Re: [Asterisk-Users] Outbound fax solution
Fax over VoIP is just not reliable in my opinion. I'd run with doing it directly to PSTN as the other poster suggested or via Hylafax. We've used Hylafax behind Asterisk very succesfully in the past. SimonOn 10/29/05, KARIM MOUSLI [EMAIL PROTECTED] wrote: my problem is to triger the transfer to sip provideri always get worng number error*** REPLY SEPARATOR***On 28/10/2005 at 20:27 Chris Mason (Lists) wrote:Teliax works for me, generally. I don't know why but no other provider does. I suspect the other translate to G729 and send SIP.--Chris MasonNetConcepts(264) 497-5670 Fax: (264) 497-8463Int:(305) 704-7249 Fax: (815)301-9759Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED]___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] SPA3000 as trunk - no caller ID - solved
Well, we figured it out. It wasn't a factory reset that fixed it either.
Here is the info:
Corrected article:
http://voipspeak.net/index.php?option=com_contenttask=viewid=24
The change that got it working was in the Peer Details. We said to put the
IP address of the asterisk server in the host field, but changing it to the
IP address of the SPA-3000 fixed the problem.
-Kerry
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of InetUID
Sent: Thursday, October 27, 2005 9:14 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] SPA3000 as trunk - no caller ID
I've had a very similar thing on my SPA-3000 and they only way to fix it was
a full default reset on the SPA and reconfigure it from scratch 8-(
Matt.
On 27/10/05, Kerry Garrison [EMAIL PROTECTED] wrote:
Upgraded to 3.1.7
Excerpts from Asterisk Log:
Oct 27 07:43:50 DEBUG[1531]: cdr_mysql: SQL command as follows: INSERT
INTO cdr
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,du
ration
,billsec,disposition,amaflags,accountcode) VALUES ('2005-10-27
07:43:50','\Garrison Kerry\
9496799285','9496799285','s','from-sip-external',
'SIP/192.168.5.200-083279d0','','Congestion','',0,0,'NO ANSWER',3,'')
Oct 27 07:43:56 VERBOSE[1531]: -- Executing GotoIf(SIP/spa3000-8d99,
0?from-pstn-reghours|s|1:) in new stack Oct 27 07:43:56 DEBUG[1531]:
Check for res for spa3000 Oct 27 07:43:56 DEBUG[1531]: Call from user
'spa3000' is 1 out of 0 Oct 27 07:43:56 DEBUG[1531]: build_route:
Contact hop:
Oct 27 07:43:56 DEBUG[1531]: Expression is '0'
Oct 27 07:43:56 VERBOSE[1531]: -- Executing GotoIf(SIP/spa3000-8d99,
0?from-pstn-reghours|s|1:) in new stack
The log is interesting in that it actually is pushing the CID across
but then something strange is happening, if I look at my CDR it shows
the
following:
The call comes in to SIP/192.168.5.200 Source is the correct source
phone number, Clid is correct CID, Dst is s, Disposition is NO ANSWER
6-7 seconds later it there is another entry The call comes in to
SIP/spa3000 Source is now empty, Clid is spa3000, Dst is 201,
Disposition is ANSWERED
Here is a link to a screenshot of the SPA3000 settings:
http://techdatapros.com/temp/spa3000.gif
-Kerry
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Re: [Asterisk-Users] Words for the Asterisk community to live by.
On 10/28/05 03:41 Leif Madsen said the following: I was sitting at my buddies house, and noticed a little sign that he We provide service which is CHEAP, FAST PERFECT. a variation on this has been applied for a long time. CHEAP, FAST and QUALITY. pick any two. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] DTMF detection
Tole spada v DTMF zgodbo... - Original Message - From: Ryan [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Sunday, October 23, 2005 6:35 AM Subject: Re: [Asterisk-Users] DTMF detection On Sat, Oct 22, 2005 at 12:53:39PM -0600, Ryan exclaimed: snip I just setup telasip and I'm having the same issue. I captured some RTP packets and realized that when I get duplicate numbers it is because an RTP packet has arrived out of order. In all my test cases it was just one packet coming 1 packet too late, but the sequence number was correct. It seems that * instead of putting the packets back in order (using the seq numbers) makes a duplicate digit. I'm not sure if this is a bug or not (I haven't read the rfc). I found this in mantis at: http://bugs.digium.com/view.php?id=4659 Unfortunately this will require upstream providers to patch asterisk before this will work (which will happen over time). ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Sipura SPA 2000 - error using second line
Kanishka Somaratne wrote: Hi I have a Sipura SPA 2000 unit and I have configured both the lines in the unit. both the lines are configured to use 729. when I make calls from the lines independently it works great. no problem at all. when line 1 is connected and when I try to make a call using line 2 while line 1 is connected I get codec error. what could be the problem , please help. I tried this with call the other codecs as well, i still get the same error, only when i am tring to make the second active call regards kanishka kanishka, The SPA-2000 cannot support two simultaneous g729 calls. You will need to allow ulaw/alaw on both users (in sip.conf) in case it needs to fall back from g729. If you need two simultaneous g729 calls, the SPA-2100 will support them. -- Kristian Kielhofner ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Repost:Generate white noise to avoid RTP timeout
I'd like to know whether is possible to play some white noise or low level background noise to keep a connection up. One of my providers have an RTP timeout which kicks in quite quickly, and I need to know how to avoid it. Are there some known means of stopping this? Regards /Obelix This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] DTMF detection
Sorry, went on wrong address Regards, Rob. - Original Message - From: Robert Rozman [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, October 29, 2005 9:22 AM Subject: Re: [Asterisk-Users] DTMF detection Tole spada v DTMF zgodbo... - Original Message - From: Ryan [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Sunday, October 23, 2005 6:35 AM Subject: Re: [Asterisk-Users] DTMF detection On Sat, Oct 22, 2005 at 12:53:39PM -0600, Ryan exclaimed: snip I just setup telasip and I'm having the same issue. I captured some RTP packets and realized that when I get duplicate numbers it is because an RTP packet has arrived out of order. In all my test cases it was just one packet coming 1 packet too late, but the sequence number was correct. It seems that * instead of putting the packets back in order (using the seq numbers) makes a duplicate digit. I'm not sure if this is a bug or not (I haven't read the rfc). I found this in mantis at: http://bugs.digium.com/view.php?id=4659 Unfortunately this will require upstream providers to patch asterisk before this will work (which will happen over time). ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] UK Pounds and pence prompt wanted
Is there a .gsm file for announcing UK pounds and pence after the credit remaining prompt, besides the dollar and cents file? /Obelix This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 - PRI - Cisco
Hi, I got a situation with not having enough knowledge of PRI and Cisco things. I wanna make a plan as follows: An Asterisk having a digium PRI card, and a Cisco 3660 also having a PRI card are available. We have two PRI lines which are connected via one basic phone number through our phone provider, so we have logically 2*32 slots concatenated, acceissible by a single 4 digit phone number - although caller must dial 8 numbers, but only first 4 numbers matter. I want to make outgoing calls through Asterisk via PRI line, and I'm currently doing this in an easy way by pluggin one PRI line into the digium card in the Asterisk machine. I want to handle incoming calls by Cisco 3660 via PRI line to give a dial-up service to users, which is being serviced too. The thing which is making this hard is to have incoming calls in two PRI lines separated by dialed number, and route half of them to asterisk. That means, I wanna give two services at the same time, one dial-up service and one in-company VoIP service via PRI. The problem is how to route calls from Cisco 3660 to asterisk, because I can't just plug the second PRI line into asterisk's PRI card, because my phone provider simply routes the calls to my number to my lines as described above. So, the only thing which distinguishes the caller is the number he/she's dialled. The first four numbers are dedicated to my lines, but he/she dials 8 numbers, so I would have 4 numbers to decide on. I would appreciate any help anyone could give, cuz it's really needed and my phone provider won't change the way it's routing my calls! Thank you in advance, Kaveh __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: SV: [Asterisk-Users] call queue
If you can, avoid it: you want to report what *people* are doing, not telephone terminals. You lose a lot of flexibility using telephones. Use PersistentAgents instead! Bye l. In data Fri, 28 Oct 2005 14:51:56 +0200, Arne Morten Johansen [EMAIL PROTECTED] ha scritto: What about making queuemembers phones instead of agents? Queues.conf: [qeuename] .Blabla. member = SIP/PhoneName -Opprinnelig melding- Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne av Baris Simsek Sendt: 28. oktober 2005 14:44 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: [Asterisk-Users] call queue hello, I want to learn that, is it 'MUST' to login call queue? I have 3 call queues, and i want to distribute incoming call to the one of them. But i don't want to callbacklogin. Because of, after a restart, all agents have to do callbacklogin. thanks... -- Assum est, versa et manduca. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Chanisavail and IAX2
Hi, Im trying to do this: exten = s,7,ChanIsAvail(IAX2/agent) I searched google and found that on cvs-head ChanisAvail(IAX2) is not working. I need both cvs-head and ChanisAvail. Any idea? Thanks. http://lists.digium.com/pipermail/asterisk-users/2005-March/096682.html __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] GSM cards / mobile phone cards for Asterisk?
