RE: [Asterisk-Users] 'Asterisk for Small Office Setup'
Surely you mean grammar? Sorry, I just had to point that out :) Personally, I'd take issue with the title -- if you need to do a small-office setup by-the-book, then chances are you're not resourceful enough to find the required information online -- and if you can't even find the basics online, once something doesn't go quite-as-planned, you're stuck up the creek without a paddle. I propose to write a five-page pamphlet based on the excellent Getting Started With Asterisk guide that simply lists what you can do with Asterisk; what you need in terms of hardware and SKILLs; then walks you through a basic setup from scratch and lists where you can find more information online. This would be a worthwhile pursuit once 1.0 is done. -Original Message- From: John Vogel [mailto:[EMAIL PROTECTED] Sent: Saturday, July 24, 2004 9:15 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] 'Asterisk for Small Office Setup' 1. Poor spelling, punctuation, grammer. A flat, hard-to-read ___ 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 with T1 PRI Configuration
[I messed up the in-reply-to of this email, sorry] Kevin P. Fleming wrote: [EMAIL PROTECTED] wrote: 1. Dial Tone - No, Yes - Precise, Yes - SCC Not relevant on a PRI. Actually the dialtone may be optional on a pri. It is mandatory for EuroISDN but in other implementations it may not be. [snip] I'm not sure of the correct answers to any of these. I do know that I want to be able to get caller ID and the number dialed for the applicationthat I'm building. Most of these questions are for trunks being delivered over a T-1, not for PRI. If you are truly ordering PRI service (which is the more powerful of the two) then these questions don't apply :-) It does very much sound like they intend to deliver a non-pri T1 trunk. There are a lot of other knobs to fiddle with for a pri. :-) Peter ___ 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 Book
I`m currently writing my second asterisk book hoping to cover all the needs for everyone, I`m currently looking for a Editor if anyone is interested someone that would be willing to read over the book recommend ways of fixing it and making it more user frendly if anyone is interested please contact [EMAIL PROTECTED] Also if anyone would like to have there name under the Contractors list please email [EMAIL PROTECTED] I`m hoping to have this next book, ready to suite everyones needs ___ 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] John Vogel
Sorry you feel Strongly about the book but, Iâm not trying to rip anyone off but as it said on the back of the book and under the author Iâm new, to writing books I am trying to help the community out, But I guess this doesnât really matter to you, But if you know so much about asterisk and the way a book should be laid out why donât you write one? BTW I am open to ideas on improving the book as I said above I am not trying to rip anyone off The Author of Asterisk For Small Office Setup Dan Cole Best of whishes to you and your avengerâs ___ 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] Layer 3 VPN Question
Ithink the question isperfectsince VLANs are a great way to provideQoS and havent really been discussed here (at least lately). Can you be more specific as to your problem? Did you set vlan tagging on the phone? Did you trunk yourswitch portsall the way back to the router (ISL or 801.q, Cisco uses ISL natively)? Thanks, Steve Totaro - Original Message - From: Kevin To: [EMAIL PROTECTED] Sent: Saturday, July 24, 2004 11:02 PM Subject: [Asterisk-Users] Layer 3 VPN Question I am trying to hook up my Cisco telephones to Asterisk using a Layer 3 switch and am having difficulties it getting it to work. I realize this may not be the proper forum for a discussion on VLAN architecture and configuration so I wont post the question here. I though I had read all the requisite information regarding the configuration for this, but perhaps I am missing something simple. Is there anyone who is knowledgeable in VLANs and networking that could offer some assistance? Thanks, Kevin
Re: [Asterisk-Users] h323 to SIP Server Load
What solution provides a higher number of simultaneous calls? I found this http://www.mera-voip.com/voip/sip-hit.php. They claim 150 with a dedicated server and relatively modest hardware. Thanks, Steve Totaro - Original Message - From: Jeremy McNamara [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, July 24, 2004 11:09 AM Subject: Re: [Asterisk-Users] h323 to SIP Server Load Steve Totaro wrote: Does anyone do any large scale SIP to H323 conversion? How many simultaneous calls can your server handle and on what hardware? I think I read on the wiki that twenty five would max out most servers. The wiki is very wrong then. 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] John Vogel
Just a word of advice. This should not be posted to the list. Even the subject is addressed to one person, so why send it to the entire list. It certainly does not make you look very professional. I can appreciate your desire to be first to the market but common sense would dictate that you get an editor and release something free of grammatic error at the least. Personally, I would offer the guy a refund or a free copy of book 2 in good faith. I think this thread belongs here because of previous posts. Caveat emptor! - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, July 24, 2004 6:22 PM Subject: [Asterisk-Users] John Vogel Sorry you feel Strongly about the book but, Iâm not trying to rip anyone off but as it said on the back of the book and under the author Iâm new, to writing books I am trying to help the community out, But I guess this doesnât really matter to you, But if you know so much about asterisk and the way a book should be laid out why donât you write one? BTW I am open to ideas on improving the book as I said above I am not trying to rip anyone off The Author of Asterisk For Small Office Setup Dan Cole Best of whishes to you and your avengerâs ___ 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] No channel type registered for 'ZAP'
