Re: [Asterisk-Users] Simple question about SIP community
Something the user list in Microsoft Messenger. I was thinking on some sort of web page that can check the registration of the sip clients on the asterisk but want to know if already exist to avoid to reinvent the wheel. That is actually quite easy and there are some projects that achive this using the Manager API of Asterisk. One is Flash based, but very pretty. I also added rudimentary support for this in DeStar, it has to made nicer and more usable, but that is easy to do. Maybe you visit the page Software Addons on the www.voip-info.org WIKI. ___ 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 server keeps crashing
All, I am very new to pbx hardware and equipment and any help will be greatly appreciated. I am now the proud owner of a TDM422p and Iaxy/S100I. The server is running debian testing so I first installed the asterisk deb package. To get the zap modules, I compiled zaptel-1.0-RC2. After some configuration, everything worked as expected. Then the server crashed -- locked up hard. Nothing in the logs, completely frozen. This is a server that typically runs months on end with no problems. After repeated crashes, I tried different slots in the computer to put the TDM422p on different IRQ settings. I also upgraded to asterisk-1.0-RC2 (compiled from source). Still had lockups, but I couldn't get the board on its own irq. So I moved it to another machine where it could have its own irq. Now that machine locks up. The lockups always occur within 1-4 hours. When running, the pbx works just as intended, but the crashes are making the system unusable. I am pulling my hair out with this problem and my SO wants me to give up the project. Any and all help will be greatly appreciated! Thanks, David ___ 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 'Under the Hood' Project
Greetings All, I have a new post on the blog. It goes a little bit more in depth on wcfxo.c and touches on zaptel.c. Two more screen shots. Loads of fun. Take a look: http://zapteldoc.blogspot.com Regards, Victor ___ 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] weird routing(?) problem with 2 Asterisk servers
traceroute A - B: traceroute to 192.168.2.44 (192.168.2.44), 30 hops max, 38 byte packets 1 192.168.11.1 (192.168.11.1) 1.964 ms 1.181 ms 0.852 ms 2 10.138.3.2 (10.138.3.2) 43.428 ms 49.634 ms 47.601 ms 3 192.168.2.44 (192.168.2.44) 53.440 ms 49.320 ms 48.968 ms traceroute B - A: traceroute to 192.168.11.6 (192.168.11.6), 30 hops max, 40 byte packets 1 192.168.2.1 (192.168.2.1) 1.873 ms 1.861 ms 2.106 ms 2 10.138.3.3 (10.138.3.3) 45.356 ms 44.139 ms 44.884 ms 3 192.168.11.6 (192.168.11.6) 43.390 ms 43.736 ms 45.823 ms 10.138.3.2-10.138.3.3 is the PPTP connection between both systems. Should bindaddr (iax.conf) or externip (sip.conf) be defined for a setup like this one? Regards, Evert Do you know where it got the 10.138.3.2 IP from? Is it configured anywhere on the server? Do you have externip defined in that config file? Evert Meulie wrote: Hi everyone! situation: Asterisk-server A: 192.168.11.6 Asterisk-server B: 192.168.2.44 server B contains a register = username:[EMAIL PROTECTED] But... when I boot it, I get: Registered to '192.168.11.6', who sees us as 10.138.3.2:4569 Why doesn't server A see server B as 192.168.2.44?? All other traffic going over these lines has no problems with this. The 192.168.2.x 192.168.11.x networks are fully 'connected' to each other... Who knows the answer...? Regards, Evert Meulie ___ 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 GW and DTMF problems
[EMAIL PROTECTED] wrote: Problem was with asterisk.. Mark had made a change in chan_sip.c that affected noncodec capabilities, it's been fixed. Do you have a bug number? Or something else to find it in the bug database? -- Andreas SikkemaRits tele.com Scheepmakersstraat 11 3011 VH Rotterdam t: +31 (0)10 2245544f: +31 (0)10 2245540 ___ 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] Dialing Out through Provider with Authentication
On Thu, 9 Sep 2004, Karl Brose wrote: In order to dial out to a sip provider, you need to configure that provider in your sip.conf file as a peer with your proper username and secret, etc. Cool! Just found that in the handbook too a second or two ago :-) Thanks for taking time to answer this. Three Cheers! Gerald ___ 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] astwind has any one got this thing to work?
On Thu, 9 Sep 2004, hank smith wrote: I have a SiS 900 PCI Fast Ethernet Adapter what do I put in there or is that what I put in the xml file? Go read: http://www.colinux.org/wiki/index.php/coLinuxNetworking Specifically: If in doubt, the name of the card can be found in colinux-daemon startup log as follows: bridged-net-daemon: Checking adapter: NDIS 5.0 driver bridged-net-daemon: Checking adapter: TAP VPN Adapter. bridged-net-daemon: No matching adapter Error initializing winPCap The correct name here is NDIS 5.0 driver and not Karta Realtek RTL8139(A) PCI Fast Ethernet Adapter. It may help to use the default console, rather than the NT-Native (as the initial window has scrollback). I tried it with winpcap v 3.0 and 3.1beta. Currently works well with 3.1 beta Deja Vu.. Is there an echo in here? -- 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] astwind has any one got this thing to work?
On Thu, 9 Sep 2004, hank smith wrote: is there going to be a gui for co linux and astwind? No. AstWind is just a Debian GNU Linux distribution with a precompiled Asterisk installation running under a CoLinux kernel. I will have to see if either there is going to be a gui or if yasr a screen reader for the blind will work with this thing. I do not know. I would assume that a blind user would probably prefer a text based interface, but I have no clue. -- 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] Store data from call to database
--- hank smith [EMAIL PROTECTED] a écrit : when you get this up up can you give the phone number? Ok, I just start the project it's for a local televisoin in french polynésia TNTV. I hope that this project will be concretized. this sounds rather interesting, and fun!!! - Original Message - From: bagattin jerome [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, September 09, 2004 7:54 PM Subject: [Asterisk-Users] Store data from call to database Hi, I use asterisk for a phone quiz game. I need to store data in a database (MySql, postgres) : telephone number, name (voice), ... and of course the answers at the quetions. What's the best way to store my data ? - script with system() command ? - AGI script - CDR - others ... Thanks Jerome Vous manquez d'espace pour stocker vos mails ? Yahoo! Mail vous offre GRATUITEMENT 100 Mo ! Créez votre Yahoo! Mail sur http://fr.benefits.yahoo.com/ Le nouveau Yahoo! Messenger est arrivé ! Découvrez toutes les nouveautés pour dialoguer instantanément avec vos amis. A télécharger gratuitement sur http://fr.messenger.yahoo.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 My Inbox is protected by SPAMfighter 1045 spam mails have been blocked so far. Download free www.spamfighter.com today! ___ 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 Vous manquez despace pour stocker vos mails ? Yahoo! Mail vous offre GRATUITEMENT 100 Mo ! Créez votre Yahoo! Mail sur http://fr.benefits.yahoo.com/ Le nouveau Yahoo! Messenger est arrivé ! Découvrez toutes les nouveautés pour dialoguer instantanément avec vos amis. A télécharger gratuitement sur http://fr.messenger.yahoo.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] Store data from call to database
--- William Suffill [EMAIL PROTECTED] a écrit : Sounds like it be best as a custom app or AGI depending how many calls you will be taking and how bad the performance hit of using an AGI vs Compiled app is for your needs OK, I first try with AGI which sound like quicker to implement. And if I performance problems I will try a custom app. For the moment I start with T2 (30 calls simultaneus max) connection but it could be increase 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 Vous manquez despace pour stocker vos mails ? Yahoo! Mail vous offre GRATUITEMENT 100 Mo ! Créez votre Yahoo! Mail sur http://fr.benefits.yahoo.com/ Le nouveau Yahoo! Messenger est arrivé ! Découvrez toutes les nouveautés pour dialoguer instantanément avec vos amis. A télécharger gratuitement sur http://fr.messenger.yahoo.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] astwind has any one got this thing to work?
yasr is text based but the interesting part is going to see if it works running on a windows platform with this version of linux with out that I can't do anything with this so I will have to see. take care. hank - Original Message - From: Greg Boehnlein [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Friday, September 10, 2004 12:00 AM Subject: Re: [Asterisk-Users] astwind has any one got this thing to work? On Thu, 9 Sep 2004, hank smith wrote: is there going to be a gui for co linux and astwind? No. AstWind is just a Debian GNU Linux distribution with a precompiled Asterisk installation running under a CoLinux kernel. I will have to see if either there is going to be a gui or if yasr a screen reader for the blind will work with this thing. I do not know. I would assume that a blind user would probably prefer a text based interface, but I have no clue. -- 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 My Inbox is protected by SPAMfighter 1050 spam mails have been blocked so far. Download free www.spamfighter.com today! ___ 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 'Under the Hood' Project
On 9 Sep 2004 at 23:24, Victor Rini wrote: Greetings All, I have a new post on the blog. It goes a little bit more in depth on wcfxo.c and touches on zaptel.c. Two more screen shots. Loads of fun. Take a look: http://zapteldoc.blogspot.com Regards, Victor Keep up the good work! And sage for firefox reads your site feed great! Cheers, Matt Riddell http://www.sineapps.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] Checking Return Codes
On Tue, 2004-09-07 at 21:48, Glenn A. Thompson wrote: Hi, I must be blind, how does one check then act upon the return code from the previous command? For instance, Answer says it can return non zero. How do I check for that. It doesn't set any other variables like Dial does. Most commands return a 0 or non zero value, and jump to priority n+101 if the return value is non-zero. In this case, all I'm really trying to do is not answer if the line has already been picked up. Do I have to make sure the channel is available before I issue the Answer cmd. Thanks, Glenn ___ 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] Dialing Out through Provider with Authentication
Hi,
I tried to make a call to extension 2001 with the setting [EMAIL PROTECTED] (Detailed:
exten = _7.,2,Dial(SIP/[EMAIL PROTECTED]/${EXTEN:1})
which does not work at all - i always get the failure message: No such host provider.com/2001 (the number i dialed) - why ??
when i try the same with a peer agent (exten = _7.,2,Dial(SIP/provider_out/${EXTEN:1}) - i always get the failure message
WARNING[-178521168]: chan_sip.c:680 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Critical Request)
What am i missing ??
I am running out if ideas !.
Johannes
Am Fr, den 10.09.2004 schrieb Begumisa Gerald M um 12:04:
On Thu, 9 Sep 2004, Karl Brose wrote:
In order to dial out to a sip provider, you need to configure that
provider in your sip.conf file as a peer with your proper username
and secret, etc.
Cool! Just found that in the handbook too a second or two ago :-)
Thanks for taking time to answer this.
Three Cheers!
Gerald
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[Asterisk-Users] Legacy Toshiba Phones
Leo wrote: Not necessarily so. Recently I discovered that Artisoft's Televantage Soft PBX can support Toshiba Strata CS digital phones (DKT 2000 and 3000) through a PCI 16-port digital station card (Toshiba part #CS-DKTU-TV). Apparently, the Strata CS is an OEM licensed version of Televantage. It would be quite cool if an Asterisk driver can be developed for the 16-port digital station card. Interesting. I just checked out the Televantage site at TrueData (http://www.truedataonline.com). In their FAQ there's a question Can TV (Televantage) use Digital Sets?. The answer includes the tidbit: The Toshiba digital station cards are a slight variation of the Intel MSI160PCI and are interoperable with other Dialogic - Intel Televantage hardware. The good news is that Intel have Linux drivers for the MSI160. Guess someone needs to find some details on how the Tosh card differs ... The bad news is that the Tosh station card doesn't come cheap ... a quick google search shows prices around $2,500! For 16 ports? Ouch! At that price, it's cheaper to throw the Tosh phones away and buy IP hardphones. -- David Gurr Congruity Ltd. Hemel Hempstead UK ___ 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] astwind has any one got this thing to work?
