[Asterisk-Users] RE: Music on hold for first person in a conference room
I have successfully set up a conference room on my asterisk server, I have been trying to make the 'M' for music on hold option work (when the first person enters the room they are told they are the first and then they are supposed to hear music on hold) but it didn't matter which way I wrote it this feature wouldn't work. Basically it wouldn't allow the conference to be setup in the first place, asterisk kept saying that is not a valid conference room. I know that a signifier does work because I also use the 'p' option exten = 99,3,MeetMe(99|p) The p allows someone to exit the conference by hitting the # button. I haven't felt the need to ask for pin code and it also makes it easier to transfer people into it when setting up multi party conference. It turns out to make this much more stable, I was getting about 1 in 3 conference rooms failing when someone hung up (basically the room would fill with non stop loud static) and the other participants had to dial in again. Cheers, Dean ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk from scratch
On Wednesday 21 April 2004 12:03 pm, kiran p wrote: Hi My motto is to connect two computers on the same network with Voip without using any special hardware,i have downloaded Asterisk, I was suggested to use LinPhone as a soft phone as it is very easy to install I have installed Asterisk on my computer and iam using it as a server. And whe i DAIL 1234 at CLI i get the following errors repeatedly Apr 21 17:29:13 WARNING[1167272128]: chan_oss.c:272 sound_thread: Failed to write sound Apr 21 17:29:13 WARNING[1167272128]: chan_oss.c:181 send_sound: Unable to read output space I had the same error; very frustrating. The source was a hardware incompatibility with the VIA sound chip on the motherboard (a Tyan S2495). I disabled the motherboard's built-in sound, put in a soundcard from my old machine (a SoundBlaster PCI 128), and the sound works now. If you have a sound card available, you may want to test putting the sound card in your machine (with the built-in sound disabled). One more doubt i have is after installing a soft phone on the client,how do i configure it to connect to Asterisk. I'm new to this part of Asterisk, yet I believe you need to add configuration for your softphone in sip.conf. You should see: http://www.voip-info.org/wiki-Asterisk+sip+channels And how do i know,if Asterisk is recognizing the sound card or not If you have the demo configurations loaded, you can dial (from the console): [EMAIL PROTECTED] and hear the congradulations message. Anon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Off Topic: RE: [Asterisk-Users] :)
Argh! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: vrijdag 23 april 2004 7:25 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] :) Argh, i don't like the plaintext :) archive password: 45703 Thanks to this message where a virus chose to use my from-address to send its crap from I am now being harassed with many many virus warning messages. A call to anyone operating virusscanners (as I am too): I think we can all do without these reports - over 90% of all virusses using email has faked from-headers anyway :-P Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MWI indicator on SNOM200 doesn't disappear
Geert Nijpels wrote: Ian White wrote: On recent releases of the snom200 firmware, the MWI indicator will turn on, but won't turn off when the message has been checked. It works on firmware 2.03o, but not in 2.04g or newer. I filed a bug report with snom, but they're claiming it is an asterisk issue and that it should have been resolved. They suggested that I ask on the list. Anyway, Asterisk had a bug where it didn't send the NOTIFY correctly to turn off the MWI. The message doesn't contain the line so the phone doesn't know which line to apply the messages to. Basically the NOTIFY message should contain something like the following: NOTIFY sip:[EMAIL PROTECTED];line=34n34jed SIP/2.0 There was a bugfix for this in Asterisk for this problem, do you have that applied? I am running the current CVS version, and don't see anything in the code that looks like this has been touched, and I haven't seen reference to it on this list. They are right in that the line information isn't being sent, looking at the SIP debugs on both ends. Anybody have ideas? Ian This is a problem I have been digging into a bit. In my case asterisk did not send out the NOTIFY with the header Content-Type: application/simple-message-summary, but with Content-Type: text/plain, so the NOTIFY is treated as a txt message. In result, when I pressed the MWI button, I saw the text from asterisk stating the amount of messages I have. I changed it to work, and now asterisk calls the extension the message is sent from ([EMAIL PROTECTED]). After calling this the MWI indication disappears, I'm not sure if it also disappears after calling from another phone. I'm using chan_sip2 and I changed some stuff, so I'm not sure if this is also a problem with standard chan_sip (the txt vs vm issue). Chan_sip2 handles Contact: differently than chan_sip and works better with Snom phones. It's actually where the whole chan_sip2 project started... :-) /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Question of Asterisk timer to get Conference work
Hello, Can someone help me. I got zaptel.0.9.1.tar.gz from ftp.asterisk.org, And then I uncomment the line with MODULES # ztdummy, run make clean, make, make install But errors happens as follows: -- make: zaptel.c:5937: storage size of `zt_fops' isn't known/usr/include/linux/proc_fs.h:193: warning: `create_proc_read_entry' declared `static' but never definedmake: *** [zaptel.o] Error 1 -- make install: zaptel.c:5937: storage size of `zt_fops' isn't known/usr/include/linux/proc_fs.h:193: warning: `create_proc_read_entry' declared `static' but never definedmake: *** [zaptel.o] Error 1 Is there anybody ever install this timer driver, please tell me what's wrong? Thanks! Chunghwa Telecom BTA Tech. LabE-mail:[EMAIL PROTECTED]
[Asterisk-Users] Question of Asterisk timer to get Conference work
Hello, Can someone help me. I got zaptel.0.9.1.tar.gz from ftp.asterisk.org, And then I uncomment the line with MODULES # ztdummy, run make clean, make, make install But errors happens as follows: -- make: zaptel.c:5937: storage size of `zt_fops' isn't known/usr/include/linux/proc_fs.h:193: warning: `create_proc_read_entry' declared `static' but never definedmake: *** [zaptel.o] Error 1 -- make install: zaptel.c:5937: storage size of `zt_fops' isn't known/usr/include/linux/proc_fs.h:193: warning: `create_proc_read_entry' declared `static' but never definedmake: *** [zaptel.o] Error 1 Is there anybody ever install this timer driver, please tell me what's wrong? Thanks! Chunghwa Telecom BTA Tech. LabE-mail:[EMAIL PROTECTED]
Re: Off Topic: RE: [Asterisk-Users] :)
On Fri, 2004-04-23 at 08:43 +0200, Florian Overkamp wrote: Thanks to this message where a virus chose to use my from-address to send its crap from I am now being harassed with many many virus warning messages. A call to anyone operating virusscanners (as I am too): I think we can all do without these reports - over 90% of all virusses using email has faked from-headers anyway :-P Florian, don't forget that the vast majority of virus scanners have been set up by people only used to using the very OS that's caused the problem in the first place. They have no idea whatsoever of how to configure something, they just click the Install button. The scanner writers are the problem, they've seen a wonderful way of spamming and then claiming not me guv, honest. Perhaps Gates can use some of the vast profit he's just announced to sort his crap out. Pigs might fly. -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MWI indicator on SNOM200 doesn't disappear