I found my error: the TDM01B (1-port FXO TDM400P bundle) ships with the single FXO module in position 4, not position 1. Thus using fxsks=4 in zaptel.conf and channel = 4 in zapata.conf fixed things. - Mike *CLI -- Executing Dial(SIP/555-83ee, ZAP/1/92262802) in new stack Jul 23 13:50:24 WARNING[-267056208]: channel.c:1860 ast_request: No channel type registered for 'ZAP' Jul 23 13:50:24 NOTICE[-267056208]: app_dial.c:696 dial_exec: Unable to create channel of type 'ZAP' == Everyone is busy/congested at this time ___ 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 Book
[EMAIL PROTECTED] wrote: (B (B if anyone would like to have there name under the (B Contractors list please email [EMAIL PROTECTED] (B (Band why don't you just list the entries on the Asterisk (BConsultants Wiki page at voip-info.org? (B (BI`m hoping to have this next book, ready to suite (Beveryones needs (B (BI don't think that trying to "suit everyone's needs" is a (Bgood strategy at all. Better pick a particular focus and (Bdo a good job on that. (B (Bjust my 2 cents (B (Brgds (Bbenjk (B (B (B-- (BSunrise Telephone Systems Ltd (B9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Shibuya-ku, Tokyo, Japan (B (B__ (BGANBARE! NIPPON! (BYahoo! JAPAN$B!!(BJOC OFFICIAL INTERNET PORTAL SITE (Bhttp://mail.ganbare-nippon.yahoo.co.jp/ (B (B___ (BAsterisk-Users mailing list (B[EMAIL PROTECTED] (Bhttp://lists.digium.com/mailman/listinfo/asterisk-users (BTo UNSUBSCRIBE or update options visit: (B http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FXS vs. FXO
Hello, Ive recently purchased Adit 600 with 3FXS and 1FXO to be connected to my * server via T100P card. This is the output of status equipment command in the Adit600: For some reason the FXO card is seen as FXS, why? Is it ok? On the card it is written FXO. Regards, Shlomi Bachar
[Asterisk-Users] Confused.
Hi all, I´m using X-Lite as SoftPhone in Asterisk. I have configured this: [101] ;Turn off silence suppression in X-Lite (Transmit Silence=YES)! ;Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed type=friend username=jozeph callerid=Jozeph Brasil 5678 host=dynamic nat=yes ; X-Lite is behind a NAT router canreinvite=no ; Typically set to NO if behind NAT ;disallow=all allow=gsm ; GSM consumes far less bandwidth than ulaw ;allow=ulaw ;allow=alaw When I try to connect X-Lite from another network follow this model: WORK (192.168.1.X) ß à FW-NAT ß à INTERNET ß à ASTERISK SERVER That machine connect OK! But, when I try to connect using this model: WORK (10.0.0.X) ß à 10.0.0.254 (ASTERISK SERVER) I can´t connect to the server why that? Asterisk Server have 2 network cards... internal and external internet card. My SIP.CONF are listen 0.0.0.0.
RE: [Asterisk-Users] FXS vs. FXO
Here is the output of the status equipment command : BootCode Version: 1.23 CardType Status SW Vers CLEI -- --- SLOT A T1x2 Present 6.1.2 SIC3DH0CAA SLOT 1 FXSx8 Present 1.09 SIC3GJ0CAA SLOT 2 FXSx8 Present 1.09 SIC3GJ0CAA SLOT 3 FXS5Gx8 Present 1.10 SIC2780JAA SLOT 4 FXSx8 Present 1.09 SIC3GJ0CAA SLOT 5 FXSx8 Not Present SLOT 6 V35x2 Not Present - Shlomi -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Shlomi Bachar Sent: Sunday, July 25, 2004 4:23 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] FXS vs. FXO Hello, Ive recently purchased Adit 600 with 3FXS and 1FXO to be connected to my * server via T100P card. This is the output of status equipment command in the Adit600: For some reason the FXO card is seen as FXS, why? Is it ok? On the card it is written FXO. Regards, Shlomi Bachar
Re: [Asterisk-Users] FXS vs. FXO
Shlomi Bachar wrote: Ive recently purchased Adit 600 with 3FXS and 1FXO to be connected to my * server via T100P card. This is the output of status equipment command in the Adit600: For some reason the FXO card is seen as FXS, why? Is it ok? On the card it is written FXO. I would ask ADIT, not the mailing list for a PBX. You could also, you know, TRY it. -- Daniel Jimenez djimenez[at]pobox[dot]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
RE: [Asterisk-Users] Layer 3 VPN Question
Shoot the questions to me offline if you'd like... -- W. Kevin Hunt CCIE #11841 www.huntbrothers.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin Sent: Saturday, July 24, 2004 10:02 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Layer 3 VPN Question I am trying to hook up my Cisco telephones to Asterisk using a Layer 3 switch and am having difficulties it getting it to work. I realize this may not be the proper forum for a discussion on VLAN architecture and configuration so I won't post the question here. I though I had read all the requisite information regarding the configuration for this, but perhaps I am missing something simple. Is there anyone who is knowledgeable in VLAN's and networking that could offer some assistance? ___ 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] Layer 3 VPN Question
