANKS
DAN.
exten = 5558,1,SayUnixTime(|GB|IM 'beep' S 'second')
exten = 5558,2,Goto(time,5558,1)
Thanks to everyone for helping me.
Now i have small problem which i am trying to fix with my less programming
knowledge. I get to hear the time in odd intervals, like 11:30:06 then
11:30:11 then 11:30:15 then 11:30:19 then 11:30:19 so the interval varies 4
and 5 seconds alternatively.
I wanted this clock to tell the time every 10 seconds and it should be the
actual system time.
ie., at 11:30:20 it should execute 5558,1 and at 11:30:30 it should execute
5558,1 that way i can hear the time every 10 seconds.
Regards
Deepak
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[Asterisk-Users] The Evil of type=friend explained, again (was Re: [Asterisk-Users] Minor Registration Problem With Polycom Soun dpoint IP 500)
David Liu wrote: Could you tell us a little bit how exactly it works? The wiki pages don't say much about type=friend, user, and peer. I tried using type=user but can't seem to register. A type=friend is simply both a type=user and type=peer using the same set of config directives. While a type=friend makes things almost trivial to get calls working in both directions, it will limit the flexibility of your config and even hinder some of the more advanced uses of Asterisk. For example: Say you want to use the same 'user' across many different Asterisk boxes, which of course will have different IP addresses. In this situation, you cannot have a host keyword in your Asterisk config stanza for the type=user, but the type=peer requires some host keyword. Thus, if you use a type=friend you will limit the use of that one username to whatever IP address is contained in the host keyword. You only need to register to Asterisk if you have a dynamic IP address or you need to blow thru a firewall/NAT device. To register you need to have a type=peer with a host=dynamic. Since in your type=friend config directive you had host=some.ip.address, while this may be this is fine to for the type=user, this same value also gets used for the type=peer, which makes it so you cannot register since the IP address is hard coded. So, either you do not need to register and things will Just Work(tm) or you will need to use separate type=user and type=peer config directives. I smell the beginnings of a Whitepaper here. Jeremy McNamara - Original Message - From: Jeremy McNamara [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, February 05, 2004 2:47 AM Subject: Re: [Asterisk-Users] Minor Registration Problem With Polycom Soun dpoint IP 500 mattf wrote: I have all of my Polycom's set to friend so I know that's not your problem. One day you too will get bitten by the type=friend's EVIL and you will see the light. Trust me, Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re:[OT] South african laws - was [Asterisk-Users] iax2 jitter buffer help
On the subject of South Africa What are the laws regarding using the Internet to carry telephone traffic? What are the laws regarding connecting digium kit to Telkom equipment? As I recall they are quite restrictive, have they been eased up a bit? Regards Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Fw: [Asterisk-Users] Possible Sip logic bug?
Anyone have comments on this? Really could use some suggestions or ideas why this is happening. Thanks. Rich Anyone recognize this as a sip logic bug? Example Case: C7960 - * - sip gateway - pstn (sip gateway config'ed with canreinvite=no, but shouldn't have an impact on this.) Outgoing call initiated from C7960. Call is completed and conversation is very much normal. All equipment on the same wire; no nat. The C7960 user hangs up the phone. Pkt flows (as observed by sniffer) are: C7960 sends sip BYE packet to * * returns 200 OK * sends INVITE to sip gateway where is BYE? sip gateway responds with 100 Trying sip gateway responds with 200 OK sip gateway responds with 200 OK sip gateway responds with 200 OK The end result, the sip gateway does not drop the pstn line until the called number hangs up. It would appear that asterisk has an issue dropping the call. When the C7960 issues the BYE, I would expect * to send a BYE to the sip g/w. Is this a * logic problem (or my logic problem)? (I'm actually running CVS-12/04/03-14:24:40 and has been very stable in this production environment. Is it time to update this one even though it is 99% sip hardphone based?) Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---End of Original Message- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Calls dropping off
I would have thought that if that was the problem, we couldn't makle or receive calls at all, or that we at least couldnt use all 3 Zap cards at the same time, but we can. The problem only happens every so often, but recently it's getting more and more frequent... management are starting to get pissed :/ No more ideas? I've tried everything else people have mentioned. Cheers, Steve On Mon, Feb 02, 2004 at 01:03:01PM -0500, Bill Hamel wrote: Hi, Have you checked for IRQ conflicts ? -b Quoting Steve Foy [EMAIL PROTECTED]: Hi, On Mon, Feb 02, 2004 at 06:04:40PM +0100, Philipp von Klitzing wrote: Steve, this really is a FAQ. You need add to EACH (!) sip user something like disallow=all allow=ulaw allow=alaw allow=gsm I do have that in my sip.conf. I am using ulaw. Calls from the SIP phones through Asterisk and out one of my X100P cards are working 95% of the time and also, incoming calls through the X100P cards to the SIP phones are the same. The only problem is that every once in a while, without any odd circustances that I can see, the call just drops and the remote user is gone. The box running asterisk isn't under heavy load, so I can't see why this is happening. I am not using g.729 or 723, just plain old ulaw, which I have got enabled in sip.conf Cheers, Steve -- Steve Foy| http://www.unite.net UNITE Solutions | Tel: 028 9077 7338 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content by The CCIS.net MailScanner, and is believed to be clean. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. -- This message was sent using IMP, the Internet Messaging Program. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Steve Foy| http://www.unite.net UNITE Solutions | Tel: 028 9077 7338 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iax2 jitter buffer help
Steve hi Yup, adsl, seems to be getting slower by the day. Maybe we can configure * to change the iax to port 21 udp ? Regards Clive On Thu, 5 Feb 2004 13:21:08 +0200 (SAST) Stephen Davies [EMAIL PROTECTED] wrote: On Thu, 5 Feb 2004 [EMAIL PROTECTED] wrote: Hi I wonder if anyone has a fix or any advice for the IAX2 jitter buffer. My internet connection here in South Africa has an international ping time of 550ms +- 50 ms. According to the scientific approach I would like to add a 100ms jitter buffer. (nevermind the latency)! I have tried playing with maxjitterbuffer and maxexcessjitterbuffer settings, I also tried from the CLI IAX2 set jitter 700 with all kinds of parameters. Hi Clive, Are you on a Telkom ADSL line? I've found it unusable for VOIP over the last two weeks - simply not enough throughput. Its only a few prioritised ports (eg port 80 - web, 21 - ftp) that have any decent throughput. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ http://www.webmail.co.za/dialup Webmail ISP - Cool Connection, Cool Price ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk + oh323 docs ?
Does anyone have any documentation on Asterisk + oh323, I am trying to allow a H323 peer to send me calls that I want to push out to SIP phones but am having trouble passing the digits dialed from the oh323 peer and dialing those digits onto a SIP client. Any docs much appreciated or even better working extensions.conf Rgds, Adam * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [OT] South african laws - was [Asterisk-Users] iax2 jitter buffer help
Basically voip is only legal if used between branch offices of a company that are connected using leased lines. Archaic.. yes, stupid... yes, but thats the law here..:( Our telco is strangling the country so they can line their pockets. On Thu, 05 Feb 2004 11:57:57 + Chris Lee [EMAIL PROTECTED] wrote: On the subject of South Africa What are the laws regarding using the Internet to carry telephone traffic? What are the laws regarding connecting digium kit to Telkom equipment? As I recall they are quite restrictive, have they been eased up a bit? Regards Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ http://www.webmail.co.za/dialup Webmail ISP - Cool Connection, Cool Price ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dialogic D300SC-E1
Hi all, by chance I have found an old Dialogic D300SC-E1 card that I would like to test with Asterisk. It should have voice capabilities on board, also. I have ABSOLUTELY no idea regarding the steps to make it work, I installed the card in a server with a working installation of *, then browsed Intel site looking for any info on that matter ... results ? none by now ! Anyone can help me starting the card ? Tnx ! -- Best regards, Alessio mailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [OT] South african laws - was [Asterisk-Users] iax2 jitter buffer help
Chris Lee wrote: On the subject of South Africa What are the laws regarding using the Internet to carry telephone traffic? Its 100% against the law, Telcom have the monopoly there still that requires ALL voice trafic to go via the Telcom network.. The Mobile phone operators there are in constant battles with them to tru and ease this so they can do some leased cost routing but they are not getting very far.. What are the laws regarding connecting digium kit to Telkom equipment? It should be fine, AFAIK they relaxed the rules about connecting third party equipment to their network a few years ago.. As I recall they are quite restrictive, have they been eased up a bit? Nope, and I don't see that they plan to ease up at all any time soon.. Which is why we scrapped the plans we had for setting up some of our facilities there.. We even had meetings with the DTI to see if they could put pressure on Telkom but seeing as the govenment own a majority share holding in Telkom why would they want to.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [OT] South african laws - was [Asterisk-Users] iax2 jitter buffer help
[EMAIL PROTECTED] wrote: Basically voip is only legal if used between branch offices of a company that are connected using leased lines. Archaic.. yes, stupid... yes, but thats the law here..:( That is provided the leased lines are operated by Telkom.. ;) There is no getting away... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Execute command in shell
Is it posible to make Asterisk execute a command on extensions.conf during a call ¿ (That's to transfer H323 call by telnetting the gatekeeper so Asterisk doesn't seem to like transferring h.323 ) Thanks! Marc ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dialogic D300SC-E1
Alessio Focardi wrote: Hi all, by chance I have found an old Dialogic D300SC-E1 card that I would like to test with Asterisk. It should have voice capabilities on board, also. I have ABSOLUTELY no idea regarding the steps to make it work, I installed the card in a server with a working installation of *, then browsed Intel site looking for any info on that matter ... results ? none by now ! Anyone can help me starting the card ? List it on http://www.ebay.com/ and take the proceeds and purchase a Digium E100P card. Seriously, Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Calls dropping off
Steve, Since I have a rather short memory and receive about 250 posting per day, I don't have a clue what has/hasn't been suggested. Here's a couple: 1. in logger.conf turn on debug, watch /var/log/asterisk/debug for size, and and hints relative to the dropped calls 2. look at /var/log/asterisk/messages for hints 3. if the problem occurs frequently enough, start a ping from the * box to one or more of the sip phones to verify you're not loosing net connections at the time of the dropped call (Spanning Tree Protocol can mess with your infrastructure without you knowing it, as one example) 4. look in /var/log/asterisk/cdr-csv/Master.csv file to see if any hints in the cdr data 5. post a relavent definition from sip.conf so we have a clue how you've defined a phone, as well as a relative Dial section from extensions.conf and zapata.conf 6. I don't recall which sip phones you're using, but some have internal logging capabilities. If your's do, turn it on and look for hints. 7. Download ethereal and sniff the asterisk nic interface, ensure you stop it right after a failure. If you need help doing the protocol analysis, then let me know. Rich I would have thought that if that was the problem, we couldn't makle or receive calls at all, or that we at least couldnt use all 3 Zap cards at the same time, but we can. The problem only happens every so often, but recently it's getting more and more frequent... management are starting to get pissed :/ No more ideas? I've tried everything else people have mentioned. Cheers, Steve On Mon, Feb 02, 2004 at 01:03:01PM -0500, Bill Hamel wrote: Hi, Have you checked for IRQ conflicts ? -b Quoting Steve Foy [EMAIL PROTECTED]: Hi, On Mon, Feb 02, 2004 at 06:04:40PM +0100, Philipp von Klitzing wrote: Steve, this really is a FAQ. You need add to EACH (!) sip user something like disallow=all allow=ulaw allow=alaw allow=gsm I do have that in my sip.conf. I am using ulaw. Calls from the SIP phones through Asterisk and out one of my X100P cards are working 95% of the time and also, incoming calls through the X100P cards to the SIP phones are the same. The only problem is that every once in a while, without any odd circustances that I can see, the call just drops and the remote user is gone. The box running asterisk isn't under heavy load, so I can't see why this is happening. I am not using g.729 or 723, just plain old ulaw, which I have got enabled in sip.conf Cheers, Steve -- Steve Foy| http://www.unite.net UNITE Solutions | Tel: 028 9077 7338 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content by The CCIS.net MailScanner, and is believed to be clean. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. -- This message was sent using IMP, the Internet Messaging Program. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Steve Foy| http://www.unite.net UNITE Solutions | Tel: 028 9077 7338 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---End of Original Message- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Fw: [Asterisk-Users] Possible Sip logic bug?