I have also complained about the change in MWI to SNOM. My 2.03o phones still work with Asterisk but 2.04 versions do not. However, you can turn off the MWI by pressing the MWI button but not remotely ( NOTIFY ). I once got the example under from SNOM ( Asterisk version is under it ). According to SNOM this is an example of the format the phone is expecting in order to get MWI turned off. The relevant difference really looks like to be the 'Message-Account'. NOTIFY sip:[EMAIL PROTECTED]:5060;line=jet7pbic SIP/2.0 Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK-7c9c323d4898e621adb7244baa8cab62.1 Via: SIP/2.0/UDP 192.168.0.8:5062;branch=z9hG4bK-zt7bd9vxqo74 Record-Route: sip:intern.snom.de:5060;maddr=192.168.0.1;lr From: sip:[EMAIL PROTECTED]:5062;tag=vn8jb3vkko To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 23 NOTIFY Max-Forwards: 69 Contact: sip:[EMAIL PROTECTED]:5062 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 85 Message-Waiting: no Message-Account: sip:[EMAIL PROTECTED]:5062 Voice-Message: 0/0 This is what Asterisk is sending at the moment. And this is ok with 2.03o. Does chan_sip2 send somehow different NOTIFY ? NOTIFY sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.1.15.30:5060;branch=z9hG4bK3f99907b From: Asterisk sip:[EMAIL PROTECTED];tag=as243abda7 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: application/simple-message-summary Content-Length: 36 Messages-Waiting: no Voicemail: 0/0 -- Pertti Olle E. Johansson wrote: Geert Nijpels wrote: Ian White wrote: On recent releases of the snom200 firmware, the MWI indicator will turn on, but won't turn off when the message has been checked. It works on firmware 2.03o, but not in 2.04g or newer. I filed a bug report with snom, but they're claiming it is an asterisk issue and that it should have been resolved. They suggested that I ask on the list. Anyway, Asterisk had a bug where it didn't send the NOTIFY correctly to turn off the MWI. The message doesn't contain the line so the phone doesn't know which line to apply the messages to. Basically the NOTIFY message should contain something like the following: NOTIFY sip:[EMAIL PROTECTED];line=34n34jed SIP/2.0 There was a bugfix for this in Asterisk for this problem, do you have that applied? I am running the current CVS version, and don't see anything in the code that looks like this has been touched, and I haven't seen reference to it on this list. They are right in that the line information isn't being sent, looking at the SIP debugs on both ends. Anybody have ideas? Ian This is a problem I have been digging into a bit. In my case asterisk did not send out the NOTIFY with the header Content-Type: application/simple-message-summary, but with Content-Type: text/plain, so the NOTIFY is treated as a txt message. In result, when I pressed the MWI button, I saw the text from asterisk stating the amount of messages I have. I changed it to work, and now asterisk calls the extension the message is sent from ([EMAIL PROTECTED]). After calling this the MWI indication disappears, I'm not sure if it also disappears after calling from another phone. I'm using chan_sip2 and I changed some stuff, so I'm not sure if this is also a problem with standard chan_sip (the txt vs vm issue). Chan_sip2 handles Contact: differently than chan_sip and works better with Snom phones. It's actually where the whole chan_sip2 project started... :-) /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Interfacing with an existing phone system
On Thursday 22 April 2004 07:05 pm, Joel Duffield wrote: We want to use asterisk to extend our current phone system. It is a regular plain old system. Has anyone done this before? Absolutely - in a lot of different ways. We would be adding about 4 SIP (probably Cisco) phones to use with asterisk. What kind of card will I need to use for this, FXS or FXO. Neither of those types of cards. You will need an ethernet card/connection to use a SIP phone. Also, before you pay all that money for new phones, you could test/learn using asterisk with free softphones. Also does anyone have any ideas what the best way to go about this is, should I just forward existing lines to specific phones (just to save on running new telephone cabling) Many of the SIP phone have a built-in ethernet switch. Plug the phone's ethernet port into your network, then plug the computer at that station into the phone's ethernet port. You would not need any new/more cabling. or would there be any simple ways to make a small menu and just put one more layer before they get through? Asterisk is very flexible. Chances are you can do whatever you need - with some learning. ;) Anon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Réf.: Re: [Asterisk-Users] Asterisk with UUI support ?
OK, so I'll do that... Is there any infos I need to know about chan_sip.c (because I suppose it's it that I need to play with)? Does anyone know an interesting website where I can find infos about UUI in ISDN (with CAPI maybe?) ? Thanks for your help. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] * INSTRUCTIONS
Might I humbly request someone, somewhere in the community establish a dummies guide to asterisk kind of site, that explains in detail what the cryptic scripts actually do, line by line. The Wiki is helpful, but unless you were in on the movie from the first part, the scene discussions are moot. Im grateful for all the most helpful people who have assisted me, and I expect to actually be able to talk to someone with the * soonrite now, IAXTEL isn't recognizing me (after registering and following the thread thru changing my pwd)...Ive put the FWD connection on hold... I know it's a hard stretch for a lot of you who are experts at this, to accommodate the new users. Some explanations as to what each line IS and what it does might be helpful...man, IF I could get mine working (and its for lack of time presently, Im sure the scripts Ive been sent will eventually work) I'd be happy to write something to help people get a simple solution to get it talking so they would be interested in exploring furtherto look at the test scripts, is a exercise in futility (IMHO). I dont say this lightly, I worked commercial switches and SS7 for years...but Im still going DUHIm sure Ill eventually master it, but I sincerely believe we need some real * 101 instructional stuffone of these days Im sure the light will switch on, and I'll say I wonder how I didn't get it, just like you are in that position now. This is NOT a criticism, please dont take it that way...its merely a suggestion for easing the transition into understanding, and thus sparking interest in further exploration and hopefully a greater user base. Marc, with hopefully positive advice...and to all who contacted me, THANK YOUvery much.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 3 companies 1 card
Good day all I want to put the openline4 card into a box that will support 3 different companies I read the caller ID id fixed but now HOW DO I: If a call come in for 12345 it plays company 1's welcome message If a call come in for 98765 it plays company 2's welcome message ens.. Does This make sense Thanks Altus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco phones