why offline? this is good info for the archives. - Original Message - From: W. Kevin Hunt [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, July 25, 2004 10:25 AM Subject: RE: [Asterisk-Users] Layer 3 VPN Question Shoot the questions to me offline if you'd like... -- W. Kevin Hunt CCIE #11841 www.huntbrothers.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin Sent: Saturday, July 24, 2004 10:02 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Layer 3 VPN Question I am trying to hook up my Cisco telephones to Asterisk using a Layer 3 switch and am having difficulties it getting it to work. I realize this may not be the proper forum for a discussion on VLAN architecture and configuration so I won't post the question here. I though I had read all the requisite information regarding the configuration for this, but perhaps I am missing something simple. Is there anyone who is knowledgeable in VLAN's and networking that could offer some assistance? ___ 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] Layer 3 VPN Question
Sorry, there seems to be an above average amount of squaking about either where to post answers (top or bottom posting) and when to take it offline. I'll be happy to answer online now that at least 2 people are interested... As far as the question, what is the exact setup you are attempting to accomplish - A.) have the switch tell the phone which vlan to tag voice traffic and which vlan to tag data traffic ? B.) have the switch force all traffic f/ the phone into VLAN X and route it to the asterisk server on VLAN Y ? C.) same as B., but have a router handle the layer 3 work D.) ??? something else W. Kevin Hunt CCIE #11841 MCSE, Linux+ SME www.huntbrothers.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Sunday, July 25, 2004 9:43 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Layer 3 VPN Question why offline? this is good info for the archives. ___ 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 for Small Office Setup'
On Sun, 25 Jul 2004, Jay Milk wrote: Surely you mean grammar? Sorry, I just had to point that out :) Personally, I'd take issue with the title -- if you need to do a small-office setup by-the-book, then chances are you're not resourceful enough to find the required information online -- and if you can't even find the basics online, once something doesn't go quite-as-planned, you're stuck up the creek without a paddle. I propose to write a five-page pamphlet based on the excellent Getting Started With Asterisk guide that simply lists what you can do with Asterisk; what you need in terms of hardware and SKILLs; then walks you through a basic setup from scratch and lists where you can find more information online. This would be a worthwhile pursuit once 1.0 is done. Jay, This would be a great pamphlet to have. I believe that Cuban (on IRC #asterisk) is working on something similar that he was using as a brief for potential Asterisk consumers. Combining your idea with his would be wonderful. I'm going to be giving a presentation at the Ohio Linuxfest (http://www.ohiolinux.org) entitled Asterisk: VoIP for the Masses and a pamphlet handout would be just awesome to give people as a take-home. Something that can give the rationale and logic behind using Asterisk, as well as the basic information for getting the system installed and working and then where to look for more information. Heck, I'd probably even foot the bill to have them professionally printed, or perhaps even allow commercial Asterisk companies (such as Digium) to help sponsor the pamphlet. For the presentation, one of the things that I'll be doing is asking for the Asterisk community to review and comment on my work before I actually present at the conference. I want to make sure that it has a good, solid peer-review process in place, and that the issues discussed are consistent with the community goals. I also want to make sure that I highlight as much pre-existing community work as possible to ensure that those resources are being taken advantage of by new 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] which DB to use? or Why Berkley DB vs. MySQL, etc
What are the criteria you use to select which DB to use with * Built in DB1/Berkley DB MySQL add in Postgres Unix odbc Brian's dbodbc Beyond just having a relational DB or not. Performance? DB size? Ease of Access? Portability? Gui/browser access ... Any comments on how to bring this all into focus? ___ 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] Need to block incoming collect calls
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Saturday 24 July 2004 12:37 pm, Osvaldo Mundim Junior wrote: All right Steve. I'll ask them.. But if anybody knows that, please post an answer to the list. This is a very important Asterisk security configuration to avoid people call you without having to pay the call.. thank you Oz Actually there's a field that gives that data. I forget exactly, but it is a type of call field. I remembered it when you said identify collect-calls as I had done that on a different phone platform a few years ago. - -- Steve They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFBA9TgljK16xgETzkRAhD0AJ9rvI0s3TdpOu195Akdbni+UQ7YjQCfXp68 qWRo8HA2Xa0Gbsh2kz74hvk= =4p18 -END PGP SIGNATURE- ___ 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] which DB to use? or Why Berkley DB vs. MySQL, etc
On Sunday 25 July 2004 11:24, Frank wrote: What are the criteria you use to select which DB to use with * Built in DB1/Berkley DB MySQL add in Postgres Unix odbc Brian's dbodbc If you don't need the relational aspects I'd probably use DB1/Berkely -- it's small, robust and works. odbc shoudl work for anything else -- I very much doubt you'll ever run into performance issues with * and odbc, and by not locking yourself into a specific DB you can always upgrade/try others. My fanatical side would choose postgres. :-) -A. ___ 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: RES: [Asterisk-Users] Play CD!