Rich, Try it again after executing: sip debug and give us the actual SIP messages. The devil is usually in the details. Rich Adamson wrote: Anyone have comments on this? Really could use some suggestions or ideas why this is happening. Thanks. Rich Anyone recognize this as a sip logic bug? Example Case: C7960 - * - sip gateway - pstn (sip gateway config'ed with canreinvite=no, but shouldn't have an impact on this.) Outgoing call initiated from C7960. Call is completed and conversation is very much normal. All equipment on the same wire; no nat. The C7960 user hangs up the phone. Pkt flows (as observed by sniffer) are: C7960 sends sip BYE packet to * * returns 200 OK * sends INVITE to sip gateway where is BYE? sip gateway responds with 100 Trying sip gateway responds with 200 OK sip gateway responds with 200 OK sip gateway responds with 200 OK The end result, the sip gateway does not drop the pstn line until the called number hangs up. It would appear that asterisk has an issue dropping the call. When the C7960 issues the BYE, I would expect * to send a BYE to the sip g/w. Is this a * logic problem (or my logic problem)? (I'm actually running CVS-12/04/03-14:24:40 and has been very stable in this production environment. Is it time to update this one even though it is 99% sip hardphone based?) Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---End of Original Message- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 bug in 6.1 evident in Asterisk
So do the 7960's have to be on the same subnet as the * box ? This seems like a major detriment to using them in a typical wan environment. - Chris Clifton - Original Message - From: Brian West [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, February 04, 2004 1:58 PM Subject: Re: [Asterisk-Users] Cisco 7960 bug in 6.1 evident in Asterisk Does the first line, backup and emergency proxy go to the * box on the same wire? Malcolm and I figured out the 7960's freak smooth out if the asterisk server isn't on the same subnet his phones kept rebooting over and over and over till we took them off the switch they were on and move them to the one with the aterisk server. bkw On Wed, 4 Feb 2004, John Todd wrote: Yes and no. The Cisco phone is on a NAT network that is quite distant from one of the Asterisk servers, but on the same wire as the other. Three lines go to the remote *, and three lines remain local on the network to the other * server. I'm running CVS as of this morning on both servers. Strangely, today the phone hasn't locked up or rebooted, though now I am getting one or two of the lines failing to REGISTER - they're simply not sending out a request, according to the network dump. sigh JT At 7:43 AM -0600 2/4/04, Brian West wrote: Question.. is the 7960 on the same subnet as your asterisk server? I have a 7960 registered with 3 diffrent asterisk servers. All 6 lines. Running 6.1 and has 12 days of uptime. bkw On Wed, 4 Feb 2004, John Todd wrote: So, I've managed to consistently lock up my Cisco 7960 (SIP 6.1) to the point where it needs to be unplugged, due to software errors. This is a first. My suspicions are that this bug in Asterisk is causing the lockups: http://bugs.digium.com/bug_view_page.php?bug_id=889 It seems unusual to me that a low volume of bogus SIP messages should lock up the 7960, but that seems to be the case. It seems this only happens on my 7960 that I have completely full of extensions (all six line buttons are lit, two of them are auto-answer.) I think this is one bug tickling another bug; bad messages from * are killing the 7960. I'd like anyone else with experiences with this type of failure with Asterisk to give me a shout; I'm going to report this to Cisco somehow, but don't have enough evidence. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re:[OT] South african laws - was [Asterisk-Users] iax2 jitter buffer help
On Thu, 5 Feb 2004, Chris Lee wrote: On the subject of South Africa What are the laws regarding using the Internet to carry telephone traffic? What are the laws regarding connecting digium kit to Telkom equipment? As I recall they are quite restrictive, have they been eased up a bit? The law is still very restrictive. Equipment should be ICASA approved for connection to the network. Digium equipment isn't. VOIP may be used on private networks. However such use is for office-to-office calls, and may not be used to bypass Telkom. This is generally understood to mean connecting in from the PSTN and then breaking back out again. Even VOIP on private networks is supposed to be dependent on getting a private telecommunications licence. In SA a private network means a network built out of Telkom data circuits. No actual private commmunications links are allowed. VPN-type networks are not included. Value Added network providers - including ISPs and suchlike are not supposed to allow the use of their service for transporting VOIP, and certainly may not market services like that. Of course they don't know and I'd guess they don't ask. Technically I guess using services like Vonage or whatever from SA is questionable too. Of course South African's have developed a certain attitude to the law, and enforcement is difficult, especially for small-scale private use. For example type-approval of equipment seems to be pretty much overlooked - see no evil, hear no evil. I'm no lawyer and perhaps Telkom/ICASA/Dept of Communications' interpretations of the law are wrong - I don't think they've really been tested in the courts. I also may have got some of the subtleties slightly wrong. You might ask why a country which could benefit so much from communication innovation has such restrictive law. It's a sad story of money, power and influence. You can read an interesting article on the SAT3 undersea cable and communications in Africa at: http://www.myadsl.co.za/forum/topic.asp?TOPIC_ID=1635 Regards, Steve Davies PS: 512k down / 256k up ADSL, capped at 3GB total inbound+outbound traffic, brutal traffic shaping which (coincidentally?) often breaks VOIP: +/- US$120 per month to you, sir. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 bug in 6.1 evident in Asterisk
Chris Clifton wrote: So do the 7960's have to be on the same subnet as the * box ? This seems like a major detriment to using them in a typical wan environment. - Chris Clifton - Original Message - From: Brian West [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, February 04, 2004 1:58 PM Subject: Re: [Asterisk-Users] Cisco 7960 bug in 6.1 evident in Asterisk Does the first line, backup and emergency proxy go to the * box on the same wire? Malcolm and I figured out the 7960's freak smooth out if the asterisk server isn't on the same subnet his phones kept rebooting over and over and over till we took them off the switch they were on and move them to the one with the aterisk server. bkw On Wed, 4 Feb 2004, John Todd wrote: Yes and no. The Cisco phone is on a NAT network that is quite distant from one of the Asterisk servers, but on the same wire as the other. Three lines go to the remote *, and three lines remain local on the network to the other * server. I'm running CVS as of this morning on both servers. Strangely, today the phone hasn't locked up or rebooted, though now I am getting one or two of the lines failing to REGISTER - they're simply not sending out a request, according to the network dump. sigh JT At 7:43 AM -0600 2/4/04, Brian West wrote: Question.. is the 7960 on the same subnet as your asterisk server? I have a 7960 registered with 3 diffrent asterisk servers. All 6 lines. Running 6.1 and has 12 days of uptime. bkw On Wed, 4 Feb 2004, John Todd wrote: So, I've managed to consistently lock up my Cisco 7960 (SIP 6.1) to the point where it needs to be unplugged, due to software errors. This is a first. My suspicions are that this bug in Asterisk is causing the lockups: http://bugs.digium.com/bug_view_page.php?bug_id=889 It seems unusual to me that a low volume of bogus SIP messages should lock up the 7960, but that seems to be the case. It seems this only happens on my 7960 that I have completely full of extensions (all six line buttons are lit, two of them are auto-answer.) I think this is one bug tickling another bug; bad messages from * are killing the 7960. I'd like anyone else with experiences with this type of failure with Asterisk to give me a shout; I'm going to report this to Cisco somehow, but don't have enough evidence. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users My remote 15 seat call center uses 79xx phones and a point to point T1. Your millage may vary with the number of users/applications your bandwidth supports. You may need to install QoS for your network to give SIP traffic top priority. It's best to have a low latency connection! Regards, TL ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk as non root
I followed the wiki instructions: http://www.voip-info.org/wiki-Asterisk+non-root Now I have a working asterisk running as user asterisk. I do however have some problems: 1: I dont have access via asterisk -r 2: The pid file is no longer being updated 3: I want to create a file in init.d so that I can use service start and stop, but need to be able to pass asterisk the gracefully command etc, any ideas welcome. maybe: asterisk -rx stop gracefully etc Regards Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 bug in 6.1 evident in Asterisk
No they do not. I am managing an installation running 7960 SIP release 6.0 and the phones are on about 4 different subnets. Half of these are on remote VPN connections at people's homes. Chris Clifton wrote: So do the 7960's have to be on the same subnet as the * box ? This seems like a major detriment to using them in a typical wan environment. - Chris Clifton - Original Message - From: Brian West [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, February 04, 2004 1:58 PM Subject: Re: [Asterisk-Users] Cisco 7960 bug in 6.1 evident in Asterisk Does the first line, backup and emergency proxy go to the * box on the same wire? Malcolm and I figured out the 7960's freak smooth out if the asterisk server isn't on the same subnet his phones kept rebooting over and over and over till we took them off the switch they were on and move them to the one with the aterisk server. bkw On Wed, 4 Feb 2004, John Todd wrote: Yes and no. The Cisco phone is on a NAT network that is quite distant from one of the Asterisk servers, but on the same wire as the other. Three lines go to the remote *, and three lines remain local on the network to the other * server. I'm running CVS as of this morning on both servers. Strangely, today the phone hasn't locked up or rebooted, though now I am getting one or two of the lines failing to REGISTER - they're simply not sending out a request, according to the network dump. sigh JT At 7:43 AM -0600 2/4/04, Brian West wrote: Question.. is the 7960 on the same subnet as your asterisk server? I have a 7960 registered with 3 diffrent asterisk servers. All 6 lines. Running 6.1 and has 12 days of uptime. bkw On Wed, 4 Feb 2004, John Todd wrote: So, I've managed to consistently lock up my Cisco 7960 (SIP 6.1) to the point where it needs to be unplugged, due to software errors. This is a first. My suspicions are that this bug in Asterisk is causing the lockups: http://bugs.digium.com/bug_view_page.php?bug_id=889 It seems unusual to me that a low volume of bogus SIP messages should lock up the 7960, but that seems to be the case. It seems this only happens on my 7960 that I have completely full of extensions (all six line buttons are lit, two of them are auto-answer.) I think this is one bug tickling another bug; bad messages from * are killing the 7960. I'd like anyone else with experiences with this type of failure with Asterisk to give me a shout; I'm going to report this to Cisco somehow, but don't have enough evidence. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 bug in 6.1 evident in Asterisk
And the * server is in your hq location ? Thanks, Chris Clifton - Original Message - From: Clif Jones [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, February 05, 2004 9:02 AM Subject: Re: [Asterisk-Users] Cisco 7960 bug in 6.1 evident in Asterisk No they do not. I am managing an installation running 7960 SIP release 6.0 and the phones are on about 4 different subnets. Half of these are on remote VPN connections at people's homes. Chris Clifton wrote: So do the 7960's have to be on the same subnet as the * box ? This seems like a major detriment to using them in a typical wan environment. - Chris Clifton - Original Message - From: Brian West [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, February 04, 2004 1:58 PM Subject: Re: [Asterisk-Users] Cisco 7960 bug in 6.1 evident in Asterisk Does the first line, backup and emergency proxy go to the * box on the same wire? Malcolm and I figured out the 7960's freak smooth out if the asterisk server isn't on the same subnet his phones kept rebooting over and over and over till we took them off the switch they were on and move them to the one with the aterisk server. bkw On Wed, 4 Feb 2004, John Todd wrote: Yes and no. The Cisco phone is on a NAT network that is quite distant from one of the Asterisk servers, but on the same wire as the other. Three lines go to the remote *, and three lines remain local on the network to the other * server. I'm running CVS as of this morning on both servers. Strangely, today the phone hasn't locked up or rebooted, though now I am getting one or two of the lines failing to REGISTER - they're simply not sending out a request, according to the network dump. sigh JT At 7:43 AM -0600 2/4/04, Brian West wrote: Question.. is the 7960 on the same subnet as your asterisk server? I have a 7960 registered with 3 diffrent asterisk servers. All 6 lines. Running 6.1 and has 12 days of uptime. bkw On Wed, 4 Feb 2004, John Todd wrote: So, I've managed to consistently lock up my Cisco 7960 (SIP 6.1) to the point where it needs to be unplugged, due to software errors. This is a first. My suspicions are that this bug in Asterisk is causing the lockups: http://bugs.digium.com/bug_view_page.php?bug_id=889 It seems unusual to me that a low volume of bogus SIP messages should lock up the 7960, but that seems to be the case. It seems this only happens on my 7960 that I have completely full of extensions (all six line buttons are lit, two of them are auto-answer.) I think this is one bug tickling another bug; bad messages from * are killing the 7960. I'd like anyone else with experiences with this type of failure with Asterisk to give me a shout; I'm going to report this to Cisco somehow, but don't have enough evidence. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 bug in 6.1 evident in Asterisk
On Thu, 5 Feb 2004, Clif Jones wrote: No they do not. I am managing an installation running 7960 SIP release 6.0 and the phones are on about 4 different subnets. Half of these are on remote VPN connections at people's homes. Currently, The Cisco 7960 SCCP can hear me but I cannot hear him. Both are in Public IP address without any firewall. What the problem should be? I tried different codecs and no change. In the asterisk side I'm using a X100P. Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 bug in 6.1 evident in Asterisk