That is a roughly what you pay for GSM gateways everywhere.On 10/28/05, Daniel Varella de Oliveira [EMAIL PROTECTED] wrote: It costs here more or less R$600,00 (about US$264,55) Our friend, Dave Cotton post a message with a good price for outside of Brazil. US$295,00 is a good price, I think. I know that guy in Sao Paolo (the correct is São Paulo), that the sitehttp://www.thehightechstore.com/plugcell.html announced. His name isDouglas Prado and he is the owner of Contacto Telecom company. Contacto isthe unique distributor of Plugcell in region of São Paulo. If you contacthim, tell about me (He knows me as Daniel ex-Nooracom company in Rio de Janeiro). Maybe you can get a discount on your negotiation. hehehehe.--[ ]'sDaniel Varella de OliveiraTecnologia IP LtdaTel.: +55 (21)3139-4091 / r. 108Rio de Janeiro - Brasil www.tecnologiaip.com.brOn Friday 28 October 2005 12:22, Tomasz Chmielewski wrote: Daniel Varella de Oliveira schrieb: Tomasz,I'm from Brazil, and we are using here a solution that is based on a box where we can connect a GSM cellphone and use this directly to a phone or PBX extension.I think that you can use some Digium's card (FXS or FXO) on your server, connect this GSM box there, and route your cellphone calls through this box.There are boxes with just one channel and others up to six channels.They have a lot compatibilities with the most common cellphones. looks interesting. do you know by chance how much such a single-cell box cost (more or less)?___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michal BielickiHalo Kwadrat Sp. z o.o.http://www.asterisk.pl/http://www.openpbx.org/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Grandstream GXP-2000
On Fri, 28 Oct 2005, Erick Baum wrote: We have 50 of these phones in one location and a couple remote phones. The problem seems to be caused by the volume settings on the phone. We have noticed that the echo seems to be worse when the volume is very high on the phone (not using speakerphone). We're still testing, but that's what we've been able to come up with so far. Which end experience the echo? The phone with the loud volume, or the other end? If it is the remote end that experience echo then I would suspect acoustic coupling from the earpiece to the microphone inside the handset. If this is the case there are a few solutions: - lower the volume (duh!) - try connecting another handset with a known good decoupling of the mic/speaker - get grandstream to use the software echo canceller when using the handset as well as when on the speaker phone. Peter ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] problem with receiving faxes over cisco as5300
On Fri, Oct 28, 2005 at 12:24:28AM +0200, Florian Meister wrote: Hi, does anybody have a working sample configuration of a cisco as53xx for receiving faxes ? Sending faxes over the as5300 works fine, but if I send a fax from pstn to asterisk (over the as5300 as pstn/voip gateway) it does not work. Disable T.38 on cisco, asterisk doesn't support it... ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Webui to show registered phones
Hi all, does anyone know if there is any app/webui that can show phones that are currently registered to *. I guess this sort of funcionality counld be grabbed from the CLI with iax2 show peers and sip show peers, but having little programming knowledge wouldn't know where to start. I'm asking because we currently have several sip phones onsite and lots of remote iax2 users who would like to see availability without dialing. plugYou can try with the Flash Operator Panel/plug http://www.asternic.org , it does all sort of things including sip and iax availability (you have to enable qualify for them). Regards, -- Nicolás Gudiño Buenos Aires - Argentina ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Prblem with 723 and 729
Hi I have G729 and G723 codecs installed, I made some calling using a SIP IP phone. when I used the codecs 723 and 729 the call volume is less and the sound is little jerky, it's like call signals coming in and out. when I use gsm or G711 it works great sound quality is crystal clear. is this some thing to do with jitter buffer , is there a way to increase the volume using asterisk. tks kani ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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: Re: Sipura SPA 2000 - error using second line
Kanishka Somaratne wrote: Hi I have a Sipura SPA 2000 unit and I have configured both the lines in the unit. both the lines are configured to use 729. when I make calls from the lines independently it works great. no problem at all. when line 1 is connected and when I try to make a call using line 2 while line 1 is connected I get codec error. what could be the problem , please help. I tried this with call the other codecs as well, i still get the same error, only when i am tring to make the second active call regards kanishkakanishka,The SPA-2000 cannot support two simultaneous g729 calls. You will need to allow ulaw/alaw on both users (in sip.conf) in case it needs to fall back from g729.If you need two simultaneous g729 calls, the SPA-2100 will support them.--Kristian Kielhofner Kristian thank you very much for the reply. what codecs does SPA-2000 support simultaneously. can it support 729 on line 1 and 723 on line 2. I tried this as well it failed. please let me know what codecs it support simultaneously. tks Kani ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zyxel omni.net USB ISDN works with Asterisk
Dear Asterisk users, Can you tell me is the Zyxel omni.net USB ISDN adapter works with Linux, and more specifically, with Asterisk chan_capi? I built an Asterisk PBX test environment on my laptop with Asterisk Management Portal, one hardphone, one ATA, and one softphone. I would connect the whole thing to an ISDN (Euro) line, but because of my laptop, I can use only USB or PCMCIA solutions. Gabor ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Opinions on IAX JitterBuffer in old-school 1 .0.0?
Wasn't that terrible thanks to asterisk-update.sh, just drudgery. As to HEAD / 1.2 I'd rather let others be early adopters, thanks. I'm trying to get business done, not lay awake at night thinking how my dialplan is going to be broken because of syntax changes or waiting for a box to core dump. I know, I know, HEAD is way better but I have to be conservative - my users are still leery about VoIP and they will nit-pick about any little thing (You mean I have to press OK after I dial? That's stupid and my all-time favorite I get confused when I have three incoming calls at one, I forget who's on each line - why can't the phone help me with that, I thought these phones were so great) Straw poll: What's the stupidest complaint you've ever had to chase down? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Friday, October 28, 2005 11:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Opinions on IAX JitterBuffer in old-school 1 .0.0? On Fri, 28 Oct 2005, Colin Anderson wrote: Does sound like you have the fix - upgrade to a newer Asterisk. *groan* Yes, it did solve the problem, 100%. I upgraded a single site to 1.0.9 and call quality is perfect. Now, on to the other 29thank GOD for SSH. As its such a big job, are you SURE you wouldn't rather move them all the CVS-HEAD, which is oh-so-nearly 1.2 beta2 and then 1.2 release...? Steve ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Trying to genereate dial tone, but stop after first digit dialed.
You could use a DISA app with no password to lead you into the context. exten = *70,1,Disa(no-password|nextcontext) [nextcontext] exten = _1NXXNXX,1,Dial(...) /edg --On Wednesday, October 26, 2005 7:36 PM -0700 Jonathan Feally [EMAIL PROTECTED] wrote: I seem to be missing something here. Basically I'm trying to do what a full CO would do in terms of *70 to disable call waiting. I have a *70 exten setup, it does the work to set the extension to not take in a second call, then does a playtones(dialrecall). This works except that all digits dialed after the *70 have the tone still playing until the dialplan kicks back in for the new exten dialed. Does somebody have a work around for this? I'd prefer to not use Background. Thanks, -Jon ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CallBack Suggestion
Hi friends, I am new in asterisk, i came for CallBack purpose, i read from Voip-info.org aboue callback with asterisk and i am near to collect all information about to start developing callback system. Just i have a samall question, Is Callback needs some special hardware? i have my PSTN phone number i want to call this number after two ring the call will be disconnect and the Callback will start to call back to the caller ID and it should prompt to enter pin id which will authunticate via freeradius.if the authuntication is valid it will give some beep for dialing the international number. Any kind of suggestion will be hearty appriciated. Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 Tel: +974 - 4883068 ICQ: 276994704 YM!: abdul_zu Fax: +974 - 4883063 Doha Qatar http://www.hatif.com __ Yahoo! FareChase: Search multiple travel sites in one click. http://farechase.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Opinions on IAX JitterBuffer in old-school 1 .0.0?