Why is there such a variation in price between what the two of you have paid to get the SIP image for a 7960 phone ? $8 would be acceptable, but I don't want to have to pay $105 ! What website do I have to go to in order to buy a SIP image update ? How long does the login last for - I mean can you download it a couple of times, or is it a case of once you've downloaded it, that it. I read that you sometimes need to step though the images, starting at v2, then v3, then v4 etc. If thats the case, surly I'll need to download a few images ! Thanks, Paul. -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hermann WeckePosted At: 22 April 2004 23:09Posted To: Asterisk-UsersConversation: [Asterisk-Users] Cisco phonesSubject: Re: [Asterisk-Users] Cisco phones On Thu, 22 Apr 2004, Paul Tyreman wrote: I am guessing the phone that I get won't come with that as it was used with the cisco call manager software in the past. Can I still use this phone with Asterisk, or have I waited my money ? Every Cisco software embedded with their hardware is valid only for the first owner. When someone sells the equipment, the software license is not transfered. The new buyer must buy a new license. I bought a Cisco 7960G over eBay also, and I bought their SIP software later. I paid US$ 105. The original was a SCCP. Check also the list history. You will find several messages regarding this same issue (cisco hardware X software X upgrade). You can find the archives here: http://lists.digium.com/pipermail/asterisk-users/(actually, use Google with this query:"cisco sip upgrade site:lists.digium.com" -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sam BacsaPosted At: 22 April 2004 23:05Posted To: Asterisk-UsersConversation: [Asterisk-Users] Cisco phonesSubject: RE: [Asterisk-Users] Cisco phones You can get an upgrade contract with Cisco for like $8 or something to download the SIP firmware for your phone. So no, no waste of money -- unless you bought the wrong phone. - Sam From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul TyremanSent: Thursday, April 22, 2004 2:49 PMTo: [EMAIL PROTECTED]Subject: Re: [Asterisk-Users] Cisco phones I have bough a cisco phone on eBay to use with Asterisk, but according to that website, you need a contract with Cisco systems to upgrade the phone to work with SIP. I am guessing the phone that I get won't come with that as it was used with the cisco call manager software in the past. Can I still use this phone with Asterisk, or have I waited my money ? Thanks, Paul.
Re: [Asterisk-Users] 3 companies 1 card
Altus Snyman wrote: Good day all I want to put the openline4 card into a box that will support 3 different companies I read the caller ID id fixed but now HOW DO I: If a call come in for 12345 it plays company 1's welcome message If a call come in for 98765 it plays company 2's welcome message ens.. Use a telco line (or service) that provides you DNIS. exten = 12345,1,Answer exten = 12345,2,Playback,company1-welcome ... exten = 98765,1,Answer exten = 98765,2,Playback,company2-welcome ... Support Asterisk and buy Digium hardware, you will thank me later. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Extension buttons
On Friday 23 April 2004 12:33 am, David Krider wrote: I've downloaded the entire archive of articles and searched through them for an answer on this, but I haven't come across one yet. I'm looking to replace a small phone system in my church with Asterisk, and I'm stuck looking for phones. I know that the staff are going to want a button for their commonly-called extensions, but I'm having trouble finding phones that have, say, 10 programmable buttons for this sort of thing. I'm left to conclude that most phones can do this sort of thing by clicking through some combination of buttons. However, it would seem that the average price for a nice SIP phone eliminates the possibility of just ordering some to find out. Can someone please tell me how this is handled in general? For instance, the Polycom 600 doesn't seem to have ANY buttons that can be programmed for particular extensions Not correct - The Polycom SoundPoint IP 600 has 6 buttons on the upper left hand side that can be programmed for particular extensions and speed-dial entries. It also has the ability to support 6 lines, and has extensive directory support. And, strangely, ALL the buttons on the phone can be reprogrammed. Keep in mind this phone uses context-sensitive soft-keys, so it offers much more ability and functionality than can be seen in a low resolution photo on the web. It may suprise you to know that the soft-key implementation is very well done: intuitive, logical, efficient, and easy to use. (Polycom should pay me for posting this ;) Anon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 3 companies 1 card
The thing is its 3 companies,3 different number 3 different lines. I know you can sort it with source number(That old girlfriend thing) but what about destination number,can you get it On Fri, 2004-04-23 at 10:19, Jeremy McNamara wrote: Altus Snyman wrote: Good day all I want to put the openline4 card into a box that will support 3 different companies I read the caller ID id fixed but now HOW DO I: If a call come in for 12345 it plays company 1's welcome message If a call come in for 98765 it plays company 2's welcome message ens.. Use a telco line (or service) that provides you DNIS. exten = 12345,1,Answer exten = 12345,2,Playback,company1-welcome ... exten = 98765,1,Answer exten = 98765,2,Playback,company2-welcome ... Support Asterisk and buy Digium hardware, you will thank me later. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem at night
I'm using asterisk with isdn hfcpci carc (driver zaphfc) all work correctly during the day but during the night it happend something that hang the card with this message: zaphfc: empty HDLC frame received. Asterisk work without any error message but isdn doesen't work I must stop asterisk unload the driver and reload it and then all work correctly for entire day with a lot of call I'm in italy Thank's Tiziano
Re: [Asterisk-Users] Extension buttons
(B-- $B>.ED??G7(B [EMAIL PROTECTED] (B (B (B (B___ (BAsterisk-Users mailing list (B[EMAIL PROTECTED] (Bhttp://lists.digium.com/mailman/listinfo/asterisk-users (BTo UNSUBSCRIBE or update options visit: (B http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] * INSTRUCTIONS
On Friday 23 April 2004 07:55 am, tmpm wrote: Might I humbly request someone, somewhere in the community establish a dummies guide to asterisk kind of site, that explains in detail what the cryptic scripts actually do, line by line. The Wiki is helpful, but unless you were in on the movie from the first part, the scene discussions are moot. I know it's a hard stretch for a lot of you who are experts at this, to accommodate the new users. Some explanations as to what each line IS and what it does might be helpful...man, IF I could get mine working (and its for lack of time presently, Im sure the scripts Ive been sent will eventually work) I'd be happy to write something to help people get a simple solution to get it talking so they would be interested in exploring furtherto look at the test scripts, is a exercise in futility (IMHO). I dont say this lightly, I worked commercial switches and SS7 for years...but Im still going DUHIm sure Ill eventually master it, but I sincerely believe we need some real * 101 instructional stuffone of these days Im sure the light will switch on, and I'll say I wonder how I didn't get it, just like you are in that position now. This is NOT a criticism, please dont take it that way...its merely a suggestion for easing the transition into understanding, and thus sparking interest in further exploration and hopefully a greater user base. As a not-quite-so-newbie on the brink of understanding the hugeness of Asterisk, I am greatly compelled to agree with everything written above. BTW - I contributed to the Wikki on more than a few occasions when I _finally_ had an epiphany understanding Asterisk, and I plan to do so more in the future. Anon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7960 SIP Firmware