it works it works it works! sorry it took it so long for the info to click thanks for the help guys!!! take care hank - Original Message - From: Greg Boehnlein [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Thursday, September 09, 2004 11:57 PM Subject: Re: [Asterisk-Users] astwind has any one got this thing to work? On Thu, 9 Sep 2004, hank smith wrote: I have a SiS 900 PCI Fast Ethernet Adapter what do I put in there or is that what I put in the xml file? Go read: http://www.colinux.org/wiki/index.php/coLinuxNetworking Specifically: If in doubt, the name of the card can be found in colinux-daemon startup log as follows: bridged-net-daemon: Checking adapter: NDIS 5.0 driver bridged-net-daemon: Checking adapter: TAP VPN Adapter. bridged-net-daemon: No matching adapter Error initializing winPCap The correct name here is NDIS 5.0 driver and not Karta Realtek RTL8139(A) PCI Fast Ethernet Adapter. It may help to use the default console, rather than the NT-Native (as the initial window has scrollback). I tried it with winpcap v 3.0 and 3.1beta. Currently works well with 3.1 beta Deja Vu.. Is there an echo in here? -- 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 My Inbox is protected by SPAMfighter 1053 spam mails have been blocked so far. Download free www.spamfighter.com today! ___ 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 testbed for teaching connecting to a PRI-ISDN
Hi, I would like to realize a voip testbed that should simulate the scenario in which two companies have an asterisk PBX connected through a PRI-ISDN to the Telco operator. I have no experience of T1/E1 connection but I think that the above could be relized with 3 asterisk boxes equipped with Digium TE405P cards. One of the box will represent the Telco, the other two, the two companies PBX. I would like to know if it is needed something between the point-point connections or it is possible to just cross-connect them. I need the testbed to be representative of the real-world difficulties in putting on an Asterisk BOX for connecting to a PRI-ISDN: is other hardware needed (e.g. echo cancellers or failover switches)? Asterisk BOX (Simulate the Telco) with Digium TE405P | \ | E1 \ T1 | \ [What to put here?] [What to put here?] | \ | E1 \ T1 | \ Asterisk BOX (Company) Asterisk BOX (Company 2) with Digium TE405P with Digium TE405P Regards, Francesco Delfino ___ 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 testbed for teaching connecting to a PRI-ISDN
You need an E1 back-to-back cable. Regards, antonio Francesco Delfino wrote: Hi, I would like to realize a voip testbed that should simulate the scenario in which two companies have an asterisk PBX connected through a PRI-ISDN to the Telco operator. I have no experience of T1/E1 connection but I think that the above could be relized with 3 asterisk boxes equipped with Digium TE405P cards. One of the box will represent the Telco, the other two, the two companies PBX. I would like to know if it is needed something between the point-point connections or it is possible to just cross-connect them. I need the testbed to be representative of the real-world difficulties in putting on an Asterisk BOX for connecting to a PRI-ISDN: is other hardware needed (e.g. echo cancellers or failover switches)? Asterisk BOX (Simulate the Telco) with Digium TE405P | \ | E1 \ T1 | \ [What to put here?] [What to put here?] | \ | E1 \ T1 | \ Asterisk BOX (Company) Asterisk BOX (Company 2) with Digium TE405P with Digium TE405P Regards, Francesco Delfino ___ 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] TDM400P lockups (FXO)
Maciej Kietlinski wrote: Are the FXOs on the 2x on ports 1-2 or 3-4? Maybe it has to do with *any* FXO on port 1... Please get back with the list with your findings. My experience led to a replacement from Digium, but the card is a TDM400P with 4 FXO...now that I think of it, during troubleshooting there was some correlation to the first port on the card (port 1)...not the first module - I swapped module positions to varying locations on the card without success, but then again they are all FXO...Maybe *is* possible that the TDM400P doesn't like an FXO module in port 1 as you are suggesting...Like I said, in the end I got a new revision board from digium, all 4 ports are still FXO and working great now... With my old revision TDM400P it was the same problem with FXO on port 1. Easiest way for me was to put FXO's on new revision card, and on old use FXS on port 1. I used info from post with: 'The card had been modified, evident from the jumper wire that been soldered between two points on the back of the card. I haven't had problems since installing the new card.' And before old card was used with FXS + 3 x FXO without problems, so it works in the same hw conf again. Now I heve no problems with TDMxxp I'll let you know how I get on. One of the cards that is giving trouble, has FXOs in positions 3 and 4. Can anyone tell me what these new revision cards are? My current ones are all Rev. E/F. Regards, Richard ___ 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] Snom 200 updates
I always just let the phone poll the Snom update server for updates but while the server is back at version 2.03o the latest stable downloadable version on the website is 2.04n.. Is Snom not distributing updates for the 200 from their server anymore?? ___ 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] Dialing Out through Provider with Authentication
On Fri, 10 Sep 2004, Johannes Hollerer wrote:
I tried to make a call to extension 2001 with the setting
[EMAIL PROTECTED] (Detailed: exten =
_7.,2,Dial(SIP/[EMAIL PROTECTED]/${EXTEN:1}) which does not work at
all - i always get the failure message: No such host
provider.com/2001 (the number i dialed) - why ??
What I understood from Karl's message is that you need to create a peer in
sip.conf. For example below:
-- sip.conf --
[myprovider]
type=peer
username=USERNAME
host=PROVIDER.COM
secret=SECRET
--
Then in extensions.conf, do the following:
--
exten = _7.,2,Dial(SIP/myprovider/${EXTEN:1})
--
This should work. What Karl meant is that using the statement below:
--
exten = _7.,2,Dial(SIP/[EMAIL PROTECTED])
--
Will only work if you are dialilng a *specific* extension on provider.com.
The statement below:
--
exten = _7.,2,Dial(SIP/[EMAIL PROTECTED]/${EXTEN:1})
--
Is illegal.
Cheers,
Gerald
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Re: [Asterisk-Users] Asterisk testbed for teaching connecting to a PRI-ISDN
On Fri, 10 Sep 2004, Francesco Delfino wrote: [...]One of the box will represent the Telco, the other two, the two companies PBX. I would like to know if it is needed something between the point-point connections or it is possible to just cross-connect them. As more experienced people prepare to reply, I'd like to give my [highly theoretical] opinion (I'm still waiting for hardware I ordered): I think it is possible to just cross connect them, as long as you get the signaling right. In my opinion, the Box simulating the telco should signal as the network side and the one representing the company should signal as the customer side... Hope that makes sense. Cheers, Gerald. ___ 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] Snom 200 updates
WipeOut schrieb: I always just let the phone poll the Snom update server for updates but while the server is back at version 2.03o the latest stable downloadable version on the website is 2.04n.. Is Snom not distributing updates for the 200 from their server anymore?? Have a look here: http://www.snom.com/download/share/ Regards Bastian ___ 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: Cisco GW and DTMF problems
Do you have a bug number? Or something else to find it in the bug database? bug #2394 Seems, the minor issue with Non-codec capabilities in sip debug still exists. Arsen. __ Do you Yahoo!? Yahoo! Mail - You care about security. So do we. http://promotions.yahoo.com/new_mail ___ 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] Netmeeting i can't hear voice
Hi. After a small war with underfined sybol error and conflicts between h323 and oh323 I successfully install h323 channel. Now, I can connect from Netmeeting to SIP and ZAP channels, but I can't here anything. When I call at phone, and try to speak, on another end of line man said, that my voice very low. Microphone volume is maximum... Is there some parameters like rxgain, txgain for h323. Or it is another problem? 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] Snom 200 updates
Bastian Schern wrote: WipeOut schrieb: I always just let the phone poll the Snom update server for updates but while the server is back at version 2.03o the latest stable downloadable version on the website is 2.04n.. Is Snom not distributing updates for the 200 from their server anymore?? Have a look here: http://www.snom.com/download/share/ Regards Bastian There are some really new versions there... So why is Snom not automatically distributing them anymore? I really liked it when the phone told me there was an update and I just had to press a button.. Now I have to go and find out if there is an update and then setup a server and load it myself.. :( ___ 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 and VoDSL
Hi, I'm new to telephony Software and Hardware, so please excuse my questioning. I plan to set up a little system, using Asterisk and VoDSL via Belcacom or Scarlet here in belgium. We are yust a little 2 man company and we are not always in our office. My idea is, to get VoDSL and set up a system that works as following: A customer sends SMS or phones to our office-numbers, if we are out, Asterisk checks the CallerID or if hidden, sends back an sms or says, that the customer should send or type in his client-number. Then Asterisk send an sms to the one, who is working on that customer, so that we can make call-backs if nesessary. Connecting Nagios to the same system, could also send sms, if something on our servers goes wrong. I also want the system to be able, that of course more than one caller can be handled at the same time. I know that Asterisk can handel this, but does sms via fixed line by Adrian Kennard work, using VoDSL?? And what other hardware do I need (except for the server with digium cards) to get this idea running?? As externals I'm planning to use 1 analog phone and maybe 2 headstets for each of us if we are in office. thx, Sascha ___ 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 and VoDSL
[EMAIL PROTECTED] wrote: Hi, I'm new to telephony Software and Hardware, so please excuse my questioning. I plan to set up a little system, using Asterisk and VoDSL via Belcacom or Scarlet here in belgium. We are yust a little 2 man company and we are not always in our office. My idea is, to get VoDSL and set up a system that works as following: A customer sends SMS or phones to our office-numbers, if we are out, Asterisk checks the CallerID or if hidden, sends back an sms or says, that the customer should send or type in his client-number. Then Asterisk send an sms to the one, who is working on that customer, so that we can make call-backs if nesessary. Connecting Nagios to the same system, could also send sms, if something on our servers goes wrong. I also want the system to be able, that of course more than one caller can be handled at the same time. I know that Asterisk can handel this, but does sms via fixed line by Adrian Kennard work, using VoDSL?? And what other hardware do I need (except for the server with digium cards) to get this idea running?? You are on the right track to get this working... Use provider who can offer you SMS service for more then one user You will need a custom AGI to inform account manager and other functions required. As for sending, SMS messages accross * server, I think SMS application does that. 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] Zaptel 'Under the Hood' Project
Hey Victor, that's really lot of fun! I'm anxious for the next chapters! Renato On Fri, 10 Sep 2004 19:27:24 +1200, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: On 9 Sep 2004 at 23:24, Victor Rini wrote: Greetings All, I have a new post on the blog. It goes a little bit more in depth on wcfxo.c and touches on zaptel.c. Two more screen shots. Loads of fun. Take a look: http://zapteldoc.blogspot.com Regards, Victor Keep up the good work! And sage for firefox reads your site feed great! Cheers, Matt Riddell http://www.sineapps.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 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] Chan zap not loaded(ast_pickup_call)
Hi all, I'm sorry, but I'm stupid because I haven't load res_parking.so. Regards, srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Sergio Serrano Enviado el: viernes, 10 de septiembre de 2004 9:35 Para: 'Asterisk Users Mailing List - Non-Commercial Discussion' Asunto: [Asterisk-Users] Chan zap not loaded(ast_pickup_call) Hi all I have installed an E100P. I have loaded zaptel and wct1xxp. My zaptel.conf is the next: span=1,1,0,ccs,hdb3,crc4,yellow bchan=1-15,17-31 dchan=16 loadzone=es defaultzone=es My zapata.conf is the next: [channels] switchtype = euroisdn language=es signalling = pri_cpe pridialplan = local prilocaldialplan = local echocancel = yes context = default group=1 channel = 1-15,17-31 When I start asterisk it says: [chan_zap.so]Sep 10 09:22:09 WARNING[1076253312]: loader.c:242 ast_load_resource: /usr/lib/asterisk/modules/chan_zap.so: undefined symbol: ast_pickup_call Sep 10 09:22:09 WARNING[1076253312]: loader.c:374 load_modules: Loading module chan_zap.so failed! Any idea? Regards, srsergio ___ 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] TDM400P lockups (FXO)