On Sat, 29 Oct 2005, Colin Anderson wrote: I know, I know, HEAD is way better Not suggesting you put it on 30 machines without testing. But it really is better you know... Steve ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Getting correct CDR info
I am using queue and dynamic (pstn) agents that log in with AgentCallBackLogin. My problem is that CDR does not make a complete picture of the calls. Below I put a truncated CDR of one call. Here, user's call (3) is forwarded to one agent (1) who hasn't answered in 22 sec, then to Agent/302 (2) who answered after 13 sec. For the second agent only the time before pick up shows up in the CDR. The actual duration of the answered outbound call to Agent/302 (the only one which is paid!!!) is lost... dstchann/lastapp/duration/billsec/disposition 1. Zap/1-1Dial 22 0 NO ANSWER 2. Zap/1-1Dial 13 0 ANSWERED 3. Agent/302 Queue 49 49 ANSWERED Tracking down the problem, I have found that queues use Local channel to place calls to agents. Local channels are destroyed when bridged to an originating channel. Therefore, the call (2) is destroyed and call 3 shows duration=billsec as it should be for that call since it is answered by the system. I have made a hack, that allows for correct reading of duration/billsec. This is achieved with an extra Local channel with /n option that saves the original channel from bridging to the Agent's channel having both legs recorded in CDR. 1. Zap/1-1Dial 12 0 NO ANSWER 2. Zap/1-1Dial 16 7 ANSWERED 3. Agent/302 Queue 69 69 ANSWERED This solution costs an extra Local channel for each call, extra contexts in the dialplan, etc. Is there any way to make queue not to call agents over Local channel or say queue to use the /n option? Thanks. Cleo __ Yahoo! FareChase: Search multiple travel sites in one click. http://farechase.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Zyxel omni.net USB ISDN works with Asterisk
Gabor Horvath wrote: Can you tell me is the Zyxel omni.net USB ISDN adapter works with Linux, and more specifically, with Asterisk chan_capi? I don't know, but I built an Asterisk PBX test environment on my laptop with Asterisk Management Portal, one hardphone, one ATA, and one softphone. I would connect the whole thing to an ISDN (Euro) line, but because of my laptop, I can use only USB or PCMCIA solutions. there are AVM PC-cards available which do work with CAPI. HTH -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] iax softphones
Dante's DIAX is pretty good IMHO. Peter Hector medina wrote: can anyone recomend a good iax softphone?? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ANNOUNCEMENT: Asterisk-Java 0.2-rc2 released
Asterisk-Java 0.2-rc2, a Java control for the Asterisk PBX, has been released. The Asterisk-Java package consists of a set of Java classes that allow you to easily build Java applications that interact with an Asterisk PBX Server. Asterisk-Java supports both interfaces that Asterisk provides for this scenario: The FastAGI protocol and the Manager API. The 0.2-rc2 release candidate focuses on the new features of the Asterisk 1.2 series though it still supports Asterisk 1.0.x. The changes include * Bug fix for variables in OriginateAction (AJ-15) * Support for FaxReceived event from spandsp (AJ-20) * Better integration with Spring Framework via SimpleMappingStrategy and AGIServerThread Asterisk-Java is used in several commercial environments and by the following Open Source projects: * Asterisk-IM A plugin for the Jive Messenger XMPP (jabber) server. It provides integrated presence between your IM client and phone, notification of incoming calls by IM and originate calls from supported IM clients. * Asterisk Desktop Manager (ADM) A desktop application that will allow for automatic on-call volume reduction, one click dial from clipboard, integrated phonebook and more. Asterisk-Java is available under Apache 2.0 license at http://www.asteriskjava.org signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] dropping extra frame of G.729 since we already have a VAD frame at the end
Hi I get the following error when i make a call from 729 to 729 dropping extra frame of G.729 since we already have a VAD frame at the end I am using asterisk 1.0.9, there was a patch for CVS vertion, is there a patch for 1.0.9 tks kani ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] CallBack Suggestion
Hello. You should not need any special hardware for callback. You will (obviously) need card to connect your box to the pstn. Do you have something setup with freeradius already? If not, you could quite easily setup something like this with ASTCC. I have a callback script @ www.aleph-com.net/astpp. Somewhere there. It is way more complicated than you need but you can cut out all the user interaction stuff. Darren Wiebe [EMAIL PROTECTED] Abdul Lateef wrote: Hi friends, I am new in asterisk, i came for CallBack purpose, i read from Voip-info.org aboue callback with asterisk and i am near to collect all information about to start developing callback system. Just i have a samall question, Is Callback needs some special hardware? i have my PSTN phone number i want to call this number after two ring the call will be disconnect and the Callback will start to call back to the caller ID and it should prompt to enter pin id which will authunticate via freeradius.if the authuntication is valid it will give some beep for dialing the international number. Any kind of suggestion will be hearty appriciated. Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 Tel: +974 - 4883068 ICQ: 276994704 YM!: abdul_zu Fax: +974 - 4883063 Doha Qatar http://www.hatif.com __ Yahoo! FareChase: Search multiple travel sites in one click. http://farechase.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Astricon - materials
marek cervenka wrote: hi, will be somewhere materials (videos, presentations) from astricon? Registered attendees will get information about the material soon. No videos where recorded this year. any chance for not registered? astricon was too far for me (europe) my english is terrible, but i can read if you have the materials, it's wrong to not use it (it can be for money) The 1.2 presentation I made together with Kevin has been available for a while at http://www.astricon.net/asterisk1-2/ and will be updated soon. nice intro to 1.2, thanks! --- Marek Cervenka Centrum Vypocetni Techniky CVT - http://cvt.fpf.slu.cz FPF SLU OPAVA - http://www.fpf.slu.cz LCNA- http://lcna.slu.cz === ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Grandstream GXP-2000
It does appear to be the remote party that is hearing the echo as a result of the loud volume on the other end. They actually had a few people calling in from the outside report that they could hear their echo. When they turned down the volume on the Grandstream, the echo seemed to go away. So I will bring up the possibility of using the AEC when using the handset. Thanks, Erick On 10/29/05, Peter Svensson [EMAIL PROTECTED] wrote: On Fri, 28 Oct 2005, Erick Baum wrote: We have 50 of these phones in one location and a couple remote phones. The problem seems to be caused by the volume settings on the phone. We have noticed that the echo seems to be worse when the volume is very high on the phone (not using speakerphone). We're still testing, but that's what we've been able to come up with so far.Which end experience the echo? The phone with the loud volume, or theother end? If it is the remote end that experience echo then I wouldsuspect acoustic coupling from the earpiece to the microphone inside the handset.If this is the case there are a few solutions:- lower the volume (duh!)- try connecting another handset with a known good decoupling of the mic/speaker- get grandstream to use the software echo canceller when using the handset as well as when on the speaker phone.Peter___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Sipura SPA 2000 - error using second line
That is simple: Sipura 2000 is not able to handle two calls in G729 at the same time. The best solution is to move both to G726.32 using the latest firmware, that is going to fix the problem The box was designed thinking on one G729 and one G711 (Fax applications) but most of the cases that was not the use for it. Regards, Carlos Alperin From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kanishka Somaratne Sent: Saturday, October 29, 2005 1:05 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Sipura SPA 2000 - error using second line Hi I have a Sipura SPA 2000 unit and I have configured both the lines in the unit. both the lines are configured to use 729. when I make calls from the lines independently it works great. no problem at all. when line 1 is connected and when I try to make a call using line 2 while line 1 is connected I get codec error. what could be the problem , please help. I tried this with call the other codecs as well, i still get the same error, only when i am tring to make the second active call regards kanishka ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Credit card machines, Asterisk and Digium any issues?
Hi, We want to be able to use any of 4 outgoing lines (POTS connected to FXO cards on a Digium TDM400P card) for a credit card machine (you know those little machines that have a phone line attached and are in many small establishments) We have 4 FXS for analog phones and 2 credit card machines. Are there any issues that I might encounter doing this? I assume it is just like a analog fax machine searching for an available line to go out on. Also if any one has done this and can shared their configuration I would really appreciate it. Thanks ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] DTMF recognition unreliable or absent, ringing lost on 2nd half of 3-way call.
On Oct 29, 2005, at 1:20 AM, Ryan wrote: On Thu, Oct 27, 2005 at 02:10:51PM -0400, Dave Grey exclaimed: The digits seem to be either not recognized at all or recognized incorrectly better than half the time. [...] I dial 7056 and it sees 7055, I dial 7056 again and it sees 75, I dial 7056 a third time and it sees 706, etc. Seems random and all over the place. Packet loss and/or ordering? This is a known issue that is fixed in CVS HEAD. Search for my prior emails to track down the bug numbers. Unfortunately this is going to require the upstream providers to upgrade to truly fix the issue. Basically the RTP packets are coming out of order and have the wrong sequence numbers. I started on a band-aid solution using ip_queue, but I have not had time to finish it up. I will post here if it ever materializes. Thanks for the response. I am using CVS HEAD, unfortunately, and still seeing these problems. When I posted the above, I was using code checked out and built on 2005-10-11. Currently, I am running: Asterisk CVS HEAD built by [EMAIL PROTECTED] on a Power Macintosh running Darwin on 2005-10-27 19:51:45 UTC. The behavior doesn't seem to have changed for me. Is there anything I should be turning on (or off) to improve matters? lyd ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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: stupidest complaint (forked from jitterbuffer thread)
On Oct 29, 2005, at 8:09 AM, Colin Anderson wrote: [...] my users are still leery about VoIP and they will nit-pick about any little thing (You mean I have to press OK after I dial? That's stupid and my all-time favorite I get confused when I have three incoming calls at one, I forget who's on each line - why can't the phone help me with that, I thought these phones were so great) Straw poll: What's the stupidest complaint you've ever had to chase down? IMO, neither of these examples are stupid, they are exactly the sort of thing that * and the whole concept of open telephony/ communications hardware and software are intended to address. A system that can be whatever the *users* want it to be is what this is all about, no? lyd ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] x100p (FXO) not being seen by asterisk (is my bestguess) .