I have just got 3 Cisco 7960 phones which I would like to connect to Asterisk... However they seem to have v3 SCCP firmware. I have tried numerous links to the Cisco Website but unable to get the SIP firmware. Has anyone managed to get a service contract or an account with download privileges? Ideally I would like to upgrade to 6.3 SIP; though it seems I might need to upgrade via v3 or v4? Any idea where I might find copies? robert AT johnson-perkins DOT com PLEASE READ: The information contained in this email is confidential and intended for the named recipient(s) only. If you are not an intended recipient of this email you must not copy, distribute or take any further action in reliance on it and you should delete it and notify the sender immediately. Email is not a secure method of communication and Nomura International plc cannot accept responsibility for the accuracy or completeness of this message or any attachment(s). Please examine this email for virus infection, for which Nomura International plc accepts no responsibility. If verification of this email is sought then please request a hard copy. Unless otherwise stated any views or opinions presented are solely those of the author and do not represent those of Nomura International plc. This email is intended for informational purposes only and is not a solicitation or offer to buy or sell securities or related financial instruments. Nomura International plc is regulated by the Financial Services Authority and is a member of the London Stock Exchange. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 SIP Firmware
you should get that from the seller of the phones, they must have a CCO login with donwload privs and give you the firmware. but if u bought them used, that's another story It's not legal to share cisco firmware without authorization... Matteo. Il ven, 2004-04-23 alle 10:38, Johnson-Perkins, Robert ha scritto: I have just got 3 Cisco 7960 phones which I would like to connect to Asterisk... However they seem to have v3 SCCP firmware. I have tried numerous links to the Cisco Website but unable to get the SIP firmware. Has anyone managed to get a service contract or an account with download privileges? Ideally I would like to upgrade to 6.3 SIP; though it seems I might need to upgrade via v3 or v4? Any idea where I might find copies? robert AT johnson-perkins DOT com PLEASE READ: The information contained in this email is confidential and intended for the named recipient(s) only. If you are not an intended recipient of this email you must not copy, distribute or take any further action in reliance on it and you should delete it and notify the sender immediately. Email is not a secure method of communication and Nomura International plc cannot accept responsibility for the accuracy or completeness of this message or any attachment(s). Please examine this email for virus infection, for which Nomura International plc accepts no responsibility. If verification of this email is sought then please request a hard copy. Unless otherwise stated any views or opinions presented are solely those of the author and do not represent those of Nomura International plc. This email is intended for informational purposes only and is not a solicitation or offer to buy or sell securities or related financial instruments. Nomura International plc is regulated by the Financial Services Authority and is a member of the London Stock Exchange. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Matteo Brancaleoni Espia System Administrator Email : [EMAIL PROTECTED] Web : http://www.espia.it Phone : +39 02 70633354 - ext 201 IAX(2): [EMAIL PROTECTED] - ext 201 Iaxtel: 1-700-56-62458 - ext 201 SIP : [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 3 companies 1 card
Altus Snyman wrote: The thing is its 3 companies,3 different number 3 different lines. I know you can sort it with source number(That old girlfriend thing) but what about destination number,can you get it Then u can separate each line out into its own context [company1] exten = s,1,Answer exten = s,1,Playback,company1-welcome ... [company2] exten = s,1,Answer exten = s,1,Playback,company2-welcome ... Jeremy McNamara On Fri, 2004-04-23 at 10:19, Jeremy McNamara wrote: Altus Snyman wrote: Good day all I want to put the openline4 card into a box that will support 3 different companies I read the caller ID id fixed but now HOW DO I: If a call come in for 12345 it plays company 1's welcome message If a call come in for 98765 it plays company 2's welcome message ens.. Use a telco line (or service) that provides you DNIS. exten = 12345,1,Answer exten = 12345,2,Playback,company1-welcome ... exten = 98765,1,Answer exten = 98765,2,Playback,company2-welcome ... Support Asterisk and buy Digium hardware, you will thank me later. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 3 companies 1 card
But who do I differentiate between the different number,how do I say: if a caller calls 1234(the destination) do: [company1] exten = s,1,Answer exten = s,1,Playback,company1-welcome ens. On Fri, 2004-04-23 at 10:44, Jeremy McNamara wrote: Altus Snyman wrote: The thing is its 3 companies,3 different number 3 different lines. I know you can sort it with source number(That old girlfriend thing) but what about destination number,can you get it Then u can separate each line out into its own context [company1] exten = s,1,Answer exten = s,1,Playback,company1-welcome ... [company2] exten = s,1,Answer exten = s,1,Playback,company2-welcome ... Jeremy McNamara On Fri, 2004-04-23 at 10:19, Jeremy McNamara wrote: Altus Snyman wrote: Good day all I want to put the openline4 card into a box that will support 3 different companies I read the caller ID id fixed but now HOW DO I: If a call come in for 12345 it plays company 1's welcome message If a call come in for 98765 it plays company 2's welcome message ens.. Use a telco line (or service) that provides you DNIS. exten = 12345,1,Answer exten = 12345,2,Playback,company1-welcome ... exten = 98765,1,Answer exten = 98765,2,Playback,company2-welcome ... Support Asterisk and buy Digium hardware, you will thank me later. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CAPI and Extensions.conf Security problem
Hi, I've installing a AVM Fritz Card in my ASterisk Box I've configured everything and its running perfectly. The problem is that everybody is allow to call through it. Explaination: All users registered in Asterisk can make a call towards the ISDN network But, everybody from the Internet, knowing the extension of CAPI in the dialplan, can call through my Asterisk to any phone number Heellp mmm please ! Thanks Ignace ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] NCS signaling
Hi, Does Asterisk support NCS signalling? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CAPI and Extensions.conf Security problem
ever heard of a 'correct dialplan' ? perhaps there's some bug in your context/extensions logic that let this happens. better review it :) Matteo. Il ven, 2004-04-23 alle 11:20, Ignace CARIA ha scritto: Hi, I've installing a AVM Fritz Card in my ASterisk Box I've configured everything and its running perfectly. The problem is that everybody is allow to call through it. Explaination: All users registered in Asterisk can make a call towards the ISDN network But, everybody from the Internet, knowing the extension of CAPI in the dialplan, can call through my Asterisk to any phone number Heellp mmm please ! Thanks Ignace ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Matteo Brancaleoni Espia System Administrator Email : [EMAIL PROTECTED] Web : http://www.espia.it Phone : +39 02 70633354 - ext 201 IAX(2): [EMAIL PROTECTED] - ext 201 Iaxtel: 1-700-56-62458 - ext 201 SIP : [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 3 companies 1 card