Are the FXOs on the 2x on ports 1-2 or 3-4? Maybe it has to do with *any* FXO on port 1... Please get back with the list with your findings. My experience led to a replacement from Digium, but the card is a TDM400P with 4 FXO...now that I think of it, during troubleshooting there was some correlation to the first port on the card (port 1)...not the first module - I swapped module positions to varying locations on the card without success, but then again they are all FXO...Maybe *is* possible that the TDM400P doesn't like an FXO module in port 1 as you are suggesting...Like I said, in the end I got a new revision board from digium, all 4 ports are still FXO and working great now... If memory serves correctly, there was a problem using an fxo module on port 1 that was diagnosed roughly thirty days after the initial tdm card was released. Seems to me that someone indicated it was a design problem with the card (apparently missing one circuit board trace or something like that), and that digium was replacing the cards for those that had the problem. The replacement tdm card had an extra wire jumper installed on it. The short-term fix (back then) was to move the fxo module to another position (if possible). Support should know all about that. 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] sip.conf from mysql
Helloall! I am trying to load sip.conf from mysql database. I have followed the instructions at http://www.voip-info.org/wiki-Asterisk+sip+mysql+peers. Seems that the authentication (user psw) works fine but I would like to get more information from mysql and I don't know how to retrieve it. Couldanybody help me? Any idea about how to do it? Regards, Victor. ___ 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] Chan zap not loaded(ast_pickup_call)
I'm sorry, but I'm stupid because I haven't load res_parking.so. And you reply to a different discussion thread. Don't use reply if you don't want to reply, create a new message instead. Hint: res_parking was renamed into res_features Hint2: get rid of /usr/lib/asterisk/modules and do a fresh installation ___ 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
On 09/09/2004 at 18:48 Josh Roberson wrote: I wrote cepstral regarding this at the beginning of the week, thought it might be relevant to post the reply: Thanks for contacting us. Our Linux package is off the site right now because we are releasing a new version, 3.02, next week. This is an incremental release. The major update of this version is a new Linux SDK. Please check back with us in 6-7 days and we should have what you're looking for. We appreciate your patience. -Craig Now hopefully, they'll hold up to it and release the new Linux SDK in a week or so... -twisted This *may* be related to my original app_cepstral that can't be integrated into CVS because of the licencing. bkw had a chat with them, iirc about making parts gpl, to solve this 'issue'.. perhaps they've done it (are doing it)... only time will tell Andy ___ 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] iaxy vs sipura
On 07/09/2004 at 23:57 Benjamin on Asterisk Mailing Lists wrote: On Tue, 07 Sep 2004 08:14:57 -0500, Brian Capouch [EMAIL PROTECTED] wrote: If you have a Linux laptop with you, then in fact the SIP devices can be configured to hide behind it. The laptop can then run an instance of asterisk that connects to the home asterisk server, Like I said: I run Asterisk on my Powerbook to do IAX to my company's Asterisk server. Keep in mind though that you don't need to have a Linux notebook to do this. A Powerbook running MacOSX runs Asterisk just fine. This may not be much of an issue for the Linux geeks and techies on the list, but if you have to send sales people and other non-tech folks on business trips and give them something to connect, then probably a Powerbook running OSX will be an easier choice since they get to keep their native MS-Office. At the risk of stating the obvious if you have a laptop not running MacOSX (ie perhaps running windows) download my asterisk live! cd ( http://www.automated.it/asterisk/ ), burn it and test it on your laptop and bung it in your laptop case along with your iaxy/sipura/whatever and errm... problem solved.. :D Andy ___ 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] Problems with 0penh323 Channel Driver
Hi, I have asterisk,openh323-v1_13_5 and pwlib-v1_6_6 installed on my PC. each time i run asterisk -c, i get the following error: [chan_oh323.so] = (OpenH323 Channel Driver) == Parsing '/etc/asterisk/rtp.conf': Found == Parsing '/etc/asterisk/oh323.conf': Found [1]WrapH323EndPoint::WrapH323EndPoint: Compile-time libraries OpenH323 v1.13.5, PWlib v1.6.6 segmentation error [EMAIL PROTECTED] root]# Can you help me? AFRIPA TELECOM, Africa Switch On ___ 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] No DTMF or Audio
I have built latest Asterisk w/ OpenH323 channel driver. We have a SIP softphone registered to the Asterisk. We can place outbound calls from the SIP phone to the PSTN via OpenH323 connection to our gatekeeper. Everything works okay - DTMF and Audio... But in the reverse - if we call from a cellphone or landline the PSTN number we can get the SIP phone to ring - we answer and can hear the originating party - but the SIP softphone is not able to transmit DTMF or audio back to the PSTN... I'm not sure if this is an issue w/ converting the signal in asterisk i.e. SIP to H323 -- or if a problem in the codec or what? The codec is G711uLaw.. Help - thanks Robert A. Huddleston, KF4BYY Cavalier Telephone LLC. 804.422.4401 [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] No DTMF or Audio
Have you configured; _ sip.conf_ ..add this line: dtmfmode=inband ..also you have uncomment the right line that matches your dhcp setup: localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks ;localnet=10.0.0.0/255.0.0.0 ; Also RFC1918;localnet=172.16.0.0/12 ; Another RFC1918 with CIDR notation;localnet=169.254.0.0/255.255.0.0 ;Zero conf local network Worked for me ;) / Stig Henning - Original Message - From: Huddleston, Robert To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Cc: Patterson, Mike Sent: Friday, September 10, 2004 2:32 PM Subject: [Asterisk-Users] No DTMF or Audio I have built latest Asterisk w/ OpenH323 channel driver. We have a SIP softphone registered to the Asterisk. We can place outbound calls from the SIP phone to the PSTN via OpenH323 connection to our gatekeeper. Everything works okay - DTMF and Audio... But in the reverse - if we call from a cellphone or landline the PSTN number we can get the SIP phone to ring - we answer and can hear the originating party - but the SIP softphone is not able to transmit DTMF or audio back to the PSTN... I'm not sure if this is an issue w/ converting the signal in asterisk i.e. SIP to H323 -- or if a problem in the codec or what? The codec is G711uLaw.. Help - thanks Robert A. Huddleston, KF4BYY Cavalier Telephone LLC. 804.422.4401 [EMAIL PROTECTED] ___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo 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 and VoDSL - Email found in subject
AFAIK, one can just send SMS via smsbug.com, but I want to be able to receive sms, without using an external sms-gateway wich should work with the sms-applikation if sms-ing is supported by VoDSL. Greetings, Sascha Am Fr, den 10.09.2004 schrieb Thorsten Neumann um 14:35: I have come across an sms platform that i am connecting to. They are http://www.smsbug.com and have very low prices (euro 0.03 per message). i am trying to integrate an AGI script to connect to their SOAP webservice (http://www.smsbug.com/api/sms.asmx). my efforts are still experimental but might allow me to use caller ID related information for e.g. pin codes, service request confirmations etc. my 2 cents /tozzi On Fri, 2004-09-10 at 12:55, Senad Jordanovic wrote: [EMAIL PROTECTED] wrote: Hi, I'm new to telephony Software and Hardware, so please excuse my questioning. I plan to set up a little system, using Asterisk and VoDSL via Belcacom or Scarlet here in belgium. We are yust a little 2 man company and we are not always in our office. My idea is, to get VoDSL and set up a system that works as following: A customer sends SMS or phones to our office-numbers, if we are out, Asterisk checks the CallerID or if hidden, sends back an sms or says, that the customer should send or type in his client-number. Then Asterisk send an sms to the one, who is working on that customer, so that we can make call-backs if nesessary. Connecting Nagios to the same system, could also send sms, if something on our servers goes wrong. I also want the system to be able, that of course more than one caller can be handled at the same time. I know that Asterisk can handel this, but does sms via fixed line by Adrian Kennard work, using VoDSL?? And what other hardware do I need (except for the server with digium cards) to get this idea running?? You are on the right track to get this working... Use provider who can offer you SMS service for more then one user You will need a custom AGI to inform account manager and other functions required. As for sending, SMS messages accross * server, I think SMS application does that. 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 ___ 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] Conference Phone
We use a nice Polycom conference phone and plugged it into the Sipura and it works crystal clear. Was cheaper than Polycom's conference phone w/ built in VOIP capabilities. Joe Dennick wrote: If it were me; I'd opt for one of the Polycom Conference phones (they are just regular analog phones), and use an FXS card to connect it to Asterisk. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chad Brown Sent: Thursday, September 09, 2004 4:13 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Conference Phone Any advice on a good conference phone that works with Asterisk? I like the Cisco line and was wondering if anyone has used the 7935 or 7936 phones. From what I can tell they dont have a sip load. Has anyone verified this or gotten an ETA from Cisco? Chad --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.745 / Virus Database: 497 - Release Date: 8/27/2004 ___ 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] Asterisk newbie questions
Hi everyone. I'm a bit of a Linux newbie, but I've been doing tech stuff for ages. I'm also brand new to *. I've been reading the Voip.org wiki, and perusing the list archives for a while since I've been asked to investigate using IP telephone / soft phones for a call-center type scenario. People (marketing folks) have pointed me at Cisco, but I really don't wanna. I'd rather be the hero and pull this off with a much smaller budget. Here is a scenario - 40 person call center, all with PC's (windows) and soft-phone. -any recommendations on hardware to run *? soft phones? 90% of calls would be IP / IAX coming to the center. I read in the list archives about an ACD application / extension to * that would probably to what I need in that regard. - thoughts? In remote locations I would also run *, and hook it up to an extension on an existing PBX. Excuse the complete newbie question, but how many 'wires' do I need to bring between the PBX and the * box to support multiple simultaneous calls? These calls would come from any extension on the TDM pbx to asterisk to the call center. In a typical scenario there would NOT be a lot of simultaneous calls unless the system we're supporting went down hard. How would / could? one configure * at the remote location to communicate with * at the call center? How would / could? one configure * at the remote location to use the existing TDM PBX as failover to call the support center via 1-800 if the IP circuit died? I know you're all banging your heads on your desks saying OY! another newbie. Thanks in advance for your wisdom and guidance. John ___ 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 question about SIP community
Have you had chance to look at Jeff Pulver's Communicator? This is a soft-phone, currently in beta, that allows you to bring together your contacts from MSN, ICQ, AOL and, importantly from your point of view, add contacts that are SIP users. I've not tried it yet with asterisk, but now you have asked the question, I'll try it out... It certainly detects FWD presence so I think it might work with Asterisk. If it doesn't I'll ask put it forward as a suggestion. Bill Seddon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scott Laird Sent: September 09, 2004 8:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Simple question about SIP community On Sep 9, 2004, at 8:53 AM, Marcello Lupo wrote: we have a community of people on an * box that use SIP softphones to talk each other. Can you suggest me the quickest and simple way to let someone know who is online without have to call one by one the persons to look if they are present or not?? Something the user list in Microsoft Messenger. I was thinking on some sort of web page that can check the registration of the sip clients on the asterisk but want to know if already exist to avoid to reinvent the wheel. thanks, The generic term for this is 'presence'. Everyone seems to agree that it's important, but I'm not aware of anyone actively working on it for Asterisk. Scott ___ 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] iaxy vs sipura
On Fri, 10 Sep 2004 14:05:09 +0200, Andy Powell [EMAIL PROTECTED] wrote: At the risk of stating the obvious if you have a laptop not running MacOSX (ie perhaps running windows) download my asterisk live! cd ( http://www.automated.it/asterisk/ ), burn it and test it on your laptop and bung it in your laptop case along with your iaxy/sipura/whatever and errm... problem solved.. :D Certainly an option, but most business folks will want to have their Outlook contacts and Excel spreadsheets in front of them when they are on the phone. Dual boot environments are not ideal in those situations. Imagine you're talking to some guy on the phone about prices and he tells you I cant' tell you what the discounts are right now because I would have to shut down the phone system to open Excel. However, you could use VMware on an Intel notebook to run both Windoze and Linux concurrently. This wouldn't be ideal for a real PBX for performance reasons, but since all you are going to use Asterisk for is to be a gateway for one single user, it's probably ok in this particular scenario. I remember there was a guy in Romania who reported he had VMware with Windoze and Asterisk on Linux running as a home PBX on his PC and it seemed to be alright. If you'd combine such a setup with a Windoze GUI tool that will start and stop the Linux environment and Asterisk at the push of a button, then you'd have a fairly convenient and workable SIP/IAX gateway solution for travelling biz folks. rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed. ___ 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] astwind has any one got this thing to work?