Hello All , On Fri, 28 Oct 2005, Mr. James W. Laferriere wrote: On Thu, 27 Oct 2005, Mr. James W. Laferriere wrote: On Thu, 27 Oct 2005, Phil Pritchard wrote: only new to asterisk, but have had some hardware exp. stay away from irq9 its tied to irq2 and will always be shared, Paul has the go.. in bios disable serial and or usb (if not using) and make sure irda is not enabled. another one is the lpt port if your not using that, there is another irq you can steel.. ALL I mean all serial/parrallel/...'everything I can find'... has been turned off in the bios . And I have recompiled a kernel with those same items turned off in it . That d??ned module wants to load at irq 9 no matter what I do . Of course there is no way to set irq's to a particular pci slot in the bios . Does anyone now howto set irq say at the boot: or in modprobe.conf ? dont share interrupts, as a rule(if you can help it)... it usually leads to system instability and usually under load. Quite well understand this point . Have heard it on this list many times . And am doing my best NOT too . UBCD ...(www.ultimatebootcd.com). has some nice tools that can probe a system to give a second appinion on interrupt conflicts, ram and hard drive errors. its my best tool for hardware problems.. IMO , The mirrors have the su??iest download schemes I have seen in some time .\IMO I have yet to burn that image but as soon as I do I'll boot it on that piece of junk I bought for near next to nothing . Which is almost what it is worth , Nothing . Thank you for your input , Every bit helps . JimL Finally got that da??ed wcfxo to load on a irq by itself (*). Had to turn off the last item of the onbord devices the ether buy an ether card to get connectivity . But even with the suggestion by 'Paul' to use a two line cord finally using a singular irq , The config's I sent last time have not changed . The x100p/wcfxo combination see the line ringing (**) . But asterisk does NOT see it on the console nor does it pick up the line . Quite frustrating when everything should be ok per every example I've seen still nothing positive to show for it . ANY suggestions/questions/... Please pipe up . Tia , JimL For everybodies info , Make sure that there isn't an entry like ... noload = chan_zap.so in /etc/asterisk/modules.conf . That was what the problem was all along . Tnx to all who helped . JimL -- +--+ | James W. Laferriere | SystemTechniques | Give me VMS | | NetworkEngineer | 3542 Broken Yoke Dr. | Give me Linux | | [EMAIL PROTECTED] | Billings , MT. 59105 | only on AXP | +--+ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Meetme streaming a recording
Hi All, * noob here :) What I'm trying to do is have a meetme number that streams a recording. Let's say there was a company pressconference live that people could join and then later, a cleaned-up version was avaialable. Live is fine, I just set up a conference where everyone comes in just monitoring it. How could I set up the later one though so the only 'person' that can be heard is the recording? Thanks for any help! Bob ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: stupidest complaint (forked from jitterbuffer thread)
On Saturday 29 October 2005 13:00, Dave Grey wrote: IMO, neither of these examples are stupid, they are exactly the sort of thing that * and the whole concept of open telephony/ communications hardware and software are intended to address. A system that can be whatever the *users* want it to be is what this is all about, no? I disagree. Technology needs to be adapted to people, yes, but most people don't have a clue what they really want. They have some vague ideas and expect that you are able to meet those needs, even though half the time they contradict one another. Having to press OK is a human interface issue. I agree that it's not acceptable. The three lines thing ... depends on what they had in mind. Showing who's on what line is certainly trivial but do they want more? -A. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: stupidest complaint (forked from jitterbuffer thread)
Andrew Kohlsmith wrote: On Saturday 29 October 2005 13:00, Dave Grey wrote: IMO, neither of these examples are stupid, they are exactly the sort of thing that * and the whole concept of open telephony/ communications hardware and software are intended to address. A system that can be whatever the *users* want it to be is what this is all about, no? I disagree. Technology needs to be adapted to people, yes, but most people don't have a clue what they really want. They have some vague ideas and expect that you are able to meet those needs, even though half the time they contradict one another. Having to press OK is a human interface issue. I agree that it's not acceptable. The three lines thing ... depends on what they had in mind. Showing who's on what line is certainly trivial but do they want more? Find me an affordable color LCD with USB interface and I can use my linksys NSLU2 to give them what they want. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: stupidest complaint (forked from jitterbuffer thread)
On Saturday 29 October 2005 13:57, Paul wrote: Find me an affordable color LCD with USB interface and I can use my linksys NSLU2 to give them what they want. Does it have to be colour? What do you define as Affordable? -A. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Pounds and pence prompt wanted
Obelix wrote: Is there a .gsm file for announcing UK pounds and pence after the credit remaining prompt, besides the dollar and cents file? /Obelix http://www.voip-info.org/tiki-index.php?page=Asterisk+sound+files+international I peeked into the archives from: http://www.desktop2door.com/asterisk/ and http://www.g7ltt.com/VoIP/vmfiles.html Found pound and pounds but no pence. I could have missed it though. You could also add your own voice to the UK male voice archive. That's what I did when I didn't find philippine(s).gsm My voice is nowhere near Allison's though. O.T. Is the Asterisk the Gaul comics still in circulation? It's been years since I read the series... -- JP Carballo http://www.netfone2x.com Bringing the world closer. Programmers confuse Christmas and Halloween because DEC 25 = OCT 31. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Pounds and pence prompt wanted
Yes there has just been a new release of Asterix (the Gaul has x at the end) . JP Carballo wrote: Obelix wrote: Is there a .gsm file for announcing UK pounds and pence after the credit remaining prompt, besides the dollar and cents file? /Obelix http://www.voip-info.org/tiki-index.php?page=Asterisk+sound+files+international I peeked into the archives from: http://www.desktop2door.com/asterisk/ and http://www.g7ltt.com/VoIP/vmfiles.html Found pound and pounds but no pence. I could have missed it though. You could also add your own voice to the UK male voice archive. That's what I did when I didn't find philippine(s).gsm My voice is nowhere near Allison's though. O.T. Is the Asterisk the Gaul comics still in circulation? It's been years since I read the series... -- voipGATE.com Support Team ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Outbound fax solution
Hylafax? At 06:27 PM 10/28/2005, you wrote: Got a client building a system that needs to send out hundreds of faxes per day (not, not junk faxes). We have just implemented an asterisk server for the client for their office and they asked if there was an outbound fax solution that would utilize VOIP providers ($0.02/minute) instead of internet based fax providers ($0.08/page). Does anyone have any thoughts on this? -Kerry ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Set outgoing MSN with chan_capi-cm
Hi List,
I setup my first asterisk today on a gentoo box.
I have a AVM Fritz PCI 2.1 running with the Fritz binary driver fcpci.
I use chan_capi-cm 0.6.0.
Calling in and out works fine, but I am unable to set the outgoing MSN.
I have setup 2 MSNs of my ISDN BRI for use with asterisk - regarding
call acception from ISDN everything works fine, only the subscribed
numbers are seen by asterisk.
I added msn=myMSN1,myMSN2 to capi.conf but asterisk always uses the
default MSN of the isdn line when calling out.
I also tried setting the Callerid per Call with the syntax described in
the README file
Dial(CAPI/g1/myMSN1,${EXTEN:1}) but this fails when dialing with CAPI
INFO 0x349c: Invalid number format
Any ideas ??
Oliver
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Re: [Asterisk-Users] Re: stupidest complaint (forked from jitterbuffer thread)
Andrew Kohlsmith wrote: On Saturday 29 October 2005 13:57, Paul wrote: Find me an affordable color LCD with USB interface and I can use my linksys NSLU2 to give them what they want. Does it have to be colour? What do you define as Affordable? Colour and backlighting are highly desirable. Size is important. Remember that such a display is often viewed while engaged in conversation. I would consider it affordable if the complete system would cost a lot less than one based on via mainboards with external svga lcd display. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Outbound fax solution
Jeff Herring wrote: Hylafax? At 06:27 PM 10/28/2005, you wrote: Got a client building a system that needs to send out hundreds of faxes per day (not, not junk faxes). We have just implemented an asterisk server for the client for their office and they asked if there was an outbound fax solution that would utilize VOIP providers ($0.02/minute) instead of internet based fax providers ($0.08/page). Does anyone have any thoughts on this? If the client is in the US you might be able to get him a good rate on landline long distance. Figure out how many lines you need to deliver all faxes on time and use hylafax with modems. THat may not sound very hightech but hylafax works. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] I need GoIAX or VoipBuster [EMAIL PROTECTED] examples?
On Sat, October 29, 2005 1:19, Kerry Garrison said: http://voipspeak.net has GOIax example for AMP. -Kerry Sorry, couldn't find it... Do you have an exact url? I only found IAX.cc/Sixtel, Teliax and Broadvoice samples... -- Francesco Peeters GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Set outgoing MSN with chan_capi-cm
On Sat, 29 Oct 2005, Oliver Welter wrote:
Hi List,
I setup my first asterisk today on a gentoo box.
I have a AVM Fritz PCI 2.1 running with the Fritz binary driver fcpci.
I use chan_capi-cm 0.6.0.
Calling in and out works fine, but I am unable to set the outgoing MSN.
I have setup 2 MSNs of my ISDN BRI for use with asterisk - regarding call
acception from ISDN everything works fine, only the subscribed numbers are
seen by asterisk.
I added msn=myMSN1,myMSN2 to capi.conf but asterisk always uses the default
MSN of the isdn line when calling out.
I also tried setting the Callerid per Call with the syntax described in the
README file
Dial(CAPI/g1/myMSN1,${EXTEN:1}) but this fails when dialing with CAPI INFO
0x349c: Invalid number format
where did you read this?
with chan_capi-cm-0.6, you can just set the callerid (SetCallerID()). The
msn= setting is not used any more and the dialstring above is not correct.
With new chan_capi from CVS-HEAD on sourceforge, you can also set another
MSN than the callerid.
Armin
Any ideas ??
Oliver
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[Asterisk-Users] zaptel + RH3?
Hmmm we have this TDM400 with one FXO module on it, we're using it for testing purposes. I can: modprobe zaptel modprobe wctdm and these get load fine as stated in /var/log/messages Oct 29 05:09:17 monitor kernel: Module 0: Installed -- AUTO FXO (FCC mode) Oct 29 05:09:17 monitor kernel: Module 1: Not installed Oct 29 05:09:17 monitor kernel: Module 2: Not installed Oct 29 05:09:17 monitor kernel: Module 3: Not installed However when I run ztcfg - I get an error: [EMAIL PROTECTED] dev]# ztcfg -vvv Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. Notice: Configuration file is /etc/zaptel.conf line 147: Unable to open master device '/dev/zap/ctl' See how it says it can't open the master device /dev/zap/ctl. Well this is because that device doesn't exist.. [EMAIL PROTECTED] dev]# ls -ald zap* crw--- 1 root root 196, 1 Oct 29 05:09 zap1 crw--- 1 root root 196, 2 Oct 29 05:09 zap2 crw--- 1 root root 196, 3 Oct 29 05:09 zap3 crw--- 1 root root 196, 4 Oct 29 05:09 zap4 crw--- 1 root root 196, 254 Oct 29 05:09 zapchannel crw--- 1 root root 196, 0 Oct 29 05:09 zapctl crw--- 1 root root 196, 255 Oct 29 05:09 zappseudo crw--- 1 root root 196, 253 Oct 29 05:09 zaptimer I can manualy create the /dev/zap directory and then make symbolic links to /dev/zap/ctl using /dev/zapctl and make a symbolic link to /dev/zap/channel by linking to /dev/zapchannel etc... This will make the card work just fine, I have tested this on another linux box and the same thing happened. If I add this symbolic link creation into the statup scripts then like I said zaptel working fine, however this is obviously not the right way to fix this issue. Suggestions? -=Linsys=- IntrusionSec.com #1 Hacker Gamez Web Site On the Internet http://www.intrusionsec.com [EMAIL PROTECTED] - When Your Life Flashes Before Your Eyes When You Die, Does That Include The Part Where Your Life Flashes Before Your Eyes? - ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] I need GoIAX or VoipBuster [EMAIL PROTECTED] examples?