No, they don't. I've got multiple 7960's functioning reliably across the Internet registering (and handling calls) just fine with * for months. And, the 7960's are behind cheap nat boxes as well. All running v6.0, but worked just as well with the v4 code. (Other 7960's are on the wire with * too.) FWIW, asterisk uses a registered IP and the sip definitions include nat=yes and canreinvite=no; think I might have tweaked the phone's config files to register every 600 seconds (don't remember for sure). We don't use the ethernet switch built into the phone, and each phone has either two or three buttons defined. The xml directory functions are programmed and working as well. Rich So do the 7960's have to be on the same subnet as the * box ? This seems like a major detriment to using them in a typical wan environment. - Chris Clifton - Original Message - From: Brian West [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, February 04, 2004 1:58 PM Subject: Re: [Asterisk-Users] Cisco 7960 bug in 6.1 evident in Asterisk Does the first line, backup and emergency proxy go to the * box on the same wire? Malcolm and I figured out the 7960's freak smooth out if the asterisk server isn't on the same subnet his phones kept rebooting over and over and over till we took them off the switch they were on and move them to the one with the aterisk server. bkw On Wed, 4 Feb 2004, John Todd wrote: Yes and no. The Cisco phone is on a NAT network that is quite distant from one of the Asterisk servers, but on the same wire as the other. Three lines go to the remote *, and three lines remain local on the network to the other * server. I'm running CVS as of this morning on both servers. Strangely, today the phone hasn't locked up or rebooted, though now I am getting one or two of the lines failing to REGISTER - they're simply not sending out a request, according to the network dump. sigh JT At 7:43 AM -0600 2/4/04, Brian West wrote: Question.. is the 7960 on the same subnet as your asterisk server? I have a 7960 registered with 3 diffrent asterisk servers. All 6 lines. Running 6.1 and has 12 days of uptime. bkw On Wed, 4 Feb 2004, John Todd wrote: So, I've managed to consistently lock up my Cisco 7960 (SIP 6.1) to the point where it needs to be unplugged, due to software errors. This is a first. My suspicions are that this bug in Asterisk is causing the lockups: http://bugs.digium.com/bug_view_page.php?bug_id=889 It seems unusual to me that a low volume of bogus SIP messages should lock up the 7960, but that seems to be the case. It seems this only happens on my 7960 that I have completely full of extensions (all six line buttons are lit, two of them are auto-answer.) I think this is one bug tickling another bug; bad messages from * are killing the 7960. I'd like anyone else with experiences with this type of failure with Asterisk to give me a shout; I'm going to report this to Cisco somehow, but don't have enough evidence. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Adding another X100P Card
History: 1. Added X100P to my system 2. Added Sipura 3. Added TDM400P (2 port) Worked fine so far 4. Now I want to add an additional X100P My question is...is the following configs files ok and is there any issue with adding the X100P (channel 4) after my 2 analog FXS channels? Thanks. Steve Here is my /etc/zaptel.conf fxsks=1,4 fxols=2-3 loadzone = us defaultzone = us Here is my /etc/asterisk/zapata.conf ; Zapata telephony interface sample configuration file ; [channels] ; ; X100P plugged into PSTN ; X100P # 1 context=incoming signalling=fxs_ks echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=1.5 txgain=1.5 immediate=no busydetect=no callprogress=no musiconhold=default usecallerid=yes callerid=asreceived channel = 1 ; ; ; ; TDM200B Port #1 plugged into analog Phone ; ; context=toll-access signalling=fxo_ls callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=1.5 txgain=1.5 immediate=no musiconhold=default usecallerid=yes callerid=Livingroom 2201 mailbox=2201 channel = 2 ; ; TDM200B Port #2 ; ; context=toll-access signalling=fxo_ls callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=1.5 txgain=1.5 immediate=no musiconhold=default usecallerid=yes callerid=Kitchen 2202 mailbox=2202 channel = 3 ; X100P # 2 context=incoming signalling=fxs_ks echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=1.5 txgain=1.5 immediate=no busydetect=no callprogress=no musiconhold=default usecallerid=yes callerid=asreceived channel = 4 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Calls dropping off
Right... It just happened there now, this came up: Feb 5 14:34:18 WARNING[1133742896]: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 3 (Response) I'm not sure if that's related to it, but it's the only thing that came up when the call got cut off. Here's the generic sip.conf stuff [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) allow=all allow=GSM allow=G729 allow=iLBC allow=SpeeX; Allow all codecs allow=ulaw Here's a sip.conf declaration: ; Andy [108] type=friend username= secret= host=dynamic dtmfmode=rfc2833 callerid=Andy McAlister 108 context=internal [EMAIL PROTECTED] qualify=yes canreinvite=no And the relevant extension.conf bit: ;Andy exten = 108,1,Dial(SIP/108,15) exten = 108,2,Playback(int-voicemail/108) exten = 108,3,Voicemail(s108) exten = 108,102,Playback(int-voicemail/108) exten = 108,103,Voicemail(s108) Any insight vastly appreciated! Cheers, Steve On Thu, Feb 05, 2004 at 06:33:06AM -0600, Rich Adamson wrote: Steve, Since I have a rather short memory and receive about 250 posting per day, I don't have a clue what has/hasn't been suggested. Here's a couple: 1. in logger.conf turn on debug, watch /var/log/asterisk/debug for size, and and hints relative to the dropped calls 2. look at /var/log/asterisk/messages for hints 3. if the problem occurs frequently enough, start a ping from the * box to one or more of the sip phones to verify you're not loosing net connections at the time of the dropped call (Spanning Tree Protocol can mess with your infrastructure without you knowing it, as one example) 4. look in /var/log/asterisk/cdr-csv/Master.csv file to see if any hints in the cdr data 5. post a relavent definition from sip.conf so we have a clue how you've defined a phone, as well as a relative Dial section from extensions.conf and zapata.conf 6. I don't recall which sip phones you're using, but some have internal logging capabilities. If your's do, turn it on and look for hints. 7. Download ethereal and sniff the asterisk nic interface, ensure you stop it right after a failure. If you need help doing the protocol analysis, then let me know. Rich I would have thought that if that was the problem, we couldn't makle or receive calls at all, or that we at least couldnt use all 3 Zap cards at the same time, but we can. The problem only happens every so often, but recently it's getting more and more frequent... management are starting to get pissed :/ No more ideas? I've tried everything else people have mentioned. Cheers, Steve On Mon, Feb 02, 2004 at 01:03:01PM -0500, Bill Hamel wrote: Hi, Have you checked for IRQ conflicts ? -b Quoting Steve Foy [EMAIL PROTECTED]: Hi, On Mon, Feb 02, 2004 at 06:04:40PM +0100, Philipp von Klitzing wrote: Steve, this really is a FAQ. You need add to EACH (!) sip user something like disallow=all allow=ulaw allow=alaw allow=gsm I do have that in my sip.conf. I am using ulaw. Calls from the SIP phones through Asterisk and out one of my X100P cards are working 95% of the time and also, incoming calls through the X100P cards to the SIP phones are the same. The only problem is that every once in a while, without any odd circustances that I can see, the call just drops and the remote user is gone. The box running asterisk isn't under heavy load, so I can't see why this is happening. I am not using g.729 or 723, just plain old ulaw, which I have got enabled in sip.conf Cheers, Steve -- Steve Foy| http://www.unite.net UNITE Solutions | Tel: 028 9077 7338 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content by The CCIS.net MailScanner, and is believed to be clean. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. -- This message was sent using IMP, the Internet Messaging Program. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options
Re: [Asterisk-Users] Cisco 7960 bug in 6.1 evident in Asterisk
The asterisk PBX is on a private subnet 192.168.20.0, the 7960's are on
192.168.{20,200,201,202,203}.0
subnets. The SIP gateways are on 192.168.{15,20,22,13}.0 subnets.
Chris Clifton wrote:
And the * server is in your hq location ?
Thanks,
Chris Clifton
- Original Message -
From: Clif Jones [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, February 05, 2004 9:02 AM
Subject: Re: [Asterisk-Users] Cisco 7960 bug in 6.1 evident in Asterisk
No they do not. I am managing an installation running 7960 SIP release
6.0 and the phones
are on about 4 different subnets. Half of these are on remote VPN
connections at people's homes.
Chris Clifton wrote:
So do the 7960's have to be on the same subnet as the * box ?
This seems like a major detriment to using them in a typical wan
environment.
- Chris Clifton
- Original Message -
From: Brian West [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, February 04, 2004 1:58 PM
Subject: Re: [Asterisk-Users] Cisco 7960 bug in 6.1 evident in Asterisk
Does the first line, backup and emergency proxy go to the * box on the
same wire? Malcolm and I figured out the 7960's freak smooth out if the
asterisk server isn't on the same subnet his phones kept rebooting over
and over and over till we took them off the switch they were on and move
them to the one with the aterisk server.
bkw
On Wed, 4 Feb 2004, John Todd wrote:
Yes and no. The Cisco phone is on a NAT network that is quite
distant from one of the Asterisk servers, but on the same wire as the
other. Three lines go to the remote *, and three lines remain local
on the network to the other * server. I'm running CVS as of this
morning on both servers. Strangely, today the phone hasn't locked up
or rebooted, though now I am getting one or two of the lines failing
to REGISTER - they're simply not sending out a request, according to
the network dump. sigh
JT
At 7:43 AM -0600 2/4/04, Brian West wrote:
Question.. is the 7960 on the same subnet as your asterisk server? I
have
a 7960 registered with 3 diffrent asterisk servers. All 6 lines.
Running
6.1 and has 12 days of uptime.
bkw
On Wed, 4 Feb 2004, John Todd wrote:
So, I've managed to consistently lock up my Cisco 7960 (SIP 6.1) to
the point where it needs to be unplugged, due to software errors.
This is a first.
My suspicions are that this bug in Asterisk is causing the lockups:
http://bugs.digium.com/bug_view_page.php?bug_id=889
It seems unusual to me that a low volume of bogus SIP messages
should
lock up the 7960, but that seems to be the case. It seems this
only
happens on my 7960 that I have completely full of extensions (all
six
line buttons are lit, two of them are auto-answer.) I think this
is
one bug tickling another bug; bad messages from * are killing the
7960.
I'd like anyone else with experiences with this type of failure
with
Asterisk to give me a shout; I'm going to report this to Cisco
somehow, but don't have enough evidence.
JT
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Re[2]: [Asterisk-Users] Dialogic D300SC-E1
Hello Jeremy, Anyone can help me starting the card ? JM List it on http://www.ebay.com/ and take the proceeds and purchase a JM Digium E100P card. It has been my first tought but guess what ? E100P is not CE certified and I'm fearing legal problems Also I dont think that someone would buy an Dialogic ISA card ... do you need one maybe ? :) JM Seriously, JM Jeremy McNamara JM ___ JM Asterisk-Users mailing list JM [EMAIL PROTECTED] JM http://lists.digium.com/mailman/listinfo/asterisk-users JM To UNSUBSCRIBE or update options visit: JMhttp://lists.digium.com/mailman/listinfo/asterisk-users -- Best regards, Alessiomailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] compact fxo device
Hi All! I´msearching fora compact external fxo device , a little box like sipura adaptor,with one or maybe two fxo. Searching google the only device that shows is the x100p, Anyone knows about a device like that? miklos
Re: [Asterisk-Users] Cisco 7960 bug in 6.1 evident in Asterisk
No they do not but apparetly his phones either didn't like the switch they were on or they have something wrong with them. bkw On Thu, 5 Feb 2004, Chris Clifton wrote: So do the 7960's have to be on the same subnet as the * box ? This seems like a major detriment to using them in a typical wan environment. - Chris Clifton - Original Message - From: Brian West [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, February 04, 2004 1:58 PM Subject: Re: [Asterisk-Users] Cisco 7960 bug in 6.1 evident in Asterisk Does the first line, backup and emergency proxy go to the * box on the same wire? Malcolm and I figured out the 7960's freak smooth out if the asterisk server isn't on the same subnet his phones kept rebooting over and over and over till we took them off the switch they were on and move them to the one with the aterisk server. bkw On Wed, 4 Feb 2004, John Todd wrote: Yes and no. The Cisco phone is on a NAT network that is quite distant from one of the Asterisk servers, but on the same wire as the other. Three lines go to the remote *, and three lines remain local on the network to the other * server. I'm running CVS as of this morning on both servers. Strangely, today the phone hasn't locked up or rebooted, though now I am getting one or two of the lines failing to REGISTER - they're simply not sending out a request, according to the network dump. sigh JT At 7:43 AM -0600 2/4/04, Brian West wrote: Question.. is the 7960 on the same subnet as your asterisk server? I have a 7960 registered with 3 diffrent asterisk servers. All 6 lines. Running 6.1 and has 12 days of uptime. bkw On Wed, 4 Feb 2004, John Todd wrote: So, I've managed to consistently lock up my Cisco 7960 (SIP 6.1) to the point where it needs to be unplugged, due to software errors. This is a first. My suspicions are that this bug in Asterisk is causing the lockups: http://bugs.digium.com/bug_view_page.php?bug_id=889 It seems unusual to me that a low volume of bogus SIP messages should lock up the 7960, but that seems to be the case. It seems this only happens on my 7960 that I have completely full of extensions (all six line buttons are lit, two of them are auto-answer.) I think this is one bug tickling another bug; bad messages from * are killing the 7960. I'd like anyone else with experiences with this type of failure with Asterisk to give me a shout; I'm going to report this to Cisco somehow, but don't have enough evidence. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Record conversation
Hi, Does anybody know if it is possible to record a conversation with asterisk ? Regards Rattana
RE: [Asterisk-Users] Record conversation
res_monitor.so: Resource for recording channels. -Original Message-From: Rattana BIV [mailto:[EMAIL PROTECTED]Sent: 05 February 2004 16:20To: [EMAIL PROTECTED]Subject: [Asterisk-Users] Record conversation Hi, Does anybody know if it is possible to record a conversation with asterisk ? Regards Rattana * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person
[Asterisk-Users] (no subject)
bkw, I realised that I was running asterisk with just asterisk no cli options changed it to safe_asterisk any my problem went away, so it might just be that it doesn't want to work in asterisk, just safe_asterisk when I some free time I'll get a coredump since there are no real informative debug traces. Thanks all Rohde - Original Message - From: Brian West [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, February 03, 2004 10:23 PM Subject: Re: [Asterisk-Users] voicemail issue How do you start asterisk? using safe_asterisk? or what cli options do you give it? bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Fw: [Asterisk-Users] Possible Sip logic bug?