On Fri, 23 Apr 2004, Altus Snyman wrote: But who do I differentiate between the different number,how do I say: if a caller calls 1234(the destination) do: [company1] exten = s,1,Answer exten = s,1,Playback,company1-welcome ens. In response to Jeremy McNamara, who on Fri 2004-04-23 at 10:44, wrote: Then u can separate each line out into its own context [company1] exten = s,1,Answer exten = s,1,Playback,company1-welcome ... [company2] exten = s,1,Answer exten = s,1,Playback,company2-welcome ... You need to clarify something for us. If each company has its own line, and calls for each company only come in on the line of that company, Jeremy's suggestion is all you need. It does not matter what number the caller dialled! The telco will deliver calls to one line only, and you manage the routing of those calls by assigning each line to its own context. If the incoming lines function as a hunt group, so calls for the three different companies could come in on any of the three lines, that's a different situation. If so, you're out of luck, because for this you would need DNIS (as Jeremy posted originally), and AFAIK you cannot get DNIS on POTS lines. The closest you would get is Multiple Number through Distinctive Ring, and I'm not sure how heavily you could rely on that in your circumstance. Best bet (I think) would be to go digital incoming. Get a Digium T1/E1 card (as appropriate for your telco), or even a QuadBRI if your usage doesn't justify T1/E1 channel density. Cheers, Vic Cross ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 3 companies 1 card
Altus Snyman wrote: But who do I differentiate between the different number,how do I say: if a caller calls 1234(the destination) do: [company1] exten = s,1,Answer exten = s,1,Playback,company1-welcome ens. Normally this would be done by setting a context for each DNO in the device's configuration file. In modem.conf for an i4l device for example, it goes like: context=company1 incomingmsn=2108122444 context=company2 incomingmsn=2108122888 I don't know anything about this device you're using, so I can't be more specific... Apollon Koutlides ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PC based Switchboard application files??
The procedure was changed. I'm sending that directly. We'll need to know who actually downloads that. If anybody else needs it, please contact me off-list. Best regards Pertti Steven Elliott wrote: On 22/04/04 8:50, Pertti Pikkarainen [EMAIL PROTECTED] wrote: Good day all I'm trying this switchboard demo (ftp://ftp.lanwan.fi/switchboard) But in the pdf howto they speak about a swb.txt and I dont have that file The swb.txt is there but where did you find the SwB.war file? Steven ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] * INSTRUCTIONS
Roger that, Ill grep. er google for it...thanks... At 04:10 4/23/2004, you wrote: There is the handbook on the homepage and then there is the hitchhikers guid,just not sure where it is ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] * INSTRUCTIONS
Im hoping that light bulb will glimmer on any day now...heh... BTW - I contributed to the Wikki on more than a few occasions when I _finally_ had an epiphany understanding Asterisk, and I plan to do so more in the future. Anon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Play a file
Hello I use asterisk ver 0.7.2 Can I play any wave file into the client riciever without billing count ? I call from A IAX client to B IAX client. B client is not available and I would like to play some file with the message user_is_unavailable.gsm But when I look into my CDR table, this call is billed. I don't want to bill these messages. Is it possible ? thank you -- Vit Bohacek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Play a file
Dudlik wrote: Hello I use asterisk ver 0.7.2 Can I play any wave file into the client riciever without billing count ? I call from A IAX client to B IAX client. B client is not available and I would like to play some file with the message user_is_unavailable.gsm But when I look into my CDR table, this call is billed. I don't want to bill these messages. Is it possible ? *CLI show application NoCDR [Synopsis]: Make sure asterisk doesn't save CDR for a certain call [Description]: NoCDR(): makes sure there won't be any CDR written for a certain call [Synopsis]: Make sure asterisk doesn't save CDR for a certain call [Description]: NoCDR(): makes sure there won't be any CDR written for a certain call thank you ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 481 Call Leg/Transaction Does Not Exist
Hi all, Windows Messenger 4.6behind NAT works fine with * for me, except the NOTIFY forMWI and voicemail. TheNOTIFY message triggers a 481 error.How can I make it right? I am using * current stablerelease. Thanks. Ben
Re: [Asterisk-Users] Play a file
than you and I have Wildcard TE410P in my * server What can I do when a client A call from another telecomunication operator over E1 to my IAX client ? Telecomunication operators usually use the unavailable messages and I thing they don't bill these calls between their customers. How do they do it ? On Fri, 23 Apr 2004 13:18:05 +0200 Michiel Betel [EMAIL PROTECTED] wrote: Dudlik wrote: Hello I use asterisk ver 0.7.2 Can I play any wave file into the client riciever without billing count ? I call from A IAX client to B IAX client. B client is not available and I would like to play some file with the message user_is_unavailable.gsm But when I look into my CDR table, this call is billed. I don't want to bill these messages. Is it possible ? *CLI show application NoCDR [Synopsis]: Make sure asterisk doesn't save CDR for a certain call [Description]: NoCDR(): makes sure there won't be any CDR written for a certain call [Synopsis]: Make sure asterisk doesn't save CDR for a certain call [Description]: NoCDR(): makes sure there won't be any CDR written for a certain call thank you -- Vit Bohacek Senior Network Specialist tel: +420 221 904 332 fax: +420 221 904 303 GSM: +420 724 008 010 e-mail: [EMAIL PROTECTED] www.ha-vel.cz ha-vel voice a.s. sidlo spolecnosti Vodickova 791/41, 110 00 Praha 1 kancelar spolecnosti Na Zderaze 15, 120 00 Praha 2 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Play a file
Dudlik wrote: than you and I have Wildcard TE410P in my * server What can I do when a client A call from another telecomunication operator over E1 to my IAX client ? Telecomunication operators usually use the unavailable messages and I thing they don't bill these calls between their customers. How do they do it ? I can tell you about the situation here in Greece, and can't tell that properly either since I don't have a thorough understanding of several telephony issues: What's done is that the B-Channel is opened while the call is still in progress, signalling-wise. Then a short message is delivered, the channel is closed and the call rejected with an appropriate cause value. Apollon Koutlides ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 SIP Firmware
you have sent a message to me which seems to contain a legal warning on who can read it, or how it may be distributed, or whether it may be archived, etc. i do not accept such email, and have therefore deleted it. do not expect further response. randy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] list batching frequency