On Sat, October 29, 2005 21:10, Francesco Peeters said: On Sat, October 29, 2005 1:19, Kerry Garrison said: http://voipspeak.net has GOIax example for AMP. -Kerry Sorry, couldn't find it... Do you have an exact url? I only found IAX.cc/Sixtel, Teliax and Broadvoice samples... /IGNORE I just found it on the Nerd Vittles site... :-) Gonna test it right now! THX! -- Francesco Peeters GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] I need GoIAX or VoipBuster [EMAIL PROTECTED] examples?
On Sat, October 29, 2005 21:58, Francesco Peeters said: On Sat, October 29, 2005 21:10, Francesco Peeters said: /IGNORE I just found it on the Nerd Vittles site... :-) Gonna test it right now! THX! I am so bloody embarrased! I decided to make a capture on my firewall and found that data was coming in the LAN, but not going out the WAN port... Investigating why that might be, I then found out that my IAX2 service in the firewall was set for TCP instead of UDP! *blushes* I changed that and it instantly worked for both GoIAX and VoipBuster (though they do give me some crap about no credits...) Sorry for the disturbance on list! ;-) -- Francesco Peeters GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Set outgoing MSN with chan_capi-cm
Hello Armin,
I also tried setting the Callerid per Call with the syntax described in the
README file
Dial(CAPI/g1/myMSN1,${EXTEN:1}) but this fails when dialing with CAPI INFO
0x349c: Invalid number format
where did you read this?
http://cvs.sourceforge.net/viewcvs.py/chan-capi/chan_capi/README?view=markup
with chan_capi-cm-0.6, you can just set the callerid (SetCallerID()). The
msn= setting is not used any more and the dialstring above is not correct.
With new chan_capi from CVS-HEAD on sourceforge, you can also set another
MSN than the callerid.
I now have found this solution (stkn on IRC pointet me that way)
exten = _9.,1,SetCallerPres(allowed)
exten = _9.,2,SetCallerid(614890)
exten = _9.,3,Dial(CAPI/g1/${EXTEN:1})
This actually does the Job - if this is not the most appropriate
approach, please point me to a better direction :9
regards
Oliver
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[Asterisk-Users] I give up - Help with TE410P
I'm trying to install two TE410P's in one box. Would like to get 3 total. I can always get one card to work. If I install only one card, I will get green lights on all ports when loop back plugs installed - everything is perfect... If I install 2 cards, I'll get yellow alarm on span 2 and 6 and 7. Flashing Red alarm on span 1,3,4,5 and 8. There is no error messages that I can find. What is the correct procedure for installing these cards? Can you give me a step-by-step on how to install these cards?I've been working on this for a week and getting frustrated.\ TIA Bart Some info (not sure what else might be needed): # cat /proc/interrupts CPU0 0:3593335 XT-PIC timer 1:518 XT-PIC i8042 2: 0 XT-PIC cascade 5: 30 XT-PIC aic7xxx 7: 37562 XT-PIC eth0 8: 1 XT-PIC rtc 9:3257067 XT-PIC acpi, wctdm 10:3257002 XT-PIC wct4xxp 11:3260152 XT-PIC wct4xxp 14: 13296 XT-PIC ide0 NMI: 0 ERR: 0 # lspci -v 00:00.0 Host bridge: Broadcom GCNB-LE Host Bridge (rev 32) Flags: fast devsel 00:00.1 Host bridge: Broadcom GCNB-LE Host Bridge Flags: fast devsel 00:02.0 SCSI storage controller: Adaptec AIC-7892P U160/m (rev 02) Subsystem: Adaptec AIC-7892P U160/m Flags: bus master, 66Mhz, medium devsel, latency 32, IRQ 5 BIST result: 00 I/O ports at d800 [disabled] [size=256] Memory at fe00 (64-bit, non-prefetchable) [size=4K] Expansion ROM at febe [disabled] [size=128K] Capabilities: [dc] Power Management version 2 00:03.0 Ethernet controller: Broadcom Corporation NetXtreme BCM5702X Gigabit Ethernet (rev 02) Subsystem: ASUSTeK Computer Inc.: Unknown device 80a9 Flags: bus master, 66Mhz, medium devsel, latency 64, IRQ 7 Memory at fd80 (64-bit, non-prefetchable) [size=64K] [virtual] Expansion ROM at febd [disabled] [size=64K] Capabilities: [40] PCI-X non-bridge device. Capabilities: [48] Power Management version 2 Capabilities: [50] Vital Product Data Capabilities: [58] Message Signalled Interrupts: 64bit+ Queue=0/3 Enable- 00:04.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device b100:0001 Flags: bus master, medium devsel, latency 32, IRQ 9 I/O ports at d400 [size=256] Memory at fd00 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 00:05.0 Communication controller: Unknown device d161:0410 (rev 02) Flags: bus master, medium devsel, latency 32, IRQ 10 Memory at fc80 (32-bit, non-prefetchable) [size=128] 00:06.0 Communication controller: Unknown device d161:0410 (rev 02) Flags: bus master, medium devsel, latency 32, IRQ 11 Memory at fc00 (32-bit, non-prefetchable) [size=128] 00:09.0 VGA compatible controller: ATI Technologies Inc Rage XL (rev 27) (prog-if 00 [VGA]) Subsystem: ATI Technologies Inc Rage XL Flags: bus master, stepping, medium devsel, latency 32, IRQ 10 Memory at fb00 (32-bit, non-prefetchable) [size=16M] I/O ports at d000 [size=256] Memory at fa80 (32-bit, non-prefetchable) [size=4K] Expansion ROM at feba [disabled] [size=128K] Capabilities: [5c] Power Management version 2 00:0f.0 ISA bridge: Broadcom CSB6 South Bridge (rev a0) Subsystem: Broadcom: Unknown device 0201 Flags: bus master, medium devsel, latency 32 00:0f.1 IDE interface: Broadcom CSB6 RAID/IDE Controller (rev a0) (prog-if 8a [Master SecP PriP]) Subsystem: Broadcom: Unknown device 0212 Flags: bus master, medium devsel, latency 64 I/O ports at ignored I/O ports at ignored I/O ports at ignored I/O ports at ignored I/O ports at 8800 [size=16] 00:0f.3 Host bridge: Broadcom GCLE-2 Host Bridge Subsystem: Broadcom: Unknown device 0230 Flags: bus master, medium devsel, latency 0 # uname -a Linux asterisk1.local 2.6.9-22.0.1.EL #1 Thu Oct 27 12:26:11 CDT 2005 i686 i686 i386 GNU/Linux # cat /proc/cpuinfo processor : 0 vendor_id : GenuineIntel cpu family : 15 model : 2 model name : Intel(R) Pentium(R) 4 CPU 2.80GHz stepping: 9 cpu MHz : 2799.826 cache size : 512 KB fdiv_bug: no hlt_bug : no f00f_bug: no coma_bug: no fpu : yes fpu_exception : yes cpuid level : 2 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic mtrr pge mca cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe cid xtpr bogomips: 5521.40 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list
Re: [Asterisk-Users] I give up - Help with TE410P
On Saturday 29 October 2005 18:06, Bart Fisher wrote: I'm trying to install two TE410P's in one box. Would like to get 3 total. I can always get one card to work. You are adjusting the 'ident' rotary switch on the others, right? -A. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] I give up - Help with TE410P
Yep - that was easy part :) and these are T1 (D4, AMI, SF, and EM Wink) BTW Bart - Original Message - From: Andrew Kohlsmith [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Saturday, October 29, 2005 3:09 PM Subject: Re: [Asterisk-Users] I give up - Help with TE410P On Saturday 29 October 2005 18:06, Bart Fisher wrote: I'm trying to install two TE410P's in one box. Would like to get 3 total. I can always get one card to work. You are adjusting the 'ident' rotary switch on the others, right? -A. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 + RH3?
[EMAIL PROTECTED] wrote on 10/29/2005 04:01:26 PM: If I add this symbolic link creation into the statup scripts then like I said zaptel working fine, however this is obviously not the right way to fix this issue. Are you doing make config when you compile Zaptel? It does all of this for you... Tim Massey ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] I give up - Help with TE410P
On Saturday 29 October 2005 18:19, Bart Fisher wrote: Yep - that was easy part :) and these are T1 (D4, AMI, SF, and EM Wink) BTW Ok, well I'll go for the obvious question: have you contacted Digium technical assistance? You have paid for support within the price of the card. -A. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] TDM01B vs. X100P
Rusty, You do defenitely not want the X100P it's discontinued and rightly so, horrible card. /Danny Rusty Dekema wrote: Hi, I apologize in advance if this is a stupid question, but I have not been able to find an answer by searching the web. I would like to add an FXO port or two to my Asterisk setup, and I am wondering if there is any good reason to spend $120 on a TDM01B or $180 on a TDM02B instead of paying $9.95 or $19.90 for one or two new, genuine, unopened X100P cards on eBay. I am not particularly worried about running out of PCI slots, as I don't envision ever needing to add any other line cards to this machine. However, if there is some kind of substantial quality compatibility difference between the two cards, I would like to know about this before wasting (even a small amount of) money on X100Ps. Thanks, Rusty ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 + RH3?