Clif and all... At the bottom of this post is the sip show debug for the problem. The underlying problem (again): when C7960 hangs up on working conversation, the C7960 sends a BYE to *, but * never sends a BYE to the sip gateway. Any suggestions would be greatly appreciated. Rich Try it again after executing: sip debug and give us the actual SIP messages. The devil is usually in the details. Rich Adamson wrote: Anyone have comments on this? Really could use some suggestions or ideas why this is happening. Thanks. Rich Anyone recognize this as a sip logic bug? Example Case: C7960 - * - sip gateway - pstn (sip gateway config'ed with canreinvite=no, but shouldn't have an impact on this.) Outgoing call initiated from C7960. Call is completed and conversation is very much normal. All equipment on the same wire; no nat. The C7960 user hangs up the phone. Pkt flows (as observed by sniffer) are: C7960 sends sip BYE packet to * * returns 200 OK * sends INVITE to sip gateway where is BYE? sip gateway responds with 100 Trying sip gateway responds with 200 OK sip gateway responds with 200 OK sip gateway responds with 200 OK The end result, the sip gateway does not drop the pstn line until the called number hangs up. It would appear that asterisk has an issue dropping the call. When the C7960 issues the BYE, I would expect * to send a BYE to the sip g/w. Is this a * logic problem (or my logic problem)? (I'm actually running CVS-12/04/03-14:24:40 and has been very stable in this production environment. Is it time to update this one even though it is 99% sip hardphone based?) Rich Note: Call is already established from C7960 (193.92) via * (193.101) to the sip gateway (193.109) which called cell phone 444-1234. The sip show channels was executed, followed by sip debug, then hung up the C7960 watching the results below. Note the C7960 sends the BYE and * confirms, but * never sends a BYE to the gateway. Sniffer on the wire confirms the exact same thing. phoenix*CLI phoenix*CLI sip show channels Peer User/ANRCall ID Seq (Tx/Rx) Lag Jitter Format 222.111.193.109 4441234 66841295427 00103/0 0ms ms ULAW 222.111.193.92 300000036bc3-8b 00102/00103 0ms ms ULAW 2 active SIP channel(s) == Spawn extension (from-sip, 64441234, 2) exited non-zero on 'SIP/3000-375c' -- Executing SetCIDNum(SIP/3000-ead2, ) in new stack -- Executing Dial(SIP/3000-ead2, SIP/[EMAIL PROTECTED]) in new sta ck -- Called [EMAIL PROTECTED] -- SIP/222.111.193.109-fb6e answered SIP/3000-ead2 -- Attempting native bridge of SIP/3000-ead2 and SIP/222.111.193.109-fb6e SIP Debugging Enabled Sip read: I BYE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 222.111.193.92:5060 From: NPI-Rich sip:[EMAIL PROTECTED];tag=00036bc38b88045b25941469-0a0c5ae b To: sip:[EMAIL PROTECTED];tag=as751f96fc Call-ID: [EMAIL PROTECTED] Date: Thu, 05 Feb 2004 15:13:50 GMT CSeq: 103 BYE User-Agent: CSCO/6 Content-Length: 0 Proxy-Authorization: Digest username=3000,realm=asterisk,uri=sip:222.111.19 3.101,response=bb01af8f1eac65d392b68147867e79e6,nonce=7660e36e,algorithm=md 5 10 headers, 0 lines Sending to 222.111.193.92 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 222.111.193.92:5060 From: NPI-Rich sip:[EMAIL PROTECTED];tag=00036bc38b88045b25941469-0a0c5ae b To: sip:[EMAIL PROTECTED];tag=as751f96fc Call-ID: [EMAIL PROTECTED] CSeq: 103 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 222.111.193.92:5060 set_destination: Parsing sip:[EMAIL PROTECTED]:5060 for address/port to send to set_destination: set destination to 222.111.193.109, port 5060 We're at 222.111.193.101 port 14308 Answering with preferred capability 4 Answering with capability 8 Answering with non-codec capability 1 11 headers, 10 lines Reliably Transmitting: INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 222.111.193.101:5060;branch=z9hG4bK40ac3b17 From: NPI-Rich sip:[EMAIL PROTECTED];tag=as3310fadb To: sip:[EMAIL PROTECTED];tag=8c44b610-98bad313 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 104 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 219 v=0 o=root 14743 14745 IN IP4 222.111.193.101 s=session c=IN IP4 222.111.193.101 t=0 0 m=audio 14308 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 (no NAT) to 222.111.193.109:5060 == Spawn extension (from-sip, 64441234, 2) exited non-zero on 'SIP/3000-ead2' Sip read: I SIP/2.0 100 Trying Call-ID: [EMAIL PROTECTED] CSeq: 104 INVITE From: NPI-Rich sip:[EMAIL PROTECTED];tag=as3310fadb To: sip:[EMAIL
Re: [Asterisk-Users] Dialogic D300SC-E1
Alessio Focardi wrote: Hello Jeremy, Anyone can help me starting the card ? JM List it on http://www.ebay.com/ and take the proceeds and purchase a JM Digium E100P card. It has been my first tought but guess what ? E100P is not CE certified and I'm fearing legal problems Also I dont think that someone would buy an Dialogic ISA card ... do you need one maybe ? :) Actually a lot of people will buy the ISA cards. They go nicely in those industral rack mounts with many slots. That card won't work with * though, or with an VoIP. It isn't full duplex. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Calls dropping off
Steve Foy wrote: Right... It just happened there now, this came up: Feb 5 14:34:18 WARNING[1133742896]: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 3 (Response) I'm not sure if that's related to it, but it's the only thing that came up when the call got cut off. Here's the generic sip.conf stuff [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) allow=all allow=GSM allow=G729 allow=iLBC allow=SpeeX; Allow all codecs allow=ulaw Here's a sip.conf declaration: ; Andy [108] type=friend username= secret= host=dynamic dtmfmode=rfc2833 callerid=Andy McAlister 108 context=internal [EMAIL PROTECTED] qualify=yes canreinvite=no And the relevant extension.conf bit: ;Andy exten = 108,1,Dial(SIP/108,15) exten = 108,2,Playback(int-voicemail/108) exten = 108,3,Voicemail(s108) exten = 108,102,Playback(int-voicemail/108) exten = 108,103,Voicemail(s108) Any insight vastly appreciated! Cheers, Steve Hmm.. From memory while back I think I had a similar problem. Try to: bind= YOUR IP ADDRESS. Ta SJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 bug in 6.1 evident in Asterisk
I had a previous error where, due to a faulty switch port, one of my 7960's was rebooting or locking fairly often. That was due to a physical, electrical error. This problem is significantly different. A fully-loaded (all six lines) 7960 will gradually stop registrations to one of my (distant) servers, and will often wedge itself, requiring reboot by power cord yanking. Or it will spontaneously reboot. I think this is due to some unusual SIP messages being sent to the phone from *, tickling a different bug in the phone that causes it to lose it's mind. See my bugnote: http://bugs.digium.com/bug_view_page.php?bug_id=889 Due to other network conditions (i.e.: the remote server 3300 miles away has a cable modem problem) I am unable to get more details. JT No they do not but apparetly his phones either didn't like the switch they were on or they have something wrong with them. bkw On Thu, 5 Feb 2004, Chris Clifton wrote: So do the 7960's have to be on the same subnet as the * box ? This seems like a major detriment to using them in a typical wan environment. - Chris Clifton - Original Message - From: Brian West [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, February 04, 2004 1:58 PM Subject: Re: [Asterisk-Users] Cisco 7960 bug in 6.1 evident in Asterisk Does the first line, backup and emergency proxy go to the * box on the same wire? Malcolm and I figured out the 7960's freak smooth out if the asterisk server isn't on the same subnet his phones kept rebooting over and over and over till we took them off the switch they were on and move them to the one with the aterisk server. bkw On Wed, 4 Feb 2004, John Todd wrote: Yes and no. The Cisco phone is on a NAT network that is quite distant from one of the Asterisk servers, but on the same wire as the other. Three lines go to the remote *, and three lines remain local on the network to the other * server. I'm running CVS as of this morning on both servers. Strangely, today the phone hasn't locked up or rebooted, though now I am getting one or two of the lines failing to REGISTER - they're simply not sending out a request, according to the network dump. sigh JT At 7:43 AM -0600 2/4/04, Brian West wrote: Question.. is the 7960 on the same subnet as your asterisk server? I have a 7960 registered with 3 diffrent asterisk servers. All 6 lines. Running 6.1 and has 12 days of uptime. bkw On Wed, 4 Feb 2004, John Todd wrote: So, I've managed to consistently lock up my Cisco 7960 (SIP 6.1) to the point where it needs to be unplugged, due to software errors. This is a first. My suspicions are that this bug in Asterisk is causing the lockups: http://bugs.digium.com/bug_view_page.php?bug_id=889 It seems unusual to me that a low volume of bogus SIP messages should lock up the 7960, but that seems to be the case. It seems this only happens on my 7960 that I have completely full of extensions (all six line buttons are lit, two of them are auto-answer.) I think this is one bug tickling another bug; bad messages from * are killing the 7960. I'd like anyone else with experiences with this type of failure with Asterisk to give me a shout; I'm going to report this to Cisco somehow, but don't have enough evidence. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] The Evil of type=friend explained, again ( wa s Re: [Asterisk-Users] Minor Registration Problem With Polycom Soun dpoin t IP 500)
Jeremy, There is one small flaw in your reasoning with the need to register. You said : You only need to register to Asterisk if you have a dynamic IP address or you need to blow thru a firewall/NAT device But this is not true if you want to maintain true presence information. If you do not register, no one who has subscribed to you will know that you are available. In many cases this is undesirable behavior. Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeremy McNamara Sent: Thursday, February 05, 2004 6:50 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] The Evil of type=friend explained, again (was Re: [Asterisk-Users] Minor Registration Problem With Polycom Soun dpoint IP 500) David Liu wrote: Could you tell us a little bit how exactly it works? The wiki pages don't say much about type=friend, user, and peer. I tried using type=user but can't seem to register. A type=friend is simply both a type=user and type=peer using the same set of config directives. While a type=friend makes things almost trivial to get calls working in both directions, it will limit the flexibility of your config and even hinder some of the more advanced uses of Asterisk. For example: Say you want to use the same 'user' across many different Asterisk boxes, which of course will have different IP addresses. In this situation, you cannot have a host keyword in your Asterisk config stanza for the type=user, but the type=peer requires some host keyword. Thus, if you use a type=friend you will limit the use of that one username to whatever IP address is contained in the host keyword. You only need to register to Asterisk if you have a dynamic IP address or you need to blow thru a firewall/NAT device. To register you need to have a type=peer with a host=dynamic. Since in your type=friend config directive you had host=some.ip.address, while this may be this is fine to for the type=user, this same value also gets used for the type=peer, which makes it so you cannot register since the IP address is hard coded. So, either you do not need to register and things will Just Work(tm) or you will need to use separate type=user and type=peer config directives. I smell the beginnings of a Whitepaper here. Jeremy McNamara - Original Message - From: Jeremy McNamara [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, February 05, 2004 2:47 AM Subject: Re: [Asterisk-Users] Minor Registration Problem With Polycom Soun dpoint IP 500 mattf wrote: I have all of my Polycom's set to friend so I know that's not your problem. One day you too will get bitten by the type=friend's EVIL and you will see the light. Trust me, Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Notice: This e-mail message, together with any attachments, contains information of Merck Co., Inc. (One Merck Drive, Whitehouse Station, New Jersey, USA 08889), and/or its affiliates (which may be known outside the United States as Merck Frosst, Merck Sharp Dohme or MSD and in Japan as Banyu) that may be confidential, proprietary copyrighted and/or legally privileged. It is intended solely for the use of the individual or entity named on this message. If you are not the intended recipient, and have received this message in error, please notify us immediately by reply e-mail and then delete it from your system. -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] sementation fault with mpg123
I'm still getting a sementation fault with mpg123. Ah, adventures in the pubic school system. Funny you should mention this... there is a large Hasidic community in our town. The boys refused to ride in busses driven by women. Eventually a separate school district was created (from the one I work in). Of course it is illegal but continues to exist while perpetually tied up in court. So it is the only segregated public school in the country. Can you give us a 'bt full' at this point? Core was generated by `asterisk -vvvfg'. Program terminated with signal 11, Segmentation fault. #0 0x0805781d in ?? () (gdb) bt full #0 0x0805781d in ?? () No symbol table info available. #1 0x41c1cf8f in ?? () No symbol table info available. #2 0x41c2d8ce in ?? () No symbol table info available. #3 0x41c28655 in ?? () No symbol table info available. #4 0x41c22410 in ?? () No symbol table info available. #5 0x08051790 in ?? () No symbol table info available. #6 0x41c1ee62 in ?? () No symbol table info available. #7 0x4002d484 in ?? () No symbol table info available. Now the plot thickens... I have several * boxes - only two run moh. One has been having these troubles (zenon on RH9 - crashes every day or so) the other runs pretty good (p4 on RH8 - crashes less than once a month). The machines not using moh never crash these are running RH9. Last night I just upgraded the RH8 to RH9 and ran up2date (and rebuilt *, of course) and it is now unstable like the other box... crapping on 'Ouch ... error while writing audio data: : Broken pipe'. John This e-mail was scanned and found clean by Monroe-Woodbury's Antivirus. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] The Evil of type=friend explained, again (was Re: [Asterisk-Users] Minor Registration Problem With Polycom Soun dpoint IP 500)
On Thursday 05 February 2004 05:50, Jeremy McNamara wrote: A type=friend is simply both a type=user and type=peer using the same set of config directives. While a type=friend makes things almost trivial to get calls working in both directions, it will limit the flexibility of your config and even hinder some of the more advanced uses of Asterisk. For example: Say you want to use the same 'user' across many different Asterisk boxes, which of course will have different IP addresses. In this situation, you cannot have a host keyword in your Asterisk config stanza for the type=user, but the type=peer requires some host keyword. Thus, if you use a type=friend you will limit the use of that one username to whatever IP address is contained in the host keyword. You only need to register to Asterisk if you have a dynamic IP address or you need to blow thru a firewall/NAT device. To register you need to have a type=peer with a host=dynamic. Since in your type=friend config directive you had host=some.ip.address, while this may be this is fine to for the type=user, this same value also gets used for the type=peer, which makes it so you cannot register since the IP address is hard coded. So, either you do not need to register and things will Just Work(tm) or you will need to use separate type=user and type=peer config directives. So, why can't you just do: [someuser] type=friend host=dynamic context=internal secret=somesecret In other words, you can have your user registered to the server AND be using a type=friend definition. This is exactly how I have some test equipment set up and it works perfectly well. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk as non root
On Thursday 05 February 2004 08:03, Chris Lee wrote: I followed the wiki instructions: http://www.voip-info.org/wiki-Asterisk+non-root Now I have a working asterisk running as user asterisk. I do however have some problems: 1: I dont have access via asterisk -r Permissions problem. User asterisk needs to have permissions to write the file /var/run/asterisk.ctl 2: The pid file is no longer being updated Again, permissions problem. 3: I want to create a file in init.d so that I can use service start and stop, but need to be able to pass asterisk the gracefully command etc, any ideas welcome. maybe: asterisk -rx stop gracefully etc Please see the multitude of init.d scripts already written that do this in /usr/src/asterisk/contrib/init.d/ -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Boards falling out...
Ejay Hire wrote: Hi. Low Temp Hot glue is what I use on my robots. Stay away from silicone (conductive) and rtv (peels traces off cheap pcb's) The only silicones that are electrically conductive are those that are loaded with some conductive material (like silver). These are rather esoteric and it is very unlikely that you would encounter them. On the other hand, many silicones are rather thermally conductive. Perhaps this is the source of this misconception. Also, Room Temperature Vulcanizing Silicones (RTV's) are, also, well, obviously, silicones. If removed with care, they should not lift traces, although, again obviously, too heavy a hand will remove traces from paper-phenolic (those brown ugly things) circuit boards. It would be really hard to lift them from epoxy-glass material (usually green). Nevertheless, hot melt glue is a fine choice. It hardens quickly, is easy to apply and is electrically and chemically inert. Stephen R. Besch ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Apple OS-X
Hi A colleague of mine read somewhere that it was possible to compile Asterisk under OS-X which he has just tried with little success. Has anybody here had any success and if so what things should my colleague take into account? Regards Martin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk as non root
Tilghman Lesher wrote: Permissions problem. User asterisk needs to have permissions to write the file /var/run/asterisk.ctl 2: The pid file is no longer being updated Again, permissions problem. I was under the impression that changing the line: ASTVARRUNDIR=$(INSTALL_PREFIX)/var/run/asterisk fixed it so that asterisk.ctl and asterisk.pid got written in the /var/run/asterisk directory, which I gave asterisk ownership of. Am I mistaken here, and if so where do I configure the source so that they get put there, thus getting the rights they need? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CallWaiting CallerID: Available on all channel types?