Try this :) ?xml version=1.0 encoding=UTF-8? colinux !-- This line needs to point to your root file system. For example change root_fs to the name of the Debian image. Inside coLinux it will be /dev/cobd0 -- block_device index=0 path=\DosDevices\c:\program files\coLinux\astwind-root-debian.fs enabled=true / !-- This line can specify a swap file if you wish, or an additional image file, it will /dev/cobd1. Additional block_devices can be specified in the same manner by increasing the index -- block_device index=1 path=\DosDevices\c:\program files\coLinux\swap_device enabled=true / !-- bootparams allows you to pass kernel boot parameters -- bootparamsroot=/dev/cobd0/bootparams !-- image allows you to specify the kernel to boot -- image path=vmlinux / !-- this line allows you to specify the amount of memory available to coLinux -- memory size=64 / !-- This allows you to modify networking parameters, see the README or website for more information -- network index=0 name=SiS NIC SISNIC type=bridged / /colinux -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of hank smith Sent: Friday, September 10, 2004 1:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] astwind has any one got this thing to work? I have a SiS 900 PCI Fast Ethernet Adapter what do I put in there or is that what I put in the xml file? - Original Message - From: Greg Boehnlein [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Thursday, September 09, 2004 9:03 PM Subject: RE: [Asterisk-Users] astwind has any one got this thing to work? On Wed, 8 Sep 2004, Chris HARIGA wrote: I make it work!! My Astwind is up and running! Now is 11:53 PM and I'm going to bed. Tomorrow morning I will post how I fix the Ethernet connection. I bet you followed the following directions! ;) From: http://www.colinux.org/wiki/index.php/coLinuxNetworking If in doubt, the name of the card can be found in colinux-daemon startup log as follows: bridged-net-daemon: Checking adapter: NDIS 5.0 driver bridged-net-daemon: Checking adapter: TAP VPN Adapter. bridged-net-daemon: No matching adapter Error initializing winPCap The correct name here is NDIS 5.0 driver and not Karta Realtek RTL8139(A) PCI Fast Ethernet Adapter. It may help to use the default console, rather than the NT-Native (as the initial window has scrollback). I tried it with winpcap v 3.0 and 3.1beta. Currently works well with 3.1 beta -- 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 My Inbox is protected by SPAMfighter 1047 spam mails have been blocked so far. Download free www.spamfighter.com today! ___ 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 smime.p7s Description: S/MIME cryptographic 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] iaxy vs sipura
I run Asterisk on Redhat 8.0 with a VM hosted by Microsoft's Virtual PC which, in turn, runs on Windows 2000 Server. Works like a charm. Can't use Zaptel cards but that's OK for me. I can put it into standby any time and it takes only a few seconds to start up the VM from its saved state and at that time the Linux session (and Asterisk) is available once again. Bill Seddon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Benjamin on Asterisk Mailing Lists Sent: September 10, 2004 2:03 PM To: Andy Powell Cc: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] iaxy vs sipura On Fri, 10 Sep 2004 14:05:09 +0200, Andy Powell [EMAIL PROTECTED] wrote: At the risk of stating the obvious if you have a laptop not running MacOSX (ie perhaps running windows) download my asterisk live! cd ( http://www.automated.it/asterisk/ ), burn it and test it on your laptop and bung it in your laptop case along with your iaxy/sipura/whatever and errm... problem solved.. :D Certainly an option, but most business folks will want to have their Outlook contacts and Excel spreadsheets in front of them when they are on the phone. Dual boot environments are not ideal in those situations. Imagine you're talking to some guy on the phone about prices and he tells you I cant' tell you what the discounts are right now because I would have to shut down the phone system to open Excel. However, you could use VMware on an Intel notebook to run both Windoze and Linux concurrently. This wouldn't be ideal for a real PBX for performance reasons, but since all you are going to use Asterisk for is to be a gateway for one single user, it's probably ok in this particular scenario. I remember there was a guy in Romania who reported he had VMware with Windoze and Asterisk on Linux running as a home PBX on his PC and it seemed to be alright. If you'd combine such a setup with a Windoze GUI tool that will start and stop the Linux environment and Asterisk at the push of a button, then you'd have a fairly convenient and workable SIP/IAX gateway solution for travelling biz folks. rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed. ___ 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] Problems with 0penh323 Channel Driver
[EMAIL PROTECTED] wrote: Hi, I have asterisk,openh323-v1_13_5 and pwlib-v1_6_6 installed on my PC. each time i run asterisk -c, i get the following error: [chan_oh323.so] = (OpenH323 Channel Driver) == Parsing '/etc/asterisk/rtp.conf': Found == Parsing '/etc/asterisk/oh323.conf': Found [1]WrapH323EndPoint::WrapH323EndPoint: Compile-time libraries OpenH323 v1.13.5, PWlib v1.6.6 segmentation error [EMAIL PROTECTED] root]# Can you help me? What versions of Asterisk, asterissk-oh323 do you use? What is the current configuration of oh323? Can you send the backtrace of the core file dumped? 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
RE: [Asterisk-Users] Digium E100P and PMX in Germany
What alarm is it. Is it red or is it yellow. If it's red then it's the /etc/zaptel config But if it's yellow then it's a problem with sync the channels Which could be a master - slave problem. Very easy to fix. Sean -Original Message- From: Jan Goericke [mailto:[EMAIL PROTECTED] Sent: 03 September 2004 12:42 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Digium E100P and PMX in Germany Thanks for the hint. I did it and zap show channels shows me the 31 channel. But when I check /proc/zaptel/1, i still get the same error as before. On Fri, 3 Sep 2004, Steven Critchfield wrote: On Fri, 2004-09-03 at 05:31, Jan Goericke wrote: Hello ml, i need some help on my zaptel configuration. My E100P only shows some YELLOW / RED alarm when I load the wct1xxp module and do a cat /proc/zaptel/1 Span 1: WCT1/0 Digium Wildcard E100P E1/PRA Card 0 HDB3/CCS YELLOW RED ... .. . My /etc/zaptel.conf is: span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 loadzone=nl defaultzone=nl I tried zaptel-1.0RC2 and the latest CVS version too. So I think it is a configuration problem. Can anyone give me a hint how to configure my E100P? Next step is to start asterisk so libpri attaches to your line and brings up the D channel. -- 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 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 newbie questions
Hi John, I'm also new to *, but if you want to set up a callcenter, with 40 people calling the same number at the same time, you probalbly will need a T-1 or E1 line wich AFAIK handles at least 30-calls. You then need at least one Digium E1/T1 card to get the calls into * and other cards to direkt them from * to the phones. I'm researching at this time on what is possible using VoDSL, but I don't dare to say that this might be an alternative for I don't know how many calls can be handled at the same time. But it would be a lot more cost effective than a E1-line here in Belgium. Greetings, Sascha By the way, me Oy! too Am Fr, den 10.09.2004 schrieb John Stegenga um 14:38: Hi everyone. I'm a bit of a Linux newbie, but I've been doing tech stuff for ages. I'm also brand new to *. I've been reading the Voip.org wiki, and perusing the list archives for a while since I've been asked to investigate using IP telephone / soft phones for a call-center type scenario. People (marketing folks) have pointed me at Cisco, but I really don't wanna. I'd rather be the hero and pull this off with a much smaller budget. Here is a scenario - 40 person call center, all with PC's (windows) and soft-phone. -any recommendations on hardware to run *? soft phones? 90% of calls would be IP / IAX coming to the center. I read in the list archives about an ACD application / extension to * that would probably to what I need in that regard. - thoughts? In remote locations I would also run *, and hook it up to an extension on an existing PBX. Excuse the complete newbie question, but how many 'wires' do I need to bring between the PBX and the * box to support multiple simultaneous calls? These calls would come from any extension on the TDM pbx to asterisk to the call center. In a typical scenario there would NOT be a lot of simultaneous calls unless the system we're supporting went down hard. How would / could? one configure * at the remote location to communicate with * at the call center? How would / could? one configure * at the remote location to use the existing TDM PBX as failover to call the support center via 1-800 if the IP circuit died? I know you're all banging your heads on your desks saying OY! another newbie. Thanks in advance for your wisdom and guidance. John ___ 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] pridialplan nationalprefix
For whom which may be interested: Here in Italy we have GSM #numbers without leading zero PSTN instead has prefix starting with '0' to have '0' recognized by * i need to insert nationalprefix=0 as Jason Williams suggested me in irc; now, you cannot have: pridialplan=natonal otherwise * will not be able to call GSM phones you need to setup: pridialplan=local prilocaldialplan=local nationalprefix=0 Maurizio ___ 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] Conference Phone
what phone did you purchase and how much - Original Message - From: Deon Rodden [EMAIL PROTECTED] To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Friday, September 10, 2004 5:59 AM Subject: Re: [Asterisk-Users] Conference Phone We use a nice Polycom conference phone and plugged it into the Sipura and it works crystal clear. Was cheaper than Polycom's conference phone w/ built in VOIP capabilities. Joe Dennick wrote: If it were me; I'd opt for one of the Polycom Conference phones (they are just regular analog phones), and use an FXS card to connect it to Asterisk. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chad Brown Sent: Thursday, September 09, 2004 4:13 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Conference Phone Any advice on a good conference phone that works with Asterisk? I like the Cisco line and was wondering if anyone has used the 7935 or 7936 phones. From what I can tell they dont have a sip load. Has anyone verified this or gotten an ETA from Cisco? Chad --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.745 / Virus Database: 497 - Release Date: 8/27/2004 ___ 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 Inbox is protected by SPAMfighter 1062 spam mails have been blocked so far. Download free www.spamfighter.com today! ___ 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] Call Parking Problem
Hi,
I'm unable to pick up parked calls after they are transfered.
I get the transfer message when I press # and then I'm told 701 The
extension I'm dialing goes to the on hold music. I'm disconnected, I hang
up, dial 701 and I see this message on the console Everyone is
busy/congested at this time
I just have the default parkedcalls file, and have this in the extensions.
[AnalogPhone]
exten = _70X,1,Dial(Zap/1/${EXTEN},20,Ttr)
include = parkedcalls
[SipPhone]
exten = _70X,1,Dial(SIP/1/${EXTEN},20,Ttr)
include = parkedcalls
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Re: [Asterisk-Users] Re: Caller-ID name lookup via anywho.com
Can anyone who is using this, give me an idea of performance impact of using this? Thank You, Matt Pusateri On Thu, 9 Sep 2004 20:00:02 -0500 (CDT), Lenny Tropiano / asterisk.org Mailing list [EMAIL PROTECTED] wrote: Did I see something on here about using an AGI script to do reverse lookups via anywho.com? I have a PRI that only gets caller-id number and no Alpha. [...] I put a copy of it here... http://www.voiping.com/calleridnamelookup.agi It was written by James Golovich [EMAIL PROTECTED] and requires the Asterisk::AGI perl bindings, but works... ___ 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] Caller id and the number of rings
Thanks. It seems like I do not have much of a choice left. Anyway, I just found out that if the usecallerid=no in zapata.conf. Asterisk does not wait for 2 rings before processing the call. On Thu, 2004-09-09 at 06:38, HengWee Chin wrote: I am wondering if there is any way or settings I can set to allow the caller id to pass thro' asterisk and let the IVR pickup the caller id information. This means that asterisk do not wait for 2 rings to process the call. Any ideas? Easy. Stop using analog interfaces. CLID on digital (Feature Group D and PRI) interfaces are done totally differently and you do not need to wait for the 2 rings for CLID. --Eric -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. _ Find it on the web with MSN Search. http://search.msn.com.sg/ ___ 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] Call Parking Problem
The 'parkedcalls' code dynamically creates and deletes entries in the
dialplan to handle the calls that have been parked, so the parking lot must
not overlap your regular extensions. The initial parking extension is
statically created on startup, thus the 'exten =' entry is matching the
parking slot digits and throwing the congestion error - remove it or edit
/etc/asterisk/features.conf to move the parking lot.
Either way, there is a fair amount missing from the extensions.conf you
posted. I suggest you troll through http://www.voip-info.org
-Original Message-
From: PHP Mechanic [mailto:[EMAIL PROTECTED]
Sent: September 10, 2004 7:27 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Call Parking Problem
Hi,
I'm unable to pick up parked calls after they are transfered.
I get the transfer message when I press # and then I'm told
701 The
extension I'm dialing goes to the on hold music. I'm
disconnected, I hang
up, dial 701 and I see this message on the console Everyone is
busy/congested at this time
I just have the default parkedcalls file, and have this in
the extensions.
[AnalogPhone]
exten = _70X,1,Dial(Zap/1/${EXTEN},20,Ttr)
include = parkedcalls
[SipPhone]
exten = _70X,1,Dial(SIP/1/${EXTEN},20,Ttr)
include = parkedcalls
{clip}
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Re: [Asterisk-Users] sip.conf from mysql
What more information? Are you talking about mailbox, nat, etc..all those other options for SIP phones? I want to do SIP from database as well but most of our phones are NAT and need that option stored in the database. Matthew - Original Message - From: Victor Alvarez [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, September 10, 2004 6:50 AM Subject: [Asterisk-Users] sip.conf from mysql Hello all! I am trying to load sip.conf from mysql database. I have followed the instructions at http://www.voip-info.org/wiki-Asterisk+sip+mysql+peers. Seems that the authentication (user psw) works fine but I would like to get more information from mysql and I don't know how to retrieve it. Could anybody help me? Any idea about how to do it? Regards, Victor. ___ 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 - RE: [Asterisk-Users] Call Parking Problem
That fixed it. Thanks
The 'parkedcalls' code dynamically creates and deletes entries in the
dialplan to handle the calls that have been parked, so the parking lot
must
not overlap your regular extensions. The initial parking extension is
statically created on startup, thus the 'exten =' entry is matching the
parking slot digits and throwing the congestion error - remove it or edit
/etc/asterisk/features.conf to move the parking lot.
Either way, there is a fair amount missing from the extensions.conf you
posted. I suggest you troll through http://www.voip-info.org
-Original Message-
From: PHP Mechanic [mailto:[EMAIL PROTECTED]
Sent: September 10, 2004 7:27 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Call Parking Problem
Hi,
I'm unable to pick up parked calls after they are transfered.
I get the transfer message when I press # and then I'm told
701 The
extension I'm dialing goes to the on hold music. I'm
disconnected, I hang
up, dial 701 and I see this message on the console Everyone is
busy/congested at this time
I just have the default parkedcalls file, and have this in
the extensions.