If you have 2.6 kernel and/or are using udev, you want to check README.Linux26 and README.udev in zaptel, if you haven't already done so. Hmmm we have this TDM400 with one FXO module on it, we're using it for testing purposes. I can: modprobe zaptel modprobe wctdm and these get load fine as stated in /var/log/messages Oct 29 05:09:17 monitor kernel: Module 0: Installed -- AUTO FXO (FCC mode) Oct 29 05:09:17 monitor kernel: Module 1: Not installed Oct 29 05:09:17 monitor kernel: Module 2: Not installed Oct 29 05:09:17 monitor kernel: Module 3: Not installed However when I run ztcfg - I get an error: [EMAIL PROTECTED] dev]# ztcfg -vvv Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. Notice: Configuration file is /etc/zaptel.conf line 147: Unable to open master device '/dev/zap/ctl' See how it says it can't open the master device /dev/zap/ctl. Well this is because that device doesn't exist.. [EMAIL PROTECTED] dev]# ls -ald zap* crw--- 1 root root 196, 1 Oct 29 05:09 zap1 crw--- 1 root root 196, 2 Oct 29 05:09 zap2 crw--- 1 root root 196, 3 Oct 29 05:09 zap3 crw--- 1 root root 196, 4 Oct 29 05:09 zap4 crw--- 1 root root 196, 254 Oct 29 05:09 zapchannel crw--- 1 root root 196, 0 Oct 29 05:09 zapctl crw--- 1 root root 196, 255 Oct 29 05:09 zappseudo crw--- 1 root root 196, 253 Oct 29 05:09 zaptimer I can manualy create the /dev/zap directory and then make symbolic links to /dev/zap/ctl using /dev/zapctl and make a symbolic link to /dev/zap/channel by linking to /dev/zapchannel etc... This will make the card work just fine, I have tested this on another linux box and the same thing happened. If I add this symbolic link creation into the statup scripts then like I said zaptel working fine, however this is obviously not the right way to fix this issue. Suggestions? -=Linsys=- IntrusionSec.com #1 Hacker Gamez Web Site On the Internet http://www.intrusionsec.com [EMAIL PROTECTED] - When Your Life Flashes Before Your Eyes When You Die, Does That Include The Part Where Your Life Flashes Before Your Eyes? - ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] I give up - Help with TE410P
Well, have you ever tried their support? They assume we are all dummies... A bunch of canned email messages to remind you to plug in the power cable. :) Ok, in a disparate act (and this might help someone body someday) I removed all the Digium card and emptied the zap*.conf files from the box and rebooted. I allowed Linux to remove the missing cards - this of course installs ztdummy. Next I shutdown and added all the cards at one time. - Booted and let Linux discover cards and allowed configuration. Copied back my zap*.conf files rebooted. This time it comes up 6 spans with green lights and 2 on first card with flashing red. I shutdown, and swap the two TE410P. Rebooted - all light green now. Since it's working, I'm done - but only go to show you these cards are flaky. Bart - Original Message - From: Andrew Kohlsmith [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Saturday, October 29, 2005 3:35 PM Subject: Re: [Asterisk-Users] I give up - Help with TE410P On Saturday 29 October 2005 18:19, Bart Fisher wrote: Yep - that was easy part :) and these are T1 (D4, AMI, SF, and EM Wink) BTW Ok, well I'll go for the obvious question: have you contacted Digium technical assistance? You have paid for support within the price of the card. -A. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] I give up - Help with TE410P
Why don't you call their support? I've called and only had a good experience. Tech Support via email is always kind of weak no matter where you go. Call them, and go through their tech support department, they have some really intelligent and knowledgeable techs down there and I'm sure they'll be able to fix this problem. -- Tom On 10/29/05, Bart Fisher [EMAIL PROTECTED] wrote: Well, have you ever tried their support? They assume we are all dummies... A bunch of canned email messages to remind you to plug in the power cable. :) Ok, in a disparate act (and this might help someone body someday) I removed all the Digium card and emptied the zap*.conf files from the box and rebooted. I allowed Linux to remove the missing cards - this of course installs ztdummy. Next I shutdown and added all the cards at one time. - Booted and let Linux discover cards and allowed configuration. Copied back my zap*.conf files rebooted. This time it comes up 6 spans with green lights and 2 on first card with flashing red. I shutdown, and swap the two TE410P. Rebooted - all light green now. Since it's working, I'm done - but only go to show you these cards are flaky. Bart - Original Message - From: Andrew Kohlsmith [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Saturday, October 29, 2005 3:35 PM Subject: Re: [Asterisk-Users] I give up - Help with TE410P On Saturday 29 October 2005 18:19, Bart Fisher wrote: Yep - that was easy part :) and these are T1 (D4, AMI, SF, and EM Wink) BTW Ok, well I'll go for the obvious question: have you contacted Digium technical assistance? You have paid for support within the price of the card. -A. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] I give up - Help with TE410P
On Saturday 29 October 2005 19:30, Bart Fisher wrote: Well, have you ever tried their support? They assume we are all dummies... A bunch of canned email messages to remind you to plug in the power cable. Actually my support from them has been great... Ok, in a disparate act (and this might help someone body someday) I removed all the Digium card and emptied the zap*.conf files from the box and rebooted. I allowed Linux to remove the missing cards - this of course installs ztdummy. allowed linux to remove the missing cards ?? what distro are you using? Next I shutdown and added all the cards at one time. - Booted and let Linux discover cards and allowed configuration. Copied back my zap*.conf files rebooted. This time it comes up 6 spans with green lights and 2 on first card with flashing red. I shutdown, and swap the two TE410P. Rebooted - all light green now. Again, what distro, what version of asterisk and whatnot? Is this [EMAIL PROTECTED] Since it's working, I'm done - but only go to show you these cards are flaky. It sounds like your system is what's flaky here... Linux doesn't need to remove the cards... Definitely something nonstandard from my point of view. I am glad it's working for you though. -A. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DID problem
Hello All, I don't know whether this has been talked about before or not but it seems that from time to time I always bump into problem with the DID. What happening at the moment is that I have a incoming number pointing to a calling card software. I have setup a inbound route in AMP and have it point to the calling card software which works completely fine. But half of the time the DID will reach the server and terminate without passing it to calling card and then a message of The service cannot be connected will come out. -- Executing AbsoluteTimeout(SIP/195.8.117.11-b7815b90, 15) in new stack -- Set Absolute Timeout to 15 -- Executing Congestion(SIP/195.8.117.11-b7815b90, ) in new stack == Spawn extension (from-sip-external, 02080359600, 2) exited non-zero on 'SIP/195.8.117.11-b7815b90' -- Executing AbsoluteTimeout(SIP/195.8.117.11-b7815b90, 15) in new stack -- Set Absolute Timeout to 15 -- Executing Congestion(SIP/195.8.117.11-b7815b90, ) in new stack == Spawn extension (from-sip-external, h, 2) exited non-zero on 'SIP/195.8.117.11-b7815b90' asterisk1*CLI And the other half of the time it will pass it on to the calling card. I am quite confused on why that is happening and would love to hear if anybody has experienced such before. Sam ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] I give up - Help with TE410P
My 2 cents: If you are running kudzu on RH or FC, new and remove hardware should be detected...in most cases. I assume other distros have something similar...? If 2 of 8 T1s are not coming up - sounds like you may have a wiring issue. Can you swap cables from a bad circuit to a good circuit? Are all of the circuits the same configuration from the carrier? As far as support, Digium's email support has ALWAYS been helpful to me - from basic questions to systematic issues. They have always been helpful and responsive. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: Saturday, October 29, 2005 4:50 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] I give up - Help with TE410P On Saturday 29 October 2005 19:30, Bart Fisher wrote: Well, have you ever tried their support? They assume we are all dummies... A bunch of canned email messages to remind you to plug in the power cable. Actually my support from them has been great... Ok, in a disparate act (and this might help someone body someday) I removed all the Digium card and emptied the zap*.conf files from the box and rebooted. I allowed Linux to remove the missing cards - this of course installs ztdummy. allowed linux to remove the missing cards ?? what distro are you using? Next I shutdown and added all the cards at one time. - Booted and let Linux discover cards and allowed configuration. Copied back my zap*.conf files rebooted. This time it comes up 6 spans with green lights and 2 on first card with flashing red. I shutdown, and swap the two TE410P. Rebooted - all light green now. Again, what distro, what version of asterisk and whatnot? Is this [EMAIL PROTECTED] Since it's working, I'm done - but only go to show you these cards are flaky. It sounds like your system is what's flaky here... Linux doesn't need to remove the cards... Definitely something nonstandard from my point of view. I am glad it's working for you though. -A. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] I give up - Help with TE410P
Yep, it CentOS 4.0 (RH) - Kudzu - also seems to be the root of my problem. I later rebooted and now back to some ports working again. I'm using a Loop-Back plug to test with - no real T1 attached until I can fix this. Swapping card does not seem to follow issues. Maybe I'll give support another :) Bart - Original Message - From: Jason Walker [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Saturday, October 29, 2005 5:09 PM Subject: RE: [Asterisk-Users] I give up - Help with TE410P My 2 cents: If you are running kudzu on RH or FC, new and remove hardware should be detected...in most cases. I assume other distros have something similar...? If 2 of 8 T1s are not coming up - sounds like you may have a wiring issue. Can you swap cables from a bad circuit to a good circuit? Are all of the circuits the same configuration from the carrier? As far as support, Digium's email support has ALWAYS been helpful to me - from basic questions to systematic issues. They have always been helpful and responsive. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: Saturday, October 29, 2005 4:50 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] I give up - Help with TE410P On Saturday 29 October 2005 19:30, Bart Fisher wrote: Well, have you ever tried their support? They assume we are all dummies... A bunch of canned email messages to remind you to plug in the power cable. Actually my support from them has been great... Ok, in a disparate act (and this might help someone body someday) I removed all the Digium card and emptied the zap*.conf files from the box and rebooted. I allowed Linux to remove the missing cards - this of course installs ztdummy. allowed linux to remove the missing cards ?? what distro are you using? Next I shutdown and added all the cards at one time. - Booted and let Linux discover cards and allowed configuration. Copied back my zap*.conf files rebooted. This time it comes up 6 spans with green lights and 2 on first card with flashing red. I shutdown, and swap the two TE410P. Rebooted - all light green now. Again, what distro, what version of asterisk and whatnot? Is this [EMAIL PROTECTED] Since it's working, I'm done - but only go to show you these cards are flaky. It sounds like your system is what's flaky here... Linux doesn't need to remove the cards... Definitely something nonstandard from my point of view. I am glad it's working for you though. -A. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Play music while on incoming connections ringing the internal user
Hi Folks, I digged all the forums but cant find an answer... I want that an incoming user (via ISDN) is entertained by some MP3 music while asterisk is ringing the internal phone to server the caller... I tried MusicOnHold but it seems to block and not ring the users. Any pointers ? THX Oliver -- Diese Nachricht wurde digital unterschrieben oliwel's public key: http://www.oliwel.de/oliwel.crt Basiszertifikat: http://www.ldv.ei.tum.de/page72 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Terrible echo with Te110P and Adit 600
I've done some very interesting testing recently:
The 64ms cards are working wonderful. $19.00 a pop is a steal. They
work great with your KB1 canceller, but any others cause HORRIBLE echo.