I have some phones that purport to handle this properly but am having quite a time figuring out just when to expect it to work and when not to. Limited to Zap channels? Zap and SIP, but in different manners? Sieving the list archives yielded more questions than answers. Pointers or discussion appreciated. Thx. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Fw: [Asterisk-Users] Possible Sip logic bug?
Rich, It is very important (at least to me) to have the whole SIP call flow. That is, I must see the initial INVITE come from the originating phone all the way to the last message. I can only speculate at this point but it appears that the second leg (destination) may never have ACK'd the call which could have Asterisk in a bad state. I cannot be sure of this without the entire flow but if this is the case, not only do you have a config problem, Asterisk has an unhandled error state.Did you answer the destination? Did it have 2-way voice path? Rich Adamson wrote: Clif and all... At the bottom of this post is the sip show debug for the problem. The underlying problem (again): when C7960 hangs up on working conversation, the C7960 sends a BYE to *, but * never sends a BYE to the sip gateway. Any suggestions would be greatly appreciated. Rich Try it again after executing: sip debug and give us the actual SIP messages. The devil is usually in the details. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Apple OS-X
Hi A colleague of mine read somewhere that it was possible to compile Asterisk under OS-X which he has just tried with little success. Has anybody here had any success and if so what things should my colleague take into account? Regards Martin *CLI show version Asterisk CVS-10/24/03-01:48:29 built by [EMAIL PROTECTED] on a Power Macintosh running Darwin *CLI I haven't tried compiling it lately, YMMV. It's somewhat choppy when you're doing things on the desktop (at least on my ~680mhz PB) - I tried and was able to get xten's softphone running through it, and also the iax client for MacOS X (see the archives.) At the moment, it's a curiosity but I'd suggest testing on a G5 or an Xserv before thinking that it's a real solution. G729 will not work, as I assume it's pre-compiled for Linux on i386 only. I don't recall if speex has a fink port to OSX, either... but the other codecs work fine. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Release phone call
Title: Message Hello all, I am trying to figure out how to have * release a phone call. We are noticing some call quality issues on people who have a "find-me" feature, and answer the call through a cell phone. Here is the call path we are seeing, and all VoIP connections are using SIP. PSTN --- Cisco 7206 --- * Server ^---| ^-| Hopefully the diagram makes sense, but in case it doesn't, let me try to explain. A call comes in from PSTN into our Cisco7206 with PRI card. It then goes to our * server, which then forwards the call back through the Cisco to a cell phone on PSTN. I am wanting to have * release the call to the Cisco once the call is connected. Any thoughts or ideas? Thanks. B. J.
Re: [Asterisk-Users] Release phone call
Title: Message I don't really have a answer for you on you issue but have a question about what "find-me" is. I see it on the feature list but am unable to find any real information about it. Is this simply call forward or is their more to it. thanks - Original Message - From: B. J. Bomar To: [EMAIL PROTECTED] Sent: Thursday, February 05, 2004 1:01 PM Subject: [Asterisk-Users] Release phone call Hello all, I am trying to figure out how to have * release a phone call. We are noticing some call quality issues on people who have a "find-me" feature, and answer the call through a cell phone. Here is the call path we are seeing, and all VoIP connections are using SIP. PSTN --- Cisco 7206 --- * Server ^---| ^-| Hopefully the diagram makes sense, but in case it doesn't, let me try to explain. A call comes in from PSTN into our Cisco7206 with PRI card. It then goes to our * server, which then forwards the call back through the Cisco to a cell phone on PSTN. I am wanting to have * release the call to the Cisco once the call is connected. Any thoughts or ideas? Thanks. B. J.
RE: [Asterisk-Users] Release phone call
Title: Message The way we have it setup is simply calling multiple numbers/channels. It is either setup manually in the configs, or through a very ugly menu interface I constructed. B. J. -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Glenn DalglieshSent: Thursday, February 05, 2004 12:31To: [EMAIL PROTECTED]Subject: Re: [Asterisk-Users] Release phone call I don't really have a answer for you on you issue but have a question about what "find-me" is. I see it on the feature list but am unable to find any real information about it. Is this simply call forward or is their more to it. thanks - Original Message - From: B. J. Bomar To: [EMAIL PROTECTED] Sent: Thursday, February 05, 2004 1:01 PM Subject: [Asterisk-Users] Release phone call Hello all, I am trying to figure out how to have * release a phone call. We are noticing some call quality issues on people who have a "find-me" feature, and answer the call through a cell phone. Here is the call path we are seeing, and all VoIP connections are using SIP. PSTN --- Cisco 7206 --- * Server ^---| ^-| Hopefully the diagram makes sense, but in case it doesn't, let me try to explain. A call comes in from PSTN into our Cisco7206 with PRI card. It then goes to our * server, which then forwards the call back through the Cisco to a cell phone on PSTN. I am wanting to have * release the call to the Cisco once the call is connected. Any thoughts or ideas? Thanks. B. J.
[Asterisk-Users] Vegastream 50 FXO with Asterisk
Anyone have any experienceconfiguringVegaStream's with Asterisk. Ihave run into afew of questions. 1. It appear that after turning on registrations I am seeing two request for registration per linesip:[EMAIL PROTECTED]sip:[EMAIL PROTECTED]What is purpose and how do I handle this?2. DTMF btw Asterisk and the Unit I was unable to get rfc2833 to work successfully with inbound or outbound DTMF. Is this a known issue? 3. How is the best way to deal with dialout and selecting a free channel on the VegaStream Any general suggestions/experiences with regard to configuring a VegaStream withasteriskwould be appricated.Thanks
Re: [Asterisk-Users] Asterisk as non root
On Thursday 05 February 2004 11:13, Chris Lee wrote: Tilghman Lesher wrote: Permissions problem. User asterisk needs to have permissions to write the file /var/run/asterisk.ctl 2: The pid file is no longer being updated Again, permissions problem. I was under the impression that changing the line: ASTVARRUNDIR=$(INSTALL_PREFIX)/var/run/asterisk fixed it so that asterisk.ctl and asterisk.pid got written in the /var/run/asterisk directory, which I gave asterisk ownership of. Am I mistaken here, and if so where do I configure the source so that they get put there, thus getting the rights they need? Check your /etc/asterisk/asterisk.conf. That file states where the varrun directory is located. If it doesn't exist, then it will use whatever path is hardcoded into asterisk. Note that changing the value in the Makefile necessitates a 'make clean install' before the paths will be updated in the binary. If you're not sure of the path hardcoded, you can run the following: strings /usr/sbin/asterisk | grep -E 'asterisk.(ctl|pid)' -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Fw: [Asterisk-Users] Possible Sip logic bug?
Sorry Clif, as a professional working with protocol analysis at corporations in more than 40 states, I should have known better. Never gave it a thought the issue could have been earlier in the call/session setup. I'll dig into that, and if still need help/suggestions will post the full debug trace. Rich - It is very important (at least to me) to have the whole SIP call flow. That is, I must see the initial INVITE come from the originating phone all the way to the last message. I can only speculate at this point but it appears that the second leg (destination) may never have ACK'd the call which could have Asterisk in a bad state. I cannot be sure of this without the entire flow but if this is the case, not only do you have a config problem, Asterisk has an unhandled error state.Did you answer the destination? Did it have 2-way voice path? Rich Adamson wrote: Clif and all... At the bottom of this post is the sip show debug for the problem. The underlying problem (again): when C7960 hangs up on working conversation, the C7960 sends a BYE to *, but * never sends a BYE to the sip gateway. Any suggestions would be greatly appreciated. Rich Try it again after executing: sip debug and give us the actual SIP messages. The devil is usually in the details. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---End of Original Message- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Data call transfer
(Please forward this to Martin Pycko in Digium)
Martin, this is all about mail that I have sent to you regarding data
call setup.
Hi Thomas,
Thanks for your hint. I have tried it but it doesn't work. Here are few
lines from my config...
;
;RAS
;
exten = 290,1,GotoIf,$[${CALLTYPE} = DIGITAL]?50:100
exten = 290,50,Dial(Zap/g2d/${EXTEN})
exten = 290,100,Dial(Zap/g2/${EXTEN})
I have captured some PRI messages from both interfaces. Here they are,
first two are captured in the moment of setup, an down are two captured
in the moment of release.
If I am right, I think the outgoing call to PBX is voice instead of
data.
INCOMING E1 PORT FROM PSTN IN THE MOMENT OF ISDN DATA CALL SETUP
Protocol Discriminator: Q.931 (8) len=32
Call Ref: len= 2 (reference 94/0x5E) (Originator)
Message type: SETUP (5)
Sending Complete (len= 4)
Bearer Capability (len= 2) [ Ext: 1 Q.931 Std: 0 Info transfer
capability: Unrestricted digital information (8)
Ext: 1 Trans mode/rate: 64kbps,
circuit-mode (16)
Ext: 0 User information layer 1: Unknown
(24)
Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Preferred
Dchan: 0
ChanSel: Reserved
Ext: 1 Coding: 0 Number Specified Channel
Type: 3
Ext: 1 Channel: 1 ]
Calling Number (len=11) [ Ext: 0 TON: Subscriber Number (4) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1)
Presentation: Presentation permitted, user
number passed network screening (1) '6658218' ]
Called Number (len= 6) [ Ext: 1 TON: Unknown Number Type (0) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1) '290' ]
-- Making new call for cr 94
-- Processing Q.931 Call Setup
-- Processing IE 33 (Sending Complete)
-- Processing IE 4 (Bearer Capability)
-- Processing IE 24 (Channel Identification)
-- Processing IE 108 (Calling Party Number)
-- Processing IE 112 (Called Party Number)
Protocol Discriminator: Q.931 (8) len=10
Call Ref: len= 2 (reference 32862/0x805E) (Terminator)
Message type: CALL PROCEEDING (2)
Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive
Dchan: 0
ChanSel: Reserved
Ext: 1 Coding: 0 Number Specified Channel
Type: 3
Ext: 1 Channel: 1 ]
Protocol Discriminator: Q.931 (8) len=14
Call Ref: len= 2 (reference 32862/0x805E) (Terminator)
Message type: ALERTING (1)
Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive
Dchan: 0
ChanSel: Reserved
Ext: 1 Coding: 0 Number Specified Channel
Type: 3
Ext: 1 Channel: 1 ]
Progress Indicator (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0)
0: 0 Location: Private network serving the local user (1)
Ext: 1 Progress Description: Inband
information or appropriate pattern now available. (8) ]
Protocol Discriminator: Q.931 (8) len=14
Call Ref: len= 2 (reference 32862/0x805E) (Terminator)
Message type: ALERTING (1)
Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive
Dchan: 0
ChanSel: Reserved
Ext: 1 Coding: 0 Number Specified Channel
Type: 3
Ext: 1 Channel: 1 ]
Progress Indicator (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0)
0: 0 Location: Private network serving the local user (1)
Ext: 1 Progress Description: Inband
information or appropriate pattern now available. (8) ]
Protocol Discriminator: Q.931 (8) len=13
Call Ref: len= 2 (reference 94/0x5E) (Originator)
Message type: STATUS (125)
Cause (len= 3) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0
Location: Public network serving the local user (2)
Ext: 1 Cause: Message not compatible with call state
(101), class = Protocol Error (6) ]
Cause data 0: 01 (1)
Call State (len= 1) [ Ext: 0 Coding: CCITT (ITU) standard (0) Call
state: Call Received (7)
-- Processing IE 8 (Cause)
-- Processing IE 20 (Call State)
Protocol Discriminator: Q.931 (8) len=14
Call Ref: len= 2 (reference 32862/0x805E) (Terminator)
Message type: CONNECT (7)
Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive
Dchan: 0
ChanSel: Reserved
Ext: 1 Coding: 0 Number Specified Channel
Type: 3
Ext: 1 Channel: 1 ]
Progress Indicator (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0)
0: 0 Location: Private network serving the local user (1)
Ext: 1 Progress Description: Called
equipment is non-ISDN. (2) ]
Protocol Discriminator: Q.931 (8) len=5
Call Ref: len= 2 (reference 94/0x5E) (Originator)
Message type: CONNECT ACKNOWLEDGE (15)
OUTGOING PORT TO PBX IN THE MOMENT OF ISDN DATA CALL SETUP
Protocol
[Asterisk-Users] question for oh323 users
Hi, I am trying to forward calls from one cisco gateway to another cisco gateway using asterisk cisco(5300)A 192.168.1.1 asterisk 192.168.1.2 cisco(5300)B 192.168.1.3 pstn --ciscoA-asterisk --ciscoB--pstn I have the below in my extension.conf [default] exten = _1905XXX,1,Dial,OH323/192.168.1.3 I keep getting error and I don't know what is wrong. I am able to see in my ciscoB accesslist, tcp packets are coming from 192.168.1.2 I get below error in my asterisk CLI Feb 5 16:17:01 ERROR[29716]: chan_oh323.c:1004 oh323_call: H323:0: Could not call 192.168.1.3. Feb 5 16:17:11 WARNING[29716]: pbx.c:1829 ast_pbx_run: Timeout, but no rule 't' in context 'default' It would be much appreciated if someone could point out what I am doing wrong or to any documentations. Many thanks. Regards, Anthony ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P - Asterisk - Asterisk - X100P setup help
You were close. Try this: phone line (PSTN) -- Asterisk X100P card -- Asterisk (Linux Box 1) ---NETWORK/INTERNET--- Asterisk (Linux Box 2)-- Asterisk X100P card-- phone line (PSTN) And use IAX2 to connect the two Asterisk boxes. Try a bit of research on loligo's site and you'll get it done. John On Thu, 2004-02-05 at 15:12, Maninder Bhatia wrote: Hi, I am very new to this forum, and to Asterisk world. I have two two X100P cards, and was trying to setup something which looks like phone line (PSTN) -- Asterisk X100P card -Asterisk (Linux Box 1) -- Asterisk X100P card -Asterisk (Linux Box 2) -- phone line (PSTN) Was requesting if someone could confirm if this could be done, and if it can be can I get the config file to do this, or the direction I should be going on. Thanks and Regards, Maninder ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fax with wildcards
Hello, does anybody know how stable is OpenCall's fax sending/receiving software? Is it still in development? I see only an old version on the ftp. Does anybody have any experienci with fax sending and the PC performance needed for this? What kind of PC hardware would I need when I would like to send concurrently faxes on one/two/three/four E1? Is it possible either? Thank you in advance, Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] simple test setup
Could someone point me to docs on how to set up a simple * test box I just d/l and installed those rpms mentioned a couple of days ago onto a fedora box I hope to get simple config, and two softphones working with each other (one windows and one linux) Hopefully, such an article would include softphone recommendations and use instructions Am I being too lazy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk as non root