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Saturday 24 July 2004 08:29 am, Jozeph Brasil wrote: I do that. But when I play MusicOnHold the music is played slowly! I don´t know why... but is how the bitrate is playing with a different number. You need to have a bitrate of 8000, and in mono. In this case sox is your friend. -Mensagem original- De: Chris Foster [mailto:[EMAIL PROTECTED] Enviada em: sábado, 24 de julho de 2004 01:37 Para: [EMAIL PROTECTED] Assunto: Re: [Asterisk-Users] Play CD! On Fri, 23 Jul 2004 23:29:19 -0300, Jozeph Brasil [EMAIL PROTECTED] wrote: Hi all, Is it possible to play a CD has MusicOnHold? Thanks, Jozeph Why don't you just rip the CD to MP3? ___ 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 - -- Steve They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFBA9/HljK16xgETzkRAn0KAJ9+N+09gOt55y0T/b9JbOf3kTbnfwCg42NP sJgL3I28ihz4G3lc7Ad4boQ= =zwLu -END PGP SIGNATURE- ___ 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] hang up when going to voicemail
Doh! The reason it changed when I upgraded is because I was compiling VM with Mysql, and I had the mailbox definitions in the voicemail.conf flat-file. I put the definition in the SQL database and it works fine, now. :-/ thanks for kicking me into the right direction :) yours, matthew Are you sure you have a mailbox for this number ? Umar -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Matthew Simpson Sent: 23 July 2004 16:34 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] hang up when going to voicemail I have a little menu set up where hitting 1, 2, or 3 places the call through to a cellular phone over IAX. That works. However, if caller hits 4 to go into voicemail, the system hangs up. Am I doing something wrong in the dial plan, or is this a CVS change? I had no trouble with this until I upgraded to about 07/21 CVS, and I'm on 07/23 [latest] now with same results. My dial plan: [txlink] exten = s,1,Answer exten = s,2,Background(/txlink/txlink-main) exten = 1,1,Dial(IAX2/:[EMAIL PROTECTED]/12149490280) exten = 1,2,Hangup exten = 2,1,Dial(IAX2/:[EMAIL PROTECTED]/14693373687) exten = 2,2,Hangup exten = 3,1,Dial(IAX2/:[EMAIL PROTECTED]/18174017579) exten = 3,2,Hangup exten = 4,1,VoiceMail(s2147649296) exten = 4,2,Hangup exten = t,1,Goto(txlink,s,2) exten = i,1,Playback(invalid) [didin] exten = 2147649296,1,Dial(SIP/2147649296,15) exten = 2147649296,2,Goto(txlink,s,1) exten = 2147649296,3,Hangup Here is console output: -- Executing Goto(SIP/2147649296-fb41, txlink|s|1) in new stack -- Goto (txlink,s,1) -- Executing Answer(SIP/2147649296-fb41, ) in new stack -- Executing BackGround(SIP/2147649296-fb41, /txlink/txlink-main) in new stack -- Playing '/txlink/txlink-main' (language 'en') == CDR updated on SIP/2147649296-fb41 -- Executing VoiceMail(SIP/2147649296-fb41, s2147649296) in new stack -- Executing Hangup(SIP/2147649296-fb41, ) in new stack == Spawn extension (txlink, 4, 2) exited non-zero on 'SIP/2147649296-fb41' ___ 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] hang up when going to voicemail
Very welcome, Glad to have helped. Umar -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Matthew Simpson Sent: 25 July 2004 17:46 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] hang up when going to voicemail Doh! The reason it changed when I upgraded is because I was compiling VM with Mysql, and I had the mailbox definitions in the voicemail.conf flat-file. I put the definition in the SQL database and it works fine, now. :-/ thanks for kicking me into the right direction :) yours, matthew Are you sure you have a mailbox for this number ? Umar -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Matthew Simpson Sent: 23 July 2004 16:34 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] hang up when going to voicemail I have a little menu set up where hitting 1, 2, or 3 places the call through to a cellular phone over IAX. That works. However, if caller hits 4 to go into voicemail, the system hangs up. Am I doing something wrong in the dial plan, or is this a CVS change? I had no trouble with this until I upgraded to about 07/21 CVS, and I'm on 07/23 [latest] now with same results. My dial plan: [txlink] exten = s,1,Answer exten = s,2,Background(/txlink/txlink-main) exten = 1,1,Dial(IAX2/:[EMAIL PROTECTED]/12149490280) exten = 1,2,Hangup exten = 2,1,Dial(IAX2/:[EMAIL PROTECTED]/14693373687) exten = 2,2,Hangup exten = 3,1,Dial(IAX2/:[EMAIL PROTECTED]/18174017579) exten = 3,2,Hangup exten = 4,1,VoiceMail(s2147649296) exten = 4,2,Hangup exten = t,1,Goto(txlink,s,2) exten = i,1,Playback(invalid) [didin] exten = 2147649296,1,Dial(SIP/2147649296,15) exten = 2147649296,2,Goto(txlink,s,1) exten = 2147649296,3,Hangup Here is console output: -- Executing Goto(SIP/2147649296-fb41, txlink|s|1) in new stack -- Goto (txlink,s,1) -- Executing Answer(SIP/2147649296-fb41, ) in new stack -- Executing BackGround(SIP/2147649296-fb41, /txlink/txlink-main) in new stack -- Playing '/txlink/txlink-main' (language 'en') == CDR updated on SIP/2147649296-fb41 -- Executing VoiceMail(SIP/2147649296-fb41, s2147649296) in new stack -- Executing Hangup(SIP/2147649296-fb41, ) in new stack == Spawn extension (txlink, 4, 2) exited non-zero on 'SIP/2147649296-fb41' ___ 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] trunk line usage report
I have googled and searched the wiki with no luck. Is there a way to get * to report on trunk line utilization. Like a busy report or a usage histogram?
[Asterisk-Users] RE: Layer 3 VPN Question
Hi, Please keep this discussion on-list. I did search the list 3 weeks ago on not much usable did show up. Here is my scenario, fyi. I have sip/iax phones registered on my * server. My ISP can also do A-Z termination and provide local did numbers and controls Qos via VLAN/CoS. If I use the right VLAN tag and CoS then traffic is prioritized over my normal internet traffic. I did buy a small switch/router (Micronet sp1678) which can set the VLAN tags and supports CoS per port but I still feel the solution is a bit clumsy. I would rather use my Linux (RH9) to do the same. I am pretty sure it can but I haven't had any time yet to investigate how, so let the list benefit from configuration experiences in this area. Freddi ___ 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] how do I play congestion tone when Zap channels are full?