I am facing the tail end AWAY from the asterisk boxso the echo is
definitely coming from somewhere between the TE110P and the Adit 600.
Interesting hunh?
I have not gotten my hands on a VX2 card yet, but the 64's are working
so well I'm not sure there is a reason too.
The Orion canceller is very nice as well, but $1000.
-Darren
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kris
Boutilier
Sent: Wednesday, October 19, 2005 12:13 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Terrible echo with Te110P and Adit 600
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Darren
Wright
Sent: Tuesday, October 18, 2005 7:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Terrible echo with Te110P and Adit 600
8 POTS lines - Adit 600 - TE110P - Dell Precision 530 Dual
Xeon running 1.0.9 and 1.2 (tried both)
The echo is insurmountable. I have tried everything, and the
pots lines are clean. If I go from an FXO on the Adit 600 straight to
an FXS, I
get no echo from an analog phone.
No echo that you can hear - remeber that echo relies on two things, a
reflected signal and a delay between the transmission and the reception
of the signal long enough for the brain to perceive it. Looping the
channel bank will not introduce any delays. Passing through Asterisk
will, by design.
I put an 128ms T1 echo canceller in between the adit and the
TE110P, and the echo was still horrible.
I finally disabled the Zapata echo cancellerand WHAMMO! It's
perfect now.
It sounds like something is confusing the zaptel canceller causing it to
distort the signal. It seems to be very sensitive to signals that are
too 'hot' (ie. too loud). Try lowering the gain on the signal going out
of the channel bank into the T1. If it's too quiet try increasing the RX
gain on the Zaptel side to compensate.
{clip}
Any ideas so I don't have to spend $1000 on an echo canceller?
I provided the patches to 1.2 that formed the basis for the kb1 echo
canceller, which is a derivative of the mark2 used in v1.0, and I still
use a 64ms Tellabs hardware echo can as well as the zaptel echo
canceller. Note that, in my case at least, the zaptel tends to handle
those echos that leak through the Tellabs gear - such as acoustic room
echos from speaker phones or cheap cordless handsets.
If you need the echo issue resolved, stick with hardware cancellation.
If you don't want to spend $1k, take a look at
http://www.voip-info.org/wiki/view/Tellabs+Hardware+Echo+Cancellers.
It's not everyones cup of tea, but it works fine for me which is why I
shared it.
The Zaptel echo can will be fixed so it performs predictably for
everyone eventually, but until then go with 3rd party T1 gear if you
want it reliably avoided.
Hope that helps.
Kris Boutilier
Information Services Coordinator
Sunshine Coast Regional District
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[Asterisk-Users] chan_zap ignoring stuff in beta1?
I just upgraded to beta1 and everything does seem to be working, however when reloading asterisk I see these error messages: -- Reloading module 'chan_zap.so' (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found Oct 29 20:33:13 WARNING[10141]: chan_zap.c:10593 setup_zap: Ignoring switchtype Oct 29 20:33:13 WARNING[10141]: chan_zap.c:10593 setup_zap: Ignoring signalling Oct 29 20:33:13 WARNING[10141]: chan_zap.c:10593 setup_zap: Ignoring toneduration Now the pri's do load and are signaling via national 2 but I would like to know why they are being ignored and how do I get it to not Ignore tone duration? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] GSM cards / mobile phone cards for Asterisk?
I have a used one for sale. 1900MHz only. $150 to my paypal account secures. Shipping included. Mark Michael Bielicki wrote: That is a roughly what you pay for GSM gateways everywhere. On 10/28/05, *Daniel Varella de Oliveira* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: It costs here more or less R$600,00 (about US$264,55) Our friend, Dave Cotton post a message with a good price for outside of Brazil. US$295,00 is a good price, I think. I know that guy in Sao Paolo (the correct is São Paulo), that the site http://www.thehightechstore.com/plugcell.html http://www.thehightechstore.com/plugcell.htmlannounced. His name is Douglas Prado and he is the owner of Contacto Telecom company. Contacto is the unique distributor of Plugcell in region of São Paulo. If you contact him, tell about me (He knows me as Daniel ex-Nooracom company in Rio de Janeiro). Maybe you can get a discount on your negotiation. hehehehe. -- [ ]'s Daniel Varella de Oliveira Tecnologia IP Ltda Tel.: +55 (21)3139-4091 / r. 108 Rio de Janeiro - Brasil www.tecnologiaip.com.br http://www.tecnologiaip.com.br On Friday 28 October 2005 12:22, Tomasz Chmielewski wrote: Daniel Varella de Oliveira schrieb: Tomasz, I'm from Brazil, and we are using here a solution that is based on a box where we can connect a GSM cellphone and use this directly to a phone or PBX extension. I think that you can use some Digium's card (FXS or FXO) on your server, connect this GSM box there, and route your cellphone calls through this box. There are boxes with just one channel and others up to six channels. They have a lot compatibilities with the most common cellphones. looks interesting. do you know by chance how much such a single-cell box cost (more or less)? ___ --Bandwidth and Colocation sponsored by Easynews.com http://Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com mailto:Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michal Bielicki Halo Kwadrat Sp. z o.o. http://www.asterisk.pl/ http://www.openpbx.org/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Pounds and pence prompt wanted
Ah, no pence huh. I guess I'll have to add that to my list of updates. Mark JP Carballo wrote: Obelix wrote: Is there a .gsm file for announcing UK pounds and pence after the credit remaining prompt, besides the dollar and cents file? /Obelix http://www.voip-info.org/tiki-index.php?page=Asterisk+sound+files+international I peeked into the archives from: http://www.desktop2door.com/asterisk/ and http://www.g7ltt.com/VoIP/vmfiles.html Found pound and pounds but no pence. I could have missed it though. You could also add your own voice to the UK male voice archive. That's what I did when I didn't find philippine(s).gsm My voice is nowhere near Allison's though. O.T. Is the Asterisk the Gaul comics still in circulation? It's been years since I read the series... -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ericsson pabx and digium card TE110P
Hi; Could some one help me: I have a problème to make call from my pabx ericsson i receive juste 4 digit from ericsson to my asterisk any idea??? thanks zaptel.conf: span=1,1,0,ccs,hdb3,crc4bchan=1-15,17-31dchan=16loadzone=frdefaultzone=fr zapata.conf: [channels]language=frswitchtype=euroisdn pridialplan=unknownprilocaldialplan=unknown hidecallerid=nothreewaycalling=yescancallforward=yescallreturn=yesechocancel=yesechocancelwhenbridged=yesrxgain=0.0txgain=0.0immediate=no context=entrant group = 0signalling=pri_netchannel = 1-15channel = 17-31 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] I give up - Help with TE410P
I understand the loopback scenario. Have you swapped the loops between circuits? Are circuits on some of your T1s but loops on others? Can you swap them to see if the green leds follow the cabling? I have kudzu enabled and do not have any issues...although I do not put more than one card in a server. When you say some of the ports are working again, can you expand on that? How about an IRQ issue? Too many for your server? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bart Fisher Sent: Saturday, October 29, 2005 5:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] I give up - Help with TE410P Yep, it CentOS 4.0 (RH) - Kudzu - also seems to be the root of my problem. I later rebooted and now back to some ports working again. I'm using a Loop-Back plug to test with - no real T1 attached until I can fix this. Swapping card does not seem to follow issues. Maybe I'll give support another :) Bart - Original Message - From: Jason Walker [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Saturday, October 29, 2005 5:09 PM Subject: RE: [Asterisk-Users] I give up - Help with TE410P My 2 cents: If you are running kudzu on RH or FC, new and remove hardware should be detected...in most cases. I assume other distros have something similar...? If 2 of 8 T1s are not coming up - sounds like you may have a wiring issue. Can you swap cables from a bad circuit to a good circuit? Are all of the circuits the same configuration from the carrier? As far as support, Digium's email support has ALWAYS been helpful to me - from basic questions to systematic issues. They have always been helpful and responsive. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: Saturday, October 29, 2005 4:50 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] I give up - Help with TE410P On Saturday 29 October 2005 19:30, Bart Fisher wrote: Well, have you ever tried their support? They assume we are all dummies... A bunch of canned email messages to remind you to plug in the power cable. Actually my support from them has been great... Ok, in a disparate act (and this might help someone body someday) I removed all the Digium card and emptied the zap*.conf files from the box and rebooted. I allowed Linux to remove the missing cards - this of course installs ztdummy. allowed linux to remove the missing cards ?? what distro are you using? Next I shutdown and added all the cards at one time. - Booted and let Linux discover cards and allowed configuration. Copied back my zap*.conf files rebooted. This time it comes up 6 spans with green lights and 2 on first card with flashing red. I shutdown, and swap the two TE410P. Rebooted - all light green now. Again, what distro, what version of asterisk and whatnot? Is this [EMAIL PROTECTED] Since it's working, I'm done - but only go to show you these cards are flaky. It sounds like your system is what's flaky here... Linux doesn't need to remove the cards... Definitely something nonstandard from my point of view. I am glad it's working for you though. -A. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] GSM cards / mobile phone cards for Asterisk?