[AnalogPhone]
exten = _70X,1,Dial(Zap/1/${EXTEN},20,Ttr)
include = parkedcalls
[SipPhone]
exten = _70X,1,Dial(SIP/1/${EXTEN},20,Ttr)
include = parkedcalls
{clip}
___
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Re: [Asterisk-Users] Conference Phone
Don't remember our costs exactly, was almost a year ago. But this would work for you: Polycom Soundstation - $110 http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemcategory=41374item=6322819848rd=1 Spira SPA-1000 - $85 http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemcategory=61840item=5716081199rd=1ssPageName=WD1V So for just over $200 (have to add shipping) you can have a nice conference phone. A couple of our customers use this solution. hank smith wrote: what phone did you purchase and how much - Original Message - From: Deon Rodden [EMAIL PROTECTED] To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Friday, September 10, 2004 5:59 AM Subject: Re: [Asterisk-Users] Conference Phone We use a nice Polycom conference phone and plugged it into the Sipura and it works crystal clear. Was cheaper than Polycom's conference phone w/ built in VOIP capabilities. Joe Dennick wrote: If it were me; I'd opt for one of the Polycom Conference phones (they are just regular analog phones), and use an FXS card to connect it to Asterisk. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chad Brown Sent: Thursday, September 09, 2004 4:13 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Conference Phone Any advice on a good conference phone that works with Asterisk? I like the Cisco line and was wondering if anyone has used the 7935 or 7936 phones. From what I can tell they dont have a sip load. Has anyone verified this or gotten an ETA from Cisco? Chad --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.745 / Virus Database: 497 - Release Date: 8/27/2004 ___ 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 Inbox is protected by SPAMfighter 1062 spam mails have been blocked so far. Download free www.spamfighter.com today! ___ 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] sip.conf from mysql
Hello, On Fri, 10 Sep 2004 09:49:43 -0500, Matthew Boehm [EMAIL PROTECTED] wrote: What more information? Are you talking about mailbox, nat, etc..all those other options for SIP phones? I want to do SIP from database as well but most of our phones are NAT and need that option stored in the database. Matthew - Original Message - From: Victor Alvarez [EMAIL PROTECTED] I am trying to load sip.conf from mysql database. I have followed the instructions at http://www.voip-info.org/wiki-Asterisk+sip+mysql+peers. Seems that the authentication (user psw) works fine but I would like to get more information from mysql and I don't know how to retrieve it. Could anybody help me? Any idea about how to do it? I never tried this, but why don't you try with res_config? It will let you store *any* configuration file in a database (sip.conf, extensions.conf, etc), and I think its available on CVS (its not an external application). You will probably have to perform a 'reload' every time you change something, but the values will be on a database. http://www.voip-info.org/tiki-index.php?page=Asterisk%20res_config -- Nicolás Gudiño Buenos Aires - Argentina ___ 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] Problem with stuttering on TE410P
Hi Guys,Im having some problems with a Wildcard TE410P card.. During a call I getsome strange messages and the voice drops out:Aug 24 16:40:17 DEBUG[1101416512]: chan_zap.c:4036 my_zt_write: Writereturned -1 (Resource temporarily unavailable) on channel 1Aug 24 16:40:17 DEBUG[1101416512]: chan_zap.c:4036 my_zt_write: Writereturned -1 (Resource temporarily unavailable) on channel 1Aug 24 16:40:17 DEBUG[1101416512]: chan_zap.c:4036 my_zt_write: Writereturned -1 (Resource temporarily unavailable) on channel 1Aug 24 16:40:17 DEBUG[1101416512]: chan_zap.c:4036 my_zt_write: Writereturned -1 (Resource temporarily unavailable) on channel 1Aug 24 16:40:17 DEBUG[1101416512]: chan_zap.c:4036 my_zt_write: Writereturned -1 (Resource temporarily unavailable) on channel 1Aug 24 16:40:17 DEBUG[1101416512]: chan_zap.c:4036 my_zt_write: Writereturned -1 (Resource temporarily unavailable) on channel 1Aug 24 16:40:17 DEBUG[1101416512]: chan_zap.c:4036 my_zt_write: Writereturned -1 (Resource temporarily unavailable) on channel 1Aug 24 16:40:17 DEBUG[1101416512]: chan_zap.c:4036 my_zt_write: Writereturned -1 (Resource temporarily unavailable) on channel 1Aug 24 16:40:17 DEBUG[1101416512]: chan_zap.c:4036 my_zt_write: Writereturned -1 (Resource temporarily unavailable) on channel 1Aug 24 16:40:17 DEBUG[1101416512]: chan_zap.c:4036 my_zt_write: Writereturned -1 (Resource temporarily unavailable) on channel 1Aug 24 16:40:17 DEBUG[1101416512]: chan_zap.c:4036 my_zt_write: Writereturned -1 (Resource temporarily unavailable) on channel 1Aug 24 16:40:17 DEBUG[1101416512]: chan_zap.c:4036 my_zt_write: Writereturned -1 (Resource temporarily unavailable) on channel 1Aug 24 16:40:17 DEBUG[1101416512]: chan_zap.c:4036 my_zt_write: Writereturned -1 (Resource temporarily unavailable) on channel 1Aug 24 16:40:17 DEBUG[1101416512]: chan_zap.c:4036 my_zt_write: Writereturned -1 (Resource temporarily unavailable) on channel 1Dump of zaptel.cfgspan=1,1,0,ccs,hdb3,crc4bchan=1-15bchan=17-31dchan=16Dump of proc/interrupts CPU0 CPU1 0: 45179382 0 IO-APIC-edge timer 1: 4 0 IO-APIC-edge keyboard 2: 0 0 XT-PIC cascade 8: 126 0 IO-APIC-edge rtc 10: 0 0 IO-APIC-level usb-ohci 12: 41 0 IO-APIC-edge PS/2 Mouse 14: 2 0 IO-APIC-edge ide0 26: 451773781 0 IO-APIC-level t4xxp 29: 5686214 0 IO-APIC-level eth1 31: 238608 0 IO-APIC-level cciss0NMI: 0 0LOC: 45179260 45179270ERR: 0MIS: 0Could anybody give me some clue as to what this error is about..Kind RegardsClaus ___ 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] sip.conf from mysql
Victor Alvarez wrote: I am trying to load sip.conf from mysql database. I have followed the instructions at _http://www.voip-info.org/wiki-Asterisk+sip+mysql+peers_. Seems that the authentication (user psw) works fine but I would like to get more information from mysql and I don't know how to retrieve it. If you can live with a less-dynamic approach, check out: http://www.voip-info.org/wiki-Asterisk+sip+conf+from+mysql Using this 'keyword' 'data' structure, you can have whatever keyword=value pairs you like in sip.conf. Use retrieve_sip_conf_from_mysql.pl to read the mysql table and write-out sip.conf. ryan ___ 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] UK Callerid bug #1719 TDM400p
should this work with the x101p? or just the tdm400? Thanks for your help Robb Edward Eastman wrote: Brilliant - thanks, took me half an hour but it's working now. Just for the record, settings as follows: The patch on http://bugs.digium.com/bug_view_page.php?bug_id=009 (ast-UK-and-DTMF-pol-CID.diff) doesn't seem to work for current cvs, but I backed up to cvs as of 31/08/04 and that worked fine. Zapata.conf: usecallerid=yes cidsignalling=v23 cidstart=polarity usecallerid=uk doesn't work, has this changed somewhere along the way, or is this something else? Caller ID detects fine, although I get this logged to asterisk console: Sep 6 13:56:22 WARNING[393238]: chan_zap.c:3369 zt_handle_event: Didn't finish Caller-ID spill. Cancelling. I'll try and add this to the wiki when I get time Thanks Ed -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Soren Rathje Sent: 06 September 2004 13:13 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] UK Callerid bug #1719 TDM400p Edward Eastman wrote: Hi Is this patch (http://bugs.digium.com/bug_view_page.php?bug_id=0001719) the best/only way to get callerid working in the UK with a tdm400p? I thought I'd seen a patch that'd gone into cvs, but maybe I was just imagining things ;) Check the bug tracker for id=9, there has been some development here. UK BT CLIP and DTMF CLIP for the TDM400 (will not work for the X100P) is now merged into one patch. /Soren ___ 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] Re: Asterisk-Users Digest, Vol 2, Issue 94
What versions of Asterisk, asterissk-oh323 do you use?
What is the current configuration of oh323?
Can you send the backtrace of the core file dumped?
Michael.
Asterisk CVS-HEAD-08/26/04-11:46:11
asterisk-oh323-0.6.3b
I think it should be the openh323 because i got the same segmentation error
there.
when i run ./sample/simple/obj_linux_x86_r/simph323 i have this:
[EMAIL PROTECTED] openh323]# ./samples/simple/obj_linux_x86_r/simph323 -l
SimpleH323 Version 1.13.5 by OpenH323 Project on Unix Linux (2.4.20-8-i686)
Local username: root
Silence compression is Enabled
Auto answer is 0
FastConnect is Enabled
H245Tunnelling is Enabled
Jitter buffer: 50-250 ms
Sound output device: /dev/dsp
Sound input device: /dev/dsp
Codecs (in preference order):
Table:
GSM-06.10{sw} 1
MS-GSM{sw} 2
G.711-uLaw-64k{sw} 3
G.711-ALaw-64k{sw} 4
SpeexNarrow-18.2k{sw} 5
SpeexNarrow-15k{sw} 6
SpeexNarrow-11k{sw} 7
SpeexNarrow-8k{sw} 8
SpeexNarrow-5.95k{sw} 9
LPC-10{sw} 10
UserInput/hookflash 11
UserInput/basicString 12
UserInput/dtmf 13
UserInput/RFC2833 14
Set:
0:
0:
GSM-06.10{sw} 1
MS-GSM{sw} 2
G.711-uLaw-64k{sw} 3
G.711-ALaw-64k{sw} 4
SpeexNarrow-18.2k{sw} 5
SpeexNarrow-15k{sw} 6
SpeexNarrow-11k{sw} 7
SpeexNarrow-8k{sw} 8
SpeexNarrow-5.95k{sw} 9
LPC-10{sw} 10
1:
UserInput/hookflash 11
UserInput/basicString 12
UserInput/dtmf 13
UserInput/RFC2833 14
Erreur de segmentation
[EMAIL PROTECTED] openh323]#
AFRIPA TELECOM, Africa Switch On
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Re: [Asterisk-Users] iaxy vs sipura
how much ram you got on the pc running the vm? also will microsoft Virtual PC run on xp home? thanks hank - Original Message - From: Bill Seddon [EMAIL PROTECTED] To: 'Benjamin on Asterisk Mailing Lists' [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial Discussion' [EMAIL PROTECTED] Sent: Friday, September 10, 2004 6:34 AM Subject: RE: [Asterisk-Users] iaxy vs sipura I run Asterisk on Redhat 8.0 with a VM hosted by Microsoft's Virtual PC which, in turn, runs on Windows 2000 Server. Works like a charm. Can't use Zaptel cards but that's OK for me. I can put it into standby any time and it takes only a few seconds to start up the VM from its saved state and at that time the Linux session (and Asterisk) is available once again. Bill Seddon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Benjamin on Asterisk Mailing Lists Sent: September 10, 2004 2:03 PM To: Andy Powell Cc: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] iaxy vs sipura On Fri, 10 Sep 2004 14:05:09 +0200, Andy Powell [EMAIL PROTECTED] wrote: At the risk of stating the obvious if you have a laptop not running MacOSX (ie perhaps running windows) download my asterisk live! cd ( http://www.automated.it/asterisk/ ), burn it and test it on your laptop and bung it in your laptop case along with your iaxy/sipura/whatever and errm... problem solved.. :D Certainly an option, but most business folks will want to have their Outlook contacts and Excel spreadsheets in front of them when they are on the phone. Dual boot environments are not ideal in those situations. Imagine you're talking to some guy on the phone about prices and he tells you I cant' tell you what the discounts are right now because I would have to shut down the phone system to open Excel. However, you could use VMware on an Intel notebook to run both Windoze and Linux concurrently. This wouldn't be ideal for a real PBX for performance reasons, but since all you are going to use Asterisk for is to be a gateway for one single user, it's probably ok in this particular scenario. I remember there was a guy in Romania who reported he had VMware with Windoze and Asterisk on Linux running as a home PBX on his PC and it seemed to be alright. If you'd combine such a setup with a Windoze GUI tool that will start and stop the Linux environment and Asterisk at the push of a button, then you'd have a fairly convenient and workable SIP/IAX gateway solution for travelling biz folks. rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed. ___ 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 Inbox is protected by SPAMfighter 1068 spam mails have been blocked so far. Download free www.spamfighter.com today! ___ 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] astwind has any one got this thing to work?
hmm really need to test this thing. On Fri, 2004-09-10 at 10:02, Greg Boehnlein wrote: On 9 Sep 2004, khurram bhatti wrote: Well I wanted to test astwind and consulted * person he gave me this comment lord help us all ... why would you want to simulate a linux system on top of a windows system in the first place? It's not a simulated linux system. CoLinux is a kernel that runs in Ring 0 of the Windows kernel, with direct access to the Processor and MMU. It runs in it's own protected memory space. The ONLY thing it uses Windows for is to actually load the kernel and handle the I/O drivers. Otherwise, CoLinux is running natively on your hardware... At the same TIME as windows. So, it isn't like running Vmware. It's a LOT faster, and if you set it up right, you can even boot your existing Linux partition. It's the best way to run Windows -AND- Linux at the same time. Blows the pants off of BoCHS and Vmware in speed. I've been running my home PBX under AstWind for about a month now. Even after the Windows kernel has crashed and the system is completely hung, CoLinux and AstWind continue to run without a problem! It's pretty amazing. Check out http://www.nacs.net/~damin/astwind.jpg ___ 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: Problem with Openh323 channel driver
Date: Fri, 10 Sep 2004 16:37:33 +0300
From: Michael Manousos [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Problems with 0penh323 Channel Driver
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=us-ascii; format=flowed
[EMAIL PROTECTED] wrote:
Hi,
I have asterisk,openh323-v1_13_5 and pwlib-v1_6_6 installed on my PC. each
time
i run asterisk -c, i get the following error:
[chan_oh323.so] = (OpenH323 Channel Driver)
== Parsing '/etc/asterisk/rtp.conf': Found
== Parsing '/etc/asterisk/oh323.conf': Found
[1]WrapH323EndPoint::WrapH323EndPoint: Compile-time libraries OpenH323
v1.13.5,
PWlib v1.6.6
segmentation error
[EMAIL PROTECTED] root]#
Can you help me?
What versions of Asterisk, asterissk-oh323 do you use?
What is the current configuration of oh323?
Can you send the backtrace of the core file dumped?
Michael.
What versions of Asterisk, asterissk-oh323 do you use?
What is the current configuration of oh323?
Can you send the backtrace of the core file dumped?
Michael.