I read the wiki and looked at the examples, but I'm
still having problems. I have a Digium 4 port card
with POTS lines plugged into all four ports. How do I
play the congestion tone the the caller when they try
and dial out but all the lines are in use?
should something like this work?
[dial-trunklocal]
; Local calls
ignorepat = 9
exten = _9NXX,1,Dial(${TRUNK1}/${EXTEN:1})
exten = _9NXX,2,Congestion
exten = _9NXX,3,Playtones(congestion)
exten = _9NXX,102,Busy
exten = _9NXX,103,Playtones(busy)
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[Asterisk-Users] Incoming SIP gateway context?
I just started service with Broadvoice.com and everything seems to work. However, apparently my understanding of incoming sip contexts is less then what I thought it was. Could someone point me in the right direction? (* on a public address, CVS-HEAD-07/12/04, C7960 phones) In my sip.conf I have: [general] port = 5060 bindaddr = 0.0.0.0 allow=ulaw tos=0x18 ;sets ip tos bits (=lowdelay and throughput) context = bogon-calls ; Send SIP callers that we don't know about here context=from-broadvoice register=303539:[EMAIL PROTECTED]/539 snip [broadvoice] ;this is referenced for outgoing calls to Broadvoice.com type=peer username=303539 snip The problem I'm having with understanding this is for incoming calls from broadvoice. If I remove the context=from-broadvoice from the above, incoming calls from broadvoice are dropped into the bogon-calls context (no service available message). I've tried several different approaches to define another context with type=user, but can never get test calls from broadvoice to be handled in anything other then the bogon-calls context. What am I missing? 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
[Asterisk-Users] Web Based Admin Interface
I am setting up an Asterix server and would like to know what you people use to administer, I would prefer a web based interface. Grateful for any suggestions, Thanks Benny ___ 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] which DB to use? or Why Berkley DB vs. MySQL, etc
On Sun, 25 Jul 2004 11:24:09 -0400, Frank [EMAIL PROTECTED] wrote: What are the criteria you use to select which DB to use with * Built in DB1/Berkley DB MySQL add in Postgres Unix odbc Brian's dbodbc Beyond just having a relational DB or not. Performance? DB size? Ease of Access? Portability? Gui/browser access IMHO, there's only two (maybe 3) worthwhile options there. One is BerkleyDB. It's simple, it works, no extra work on your part. If you're going to be in a very active environment with LOTS of DB access going on, you go with options 23. MySQL or PostgreSQL. With regards to Asterisk, I'd say they're both about equal. It comes down to personal preference for the most part. If you prefer (or are already running) MySQL, use that. If you prefer (or are already running) PostgreSQL, use that. Other than that, I don't think there are really all that many other options. ___ 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] Incoming SIP gateway context?
The problem I'm having with understanding this is for incoming calls from broadvoice. If I remove the context=from-broadvoice from the above, incoming calls from broadvoice are dropped into the bogon-calls context (no service available message). just add the context = from-broadvoice to the [broadvoice] section like this: [general] ... context = bogon-calls ; Send SIP callers that we don't know about here ... [broadvoice] type=friend username=303539 host=... context=from-broadvoice i also have a fromuser (value equals username), fromdomain (value equals host) and insecure=very entry in that section to direct incoming calls from sipgate to the right context. as there is no way (other than the originating host) to identify such calls the we context used there should be quite limited. I've tried several different approaches to define another context with type=user, but can never get test calls from broadvoice to be handled in anything other then the bogon-calls context. i use type=friend to handle incoming and outgoing connections in the same section, but you can also define one with type=peer and one with type=user. hope that helps, stefan ___ 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] X100P Inbound Issue
Hello, After much searching of voip-info.org and google, I'm finally giving in and asking the list. The setup I have is this:- Single X100P card in a Debian system Inbound/Outbound POTS line connects to the X100P Sipura 2000 and Budgetone 100 on the LAN 1 Cordless and one conventional phone connected to the sipura Account on Stanaphone.com for eitherbound SIP calls. (I have other SIPaccounts as well - all work flawlessly) I have a simple dialplan - an incoming call rings all phones and goes to voicemail if not answered. When I dial '8' followed by a number - the call routes out via Stanaphone fine. No issues. When I call the Stanaphone number - all phones ring as expected, I can answer the call and talk fine. no issues at all. When I dial '9' followed by a number - the call routes out via the POTS line just fine. No issues. However, inbound calls on the POTS line are the issue. When a call comes in, * detects it and starts ringing all of the extensions. However, when I pickup the extension - it gets immediately disconnected. Other SIP extensions keep ringing - and the caller still hears the ring tone. Caller hangs up - SIP