How many sim does it take? I am interested sent me across some detail . Sam -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Phillips Sent: 30 October 2005 01:49 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] GSM cards / mobile phone cards for Asterisk? I have a used one for sale. 1900MHz only. $150 to my paypal account secures. Shipping included. Mark Michael Bielicki wrote: That is a roughly what you pay for GSM gateways everywhere. On 10/28/05, *Daniel Varella de Oliveira* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: It costs here more or less R$600,00 (about US$264,55) Our friend, Dave Cotton post a message with a good price for outside of Brazil. US$295,00 is a good price, I think. I know that guy in Sao Paolo (the correct is São Paulo), that the site http://www.thehightechstore.com/plugcell.html http://www.thehightechstore.com/plugcell.htmlannounced. His name is Douglas Prado and he is the owner of Contacto Telecom company. Contacto is the unique distributor of Plugcell in region of São Paulo. If you contact him, tell about me (He knows me as Daniel ex-Nooracom company in Rio de Janeiro). Maybe you can get a discount on your negotiation. hehehehe. -- [ ]'s Daniel Varella de Oliveira Tecnologia IP Ltda Tel.: +55 (21)3139-4091 / r. 108 Rio de Janeiro - Brasil www.tecnologiaip.com.br http://www.tecnologiaip.com.br On Friday 28 October 2005 12:22, Tomasz Chmielewski wrote: Daniel Varella de Oliveira schrieb: Tomasz, I'm from Brazil, and we are using here a solution that is based on a box where we can connect a GSM cellphone and use this directly to a phone or PBX extension. I think that you can use some Digium's card (FXS or FXO) on your server, connect this GSM box there, and route your cellphone calls through this box. There are boxes with just one channel and others up to six channels. They have a lot compatibilities with the most common cellphones. looks interesting. do you know by chance how much such a single-cell box cost (more or less)? ___ --Bandwidth and Colocation sponsored by Easynews.com http://Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com mailto:Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michal Bielicki Halo Kwadrat Sp. z o.o. http://www.asterisk.pl/ http://www.openpbx.org/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Webui to show registered phones
Hi For those who are insterested in monitoring and managing easilly the asterisk server.. this is a solution for multitenant hosted PBX o single tenant is windows based (the admin of couse) and http://www.cripiland.com/screenshots/manager3.jpg http://www.cripiland.com/screenshots/manager4.jpg http://www.cripiland.com/screenshots/manager1.jpg http://www.cripiland.com/screenshots/manager2.jpg regards Saul Matt Gibson wrote: Hi Guys, Here's what I use to view the current IAX and SIP peer status. It's not very pretty, but it works. I also have an included script (vm.php) that will show the current voicemail usage for a box. Uses php asterisk library to work through asterisk manager. Configure your options in cfg.php Matt Nicolás Gudiño wrote: Hi all, does anyone know if there is any app/webui that can show phones that are currently registered to *. I guess this sort of funcionality counld be grabbed from the CLI with iax2 show peers and sip show peers, but having little programming knowledge wouldn't know where to start. I'm asking because we currently have several sip phones onsite and lots of remote iax2 users who would like to see availability without dialing. plugYou can try with the Flash Operator Panel/plug http://www.asternic.org , it does all sort of things including sip and iax availability (you have to enable qualify for them). Regards, -- Nicolás Gudiño Buenos Aires - Argentina ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Play music while on incoming connections ringing the internal user
On Oct 29, 2005, at 8:39 PM, Oliver Welter wrote: Hi Folks, I digged all the forums but cant find an answer... I want that an incoming user (via ISDN) is entertained by some MP3 music while asterisk is ringing the internal phone to server the caller... I tried MusicOnHold but it seems to block and not ring the users. Any pointers ? THX I was about to post the complete output of show application dial from the CLI, but then I thought it would seem like I was being a smart-ass. It is extremely informative, though, and I (as someone new to asterisk) find myself returning to study it frequently. I highly recommend going over it. In any case, excerpted from that text: 'm[(class)]' -- provide hold music to the calling party until answered (optionally with the specified class. In other words, Dial(technology/resource,,m) will do what you want. lyd ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 1.2
Hi Is there a release date for asterisk 1.2. I thought it'll be released this month. can we upgrade from asterisk 1.0.9 or have to do a fresh installation once it's released. tks kani ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Outbound fax solution
Do you actualy send faxes through them? Regards, Chris - Original Message - From: Chris Mason (Lists) [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, October 28, 2005 7:27 PM Subject: Re: [Asterisk-Users] Outbound fax solution Teliax works for me, generally. I don't know why but no other provider does. I suspect the other translate to G729 and send SIP. -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Access to channel data...
Hi - am looking for someone who can help me understand how to get access to channel data via AGI.
My objective is to write an AGI Script to monitor channel status on an originated call prior to passing it to a queue.
Current approach:
1. Originate via AMI...
set msg(Channel) Local/[EMAIL PROTECTED]/n
set msg(Exten) 0021$numberDial
set msg(Account) $agentid
set msg(Callerid) $axtn
set msg(Priority) 1
[default-agi]
exten = _1128,1,agi(OutBoundCall.agi)
exten = _0021X.,1,Dial(IAX2/id:[EMAIL PROTECTED]/${EXTEN},20,g)
2. Capture channel status via agi then initiate transfer into a queue
but channel data returning into my AGI script appears a little removed from the action...
Local/1128-abcd,1 rings Local/1128-abcd,2
I issue ANSWER from AGI and see data associated with Local/1128-abcd,2
however, the oubbound channel is IAX/provider-1 which is bridged with Local/1128-abcd,1
according to show channels verbose, and my AGI is looking at Local/1128-abcd,2
What I'd really like to see is the status when IAX/provider-1 gets linked or achieves progress.
The plan is to detect this in the AGI and initiate a transfer into a queue at this point.
If no progress is made in connecting the call, I want to drop it programmatically.
If anyone could point me in the direction of some additional docco that might
help nudge me in the right direction, I would be very grateful.
cheers,
Mark Edwards.
___
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Re: [Asterisk-Users] DTMF recognition unreliable or absent, ringing lost on 2nd half of 3-way call.
Thanks for the response. I am using CVS HEAD, unfortunately, and still seeing these problems. When I posted the above, I was using code checked out and built on 2005-10-11. Currently, I am running: Asterisk CVS HEAD built by [EMAIL PROTECTED] on a Power Macintosh running Darwin on 2005-10-27 19:51:45 UTC. The behavior doesn't seem to have changed for me. Is there anything I should be turning on (or off) to improve matters? The problem is caused on the provider side, not in your asterisk. The packets they are sending to your asterisk have the wrong sequence numbers. I finished some code lastnight to *help* work-around this issue, I just need to put the finishing touches on it and throw it out on my site. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] GSM cards / mobile phone cards for Asterisk?
One. Sam Tam wrote: How many sim does it take? I am interested sent me across some detail . Sam -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Phillips Sent: 30 October 2005 01:49 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] GSM cards / mobile phone cards for Asterisk? I have a used one for sale. 1900MHz only. $150 to my paypal account secures. Shipping included. Mark Michael Bielicki wrote: That is a roughly what you pay for GSM gateways everywhere. On 10/28/05, *Daniel Varella de Oliveira* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: It costs here more or less R$600,00 (about US$264,55) Our friend, Dave Cotton post a message with a good price for outside of Brazil. US$295,00 is a good price, I think. I know that guy in Sao Paolo (the correct is São Paulo), that the site http://www.thehightechstore.com/plugcell.html http://www.thehightechstore.com/plugcell.htmlannounced. His name is Douglas Prado and he is the owner of Contacto Telecom company. Contacto is the unique distributor of Plugcell in region of São Paulo. If you contact him, tell about me (He knows me as Daniel ex-Nooracom company in Rio de Janeiro). Maybe you can get a discount on your negotiation. hehehehe. -- [ ]'s Daniel Varella de Oliveira Tecnologia IP Ltda Tel.: +55 (21)3139-4091 / r. 108 Rio de Janeiro - Brasil www.tecnologiaip.com.br http://www.tecnologiaip.com.br On Friday 28 October 2005 12:22, Tomasz Chmielewski wrote: Daniel Varella de Oliveira schrieb: Tomasz, I'm from Brazil, and we are using here a solution that is based on a box where we can connect a GSM cellphone and use this directly to a phone or PBX extension. I think that you can use some Digium's card (FXS or FXO) on your server, connect this GSM box there, and route your cellphone calls through this box. There are boxes with just one channel and others up to six channels. They have a lot compatibilities with the most common cellphones. looks interesting. do you know by chance how much such a single-cell box cost (more or less)? ___ --Bandwidth and Colocation sponsored by Easynews.com http://Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com mailto:Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michal Bielicki Halo Kwadrat Sp. z o.o. http://www.asterisk.pl/ http://www.openpbx.org/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