On Thu, 2004-02-05 at 14:03, Chris Lee wrote: I followed the wiki instructions: http://www.voip-info.org/wiki-Asterisk+non-root Glad someone's finding it useful :) Now I have a working asterisk running as user asterisk. I do however have some problems: 1: I dont have access via asterisk -r root should have access using asterisk -r (does for me anyway) 2: The pid file is no longer being updated If this is an upgrade to a previous install, then check /etc/asterisk/asterisk.conf to see whether the change to ASTVARRUNDIR has taken effect in the config file... 3: I want to create a file in init.d so that I can use service start and stop, but need to be able to pass asterisk the gracefully command etc, any ideas welcome. maybe: asterisk -rx stop gracefully etc pass F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk-Users digest, Vol 1 #2749 - 7 msgs
You can have a look at wiki on iax trunking plus notes on setting up x100p card. David Kwok Message: 5 From: Maninder Bhatia [EMAIL PROTECTED] To: [EMAIL PROTECTED] Date: Thu, 5 Feb 2004 16:12:07 -0500 Subject: [Asterisk-Users] X100P - Asterisk - Asterisk - X100P setup help Reply-To: [EMAIL PROTECTED] I am very new to this forum, and to Asterisk world. I have two two X100P cards, and was trying to=20 setup something which looks like=20 phone line (PSTN) -- Asterisk X100P card -Asterisk (Linux Box 1) = -- =20 Asterisk X100P card -Asterisk (Linux Box 2) -- phone line (PSTN) Was requesting if someone could confirm if this could be done, and if = it can be can I get the config file to do this, or the direction I should be going = on.=20 smime.p7s Description: S/MIME Cryptographic Signature
[Asterisk-Users] Voiceglo questions
Hi, We're just about to bring up Asterisk in a small business setting with a broadband carrier. In this case, we have no reason to have any POTS lines to make incoming and outgoing calls using our SIP phones (Cisco 7960, 7940 and Grandstream 102.) We're probably selecting Voiceglo simply because we can transfer our existing local lines from an area code they handle (925). We've talked to a *lot* of broadband carriers, all of whom are stunningly unable to answer our basic questions about our proposed architecture. The one notable exception to this is Nufone which, unfortunately, doesn't service our local area code. A couple of questions for Voiceglo/Asterisk users: 1. Can someone confirm whether Voiceglo needs to use SIP or can it handle IAX? This link seems to indicate it uses SIP: http://www.mail-archive.com/[EMAIL PROTECTED]/msg20561.html although other messages on the mailing list indicate that Voiceglo is using Asterisk in its internal architecture. 2. Voiceglo's support keeps telling us we need to purchase an MTA (Multimedia Terminal Adapter), essentially an analog to digital box, described here: http://www.voiceglo.com/pages/Products_equipment.html Since we're using SIP phones and Asterisk, we have no need for this, right? 3. Any words of warning or praise from clients of Voiceglo? Thanks for any advice/help. Michael Swan Neon Software, Inc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voiceglo questions
quote who=Michael Swan 1. Can someone confirm whether Voiceglo needs to use SIP or can it handle IAX? This link seems to indicate it uses SIP: http://www.mail-archive.com/[EMAIL PROTECTED]/msg20561.html although other messages on the mailing list indicate that Voiceglo is using Asterisk in its internal architecture. The MTAs they sell use SIP. Their softphone uses IAX1. 2. Voiceglo's support keeps telling us we need to purchase an MTA (Multimedia Terminal Adapter), essentially an analog to digital box, described here: http://www.voiceglo.com/pages/Products_equipment.html Since we're using SIP phones and Asterisk, we have no need for this, right? They support connecting via equipment/software purchased through them. You are on your own, when connecting your own Asterisk implementation to their network. 3. Any words of warning or praise from clients of Voiceglo? They (atleast under SIP) use DTMF inband detection for DTMF after initial call setup. They also use g729. This means that while someone is talking you will hear a DTMF every once in a while. Also, when trying to get through DTMF menus is difficult. You get missed or double digits. -- END OF LINE -MCP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] has Allison recorded Do Not Disturb
I can't find Allison saying Do Not Disturb Anybody got this If not, is there a place to submit generic requests for sounds ??? -Lance ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] http://www.oneunified.net
Has anyone had good or bad experiance with http://www.oneunified.net. I need a DID for incomming calls only. Nufone does not have service in my area(614-XXX) :( Anyone have worked with these people. Good comments bad comments. Should we create a area in the WIKI for all of the VOIP providers so we can leave comments about them someplace, and not take up mailling list time? Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OT Asterisk Sales Questions (Not for Asterisk itself)
I have a few questions, and I'm hoping all of you nice people have an answer and can share the info. I don't need exact numbers, just asking for general info like yes, no, not as good as expected, etc. Has/is anyone selling Asterisk commercially? And is it successful or a flop? Has anyone sold an Asterisk box branded as something other than as an Asterisk box? And was it successful or a flop? (example: XYZ, Inc PBX) You can just e-mail me directly at [EMAIL PROTECTED] Rohde ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Randomly Stopping
Recently we established connection between two Asterisk systems using IAX. Since then, we have observed that these boxes randomly stop working. Checking the processes shows the Asterisk process is not running. There is no error message on the remote consoles that we use to monitor these boxes. Has anyone else experienced this phenomenon? Prior to the IAX interconnection, these boxes operated without any failure. TIA / Dam Zener ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voiceglo questions
On Thu, 5 Feb 2004, Michael Swan wrote: We're just about to bring up Asterisk in a small business setting with a broadband carrier. In this case, we have no reason to have any POTS lines to make incoming and outgoing calls using our SIP phones (Cisco 7960, 7940 and Grandstream 102.) We're probably selecting Voiceglo simply because we can transfer our existing local lines from an area code they handle (925). We've talked to a *lot* of broadband carriers, all of whom are stunningly unable to answer our basic questions about our proposed architecture. The one notable exception to this is Nufone which, unfortunately, doesn't service our local area code. A couple of questions for Voiceglo/Asterisk users: 1. Can someone confirm whether Voiceglo needs to use SIP or can it handle IAX? This link seems to indicate it uses SIP: http://www.mail-archive.com/[EMAIL PROTECTED]/msg20561.html although other messages on the mailing list indicate that Voiceglo is using Asterisk in its internal architecture. Voiceglo uses * servers (It identifies itself in the SIP headers, which you'll see when you turn debugging on). Their free webphone gadget uses IAX, while the pay-to-use services use SIP. (g711 and g729 codecs) 2. Voiceglo's support keeps telling us we need to purchase an MTA (Multimedia Terminal Adapter), essentially an analog to digital box, described here: http://www.voiceglo.com/pages/Products_equipment.html Since we're using SIP phones and Asterisk, we have no need for this, right? 3. Any words of warning or praise from clients of Voiceglo? I actually just signed up with them Tuesday of last week. My USB phone hasn't arrived yet, which means 9 of my 14 day risk free guarantee has already passed -- and I wouldn't have been able to even try the service yet if not for my determination to make it work. As you have discovered, they are very clear about not supporting anything but the MTA and USB phone available through them. The support people seem to be unaware of SIP or the fact that they aren't the only ones who use it. When I have called their support with any type of question, they keep telling me that I have to wait until I receive the USB phone and CD. Whatever. If you sign up for the USB phone, they'll send you a Windows registry file which contains your username and password (it's plaintext). You can use these to make any SIP client connect to their server. I haven't heard from anybody who got the MTA; it may be more difficult to discover your login info this route because the MTA probably comes pre-configured. I've been having some struggles with it so far: - when I use the xten softphone to call through my * and into voiceglo, the call fails because of some g729/ulaw codec issue. But when I connect the xten softphone directly to voiceglo, it works fine. Stranger still, if I use the SJ Labs softphone to call through my * and into voiceglo, it works. I don't know why. I also don't have a g729 license yet, and that may fix the issue * has when I use the xten client. - dmtf doesn't work reliably (not at all for me). When I call my voiceglo number and * answers, the menus and such don't work because of this. (they do work when I call in through iaxtel, FWD, etc) I haven't gotten around to bothering them about this yet. Since I'm using this for home use, it isn't a huge deal right now. And it still beats signing up for Vonage and paying $40 whenever I decide to end my service. The voice quality has been fairly good in my experience, but I wouldn't rate the support very highly. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] http://www.oneunified.net
Oneunified.net seems a little bit on the high side for pricing, no? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Thursday, February 05, 2004 5:43 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] http://www.oneunified.net Has anyone had good or bad experiance with http://www.oneunified.net. I need a DID for incomming calls only. Nufone does not have service in my area(614-XXX) :( Anyone have worked with these people. Good comments bad comments. Should we create a area in the WIKI for all of the VOIP providers so we can leave comments about them someplace, and not take up mailling list time? Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] http://www.oneunified.net
Should we create a area in the WIKI for all of the VOIP providers so we can leave comments about them someplace, and not take up mailling list time? Many of the providers already have a page on the Wiki. (You can create one if not) Please feel free to add comments to these pages about your expericences using them. http://www.voip-info.org/wiki-VOIP+Service+Providers Jim James H. Thompson [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voiceglo questions
I cannot recommend using Voiceglo for a business. Unless ringing and DTMF start working in a sensible way. Call quality has been reasonable. If you ignore that I would recommend: g729 licenses for Asterisk Broadband? Are you going to QoS to the broadband connection? How broad is your broadband? It is very likely that some customers will notice you are on a VoIP system, because call quality with Voiceglo is very similar to a cellular phone. I would plan to test call quality with Voiceglo in a production situation quite thoroughly. cameron. On Thu, 5 Feb 2004, Michael Swan wrote: Hi, We're just about to bring up Asterisk in a small business setting with a broadband carrier. In this case, we have no reason to have any POTS lines to make incoming and outgoing calls using our SIP phones (Cisco 7960, 7940 and Grandstream 102.) We're probably selecting Voiceglo simply because we can transfer our existing local lines from an area code they handle (925). We've talked to a *lot* of broadband carriers, all of whom are stunningly unable to answer our basic questions about our proposed architecture. The one notable exception to this is Nufone which, unfortunately, doesn't service our local area code. A couple of questions for Voiceglo/Asterisk users: 1. Can someone confirm whether Voiceglo needs to use SIP or can it handle IAX? This link seems to indicate it uses SIP: http://www.mail-archive.com/[EMAIL PROTECTED]/msg20561.html although other messages on the mailing list indicate that Voiceglo is using Asterisk in its internal architecture. 2. Voiceglo's support keeps telling us we need to purchase an MTA (Multimedia Terminal Adapter), essentially an analog to digital box, described here: http://www.voiceglo.com/pages/Products_equipment.html Since we're using SIP phones and Asterisk, we have no need for this, right? 3. Any words of warning or praise from clients of Voiceglo? Thanks for any advice/help. Michael Swan Neon Software, Inc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 bug in 6.1 evident in Asterisk
So Star-Six-Settings won't reboot the phone in this state? cameron. On Thu, 5 Feb 2004, John Todd wrote: I had a previous error where, due to a faulty switch port, one of my 7960's was rebooting or locking fairly often. That was due to a physical, electrical error. This problem is significantly different. A fully-loaded (all six lines) 7960 will gradually stop registrations to one of my (distant) servers, and will often wedge itself, requiring reboot by power cord yanking. Or it will spontaneously reboot. I think this is due to some unusual SIP messages being sent to the phone from *, tickling a different bug in the phone that causes it to lose it's mind. See my bugnote: http://bugs.digium.com/bug_view_page.php?bug_id=889 Due to other network conditions (i.e.: the remote server 3300 miles away has a cable modem problem) I am unable to get more details. JT No they do not but apparetly his phones either didn't like the switch they were on or they have something wrong with them. bkw On Thu, 5 Feb 2004, Chris Clifton wrote: So do the 7960's have to be on the same subnet as the * box ? This seems like a major detriment to using them in a typical wan environment. - Chris Clifton - Original Message - From: Brian West [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, February 04, 2004 1:58 PM Subject: Re: [Asterisk-Users] Cisco 7960 bug in 6.1 evident in Asterisk Does the first line, backup and emergency proxy go to the * box on the same wire? Malcolm and I figured out the 7960's freak smooth out if the asterisk server isn't on the same subnet his phones kept rebooting over and over and over till we took them off the switch they were on and move them to the one with the aterisk server. bkw On Wed, 4 Feb 2004, John Todd wrote: Yes and no. The Cisco phone is on a NAT network that is quite distant from one of the Asterisk servers, but on the same wire as the other. Three lines go to the remote *, and three lines remain local on the network to the other * server. I'm running CVS as of this morning on both servers. Strangely, today the phone hasn't locked up or rebooted, though now I am getting one or two of the lines failing to REGISTER - they're simply not sending out a request, according to the network dump. sigh JT At 7:43 AM -0600 2/4/04, Brian West wrote: Question.. is the 7960 on the same subnet as your asterisk server? I have a 7960 registered with 3 diffrent asterisk servers. All 6 lines. Running 6.1 and has 12 days of uptime. bkw On Wed, 4 Feb 2004, John Todd wrote: So, I've managed to consistently lock up my Cisco 7960 (SIP 6.1) to the point where it needs to be unplugged, due to software errors. This is a first. My suspicions are that this bug in Asterisk is causing the lockups: http://bugs.digium.com/bug_view_page.php?bug_id=889 It seems unusual to me that a low volume of bogus SIP messages should lock up the 7960, but that seems to be the case. It seems this only happens on my 7960 that I have completely full of extensions (all six line buttons are lit, two of them are auto-answer.) I think this is one bug tickling another bug; bad messages from * are killing the 7960. I'd like anyone else with experiences with this type of failure with Asterisk to give me a shout; I'm going to report this to Cisco somehow, but don't have enough evidence. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED]
RE: [Asterisk-Users] AutoAttendent ON/OFF control by Attendent
I know this is fairly old thread, but I have a question regarding this.