Asterisk CVS-HEAD-08/26/04-11:46:11
asterisk-oh323-0.6.3b
I think it should be the openh323 because i got the same segmentation error
there.
when i run ./sample/simple/obj_linux_x86_r/simph323 i have this:
[EMAIL PROTECTED] openh323]# ./samples/simple/obj_linux_x86_r/simph323 -l
SimpleH323 Version 1.13.5 by OpenH323 Project on Unix Linux (2.4.20-8-i686)
Local username: root
Silence compression is Enabled
Auto answer is 0
FastConnect is Enabled
H245Tunnelling is Enabled
Jitter buffer: 50-250 ms
Sound output device: /dev/dsp
Sound input device: /dev/dsp
Codecs (in preference order):
Table:
GSM-06.10{sw} 1
MS-GSM{sw} 2
G.711-uLaw-64k{sw} 3
G.711-ALaw-64k{sw} 4
SpeexNarrow-18.2k{sw} 5
SpeexNarrow-15k{sw} 6
SpeexNarrow-11k{sw} 7
SpeexNarrow-8k{sw} 8
SpeexNarrow-5.95k{sw} 9
LPC-10{sw} 10
UserInput/hookflash 11
UserInput/basicString 12
UserInput/dtmf 13
UserInput/RFC2833 14
Set:
0:
0:
GSM-06.10{sw} 1
MS-GSM{sw} 2
G.711-uLaw-64k{sw} 3
G.711-ALaw-64k{sw} 4
SpeexNarrow-18.2k{sw} 5
SpeexNarrow-15k{sw} 6
SpeexNarrow-11k{sw} 7
SpeexNarrow-8k{sw} 8
SpeexNarrow-5.95k{sw} 9
LPC-10{sw} 10
1:
UserInput/hookflash 11
UserInput/basicString 12
UserInput/dtmf 13
UserInput/RFC2833 14
Erreur de segmentation
[EMAIL PROTECTED] openh323]#
AFRIPA TELECOM, Africa Switch On
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[Asterisk-Users] Net2Phone, Asterisk, and 404 Not Found
Hi! Net2Phone is getting a common SIP status code, 404 Not Found, when trying to place a call to our Asterisk server. We're hoping someone on the list can shed some light on why this is happening. We can process a call from Asterisk to Net2Phone without any problems. Net2Phone sends the INVITE but immediately gets the 404 Not Found. The To: field of the INVITE contains the E.164 formatted number with a plus + sign before the 11 digits and we were thinking that the presence of that plus sign had something to do with the 404 problem. But I guess the plus sign is part of the SIP standard. I don't think we've seen the INVITE but I'll dig further on that. Has anyone connected Asterisk to a different SIP proxy and used SIP to communicate between the two? Can anyone further explain why our Asterisk is not replying to Net2Phone's INVITE? Here is the entry from our sip.conf file: [net2phone3] context = n2p-in host=Net2Phone's IP Address disallow=g723.1 allow=g729 type=friend dtmfmode=rfc2833 Thanks in advance! 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] sip.conf from mysql
Hi, First of all thank you Matthew, Nicolas and Ryan for your response. I would like to get information like context, mailbox, callgroup, pickupgroup, codecs... also nat! If I make the substitution of the text file i wouldn't like to miss information in the process. retrieve_sip_conf_from_mysql.pl seems to be a good B plan. I will have to recharge sip.conf manually but.. If this is the way, I willfollow it. Anyway if plan A works with user and password, Why it can't work with the rest of parameters?? I'll continue my work on Monday. Have a nice weekend! Victor. ___ 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] Red Alarm
I posted a while ago, about the FXO card entering a non-operational state. While in a call, all of a sudden, there's this loud noise, and the card remains like that until I reload the wcfxo module. There's no way to dial in or out the FXO unless the module is reloaded. I made some progress... I was looking for an indication in the system that the problem was occurring. Well, I found it. When the FXO blows up asterisk gets the following event: WARNING[213006]: Detected alarm on channel 1: Red Alarm Last night I was talking on the FXO extension, and it happened. Just to confirm, I went to /var/log/asterisk and looked at the messages file. There it was: the freaking Red Alarm. I noticed that the Red Alarm also happens when I unplug the phone line. When I plug it back, there's this notice: NOTICE[229390]: Alarm cleared on channel 1 I'm wondering if it's the FXO that causes the problem, or if it's the line that puts the FXO in the inoperable state. I don't know if it makes any difference, but the line is connected to a PBX, not directly to the telco. So, instead of shutting down asterisk, reloading the module, and starting asterisk again, I will take a look at the drivers source to see if I manage to make it self-restart when it detects the Red Alarm. Any thoughts? Suggestions? Marconi. ___ 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] DevKit TDM400P module won't load
Maybe this is silly but I had a similar problem when I installed my kit. The problem was my motherboard didn't provide 3.3V... Rgds, Renato On Fri, 10 Sep 2004 13:16:26 +1200, Colin Haxton [EMAIL PROTECTED] wrote: Hi Lyle, I don't have lspci on my system. It's a dump of what is in the /proc/pci anyway. Yep, zaptel loads fine. The X100P loads and is working well, with Asterisk. It's actually in another machine, I moved this TDM400p to a machine that I can play around on. I haven't loaded wcfxo as that's on the X100P and I don't have a FXO on this card. Only one FXS. lsmod says wcfxs 38432 0 zaptel189188 1 wcfxs crc_ccitt 2560 1 zaptel But dmesg reports these errors. Any other ideas? :) Colin Lyle Giese wrote: What does lspci -v show? I just looked at my /proc/dev and it shows two Communication Controller: Tiger Jet Network Inc in there. I have a TDM22b and a X100P on a 2.4.x kernel. Did you modprobe zaptel first? Then wcfxs and then wcfxo? Lyle - Original Message - From: Colin Haxton [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, September 09, 2004 7:57 PM Subject: [Asterisk-Users] DevKit TDM400P module won't load Hi all, I have just purchased the DevKit from Digium and received a X100P and a TDM400P (it has one FXS module). The problem is that I can't get the kernel module (wcfxs) to load and run. I have searched the archives and can't find anything about this. Do the messages below ring any bells with anyone ? There is no interrupt clash (see pci list below). I am running on a AMD 1.2G processor, 2.6 kernel. The motherboard is a MSI K7T Turbo2, which is PCI 2.2, I flash upgraded the bios to the latest version just in case. Can anyone help? I am at a loss what to try next and I don't want to end up throwing away my new toy. :) Thanks, Colin /--dmesg- zaptel: no version for struct_module found: kernel tainted. Zapata Telephony Interface Registered on major 196 ACPI: PCI interrupt :00:08.0[A] - GSI 17 (level, low) - IRQ 17 Freshmaker version: 71 00 != ff 01 != ff 02 != ff 03 != ff 04 != ff 05 != ff snip f2 != ff f3 != ff f4 != ff f5 != ff f6 != ff f7 != ff f8 != ff f9 != ff fa != ff fb != ff fc != ff fd != ff fe != ff Freshmaker failed register test wcfxs: probe of :00:08.0 failed with error -5 / /---proc/pci- PCI devices found: Bus 0, device 0, function 0: Host bridge: VIA Technologies, Inc. VT8363/8365 [KT133/KM133] (rev 3). Master Capable. Latency=8. Prefetchable 32 bit memory at 0xe000 [0xe3ff]. Bus 0, device 1, function 0: PCI bridge: VIA Technologies, Inc. VT8363/8365 [KT133/KM133 AGP] (rev 0). Master Capable. No bursts. Min Gnt=12. Bus 0, device 7, function 0: ISA bridge: VIA Technologies, Inc. VT82C686 [Apollo Super South] (rev 64). Bus 0, device 7, function 1: IDE interface: VIA Technologies, Inc. VT82C586A/B/VT82C686/A/B/VT823x/A/C PIPC Bus Master IDE (rev 6). Master Capable. Latency=32. I/O at 0xd000 [0xd00f]. Bus 0, device 7, function 2: USB Controller: VIA Technologies, Inc. VT82x UHCI USB 1.1 Controller (rev 26). IRQ 11. Master Capable. Latency=32. I/O at 0xd400 [0xd41f]. Bus 0, device 7, function 3: USB Controller: VIA Technologies, Inc. VT82x UHCI USB 1.1 Controller (#2) (rev 26). IRQ 11. Master Capable. Latency=32. I/O at 0xd800 [0xd81f]. Bus 0, device 7, function 4: Host bridge: VIA Technologies, Inc. VT82C686 [Apollo Super ACPI] (rev 64). IRQ 7. Bus 0, device 8, function 0: Network controller: Individual Computers - Jens Schoenfeld Intel 537 (rev 0). IRQ 17. Master Capable. Latency=32. Min Gnt=1.Max Lat=128. I/O at 0xdc00 [0xdcff]. Non-prefetchable 32 bit memory at 0xe400 [0xe4000fff]. Bus 0, device 13, function 0: Ethernet controller: 3Com Corporation 3c905B 100BaseTX [Cyclone] (rev 36). IRQ 18. Master Capable. Latency=32. Min Gnt=10.Max Lat=10. I/O at 0xe000 [0xe07f]. Non-prefetchable 32 bit memory at 0xe8001000 [0xe800107f]. Bus 1, device 0, function 0: VGA compatible controller: nVidia Corporation NV17 [GeForce4 MX 440-SE] (rev 163). IRQ 16. Master Capable. Latency=32. Min Gnt=5.Max Lat=1. Non-prefetchable 32 bit memory at 0xe500 [0xe5ff]. Prefetchable 32 bit memory at 0xd000 [0xd7ff].
RE: [Asterisk-Users] Red Alarm
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Marconi Rivello Sent: Friday, September 10, 2004 1:47 PM To: Asterisk Subject: [Asterisk-Users] Red Alarm I made some progress... I was looking for an indication in the system that the problem was occurring. Well, I found it. When the FXO blows up asterisk gets the following event: WARNING[213006]: Detected alarm on channel 1: Red Alarm Yeah, basically a RedAlarm is when the Card doesn't see a dialtone. So, instead of shutting down asterisk, reloading the module, and starting asterisk again, I will take a look at the drivers source to see if I manage to make it self-restart when it detects the Red Alarm. No, you don't want to do this. Fix the problem so the red alarm doesn't occur. We had the same issue when we were running Promise Array card in there. What hardware are you running? CPU, Motherboard, type of computer, extra PCI cards, is it on it's own IRQ, etc.. Those are where you want to start. - Brent ___ 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] sip.conf from mysql
Apparently, Plan A is hard coded to only select out certain info from the database. If you know C you could probably take a crack at adding some more code. This is what I am going to do here in a bit or over the weekend. Matthew - Original Message - From: Victor Alvarez [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, September 10, 2004 12:44 PM Subject: Re: [Asterisk-Users] sip.conf from mysql Hi, First of all thank you Matthew, Nicolas and Ryan for your response. I would like to get information like context, mailbox, callgroup, pickupgroup, codecs... also nat! If I make the substitution of the text file i wouldn't like to miss information in the process. retrieve_sip_conf_from_mysql.pl seems to be a good B plan. I will have to recharge sip.conf manually but.. If this is the way, I will follow it. Anyway if plan A works with user and password, Why it can't work with the rest of parameters?? I'll continue my work on Monday. Have a nice weekend! Victor. ___ 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] TDM400P lockups (FXO)
It sounds like my lockups may be related since my TDM422b card has the FXS FXS FXO FXO configuration and doesn't have an FXO in position 1 either. My card is identified in software as Rev E/F and has the wire jumper on the back. David Richard Scobie said: Maciej Kietlinski wrote: Are the FXOs on the 2x on ports 1-2 or 3-4? Maybe it has to do with *any* FXO on port 1... Please get back with the list with your findings. My experience led to a replacement from Digium, but the card is a TDM400P with 4 FXO...now that I think of it, during troubleshooting there was some correlation to the first port on the card (port 1)...not the first module - I swapped module positions to varying locations on the card without success, but then again they are all FXO...Maybe *is* possible that the TDM400P doesn't like an FXO module in port 1 as you are suggesting...Like I said, in the end I got a new revision board from digium, all 4 ports are still FXO and working great now... With my old revision TDM400P it was the same problem with FXO on port 1. Easiest way for me was to put FXO's on new revision card, and on old use FXS on port 1. I used info from post with: 'The card had been modified, evident from the jumper wire that been soldered between two points on the back of the card. I haven't had problems since installing the new card.' And before old card was used with FXS + 3 x FXO without problems, so it works in the same hw conf again. Now I heve no problems with TDMxxp I'll let you know how I get on. One of the cards that is giving trouble, has FXOs in positions 3 and 4. Can anyone tell me what these new revision cards are? My current ones are all Rev. E/F. Regards, Richard ___ 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] UK Callerid bug #1719 TDM400p
The ast-UK-and-DTMF-pol-CID.diff patch will only work for the tdm400 FXO modules due to the fact that the x101p is not capable of detecting polarity reversal events. Dan On Fri, 2004-09-10 at 17:38, Robert Boardman wrote: should this work with the x101p? or just the tdm400? Thanks for your help Robb Edward Eastman wrote: Brilliant - thanks, took me half an hour but it's working now. Just for the record, settings as follows: The patch on http://bugs.digium.com/bug_view_page.php?bug_id=009 (ast-UK-and-DTMF-pol-CID.diff) doesn't seem to work for current cvs, but I backed up to cvs as of 31/08/04 and that worked fine. Zapata.conf: usecallerid=yes cidsignalling=v23 cidstart=polarity usecallerid=uk doesn't work, has this changed somewhere along the way, or is this something else? Caller ID detects fine, although I get this logged to asterisk console: Sep 6 13:56:22 WARNING[393238]: chan_zap.c:3369 zt_handle_event: Didn't finish Caller-ID spill. Cancelling. I'll try and add this to the wiki when I get time Thanks Ed -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Soren Rathje Sent: 06 September 2004 13:13 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] UK Callerid bug #1719 TDM400p Edward Eastman wrote: Hi Is this patch (http://bugs.digium.com/bug_view_page.php?bug_id=0001719) the best/only way to get callerid working in the UK with a tdm400p? I thought I'd seen a patch that'd gone into cvs, but maybe I was just imagining things ;) Check the bug tracker for id=9, there has been some development here. UK BT CLIP and DTMF CLIP for the TDM400 (will not work for the X100P) is now merged into one patch. /Soren ___ 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] call quality monitoring
I need to debug a call quality issue with remote users on the other end of a satellite link. The symptoms are: we here on the Internet side can hear them just fine. On their end, things work sorta OK most times, but they often suffer from severe dropouts and digital warbling, both of which I attribute to them missing packets. Often times they can't make out a word we are saying while we can hear them crystal clearly. Various pings and other network tests indicate that the underlying network is functioning as well as can be expected for a sat link. In fact, the overall jitter seems to be pretty low (avg 20ms). Packet loss is around 1-2%, and latency is around 700ms on average. I'm left to assume that the jitter buffer on that end isn't functioning properly. Both ends of the call have the same jitter buffer settings. The call is carried by IAX2 and encoded with ILBC. The iax.conf files on each end start like this: [general] trunk=no notransfer=yes iaxcompat=no bandwidth=low disallow=all allow=ilbc jitterbuffer=yes dropcount=3 maxjitterbuffer=500 maxexcessbuffer=150 minexcessbuffer=40 jittershrinkrate=1 Of course, perhaps the jitter buffer isn't to blame, but given that one side of the call sounds perfect, I can't think of anything else obvious that would cause this. Is there any way to extract from asterisk some idea of why it thinks the calls sound bad? For example, when the jitter buffer notices that packets are discarded because they are too late, when excessive packets are completely missing, etc. I've been collecting a giant debug log for a while now, so I could pretty easily sift through it if there's something good to look for. Thanks. -- Matt Ranney - [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] Net2Phone, Asterisk, and 404 Not Found
did you try to add canreinvite=yes to [net2phone3] ?? Marc [EMAIL PROTECTED] wrote: Hi! Net2Phone is getting a common SIP status code, 404 Not Found, when trying to place a call to our Asterisk server. We're hoping someone on the list can shed some light on why this is happening. We can process a call from Asterisk to Net2Phone without any problems. Net2Phone sends the INVITE but immediately gets the 404 Not Found. The To: field of the INVITE contains the E.164 formatted number with a plus + sign before the 11 digits and we were thinking that the presence of that plus sign had something to do with the 404 problem. But I guess the plus sign is part of the SIP standard. I don't think we've seen the INVITE but I'll dig further on that. Has anyone connected Asterisk to a different SIP proxy and used SIP to communicate between the two? Can anyone further explain why our Asterisk is not replying to Net2Phone's INVITE? Here is the entry from our sip.conf file: [net2phone3] context = n2p-in host=/Net2Phone's IP/ /Address/ disallow=g723.1 allow=g729 type=friend dtmfmode=rfc2833 Thanks in advance! 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 ___ 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] SpanDSP/RxFax anomalies...