extensions keep ringing. Phone I picket up I now return to the hook. * then 'calls me back' ! Does anybody have any idea what's going on? I have put some snippets from the configs below.. Any insight would be very much appreciated! Michael. EXAMPLE FROM: zapata.conf [channels] busydetect=1busycount=7callprogress=yesrelaxdtmf=yescallwaiting=yescallwaitingcallerid=yesthreewaycalling=yestransfer=yescancallforward=yesusecallerid=yesechocancel=yesechocancelwhenbridged=yesrxgain=0.0txgain=0.0group=1pickupgroup=1-4immediate=nocontext=from-bellsignalling=fxs_kscallerid=asreceivedchannel=1 EXAMPLE FROM: extensions.conf [from-bell]exten = _.,1,Dial(SIP/001SIP/002SIP/003,30,t)exten = _.,2,Answerexten = _.,3,Wait(1)exten = _.,4,Voicemail(u099)exten = h,1,Hangup EXAMPLE FROM: sip.conf [002] ; Line 1 on adaptertype=friendusername=002secret=some password host=dynamiccontext=extensionsmailbox=099incominglimlit=2canreinvite=no
[Asterisk-Users] Can not make progdocs
Not even sure how important this is considering the state of many of the online docs... I have doxygen installed as is noted for the requirements for 'make progdocs', but the make doesn't find dot. I have no idea where dot went, is or should have been... I am installing und Suse 9.0 and it's rough. If you forget something duringthe initial install, adding the package later doesn't seem to work. I have been through several installs getting things to work. This seems to be the last 'bug' up to now. Lyle ___ 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] Can not make progdocs
The 'dot' program comes with the GraphViz package. You can also edit the contrib/asterisk-ng-doxygen file and set HAVE_DOT = NO (the default is YES) James On Sun, 25 Jul 2004, Lyle Giese wrote: Not even sure how important this is considering the state of many of the online docs... I have doxygen installed as is noted for the requirements for 'make progdocs', but the make doesn't find dot. I have no idea where dot went, is or should have been... I am installing und Suse 9.0 and it's rough. If you forget something duringthe initial install, adding the package later doesn't seem to work. I have been through several installs getting things to work. This seems to be the last 'bug' up to now. ___ 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] pound key tone generated after call answered?
Hello, I've been working on an * dialer application, whereby a requirement is that if no one answers the call, a message must be left on voicemail. I've been using the record(tmp.gsm) function with silence detection enabled to wait for the greeting to finish before speaking. However, on voicemail systems where you can interrupt the greeting with a pound (#) key to access your voicemail (ie. verizon wireless), the call placed by asterisk appears to be immediately generating a tone (upon answer) that the voicemail system is interpreting as a pound key (#). I know this because if i listen to tmp.gsm, i hear the tail end of please enter your password, then press pound., and NOT hello, this is steve, please leave message as i expected. also of note is that this only happens when dialing using IAX2 with a call termination provider, and not with ZAP and a POTS line. also note that i've tried two different IAX2 providers with the same resuls. (using zap and pots lines is not preferrable in this case) one more note is that i've disabled the 'beep' before the recording starts (commented out lines in app_record.c and re-built code), just in case that was causing it -- no change in behavior, though. any thoughts on what could be causing this, or how to further troubleshoot? 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
[Asterisk-Users] Broadvoice problems again
I had my asterisk configuration working very well with broadvoice, but it stopped working this afternoon. I plugged the Cisco 7960 phone I used for my origional signup (just a few days before they offered generic BYOD) and it works fine. I did notice it seems to do all of its comunication through proxy.broadvoice.com (I used tcpdump). I have never contacted broadvoice about using asterisk (it seems their support is busy enough), and just downloaded the phones configuration file via tftp before the phone did. Is anyone else having this problem? I read the messages about the failure a few days ago (as I experienced it as well) where asterisk wasn't using the redundant servers. I did a tcpdump on my asterisk traffic and it apeared to be using both addresses for sip.broadvoice.com. I am running today's cvs head (but this failure was present before, but I didn't run tcpdump then). Thank you, Charlie ___ 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] Busydetect problems
Hi guys. I have a XP100P Clone , and the busydetect dont work for me.. PSTN---Asterisk---Sip---AsteriskPBX Any call from pstn side dont disconnect ... I have no disconnect supervision and busydetect dont work... Please Help me. Zapata.conf [channels]echocancel=yesusecallerid=nohidecallerid=norxgain=0.0txgain=0.0signalling=fxs_kscallprogress=nocontext=entradachannel=1musiconhold=defaultbusydetect=yesbusycount=7;echocancel=yes;usecallerid=yes;hidecallerid=no I try compile asterisk with BUSYDETECT_MARTIN and change dsp.c for my busy tone but no results... Sorry my bad english. Fabricio Chicon Pereira da Silva[EMAIL PROTECTED]http://www.freenetworks.com.br
[Asterisk-Users] Sergio, check your date...