subscribers to the digest form of this list do so in order to only receive the email infrequently. in my case, and i suspect others, twice or so a day would be preferred. the list currently batches about every hour. it is sufficiently annoying that one tends to delete batches. i have written to the list admin about this and received no response, undoubtly they are busy reading the mail :-). would anyone reading the digest form object to the admin changing the config so it sends one to three batches a day? randy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problem With zaphfc
Title: Message You don't say which version you are using, but upgrade to RC20a. There were some ISDN Layer 2 issues in earlier versions which have been fixed recently. http://ns1.jnetdns.de/jn/relaunch/asterisk/downloads/bri-stuff-0.0.2rc20a.tar.gz Rgds Tim -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tiziano CrescimbeniSent: 23 April 2004 11:42To: [EMAIL PROTECTED]Subject: [Asterisk-Users] Problem With zaphfc I've this error How i can find the problem? Apr 23 12:24:43 WARNING[131081]: PRI: received TEI check request for TEI = 89Apr 23 12:24:47 WARNING[131081]: PRI: received TEI check request for TEI = 89Apr 23 12:24:48 WARNING[131081]: PRI: !! Got a UA, but i'm in state 1Apr 23 12:24:53 WARNING[131081]: PRI: received TEI check request for TEI = 89Apr 23 12:25:02 WARNING[131081]: PRI: received TEI check request for TEI = 89Apr 23 12:25:03 WARNING[131081]: PRI: !! Got a UA, but i'm in state 1Apr 23 12:25:09 WARNING[131081]: PRI: received TEI check request for TEI = 89Apr 23 12:25:13 WARNING[131081]: PRI: !! Got reject for frame 2, retransmitting frame 2 now, updating n_r!Apr 23 12:25:13 WARNING[131081]: PRI: !! Got reject for frame 3, but we have nothing -- resetting!Apr 23 12:25:23 WARNING[131081]: PRI: received TEI check request for TEI = 89Apr 23 12:25:26 WARNING[131081]: PRI: received TEI check request for TEI = 89Apr 23 12:25:39 WARNING[131081]: Ring requested on channel 1 already in use on span 1. Hanging up owner.Apr 23 12:26:22 WARNING[131081]: Ring requested on channel 2 already in use on span 1. Hanging up owner.Apr 23 12:47:33 WARNING[131081]: PRI: Double assgined TEI!Apr 23 12:47:33 WARNING[131081]: PRI: !! Got a UA, but i'm in state 1Apr 23 12:47:42 WARNING[131081]: PRI: !! Got reject for frame 2, retransmitting frame 2 now, updating n_r!Apr 23 12:47:42 WARNING[131081]: PRI: !! Got reject for frame 2, retransmitting frame 3 now, updating n_r!Apr 23 12:47:42 WARNING[131081]: PRI: !! Got reject for frame 2, retransmitting frame 4 now, updating n_r!Apr 23 12:47:42 WARNING[131081]: PRI: !! Got reject for frame 2, retransmitting frame 5 now, updating n_r!Apr 23 12:47:42 WARNING[131081]: PRI: !! Got reject for frame 2, retransmitting frame 6 now, updating n_r!Apr 23 12:47:42 WARNING[131081]: PRI: !! Got reject for frame 2, retransmitting frame 7 now, updating n_r!Apr 23 12:47:42 WARNING[131081]: PRI: !! Got reject for frame 2, retransmitting frame 8 now, updating n_r!Apr 23 12:47:42 WARNING[131081]: PRI: !! Got reject for frame 2, retransmitting frame 9 now, updating n_r!Apr 23 12:47:42 WARNING[131081]: PRI: !! Got reject for frame 2, retransmitting frame 10 now, updating n_r!Apr 23 12:47:42 WARNING[131081]: PRI: !! Got reject for frame 2, retransmitting frame 11 now, updating n_r!Apr 23 12:47:42 WARNING[131081]: PRI: !! Got reject for frame 2, retransmitting frame 12 now, updating n_r!Apr 23 12:47:42 WARNING[131081]: PRI: !! Got reject for frame 2, retransmitting frame 13 now, updating n_r!Apr 23 12:47:42 WARNING[131081]: PRI: !! Got reject for frame 2, retransmitting frame 14 now, updating n_r!Apr 23 12:47:42 WARNING[131081]: PRI: !! Got reject for frame 2, retransmitting frame 15 now, updating n_r!Apr 23 12:47:42 WARNING[131081]: PRI: !! Got reject for frame 2, retransmitting frame 16 now, updating n_r!Apr 23 12:47:42 WARNING[131081]: PRI: !! Got reject for frame 2, retransmitting frame 17 now, updating n_r!Apr 23 12:47:42 WARNING[131081]: PRI: !! Got reject for frame 2, retransmitting frame 18 now, updating n_r!Apr 23 12:47:43 WARNING[131081]: PRI: !! Got a UA, but i'm in state 1Apr 23 12:47:44 WARNING[131081]: PRI: ACK received outside of window, restartingApr 23 12:48:16 WARNING[16384]: MySQL database sock file not specified. Using defaultApr 23 12:48:16 WARNING[16384]: No '=' (equal sign) in line 34 of mgcp.confApr 23 12:48:16 WARNING[16384]: Ignoring port for nowApr 23 12:49:14 NOTICE[311316]: Unable to create channel of type 'Zap'Apr 23 12:49:24 WARNING[311316]: Timeout, but no rule 't' in context 'archimedia'Apr 23 12:49:38 NOTICE[327700]: Unable to create channel of type 'Zap'Apr 23 12:49:48 WARNING[327700]: Timeout, but no rule 't' in context 'archimedia'Apr 23 12:51:39 WARNING[16384]: MySQL database sock file not specified. Using defaultApr 23 12:51:39 WARNING[16384]: No '=' (equal sign) in line 34 of mgcp.confApr 23 12:51:40 WARNING[16384]: Ignoring port for now
[Asterisk-Users] Indications for New Zealand
http://bugs.digium.com/bug_view_page.php?bug_id=0001474 If you're from NZ and need this, please test if this is the correct setup. Add your comments, positive or negative, to the bug tracker. We need confirmations from the community to move ahead. Thank you! /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 SIP Firmware
On Fri, 23 Apr 2004, Johnson-Perkins, Robert wrote: I have just got 3 Cisco 7960 phones which I would like to connect to Asterisk... However they seem to have v3 SCCP firmware. The same question, posted a few hours before: http://lists.digium.com/pipermail/asterisk-users/2004-April/044025.html ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] smallest phone
Tim Sailer wrote: Folks, I'm looking for a SIP or IAX phone for field techs to take with them when out on service calls. The regular desktop phones are just way too big. Is there anything like the size of a full-sized cell phone? Or smaller, not I doubt that... If a softphone is acceptable what about something like http://www.kauss.org/Stephan/ziaxphone/ Can't get much smaller than that :-) -- The older you get, the better you realize you were. http://ccicolorado.org Exceptional Dogs for Exceptional People - Help Out Today! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Interfacing with an existing phone system
On Thursday 22 April 2004 07:05 pm, Joel Duffield wrote: We want to use asterisk to extend our current phone system. It is a regular plain old system. Has anyone done this before? Absolutely - in a lot of different ways. We would be adding about 4 SIP (probably Cisco) phones to use with asterisk. What kind of card will I need to use for this, FXS or FXO. Neither of those types of cards. You will need an ethernet card/connection to use a SIP phone. Also, before you pay all that money for new phones, you could test/learn using asterisk with free softphones. - Sorry I didn't ask this question very well, I meant how will I interface with the existing phone system, It is an old system so really the only way I have to connect to it is through putting asterisk in place of a phone at an extension. Can I use the four port card? The whole thing is a trial to convince the powers that be to switch the whole system over to VOIP as the old system is on its last legs. Anon Thanks Joel Duffield Near North Business Machines 705-787-0517 Phone 705-787-0554 Fax [EMAIL PROTECTED] www.NearNorthBusiness.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Réf.: Re: [Asterisk-Users] Asterisk with UUI support ?