I've recently started playing with the RxFax application on my Asterisk box. I've had success, mostly, but I've had some failures, too... The most recent failure is specific to receiving from a particular fax machine -- a Canon Laser Class 9000S. The TIF images received are readable, but the aspect ratio is stretched horizonatlly (or squished vertically). Is this a problem anyone else has seen before? Is there a workaround? Thanks, Rob ___ 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] UK Callerid bug #1719 TDM400p
thanks for the reply Dan Does anyone know if the history buffer CID patch still works with the latest cvs? Robb Dan Tucny wrote: The ast-UK-and-DTMF-pol-CID.diff patch will only work for the tdm400 FXO modules due to the fact that the x101p is not capable of detecting polarity reversal events. Dan On Fri, 2004-09-10 at 17:38, Robert Boardman wrote: should this work with the x101p? or just the tdm400? Thanks for your help Robb Edward Eastman wrote: Brilliant - thanks, took me half an hour but it's working now. Just for the record, settings as follows: The patch on http://bugs.digium.com/bug_view_page.php?bug_id=009 (ast-UK-and-DTMF-pol-CID.diff) doesn't seem to work for current cvs, but I backed up to cvs as of 31/08/04 and that worked fine. Zapata.conf: usecallerid=yes cidsignalling=v23 cidstart=polarity usecallerid=uk doesn't work, has this changed somewhere along the way, or is this something else? Caller ID detects fine, although I get this logged to asterisk console: Sep 6 13:56:22 WARNING[393238]: chan_zap.c:3369 zt_handle_event: Didn't finish Caller-ID spill. Cancelling. I'll try and add this to the wiki when I get time Thanks Ed -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Soren Rathje Sent: 06 September 2004 13:13 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] UK Callerid bug #1719 TDM400p Edward Eastman wrote: Hi Is this patch (http://bugs.digium.com/bug_view_page.php?bug_id=0001719) the best/only way to get callerid working in the UK with a tdm400p? I thought I'd seen a patch that'd gone into cvs, but maybe I was just imagining things ;) Check the bug tracker for id=9, there has been some development here. UK BT CLIP and DTMF CLIP for the TDM400 (will not work for the X100P) is now merged into one patch. /Soren ___ 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] IAX2 dropping call?
On 9 Sep 2004 at 15:35, [EMAIL PROTECTED] wrote:
I am using CVS-HEAD-08/29/04-22:41:39
I have notransfer=yes in my iax.conf
I have been on the phone most of the day...dropped twice so far.
Paul Seniuk
-Original Message-
From: Kris.Boutilier [mailto:[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Hello all,
I updated from CVS 3 days ago and now my IAX2 gateway is dropping
calls without warning.
{clip}
Which version were you running with before the CVS update?
I have been having the same type of problem and it seems to be related
to allowing native bridging in IAX2 (setting 'notransfer=no'). I have
no NAT or other complexites in the way, it just inexplicably drops the
call. I'm running 'CVS-HEAD-08/13/04-10:37:13'.
Kris Boutilier
My wife has been complaining about the same thing (also after a cvs
update). My problem is on FXO-FXS and vice versa calls though, no
IAX. I have since increased busycount from 6 to 8 which seems to be
working at the moment, I'll post again if it resurfaces.
I posted to the -dev list the other night (although I was a little
drunk) about whether the busydetect code recognizes the cadences as
well as the tone. Reason being that there are definitely not 6 x
busy length tones being played that would cause it to be hung
up...not even one. I think (without looking at the code) that what
it is doing is looking for the tone and increasing a var.
Cheers,
Matt Riddell
http://www.sineapps.com
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Re: [Asterisk-Users] Simple question about SIP community
Marcello Lupo wrote: Hi to all, we have a community of people on an * box that use SIP softphones to talk each other. Can you suggest me the quickest and simple way to let someone know who is online without have to call one by one the persons to look if they are present or not?? Something the user list in Microsoft Messenger. I was thinking on some sort of web page that can check the registration of the sip clients on the asterisk but want to know if already exist to avoid to reinvent the wheel. thanks, Bye, MArcello I would suggest you check out the Flash Operator Panel at www.asternic.org/ . It gives you an overview of who is on the phone and what lines/channels are in use. If you configure it properly, you can even use it to make internal calls. Just simply click on the person you want to talk to, and both of your phones will start ringing. ___ 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] Number of TDM405 Cards in one server
Has any one put 3 or more TDM405P or TDM410P cards in a single server? I would like to fit as many as 6 into one box. I am concerned about several things such as power requirements and the amount of cooling as well as CPU and memory utilization. Is there a difference in the power consumption and heat between the 5.0v and 3.3v boards would one be better than the other for such a dense situation? I have not been able to find any recommendations from digium on this side of things. Has any one implemented this in the past and what did you have to worry about. What type of motherboard/system/memory/cpu did you use or what do you think would be best. I get to start from scratch on this but I would like it to be as dense as possible since it is going into a Colocation Rack at a data center and space is money. Thanks 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
Re: [Asterisk-Users] Number of TDM405 Cards in one server
From what I have heard, read and seen, the most you will ever want to do is two, and that is only in certain situations, i.e. you are not doing much/any transcoding, IVR's, a bunch of conferences, etc. A better solution would be multiple 1U servers, potentially, even though I realize, space is money. Someone else might have a different opinion here, but 288 channels seems like a few too many, given what I have seen of these boards. They're good hardware, but somewhat demanding on the machine, in unseen ways. On Fri, 10 Sep 2004 13:39:33 -0600, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Has any one put 3 or more TDM405P or TDM410P cards in a single server? I would like to fit as many as 6 into one box. I am concerned about several things such as power requirements and the amount of cooling as well as CPU and memory utilization. Is there a difference in the power consumption and heat between the 5.0v and 3.3v boards would one be better than the other for such a dense situation? I have not been able to find any recommendations from digium on this side of things. Has any one implemented this in the past and what did you have to worry about. What type of motherboard/system/memory/cpu did you use or what do you think would be best. I get to start from scratch on this but I would like it to be as dense as possible since it is going into a Colocation Rack at a data center and space is money. Thanks 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] Number of TDM405 Cards in one server
When we need that many T1s, we use routers. Much less complex and roughly the same cost. William -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Friday, September 10, 2004 12:40 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Number of TDM405 Cards in one server Has any one put 3 or more TDM405P or TDM410P cards in a single server? I would like to fit as many as 6 into one box. I am concerned about several things such as power requirements and the amount of cooling as well as CPU and memory utilization. Is there a difference in the power consumption and heat between the 5.0v and 3.3v boards would one be better than the other for such a dense situation? I have not been able to find any recommendations from digium on this side of things. Has any one implemented this in the past and what did you have to worry about. What type of motherboard/system/memory/cpu did you use or what do you think would be best. I get to start from scratch on this but I would like it to be as dense as possible since it is going into a Colocation Rack at a data center and space is money. Thanks 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] Number of TDM405 Cards in one server
I would think the first issue regarding the number of cards is that each card has to have a seperate and unique IRQ and cann't share IRQ's with anything else. So from that requirement, six would seem out of the question. As far as the rest, there are limits on number of calls, but they are more related to the translating from one codec to another codec and so on. Dig around in the wiki for info on number of calls supported. Lyle - Original Message - From: [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' [EMAIL PROTECTED] Sent: Friday, September 10, 2004 2:39 PM Subject: [Asterisk-Users] Number of TDM405 Cards in one server Has any one put 3 or more TDM405P or TDM410P cards in a single server? I would like to fit as many as 6 into one box. I am concerned about several things such as power requirements and the amount of cooling as well as CPU and memory utilization. Is there a difference in the power consumption and heat between the 5.0v and 3.3v boards would one be better than the other for such a dense situation? I have not been able to find any recommendations from digium on this side of things. Has any one implemented this in the past and what did you have to worry about. What type of motherboard/system/memory/cpu did you use or what do you think would be best. I get to start from scratch on this but I would like it to be as dense as possible since it is going into a Colocation Rack at a data center and space is money. Thanks 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] IAX2 dropping call?
Matt,
That interesting. We have even had the problem occur with SIP clients
Using SNOM handsets. The gateway has a PRI, so I dont think busycount
Even applies too me?
Cheers,
Paul Seniuk
-Original Message-
From: matt.riddell [mailto:[EMAIL PROTECTED]
Sent: September 10, 2004 1:20 PM
To: asterisk-users
Subject: RE: [Asterisk-Users] IAX2 dropping call?
On 9 Sep 2004 at 15:35, [EMAIL PROTECTED] wrote:
I am using CVS-HEAD-08/29/04-22:41:39
I have notransfer=yes in my iax.conf
I have been on the phone most of the day...dropped twice so far.
Paul Seniuk
-Original Message-
From: Kris.Boutilier [mailto:[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
Hello all,
I updated from CVS 3 days ago and now my IAX2 gateway is dropping
calls without warning.
{clip}
Which version were you running with before the CVS update?
I have been having the same type of problem and it seems to be
related
to allowing native bridging in IAX2 (setting 'notransfer=no'). I
have
no NAT or other complexites in the way, it just inexplicably drops
the
call. I'm running 'CVS-HEAD-08/13/04-10:37:13'.
Kris Boutilier
My wife has been complaining about the same thing (also after a cvs
update). My problem is on FXO-FXS and vice versa calls though, no
IAX. I have since increased busycount from 6 to 8 which seems to be
working at the moment, I'll post again if it resurfaces.
I posted to the -dev list the other night (although I was a little
drunk) about whether the busydetect code recognizes the cadences as
well as the tone. Reason being that there are definitely not 6 x
busy length tones being played that would cause it to be hung
up...not even one. I think (without looking at the code) that what
it is doing is looking for the tone and increasing a var.