It isn't August yet... At 05:43 PM 8/22/2004, you wrote: It's more easy download tarball and compile it. srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] En nombre de Yan Enviado el: jueves, 22 de julio de 2004 13:31 Para: [EMAIL PROTECTED] Asunto: [Asterisk-Users] debian install zaptel Hi: Did anyone use apt-get install zaptel successfully? After apt-get instal zaptel, use modprobe zaptel, get a FATAL modul zaptel not found. Thanks. Yan Best Regards, Karl J. Vesterling E-Mail: [EMAIL PROTECTED] Yahoo Messenger: karl_vesterling ICQ: 1548052 AOL Instant Messenger: n2vqm Telephone: Washington DC: (202) 448-3009 Extension 0 Annapolis MD: (240) 524-6706 Extension 0 Seattle WA: (360) 516-1822 Extension 0 Niagara Falls NY: (716) 286-9175 Extension 0 Buffalo NY: (716) 608-1121 Extension 0 United Kingdom: 0870 3403428 Extension 0
Re: [Asterisk-Users] MSSQL ODBC CDR
Yes, It is in CVS now. Duane Cox on 2004/7/23 09:44 wrote: Thanks, I _finally_ got unixODBC and FreeTDS working with MSSQL. I hate to through all that hard work out the door, but I like your idea better. Is it in cvs now, ready to go? I read that mark was waiting on a fix... ? Thanks for the link. Thanks Duane Cox -- Jim Kou Malico Inc. No.5, Ming-Lung Road, Yang-Mei, Tao-Yuang, Taiwan 32643 Tel : 886-3-472-8155#218 Fax : 886-3-472-5979 Site: http://www.malico.com.tw _/ _/ _/ _/ _/_/_/ _/_/_/ _/_/_/ _/_/ _/_/ _/ _/ _/ _/ _/ _/_/ _/ _/ _/ _/_/_/_/ _/ _/ _/ _/_/ _/ _/ _/_/ _/_/_/ _/_/_/ _/_/_/ _/_/_/ ___ 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] Broadvoice problems again
First, let me apologize for following up on my own message. I traced the connection the tftp configured 7960 phone was using and want to see how close I can make asterisk approximate this for the best reliability. The phone does an srv lookup for _sip._udp.proxy.broadvoice.com which returns proxy.lax.broadvoice.com and proxy.dca.broadvoice.com It then sends all requests to register for [EMAIL PROTECTED] via BOTH proxies. If anyone would find the packet dump usefully please contact me off list. Thank you, Charlie Hedlin Charlie Hedlin wrote: I had my asterisk configuration working very well with broadvoice, but it stopped working this afternoon. I plugged the Cisco 7960 phone I used for my origional signup (just a few days before they offered generic BYOD) and it works fine. I did notice it seems to do all of its comunication through proxy.broadvoice.com (I used tcpdump). I have never contacted broadvoice about using asterisk (it seems their support is busy enough), and just downloaded the phones configuration file via tftp before the phone did. Is anyone else having this problem? I read the messages about the failure a few days ago (as I experienced it as well) where asterisk wasn't using the redundant servers. I did a tcpdump on my asterisk traffic and it apeared to be using both addresses for sip.broadvoice.com. I am running today's cvs head (but this failure was present before, but I didn't run tcpdump then). Thank you, Charlie ___ 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] Broadvoice problems again
I had my asterisk configuration working very well with broadvoice, but it stopped working this afternoon. I plugged the Cisco 7960 phone I used for my origional signup (just a few days before they offered generic BYOD) and it works fine. I did notice it seems to do all of its comunication through proxy.broadvoice.com (I used tcpdump). I have never contacted broadvoice about using asterisk (it seems their support is busy enough), and just downloaded the phones configuration file via tftp before the phone did. Is anyone else having this problem? I just signed up with them yesterday and the config sheet they emailed me indicates the registrar IP is sip.broadvoice.com, and the proxy IP is proxy.broadvoice.com. However, for the last hour or so their site has been unreachable with an icmp destination unreachable coming from 199.232.42.62, which belongs to Cambridge Entrepreneurial Network in Quincy MA. Would guess either someone upgrading hardware or a failure near broadvoice. 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
Re: [Asterisk-Users] Broadvoice problems again
I am having issues also. I called them and I was told that they are doing upgrades on their network. Zac On Mon, 2004-07-26 at 04:55, Charlie Hedlin wrote: I had my asterisk configuration working very well with broadvoice, but it stopped working this afternoon. I plugged the Cisco 7960 phone I used for my origional signup (just a few days before they offered generic BYOD) and it works fine. I did notice it seems to do all of its comunication through proxy.broadvoice.com (I used tcpdump). I have never contacted broadvoice about using asterisk (it seems their support is busy enough), and just downloaded the phones configuration file via tftp before the phone did. Is anyone else having this problem? I read the messages about the failure a few days ago (as I experienced it as well) where asterisk wasn't using the redundant servers. I did a tcpdump on my asterisk traffic and it apeared to be using both addresses for sip.broadvoice.com. I am running today's cvs head (but this failure was present before, but I didn't run tcpdump then). Thank you, Charlie ___ 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 with T1 PRI Configuration
On Sun, 2004-07-25 at 04:56, Peter Svensson wrote: [snip] I'm not sure of the correct answers to any of these. I do know that I want to be able to get caller ID and the number dialed for the applicationthat I'm building. Most of these questions are for trunks being delivered over a T-1, not for PRI. If you are truly ordering PRI service (which is the more powerful of the two) then these questions don't apply :-) It does very much sound like they intend to deliver a non-pri T1 trunk. There are a lot of other knobs to fiddle with for a pri. :-) I'm digging out from under a weekend of email and maybe this has been covered already, but most of the time you are talking to a sales guy who has been given a check list and told to get these answers. It assumes that the installer of the PBX knows which ones to ignore for what is being ordered. Even when you graduate up to a entry level tech that proof reads the checklist to verify there is nothing in conflict, you may get asked similar questions again. The other answers where spot on. This hopefully will give you confidence in why so much was not relevant. -- Steven Critchfield [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] Asterisk CDR UniqueID