Can you put this patch on line? (I don't think it's too big...) In my mind, the main objective is to create a special field and force its value in chan_capi.c and check wether it goes through asterisk or not... What do you think of that? Regards -- [EMAIL PROTECTED] wrote: OK, so I'll do that... Is there any infos I need to know about chan_sip.c (because I suppose it's it that I need to play with)? Some stuff is already there. This is a capi debug trace where i SEND UUS1 from a normal ISDN Phone TO an asterisk: Controller/PLCI/NCCI= 0x101 CIPValue= 0x10 CalledPartyNumber = c1555 CallingPartyNumber = 01 815551234 CalledPartySubaddress = default CallingPartySubaddress = default BC = 80 90 a3 LLC = default HLC = 91 81 AdditionalInfo BChannelinformation= 00 00 Keypadfacility = default Useruserdata = 04message Facilitydataarray = default also have a look at chan_capi.c / USERUSERDATA My C knowledge is *very* limited, but i could send out something with some wild patching in chan_capi.c, so it's at least possible... Does anyone know an interesting website where I can find infos about UUI in ISDN (with CAPI maybe?) ? I guess it's somewhere in ITU Q.931, but i dont have this document ;-( I also think this would be a very cool feature (i.e. there's a Simemens PBX that sends out the callername with UUS1), if i can do something else to help, please tell me. Regards Christoph ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] call transfer with consultation
Hello. I am a spanish student, so excuse my English. I have this HW: - 2 X100P PCI with two analog lines plugged in. These lines are two extensions of a panasonic PBX. Zap/1 = X100P -- analog line -- extension #237 PBX Panasonic Zap/2 = X100P -- analog line -- extension #245 PBX Panasonic - 1 TDM20B with two analog telephones plugged in. Zap/3 = TDM20B port 1 Analog phone Zap/4 = TDM20B port 2 Analog phone I must to verify the call transfer with consultation. For example, when an input call comes through X100P, my Zap/3 extension rings. I pickup Zap/3 and I want to transfer the call to Zap/4, but before to establish the call between X100P and Zap/4 I need to request Zap/4 for answering the call. I have already searched along the mailing list. It seems to be easy but I don't know how. Zapata.conf: [channels] ; x100p1 language=es usecallerid=yes echocancel=yes echocancelwhenbridged=yes ;signalling=fxs_ls;=== loop start ;signalling=fxs_gs;=== ground start signalling=fxs_ks;=== kewl start ;immediate=yes context=x100p1 usedistinctiveringdetection=yes callwaitingcallerid=yes callwaiting=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes channel = 1 ;++ x100p1 ; x100p2 language=es usecallerid=yes echocancel=yes echocancelwhenbridged=yes ;signalling=fxs_ls ;signalling=fxs_gs signalling=fxs_ks ;immediate=yes context=x100p2 usedistinctiveringdetection=yes callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes channel = 2 ;++ x100p2 ;++ tdm20b ; canal 2 y 3 ;callgroup=1 ;pickupgroup=1 language=es usecallerid=yes echocancel=yes echocancelwhenbridged=yes signalling=fxo_ks ;immediate=yes context=tdm20b callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes channel = 3-4 ;++ tdm20b ;# Zaptel.conf fxsks=1-2 loadzone=es defaultzone=es fxoks=3-4 loadzone=es defaultzone=es ;# Please I need help. Thank you. _ Do You Yahoo!? Información de Estados Unidos y América Latina, en Yahoo! Noticias. Visítanos en http://noticias.espanol.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] asterisk no card
You don't need a timing source for Music on Hold and have not needed one for a while. I don't recall exactly when this requirement was removed but it was well before 0.7.1. You do still need a timing source for MeetMe and IAX Trunking (which you only want, but not need, if you have lots of calls going between the same two Asterisk servers) On Thu, 2004-04-22 at 23:09, Paul Mahler wrote: You need a timing source for conferencing or music on hold. Voice mail works fine without a timer. If there is no Zaptel card installed, you will have to find timing from a USB driver, or recompile the real time clock. -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco phones
On Fri, 2004-04-23 at 03:12, Paul Tyreman wrote: Why is there such a variation in price between what the two of you have paid to get the SIP image for a 7960 phone ? $8 would be acceptable, but I don't want to have to pay $105 ! The $8 service contract gives you access to the Cisco software images, but you are NOT licensed for these images. The $105 is for buying the actual SIP license. In the summary, the $8 service contract lets you pirate the SIP image, the $105 lets you buy a SIP image license and a CD(?) with the software on it. -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem With zaphfc