Cheers,
Matt Riddell
http://www.sineapps.com
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[Asterisk-Users] What would be required for this?
Hey All, I have a question that I'm curious about. I want to set up a 4 phone system in my home with 2 actual lines coming into the house. Both or just regular lines (not sure of this matters?), one being VoIP and the other just a regular analog line. For now though I just want the VoIP line coming in, but would like the ability to expand to 2 lines in the future. What type of hardware is required for this, and how much would it cost? For now though, this is what I want to do and for as cheap as possible.. I have a VoIP line that has free long distance on it and I want to be able to dial into Astrisk from my cell to be able to reach any number I want (eg extention that dials an outside line). Any ideas on how to go about this? Thanks in advance! ___ 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] Sangoma S508 Rev-B
Benedict P. Barszcz wrote: Can I use this card with asterisk in any way but without subscription to a Frame Relay account? Perhaps in similiar manner as T1/E1 between a channel bank and an asterisk server. Or perhaps there is way to make it behave like a kind of an FXS interface (to anything). No, this a high-speed synchronous serial interface card. There are no functions in Asterisk that could be used with this card. ___ 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 Dropped Calls
When sending calls to my Long Distance Provider I've come across this problem. After about 3 or 4 seconds into a call, it gets cut off. This is what I have concluded after doing a trace. 1. An invite is sent to the Asterix PBX 2. Asterix sends back a 100 trying. 3. Asterix then sends a 200 OK, with session description. 4. They ACKnowledge the Asterix 200 OK 5. Asterix then sends a 183 Session Progress, with description; this message is equivalent to a ringing and I'm not sure why the Asterix sends this message. 6. At this point the Asterix sends a total of six 200 OKs which they never respond to. Their terminating device has already setup the call it doesn't respond to the six 200 OKs. 7. It appears since the Aterix doesn't receive a reply to it's 200 OKs it gives up and sends a BYE and releases the call. Now this doesn't happen on every call. Just certain Destinations. Any Ideas? ___ 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] moh cell phones
Hello, MOH always is choppy when someone calls from a cell phone to my pots or nufone 866. It sounds fine when it originates from a land line. I use zaptel hardware, and plenty of resources. I have tried to use different songs. None have the id3 tags, I tried the custom settings with -q -r 8000 -f 8192 -b 2048 --mono -s. Tried permanent resampling to 8khz, 16bit, filterd with lame -q1. I removed my packaged mpg123.59r-15 from debian testing, compiled mpg123r (patched). NO SOUP FOR ME. thanks a billion. J __ Do you Yahoo!? New and Improved Yahoo! Mail - 100MB free storage! http://promotions.yahoo.com/new_mail ___ 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] Number of TDM405 Cards in one server
First, START NEW MESSAGES. don't respond to something totally different and then remove the contents. You message has NOTHING to do with the message your mail client said you responded to. In-Reply-To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] 7 Dialing gives a busy signal On Fri, 2004-09-10 at 14:39, [EMAIL PROTECTED] wrote: Has any one put 3 or more TDM405P or TDM410P cards in a single server? I would like to fit as many as 6 into one box. Do you want a TDM400 series card or a TE400 series card. Quick mention of Digium part numbers, TDM/S/X are analog cards, T/E/TE are T1 or E1 or T1 and E1 capable cards. the first number is a port capacity. From Digiums site, you get this The Wildcard TDM400P is a half-length PCI 2.2 compliant card that supports from one to four telephone interfaces for connecting analog telephones or analog lines to a PC. ... The naming convention for the TDM bundles is as follows: TDM X Y B. Where TDM denotes that the card is TDM, X denotes the number of FXS modules, Y denotes the number of FXO modules, and B indicates that that this product is a bundle. So you see there isn't a TDM405P or TDM410P. There are however TE405P and TE410P cards. When you get to T1 or E1 configurations, you shouldn't look at more than 2 cards per server, and 2 cards should probably be only undertaken with extreme care and caution. Having a simple hardware failure take down 96 lines is bad, but not as bad as taking down 192 or 288. If you are routing 288 calls, your downtime cost to repair a single box will quickly exceed the cost of redundant servers. -- 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
RE: [Asterisk-Users] IAX2 dropping call?
On Sat, 11 Sep 2004 [EMAIL PROTECTED] wrote: I posted to the -dev list the other night (although I was a little drunk) about whether the busydetect code recognizes the cadences as well as the tone. Reason being that there are definitely not 6 x busy length tones being played that would cause it to be hung up...not even one. I think (without looking at the code) that what it is doing is looking for the tone and increasing a var. I always found the busydetect code much more inclined to hang up on women than on men. The voice pitch I suppose. Just what us comms geeks with longsuffering guinea-pig wives/GFs at home don't need. 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] RDNIS and Q.931
Does anyone know what Q.931 Information Element that * pulls the RDNIS variable from? Jody N. Rudolph Heartland Communications Internet Services, Inc 1301 Boadway Paducah, KY 42001 [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] IAX2 dropping call?
Steve, Are you for real about the voice pitch? I am both laughing and fascinated at the same time!?!?! :P Paul Seniuk -Original Message- From: steve [mailto:[EMAIL PROTECTED] Sent: September 10, 2004 2:36 PM To: asterisk-users Subject: RE: [Asterisk-Users] IAX2 dropping call? On Sat, 11 Sep 2004 [EMAIL PROTECTED] wrote: I posted to the -dev list the other night (although I was a little drunk) about whether the busydetect code recognizes the cadences as well as the tone. Reason being that there are definitely not 6 x busy length tones being played that would cause it to be hung up...not even one. I think (without looking at the code) that what it is doing is looking for the tone and increasing a var. I always found the busydetect code much more inclined to hang up on women than on men. The voice pitch I suppose. Just what us comms geeks with longsuffering guinea-pig wives/GFs at home don't need. 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 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] Valet Park Application
I love the functionality of the Valet Park Application. I have a question regarding its operation. The problem I am having when there is a call already parked on specific park extension. If a caller uses 'blind' transfer on a Cisco Phone the caller gets disconnected. Can any offer any suggestions on how to prevent the transfer from taking place? ___ 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] Number of TDM405 Cards in one server
[snip] From Digiums site, you get this The Wildcard TDM400P is a half-length PCI 2.2 compliant card that supports from one to four telephone interfaces for connecting analog telephones or analog lines to a PC. ... The naming convention for the TDM bundles is as follows: TDM X Y B. Where TDM denotes that the card is TDM, X denotes the number of FXS modules, Y denotes the number of FXO modules, and B indicates that that this product is a bundle. So you see there isn't a TDM405P or TDM410P. There are however TE405P and TE410P cards. When you get to T1 or E1 configurations, you shouldn't look at more than 2 cards per server, and 2 cards should probably be only undertaken with extreme care and caution. Having a simple hardware failure take down 96 lines is bad, but not as bad as taking down 192 or 288. If you are routing 288 calls, your downtime cost to repair a single box will quickly exceed the cost of redundant servers. Quick question if one TDM400P card has 4xFXS port that is only four internal lines, am I right? So how do you calculate that number 96 line, I think you are referring to T1 capacity isn't it? -- #Joseph ___ 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 dropping call?
On Fri, 10 Sep 2004 [EMAIL PROTECTED] wrote: Steve, Are you for real about the voice pitch? I am both laughing and fascinated at the same time!?!?! :P Paul Seniuk Yeah - I'm quite serious. I was trying to get busydetection working for the UK, so I had loads of debugging in the code - and my wife's voice triggered or nearly triggered the busy signal detector much much more than mine. 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] Proposal regarding the *80 vertical service code
I can't seem to get *80 to do its thing on a Zap channel. Looks like *8 is being seen by asterisk first, and *80 is basically inaccessible. What *80 is intended to do, by the documentation on the wiki and by inspection of the source code, is add the last callerid to the blacklist. Looking at the source, I see the same behavior coded in chan_zap, chan_mgcp, and chan_skinny. While *8 isn't hard-coded here, it does seem to be hard-coded in res_features, or at least has the default pickup extension defined there. Looking at the non-asterisk-specific CLASS/VSC page on the wiki, the normal behavior of *80 is defined as Selective Call Rejection Deactivation, which doesn't seem to jive with asterisks intended *80 behavior. On the other hand, *60 is defined as Selective Call Rejection Activation, which does seem to make some sense... Given the above, wouldn't it make sense to move this feature to *60? It wouldn't be 'blocked' by the default call pickup extension, and it would align more logically with the standard VSC definitions... Rob ___ 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] Number of TDM405 Cards in one server
I will not be using all of the T1s for voice. I will be using a combination of voice and data and I don't expect that all of the lines will ever be full. Since the people how answered only recommend 1 TE4**P card (thanks Steven) in a box I imagine that the solution is to setup peering between separate asterisk boxes in order to create a single overall application. So if I did do two cards any recommendations on whether I should use the 3.3v or 5.0v cards? Or on motherboard/memory/cpu specs? Obviously I would make sure that there are plenty of IRQs on the motherboard to handle the cards. Michael -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Critchfield Sent: Friday, September 10, 2004 2:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Number of TDM405 Cards in one server First, START NEW MESSAGES. don't respond to something totally different and then remove the contents. You message has NOTHING to do with the message your mail client said you responded to. In-Reply-To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] 7 Dialing gives a busy signal On Fri, 2004-09-10 at 14:39, [EMAIL PROTECTED] wrote: Has any one put 3 or more TDM405P or TDM410P cards in a single server? I would like to fit as many as 6 into one box. Do you want a TDM400 series card or a TE400 series card. Quick mention of Digium part numbers, TDM/S/X are analog cards, T/E/TE are T1 or E1 or T1 and E1 capable cards. the first number is a port capacity. From Digiums site, you get this The Wildcard TDM400P is a half-length PCI 2.2 compliant card that supports from one to four telephone interfaces for connecting analog telephones or analog lines to a PC. ... The naming convention for the TDM bundles is as follows: TDM X Y B. Where TDM denotes that the card is TDM, X denotes the number of FXS modules, Y denotes the number of FXO modules, and B indicates that that this product is a bundle. So you see there isn't a TDM405P or TDM410P. There are however TE405P and TE410P cards. When you get to T1 or E1 configurations, you shouldn't look at more than 2 cards per server, and 2 cards should probably be only undertaken with extreme care and caution. Having a simple hardware failure take down 96 lines is bad, but not as bad as taking down 192 or 288. If you are routing 288 calls, your downtime cost to repair a single box will quickly exceed the cost of redundant servers. -- 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 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] Valet Park Application
No it doesn't/shouldn't.. If a call is already parked in that location you shouldn't be able to complete the transfer and you'll have to press resume and try again. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Kevin Sent: Friday, September 10, 2004 3:47 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Valet Park Application I love the functionality of the Valet Park Application. I have a question regarding its operation. The problem I am having when there is a call already parked on specific park extension. If a caller uses 'blind' transfer on a Cisco Phone the caller gets disconnected. Can any offer any suggestions on how to prevent the transfer from taking place? ___ 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] Definity - Asterisk w/callerid
Hi there, So I've finally got our Definity and * box talking back and forth, but can't figure out how get callerid sent from the Definity to *. Has anyone had any success with this? I've tried every combination of zapata.conf variables pertaining to callerid with the same results: Accepting call from '' to '' on channel 0/1, span 1. On the SIP phones I receive the callerid as 'asterisk'. Is this an * default when none is available? Callerid gets sent to the Definity from * just fine, (although there's some funky character prepended to the id on my 8411d phone) All the send id/number options are enabled on the Definity's trunk definition. Here are my current configs: (Using PRI w/T100P -TN464 + 120A2 CSU) zaptel.conf: span=1,1,0,esf,b8zs bchan=1-23 dchan=24 (also tried d4 with same results) loadzone=us defaultzone=us zapata.conf: [channels] context = default switchtype = national overlapdial = no ;musiconhold = default signalling = pri_net ;rxwink = 300 ;callwaiting = yes ;callwaitingcallerid = yes ;threewaycalling = yes ;transfer = yes ;cancallforward = yes ;callreturn = yes ;echocancel = yes ;echocancelwhenbridged =yes ;rxgain = 0.0 ;txgain = 0.0 group = 1 ;immediate = no ;hidecallerid = no usecallerid = yes callerid = asreceived ;restrictcid = no ;usecallingpres = yes channel = 1-23 Thanks, Rob ___ 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 dropping call?
On 10 Sep 2004 at 22:51, [EMAIL PROTECTED] wrote: On Fri, 10 Sep 2004 [EMAIL PROTECTED] wrote: Steve, Are you for real about the voice pitch? I am both laughing and fascinated at the same time!?!?! :P Paul Seniuk Yeah - I'm quite serious. I was trying to get busydetection working for the UK, so I had loads of debugging in the code - and my wife's voice triggered or nearly triggered the busy signal detector much much more than mine. Yeah see it should be looking for the tone and the cadences set in indications.conf. Otherwise why are we not just putting in a frequency in Hz? Surely this is a bug? Oh and btw no this shouldn't apply to the SIP phone. Matt Riddell http://www.sineapps.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