The following line:
exten = s,2,GotoIf($[${autoattendant} = 1]?auto|1)
Is basically saying goto context priority 1. So the last line also has
a goto to statement. When is this being trigered. So could you use the
same line but instead say:
exten = s,2,GotoIf($[${autoattendant} = 1]?4:3)
Just curious
-gcc
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tilghman
Lesher
Posted At: Monday, December 15, 2003 12:45 PM
Posted To: Asterisk User Group
Conversation: [Asterisk-Users] AutoAttendent ON/OFF control by Attendent
Subject: Re: [Asterisk-Users] AutoAttendent ON/OFF control by Attendent
On Monday 15 December 2003 10:57, Sri wrote:
Hi All
This is one scenario I would like to have some help. I have searched
the digium lists and could not find any posts on this.
How can an Attendant switch on or off the AutoAttendant from her
phone? Eg. 8am - Attendent enters office - switches OFF auto
attendent. He/She takes in all the incoming calls and answers.
12pm - out of lunch. Needs to put the system back into Auto.
1 pm - return from lunch. Needs to switch OFF auto attendent
5 pm- Puts Auto attendent ON.
I am sure there can be a script built that should change
extensions.conf. and reloading asterisk on the attendent activating
based on a clock that kicks in 8 am, 12 pm, 1 pm and 5 pm. I dont want
this to be time restricted. the attendent should have control. Is
there a better way ? this could be even done through the phone of the
attendent eg, like *80-1 (ON) *80 - 2 (OFF)...
exten = *801,1,DBPut(auto/attendant=1)
exten = *802,1,DBPut(auto/attendant=0)
exten = s,1,DBGet(autoattendant=auto/attendant)
exten = s,2,GotoIf($[${autoattendant} = 1]?auto|1)
exten = s,3,Dial(Zap/23,30,t)
exten = s,4,Goto(auto|1)
-Tilghman
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Re: [Asterisk-Users] has Allison recorded Do Not Disturb
cvs checkout asterisk-sounds bkw On Thu, 5 Feb 2004, Lance Arbuckle wrote: I can't find Allison saying Do Not Disturb Anybody got this If not, is there a place to submit generic requests for sounds ??? -Lance ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Asterisk-Dev] DISA
Hi All! Okay, let me understand this. No offense intended. I've struggled for about a month with an Asterisk system, seeking to establish at least the minimal functionality of the PBX we wanted to retire (Nortel SL1). My objective was to try and use Asterisk as a replacement/backup telephone switch. Although no one I've spoken with has said as much, it appears (based on my nearly constant efforts and the reems of downloaded code I've gone through) that the Asterisk application lack's the one capability that a large segment of the telephone market (CLEC's, as in my case, but virtually all service providers, etc...) require. Apparently, Asterisk doesn't really work like a traditional pbx in that you really can't (for example) select a line (say from a Norstar using a T1 connected to a Digium T410p) go off hook and get dialtone. Nor (and I understand the security issues of a DISA environment) does it appear that users can readily dial into an Asterisk system, get dialtone, and dial a call. I've reviewed, massaged, monkeyed with app_disa.c, and while it is a well done and serviceable application, it lacks the flexibility necessary to adequately address real world uses. Anyway, before I trash the project entirely and sell the equipment, I wanted to make sure that I really inderstood that Asterisk isn't (at present) capable of volume call switching in a DISA application. [moved to asterisk-users, as it is more on-topic there] I would disagree with your summary. Asterisk does not work well in an environment where you're connecting two lines together without dialing anything, though I can't say that I've tried this: exten = 1234,1,Dial(Zap/1-2/w) Where Zap 1-2 was a non-PRI T1 channel that had dialtone on it. Perhaps that would work. But that method would be foolish and somewhat crippled in functionality. But, to your argument, you certainly can create dialplan rules that just connect one line to another. If you have given the description above to people who said that it could not be done, then I suspect you have talked to people who have not done it and who were only marginally clued in as to how Asterisk works. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] help *** newbie
Es posible que no tengas bien configurada tu interface zapata.??? etc/asterisk/zapata.conf jorge On Thu, 2004-02-05 at 00:29, FRANCISCO PEREZ-LANDAETA wrote: can anyone help me on this ? i am having problems configuring the asterisk. i have included an attachment because for some reason i could not cut and past from the terminal to my hotmail account. your help is appreciated. thanks, *** please look at the errors francisco _ Check out the new MSN 9 Dial-up fast reliable Internet access with prime features! http://join.msn.com/?pgmarket=en-uspage=dialup/homeST=1 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Execute command in shell
I've seen its possible to use the System applications, but what about passing arguments to the command ? Thanks for your help! -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Marc Fargas Enviado el: jueves, 05 de febrero de 2004 13:37 Para: [EMAIL PROTECTED] Asunto: [Asterisk-Users] Execute command in shell Is it posible to make Asterisk execute a command on extensions.conf during a call ¿ (That's to transfer H323 call by telnetting the gatekeeper so Asterisk doesn't seem to like transferring h.323 ) Thanks! Marc ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 0.7.2 RPMS Updated
On Wed, 4 Feb 2004, Greg Boehnlein wrote: On Wed, 4 Feb 2004, Chris Tooley wrote: Well, I don't really know all that much about SuSE either. I just installed it about 19 hours ago for the first time. Well, depending on the version of RPM that they installed, you'll either need to issue rpm -ba asterisk.spec or rpmbuild -ba asterisk.spec. For all I know, you might just be able to install the RPMS for RH9 on Suse. It should complain if dependncies aren't met. I just realized that due to a logic error in my build-asterisk-distrib script, I did not upload the kernel-modules-zaptel RPMS for the 0.7.2 release. I have corrected that error, and they are now available. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 bug in 6.1 evident in Asterisk
Correct. Wedged hard. JT So Star-Six-Settings won't reboot the phone in this state? cameron. On Thu, 5 Feb 2004, John Todd wrote: I had a previous error where, due to a faulty switch port, one of my 7960's was rebooting or locking fairly often. That was due to a physical, electrical error. This problem is significantly different. A fully-loaded (all six lines) 7960 will gradually stop registrations to one of my (distant) servers, and will often wedge itself, requiring reboot by power cord yanking. Or it will spontaneously reboot. I think this is due to some unusual SIP messages being sent to the phone from *, tickling a different bug in the phone that causes it to lose it's mind. See my bugnote: http://bugs.digium.com/bug_view_page.php?bug_id=889 Due to other network conditions (i.e.: the remote server 3300 miles away has a cable modem problem) I am unable to get more details. JT No they do not but apparetly his phones either didn't like the switch they were on or they have something wrong with them. bkw On Thu, 5 Feb 2004, Chris Clifton wrote: So do the 7960's have to be on the same subnet as the * box ? This seems like a major detriment to using them in a typical wan environment. - Chris Clifton - Original Message - From: Brian West [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, February 04, 2004 1:58 PM Subject: Re: [Asterisk-Users] Cisco 7960 bug in 6.1 evident in Asterisk Does the first line, backup and emergency proxy go to the * box on the same wire? Malcolm and I figured out the 7960's freak smooth out if the asterisk server isn't on the same subnet his phones kept rebooting over and over and over till we took them off the switch they were on and move them to the one with the aterisk server. bkw On Wed, 4 Feb 2004, John Todd wrote: Yes and no. The Cisco phone is on a NAT network that is quite distant from one of the Asterisk servers, but on the same wire as the other. Three lines go to the remote *, and three lines remain local on the network to the other * server. I'm running CVS as of this morning on both servers. Strangely, today the phone hasn't locked up or rebooted, though now I am getting one or two of the lines failing to REGISTER - they're simply not sending out a request, according to the network dump. sigh JT At 7:43 AM -0600 2/4/04, Brian West wrote: Question.. is the 7960 on the same subnet as your asterisk server? I have a 7960 registered with 3 diffrent asterisk servers. All 6 lines. Running 6.1 and has 12 days of uptime. bkw On Wed, 4 Feb 2004, John Todd wrote: So, I've managed to consistently lock up my Cisco 7960 (SIP 6.1) to the point where it needs to be unplugged, due to software errors. This is a first. My suspicions are that this bug in Asterisk is causing the lockups: http://bugs.digium.com/bug_view_page.php?bug_id=889 It seems unusual to me that a low volume of bogus SIP messages should lock up the 7960, but that seems to be the case. It seems this only happens on my 7960 that I have completely full of extensions (all six line buttons are lit, two of them are auto-answer.) I think this is one bug tickling another bug; bad messages from * are killing the 7960. I'd like anyone else with experiences with this type of failure with Asterisk to give me a shout; I'm going to report this to Cisco somehow, but don't have enough evidence. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED]
Re: [Asterisk-Users] Cisco 7960 bug in 6.1 evident in Asterisk
Eek. cameron. On Thu, 5 Feb 2004, John Todd wrote: Correct. Wedged hard. JT So Star-Six-Settings won't reboot the phone in this state? cameron. On Thu, 5 Feb 2004, John Todd wrote: I had a previous error where, due to a faulty switch port, one of my 7960's was rebooting or locking fairly often. That was due to a physical, electrical error. This problem is significantly different. A fully-loaded (all six lines) 7960 will gradually stop registrations to one of my (distant) servers, and will often wedge itself, requiring reboot by power cord yanking. Or it will spontaneously reboot. I think this is due to some unusual SIP messages being sent to the phone from *, tickling a different bug in the phone that causes it to lose it's mind. See my bugnote: http://bugs.digium.com/bug_view_page.php?bug_id=889 Due to other network conditions (i.e.: the remote server 3300 miles away has a cable modem problem) I am unable to get more details. JT No they do not but apparetly his phones either didn't like the switch they were on or they have something wrong with them. bkw On Thu, 5 Feb 2004, Chris Clifton wrote: So do the 7960's have to be on the same subnet as the * box ? This seems like a major detriment to using them in a typical wan environment. - Chris Clifton - Original Message - From: Brian West [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, February 04, 2004 1:58 PM Subject: Re: [Asterisk-Users] Cisco 7960 bug in 6.1 evident in Asterisk Does the first line, backup and emergency proxy go to the * box on the same wire? Malcolm and I figured out the 7960's freak smooth out if the asterisk server isn't on the same subnet his phones kept rebooting over and over and over till we took them off the switch they were on and move them to the one with the aterisk server. bkw On Wed, 4 Feb 2004, John Todd wrote: Yes and no. The Cisco phone is on a NAT network that is quite distant from one of the Asterisk servers, but on the same wire as the other. Three lines go to the remote *, and three lines remain local on the network to the other * server. I'm running CVS as of this morning on both servers. Strangely, today the phone hasn't locked up or rebooted, though now I am getting one or two of the lines failing to REGISTER - they're simply not sending out a request, according to the network dump. sigh JT At 7:43 AM -0600 2/4/04, Brian West wrote: Question.. is the 7960 on the same subnet as your asterisk server? I have a 7960 registered with 3 diffrent asterisk servers. All 6 lines. Running 6.1 and has 12 days of uptime. bkw On Wed, 4 Feb 2004, John Todd wrote: So, I've managed to consistently lock up my Cisco 7960 (SIP 6.1) to the point where it needs to be unplugged, due to software errors. This is a first. My suspicions are that this bug in Asterisk is causing the lockups: http://bugs.digium.com/bug_view_page.php?bug_id=889 It seems unusual to me that a low volume of bogus SIP messages should lock up the 7960, but that seems to be the case. It seems this only happens on my 7960 that I have completely full of extensions (all six line buttons are lit, two of them are auto-answer.) I think this is one bug tickling another bug; bad messages from * are killing the 7960. I'd like anyone else with experiences with this type of failure with Asterisk to give me a shout; I'm going to report this to Cisco somehow, but don't have enough evidence. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___
RE: [Asterisk-Users] simple test setup
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian Johnson Sent: Friday, 6 February 2004 9:00 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] simple test setup Could someone point me to docs on how to set up a simple * test box I just d/l and installed those rpms mentioned a couple of days ago onto a fedora box I hope to get simple config, and two softphones working with each other (one windows and one linux) Hopefully, such an article would include softphone recommendations and use instructions Am I being too lazy Yes! http://www.automated.it/guidetoasterisk.htm cd /usr/src/asterisk make samples ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: DISA
So, to boil your problem down to what I think is the problem:
When you attach an inbound call to the DISA application, it does not
produce a dialtone fast enough.
Is that the summary that I understand from your comments below? If
so, then we have narrowed things down a bit.
To the end of your actual problem, if I interpret it correctly: for
experimentation, try adding an Answer command right before the DISA
and see what you get. My experience with DISA (on VoIP and PRI, at
least) is that it gives an _immediate_ dialtone, without entry of any
keys or delay.
The unchangeable timer is the delay between the last keystroke of
entering something within the DISA and the DISA deciding to act upon
the string. As to my usage of the word unchangeable: this is open
source. Everything is changeable. My comments referenced what can
be done within the dialplan.
PS: Your mail program is confusing who said what, as well. I did not
say everything you attribute to me, and that is not clear from
looking at your message. James Sharp wrote the first two paragraphs
you say that are my quotes.
JT
At 7:43 PM -0600 2/5/04, Ed Devine wrote:
Andrew,
Thanks for your interest and courteous response. My company is a facilities
based CLEC. By way of background, I'm new to Asterisk and Digium, but I have
a good deal of past experience with Dialogic and NMS products in the telco
environment. I spend most of my time working with Nortel DMS-XXX switchgear
and managing the company ISP facilities.