Hey All, We are running a small SIP/IAX termination service at the moment (planning on growing it) with 2 asterisk machines. One terminates the SIP/IAX calls from our customers and one is our gateway to our upstream provider. Both machines are logging CDR data to the same postgres table using the cdr_psql module. The problem I am having is I'm trying to work out how to link the CDR records into a single 'call stream', rather than having separate records per machine the call passes through. Reading around on the Wiki and doing a bit of googling, I've worked out that what I'm trying to do is the normalization step of CDR mediation, but I have not been able to find out any specifics about how to go about it. I would have thought that the originating asterisk machine would generate the UniqueID for the call (Message-ID in SMTP terms) and pass that along the call path, but each machine is using the epoch timestamp of when it sees (or records, not sure which) of the call. I know I can use the NoCDR app on the first machine in the chain, but that does not scale if we need to add more machines to our network. Does: a) anyone have any idea of what I'm trying to explain, and b) have any pointers of where I can find more information about doing this? Thanks Darryl -- If you want to live up to the whole There is more than one way to do it slogan, you have to give someone a swiss army chainsaw ... ___ 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] Broadvoice problems again
I called them also. They are upgrading sip.broadvoice.com I hope it works for them. They seem good so far. -- David Hickman Pots 314-865-4752 x1 business x31 home FWD 23633 IAXTEL 700-865-4752 AOLIM fsckrmrf ICQ 7059948 Yahoo dhickman THIS IS INSANE! I THOUGHT TECHNOLOGY WAS SUPPOSED TO SIMPLIFY MY LIFE!! On Jul 25, 2004, at 15:30, Zac Amsler wrote: I am having issues also. I called them and I was told that they are doing upgrades on their network. Zac On Mon, 2004-07-26 at 04:55, Charlie Hedlin wrote: I had my asterisk configuration working very well with broadvoice, but it stopped working this afternoon. I plugged the Cisco 7960 phone I used for my origional signup (just a few days before they offered generic BYOD) and it works fine. I did notice it seems to do all of its comunication through proxy.broadvoice.com (I used tcpdump). I have never contacted broadvoice about using asterisk (it seems their support is busy enough), and just downloaded the phones configuration file via tftp before the phone did. Is anyone else having this problem? I read the messages about the failure a few days ago (as I experienced it as well) where asterisk wasn't using the redundant servers. I did a tcpdump on my asterisk traffic and it apeared to be using both addresses for sip.broadvoice.com. I am running today's cvs head (but this failure was present before, but I didn't run tcpdump then). Thank you, Charlie ___ 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] how do I play congestion tone when Zap channels are full?
Hello Joe,
Monday, July 26, 2004, 2:33:07 AM, you wrote:
JB I read the wiki and looked at the examples, but I'm
JB still having problems. I have a Digium 4 port card
JB with POTS lines plugged into all four ports. How do I
JB play the congestion tone the the caller when they try
JB and dial out but all the lines are in use?
JB should something like this work?
JB [dial-trunklocal]
JB ; Local calls
ignorepat = 9
exten = _9NXX,1,Dial(${TRUNK1}/${EXTEN:1})
exten = _9NXX,2,Congestion
exten = _9NXX,3,Playtones(congestion)
exten = _9NXX,102,Busy
exten = _9NXX,103,Playtones(busy)
You had to Answer() call before trying to Playtones().
--
Best regards,
Olegmailto:[EMAIL PROTECTED]
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Re: [Asterisk-Users] Incoming SIP gateway context?
On Sun, 25 Jul 2004, Rich Adamson wrote: I just started service with Broadvoice.com and everything seems to work. However, apparently my understanding of incoming sip contexts is less then what I thought it was. Could someone point me in the right direction? (* on a public address, CVS-HEAD-07/12/04, C7960 phones) In my sip.conf I have: [general] port = 5060 bindaddr = 0.0.0.0 allow=ulaw tos=0x18 ;sets ip tos bits (=lowdelay and throughput) context = bogon-calls ; Send SIP callers that we don't know about here context=from-broadvoice register=303539:[EMAIL PROTECTED]/539 snip [broadvoice] ;this is referenced for outgoing calls to Broadvoice.com type=peer username=303539 snip this doesn't address your question (I think the other post did) but it anticipates your next question.. Add dtmfmode=general to BOTH the general and broadvoice contexts in sip.conf. Asterisk seems to make an incorrect assumption about dtmf with broadvoice (on calls inbound to your box, that is) unless you set it in the general section as well. 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] Broadvoice problems again
[EMAIL PROTECTED] (Rich Adamson) writes: However, for the last hour or so their site has been unreachable with an icmp destination unreachable coming from 199.232.42.62, which belongs to Cambridge Entrepreneurial Network in Quincy MA. Would guess either someone upgrading hardware or a failure near broadvoice. I am having the same problem with sip.broadvoice.com. My asterisk /var/log/asterisk/messages has over 1200 lines of gripes about them starting at Jul 25 16:49:40 (PDT) and continuing to the present. Does broadvoice have a status page somewhere with real info on it. (Like what is causing this extended outage?) Ob-asterisk. I should really see how hard it would be to hack asterisk to register with all addresses if a hostname has multiple aliases. It seems that some of these outages could be weathered if asterisk were to keep tabs on all the sip servers a provider offered, and then actively uses whichever one was up and had the lowest round-trip-delay. -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ openbsd amd64 http://www.wsrcc.com/wolfgang/ftp/asterisk-openbsd35.patch ___ 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