I've been using a variation of a dialplan that I got from John Todd (see
below). The problem I've allways encountered is that for Asterisk to work in
our environment, it must allow the following:
Scenario:
user (whether automatic dialer, PBX ARS, PBX LCR, or even manually dialing
from any phone) accesses the switch (asterisk system) via a 10 digit did
number.
dial 972-NXX-
the switch answers and returns dialtone immediately
dial 228 1XX
(we use a seven digit authorization code sometimes in conjunction with
caller-id to verify that this is a valid account, etc...) followed
immediately by the 10 or 11 digit number you want to reach. The switch
selects an outbound trunk, strips the MSD if necessary, and ships the dialed
number digits.
The problem I've encountered is that inbound disa calls don't return
dialtone unless you enter something or until the unchangeable (John's word,
not mine) timer values time out.
John Todd has been most helpful, and his brief communications have been
incredible insights into how Asterisk works, he recently sent the following:
John's stuff starts here
app_disa will give answer and give you dial tone, wait for an
authentication code, then dump you into a context where you can make your
outgoing calls. Unfortunately, it needs a # at the end of the
authentication code.
A quick glance at the code suggests that it could be changed to expect a
fixed 7 digit access code.
It would be easy enough just to cut the first seven digits off the number
and run it through a comparison pass, and not use the authentication
routines at all.
;
; for North American numbers...
;
[main1]
;
; Take any number, and give it to the DISA. The DISA
; just then takes anything typed in within the (unchangeable)
; timer values, and hands it off to main2 to be post-processed.
; I include the standard i,h,t values for pedantic reasons.
;
exten = _X.,1,DISA(no-password,main2)
exten = _X.,2,Hangup
;
exten = h,1,Hangup
exten = i,1,Congestion
exten = i,2,Hangup
exten = t,1,Congestion
exten = t,2,Hangup
;
;
[main2]
;
; Now, set the AUTHCODE to be the first seven digits of EXTEN
;
exten = _XXX1XX,1,SetVar(AUTHCODE=${EXTEN:0:7})
;
; ...and then forward this call out to a new context and extension,
; where the new extension is the 7th through 17th digit of the old EXTEN,
; which should translate into 1-123-456-7890 or whatever it was that
; the user entered as the desired destination phone number.
;
exten = _XXX1XX,2,Goto(main-dial-routine,${EXTEN:7:17},1)
;
exten = h,1,Hangup
exten = i,1,Congestion
exten = i,2,Hangup
exten = t,1,Congestion
exten = t,2,Hangup
;
; end of example
This would end up (if the user entered the appropriate 7 digits and
1-npa-xxx- phone number) with passing the authentication code to
the main-dial-routine contained in ${AUTHCODE} and the ${EXTEN} set
to the number dialed.
You could also use the Cut application to perform a similar purpose
to my example using substring identifiers, if you wanted to put a
pound or star (or for those telephonically exotic among you, the
A/B/C/D) key separator in between the passcode digits and the phone
number.
JT
The upshot of my attempts was that, unless something is entered, dialtone
takes 5 seconds.
The same effect is apparent whether dialing inbound via the 10 digit did, or
when selecting a line from the Norstar
attached to the Digium T410P. If I dial in, the asterisk won't provide
dialtone unless I enter
RE: [Asterisk-Users] simple test setup
The Mepis linux distro is pre-configured for Asterisk. It's at www.mepis.org Start with two extensions that talk with each locally. Have all the test equipment on the same sub-net. Don't try to go through NAT. Paul Paul Mahler mail:[EMAIL PROTECTED] phone: 650.207.9855 fax: 877.408.0105 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Thursday, February 05, 2004 6:03 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] simple test setup -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian Johnson Sent: Friday, 6 February 2004 9:00 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] simple test setup Could someone point me to docs on how to set up a simple * test box I just d/l and installed those rpms mentioned a couple of days ago onto a fedora box I hope to get simple config, and two softphones working with each other (one windows and one linux) Hopefully, such an article would include softphone recommendations and use instructions Am I being too lazy Yes! http://www.automated.it/guidetoasterisk.htm cd /usr/src/asterisk make samples ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] simple test setup
Notice he did indicate he installed from the rpm's, so he's not using the source. But I agree on the lazy part! There are tons of resources available. Try http://www.voip-info.org and look at the config file section. Then try to create what you need (they're not hard for proof-of-concept work). Worst case, download the asterisk source and just do a make config. That'll create your configs for you. Should the rpm's at least install the *.conf.sample files? Just a thought -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Thursday, February 05, 2004 9:03 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] simple test setup -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian Johnson Sent: Friday, 6 February 2004 9:00 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] simple test setup Could someone point me to docs on how to set up a simple * test box I just d/l and installed those rpms mentioned a couple of days ago onto a fedora box I hope to get simple config, and two softphones working with each other (one windows and one linux) Hopefully, such an article would include softphone recommendations and use instructions Am I being too lazy Yes! http://www.automated.it/guidetoasterisk.htm cd /usr/src/asterisk make samples ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: DISA
On Thu, 5 Feb 2004, John Todd wrote: So, to boil your problem down to what I think is the problem: When you attach an inbound call to the DISA application, it does not produce a dialtone fast enough. snip [main1] ; ; Take any number, and give it to the DISA. The DISA ; just then takes anything typed in within the (unchangeable) ; timer values, and hands it off to main2 to be post-processed. ; I include the standard i,h,t values for pedantic reasons. ; exten = _X.,1,DISA(no-password,main2) exten = _X.,2,Hangup ; exten = h,1,Hangup exten = i,1,Congestion exten = i,2,Hangup exten = t,1,Congestion exten = t,2,Hangup Not to point out the obvious, but isn't the delay he's seeing caused by the _X. and the digittimeout? Couldn't this be resolved by using a more specific match on the DISA instead of _X. ? Steve [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Current version of gastman precompiled binary
Looking for a current precompiled Win32 binary for gastman, don't have a build environment for Windows. Also does gastman compile under Linux and is there a current binary as well... Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: DISA
quote who=Steve Creel [main1] ; ; Take any number, and give it to the DISA. The DISA ; just then takes anything typed in within the (unchangeable) ; timer values, and hands it off to main2 to be post-processed. ; I include the standard i,h,t values for pedantic reasons. ; exten = _X.,1,DISA(no-password,main2) exten = _X.,2,Hangup ; exten = h,1,Hangup exten = i,1,Congestion exten = i,2,Hangup exten = t,1,Congestion exten = t,2,Hangup Not to point out the obvious, but isn't the delay he's seeing caused by the _X. and the digittimeout? Couldn't this be resolved by using a more specific match on the DISA instead of _X. ? I think that would be right. I would have used: exten = s,1,DISA(no-password,main2) exten = s,2,Hangup -- END OF LINE -MCP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sip transfers
My polycom phones have a transfer button on them and it used to work, now it puts the call on hold, you are allowed to call the other extension and tell them the call is there, but when you hang up, the call stays on hold, and the extension you are trying to transfer to gets nothing. Ideas? Thanks Sean Garland [EMAIL PROTECTED] http://www.siskiyoutech.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Answer from a specific Number
I was trying to only have Asterisk only answer with extension when it came from a specific Caller-id number, it works from all numbers with my example below: include = parkedcalls exten = s,1,Answer exten = s,2,DigitTimeout(10) exten = s,3,ResponseTimeout(20) exten = s,4,Background(vm-extension) Modified to: include = parkedcalls exten = s,1,Answer/6145551212 exten = s,2,DigitTimeout(10) exten = s,3,ResponseTimeout(20) exten = s,4,Background(vm-extension) I thought be adding the /6145551212 after the Answer above would do what I wanted but it doesn't. Could someone advise me of the right example? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] zaptel on Debian
Does anyone have the zaptel modules built for Debian 2.4.24 kernel? When I try to compile I get : ./gendigits gcc -I/include -O2 -DMODULE -D__KERNEL__ -DEXPORT_SYMTAB -Wall -I. -Wstrict-prototypes -fo mit-frame-pointer -DSTANDALONE_ZAPATA -c zaptel.c In file included from /usr/include/asm/smp.h:18, from /usr/include/linux/smp.h:17, from /usr/include/linux/sched.h:23, from /usr/include/linux/module.h:10, from zaptel.c:44: /usr/include/asm/mpspec.h:6:25: mach_mpspec.h: No such file or directory In file included from /usr/include/asm/smp.h:18, from /usr/include/linux/smp.h:17, from /usr/include/linux/sched.h:23, from /usr/include/linux/module.h:10, from zaptel.c:44: /usr/include/asm/mpspec.h:8: error: `MAX_MP_BUSSES' undeclared here (not in a function) /usr/include/asm/mpspec.h:9: error: `MAX_MP_BUSSES' undeclared here (not in a function) /usr/include/asm/mpspec.h:10: error: `MAX_MP_BUSSES' undeclared here (not in a function) /usr/include/asm/mpspec.h:12: error: `MAX_MP_BUSSES' undeclared here (not in a function) /usr/include/asm/mpspec.h:19: error: `MAX_APICS' undeclared here (not in a function) /usr/include/asm/mpspec.h:20: error: `MAX_MP_BUSSES' undeclared here (not in a function) /usr/include/asm/mpspec.h:20: error: conflicting types for `mp_bus_id_to_type' /usr/include/asm/mpspec.h:8: error: previous declaration of `mp_bus_id_to_type' /usr/include/asm/mpspec.h:22: error: `MAX_IRQ_SOURCES' undeclared here (not in a function) /usr/include/asm/mpspec.h:24: error: `MAX_MP_BUSSES' undeclared here (not in a function) /usr/include/asm/mpspec.h:24: error: conflicting types for `mp_bus_id_to_pci_bus' /usr/include/asm/mpspec.h:12: error: previous declaration of `mp_bus_id_to_pci_bus' /usr/include/asm/mpspec.h:54: error: `MAX_APICS' undeclared here (not in a function) In file included from /usr/include/asm/smp.h:20, from /usr/include/linux/smp.h:17, from /usr/include/linux/sched.h:23, from /usr/include/linux/module.h:10, from zaptel.c:44: /usr/include/asm/io_apic.h:120: error: `MAX_IRQ_SOURCES' undeclared here (not in a functio n) /usr/include/asm/io_apic.h:120: error: conflicting types for `mp_irqs' /usr/include/asm/mpspec.h:22: error: previous declaration of `mp_irqs' make[2]: *** [zaptel.o] Error 1 make[2]: Leaving directory `/usr/src/modules/zaptel' make[1]: *** [binary-modules] Error 2 make[1]: Leaving directory `/usr/src/modules/zaptel' make: *** [kdist_image] Error 2 I'm too tired to fight with this. If someone has the modules for this kernel, can you tar them up and send them to me? Tim -- Tim Sailer Coastal Internet, Inc. Network and Systems Operations PO Box 726 http://www.buoy.comMoriches, NY 11955 [EMAIL PROTECTED] (631) 399-2910 (888) 924-3728 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] zaptel on Debian
do you have the kernel source installed? -Original Message- From: Tim Sailer [mailto:[EMAIL PROTECTED] Sent: Thursday, February 05, 2004 10:33 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] zaptel on Debian Does anyone have the zaptel modules built for Debian 2.4.24 kernel? When I try to compile I get : ./gendigits gcc -I/include -O2 -DMODULE -D__KERNEL__ -DEXPORT_SYMTAB -Wall -I. -Wstrict-prototypes -fo mit-frame-pointer -DSTANDALONE_ZAPATA -c zaptel.c In file included from /usr/include/asm/smp.h:18, from /usr/include/linux/smp.h:17, from /usr/include/linux/sched.h:23, from /usr/include/linux/module.h:10, from zaptel.c:44: /usr/include/asm/mpspec.h:6:25: mach_mpspec.h: No such file or directory In file included from /usr/include/asm/smp.h:18, from /usr/include/linux/smp.h:17, from /usr/include/linux/sched.h:23, from /usr/include/linux/module.h:10, from zaptel.c:44: /usr/include/asm/mpspec.h:8: error: `MAX_MP_BUSSES' undeclared here (not in a function) /usr/include/asm/mpspec.h:9: error: `MAX_MP_BUSSES' undeclared here (not in a function) /usr/include/asm/mpspec.h:10: error: `MAX_MP_BUSSES' undeclared here (not in a function) /usr/include/asm/mpspec.h:12: error: `MAX_MP_BUSSES' undeclared here (not in a function) /usr/include/asm/mpspec.h:19: error: `MAX_APICS' undeclared here (not in a function) /usr/include/asm/mpspec.h:20: error: `MAX_MP_BUSSES' undeclared here (not in a function) /usr/include/asm/mpspec.h:20: error: conflicting types for `mp_bus_id_to_type' /usr/include/asm/mpspec.h:8: error: previous declaration of `mp_bus_id_to_type' /usr/include/asm/mpspec.h:22: error: `MAX_IRQ_SOURCES' undeclared here (not in a function) /usr/include/asm/mpspec.h:24: error: `MAX_MP_BUSSES' undeclared here (not in a function) /usr/include/asm/mpspec.h:24: error: conflicting types for `mp_bus_id_to_pci_bus' /usr/include/asm/mpspec.h:12: error: previous declaration of `mp_bus_id_to_pci_bus' /usr/include/asm/mpspec.h:54: error: `MAX_APICS' undeclared here (not in a function) In file included from /usr/include/asm/smp.h:20, from /usr/include/linux/smp.h:17, from /usr/include/linux/sched.h:23, from /usr/include/linux/module.h:10, from zaptel.c:44: /usr/include/asm/io_apic.h:120: error: `MAX_IRQ_SOURCES' undeclared here (not in a functio n) /usr/include/asm/io_apic.h:120: error: conflicting types for `mp_irqs' /usr/include/asm/mpspec.h:22: error: previous declaration of `mp_irqs' make[2]: *** [zaptel.o] Error 1 make[2]: Leaving directory `/usr/src/modules/zaptel' make[1]: *** [binary-modules] Error 2 make[1]: Leaving directory `/usr/src/modules/zaptel' make: *** [kdist_image] Error 2 I'm too tired to fight with this. If someone has the modules for this kernel, can you tar them up and send them to me? Tim -- Tim Sailer Coastal Internet, Inc. Network and Systems Operations PO Box 726 http://www.buoy.comMoriches, NY 11955 [EMAIL PROTECTED] (631) 399-2910 (888) 924-3728 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