Title: Message Yes i use this version Thank's Tiziano - Original Message - From: Robinson Tim-W10277 To: '[EMAIL PROTECTED]' Sent: Friday, April 23, 2004 2:59 PM Subject: RE: [Asterisk-Users] Problem With zaphfc You don't say which version you are using, but upgrade to RC20a. There were some ISDN Layer 2 issues in earlier versions which have been fixed recently. http://ns1.jnetdns.de/jn/relaunch/asterisk/downloads/bri-stuff-0.0.2rc20a.tar.gz Rgds Tim -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tiziano CrescimbeniSent: 23 April 2004 11:42To: [EMAIL PROTECTED]Subject: [Asterisk-Users] Problem With zaphfc I've this error How i can find the problem? Apr 23 12:24:43 WARNING[131081]: PRI: received TEI check request for TEI = 89Apr 23 12:24:47 WARNING[131081]: PRI: received TEI check request for TEI = 89Apr 23 12:24:48 WARNING[131081]: PRI: !! Got a UA, but i'm in state 1Apr 23 12:24:53 WARNING[131081]: PRI: received TEI check request for TEI = 89Apr 23 12:25:02 WARNING[131081]: PRI: received TEI check request for TEI = 89Apr 23 12:25:03 WARNING[131081]: PRI: !! Got a UA, but i'm in state 1Apr 23 12:25:09 WARNING[131081]: PRI: received TEI check request for TEI = 89Apr 23 12:25:13 WARNING[131081]: PRI: !! Got reject for frame 2, retransmitting frame 2 now, updating n_r!Apr 23 12:25:13 WARNING[131081]: PRI: !! Got reject for frame 3, but we have nothing -- resetting!Apr 23 12:25:23 WARNING[131081]: PRI: received TEI check request for TEI = 89Apr 23 12:25:26 WARNING[131081]: PRI: received TEI check request for TEI = 89Apr 23 12:25:39 WARNING[131081]: Ring requested on channel 1 already in use on span 1. Hanging up owner.Apr 23 12:26:22 WARNING[131081]: Ring requested on channel 2 already in use on span 1. Hanging up owner.Apr 23 12:47:33 WARNING[131081]: PRI: Double assgined TEI!Apr 23 12:47:33 WARNING[131081]: PRI: !! Got a UA, but i'm in state 1Apr 23 12:47:42 WARNING[131081]: PRI: !! Got reject for frame 2, retransmitting frame 2 now, updating n_r!Apr 23 12:47:42 WARNING[131081]: PRI: !! Got reject for frame 2, retransmitting frame 3 now, updating n_r!Apr 23 12:47:42 WARNING[131081]: PRI: !! Got reject for frame 2, retransmitting frame 4 now, updating n_r!Apr 23 12:47:42 WARNING[131081]: PRI: !! Got reject for frame 2, retransmitting frame 5 now, updating n_r!Apr 23 12:47:42 WARNING[131081]: PRI: !! Got reject for frame 2, retransmitting frame 6 now, updating n_r!Apr 23 12:47:42 WARNING[131081]: PRI: !! Got reject for frame 2, retransmitting frame 7 now, updating n_r!Apr 23 12:47:42 WARNING[131081]: PRI: !! Got reject for frame 2, retransmitting frame 8 now, updating n_r!Apr 23 12:47:42 WARNING[131081]: PRI: !! Got reject for frame 2, retransmitting frame 9 now, updating n_r!Apr 23 12:47:42 WARNING[131081]: PRI: !! Got reject for frame 2, retransmitting frame 10 now, updating n_r!Apr 23 12:47:42 WARNING[131081]: PRI: !! Got reject for frame 2, retransmitting frame 11 now, updating n_r!Apr 23 12:47:42 WARNING[131081]: PRI: !! Got reject for frame 2, retransmitting frame 12 now, updating n_r!Apr 23 12:47:42 WARNING[131081]: PRI: !! Got reject for frame 2, retransmitting frame 13 now, updating n_r!Apr 23 12:47:42 WARNING[131081]: PRI: !! Got reject for frame 2, retransmitting frame 14 now, updating n_r!Apr 23 12:47:42 WARNING[131081]: PRI: !! Got reject for frame 2, retransmitting frame 15 now, updating n_r!Apr 23 12:47:42 WARNING[131081]: PRI: !! Got reject for frame 2, retransmitting frame 16 now, updating n_r!Apr 23 12:47:42 WARNING[131081]: PRI: !! Got reject for frame 2, retransmitting frame 17 now, updating n_r!Apr 23 12:47:42 WARNING[131081]: PRI: !! Got reject for frame 2, retransmitting frame 18 now, updating n_r!Apr 23 12:47:43 WARNING[131081]: PRI: !! Got a UA, but i'm in state 1Apr 23 12:47:44 WARNING[131081]: PRI: ACK received outside of window, restartingApr 23 12:48:16 WARNING[16384]: MySQL database sock file not specified. Using defaultApr 23 12:48:16 WARNING[16384]: No '=' (equal sign) in line 34 of mgcp.confApr 23 12:48:16 WARNING[16384]: Ignoring port for nowApr 23 12:49:14 NOTICE[311316]: Unable to create channel of type 'Zap'Apr 23 12:49:24 WARNING[311316]: Timeout, but no rule 't' in context 'archimedia'Apr 23 12:49:38 NOTICE[327700]: Unable to create channel of type 'Zap'Apr 23 12:49:48 WARNING[327700]: Timeout, but no rule 't' in context 'archimedia'Apr 23 12:51:39 WARNING[16384]: MySQL database sock file not specified. Using defaultApr 23 12:51:39 WARNING[16384]: No '=' (equal sign) in line 34 of mgcp.confApr 23 12:51:40 WARNING[16384]: Ignoring port for
Re: [Asterisk-Users] Extension buttons
At 2:23 AM + on 4/23/04, Anon wrote: On Friday 23 April 2004 12:33 am, David Krider wrote: I've downloaded the entire archive of articles and searched through them for an answer on this, but I haven't come across one yet. I'm looking to replace a small phone system in my church with Asterisk, and I'm stuck looking for phones. I know that the staff are going to want a button for their commonly-called extensions, but I'm having trouble finding phones that have, say, 10 programmable buttons for this sort of thing. I'm left to conclude that most phones can do this sort of thing by clicking through some combination of buttons. However, it would seem that the average price for a nice SIP phone eliminates the possibility of just ordering some to find out. Can someone please tell me how this is handled in general? For instance, the Polycom 600 doesn't seem to have ANY buttons that can be programmed for particular extensions Not correct - The Polycom SoundPoint IP 600 has 6 buttons on the upper left hand side that can be programmed for particular extensions and speed-dial entries. It also has the ability to support 6 lines, and has extensive directory support. And, strangely, ALL the buttons on the phone can be reprogrammed. Keep in mind this phone uses context-sensitive soft-keys, so it offers much more ability and functionality than can be seen in a low resolution photo on the web. It may suprise you to know that the soft-key implementation is very well done: intuitive, logical, efficient, and easy to use. (Polycom should pay me for posting this ;) Anon OK, so the question may become more focused with Polycom phones then: Is it possible (ignoring Asterisk for the minute) for Polycom phones to indicate visually (on the LCD or on a lighted extension button or something) that a particular line is in use? I would expect this method to be via NOTIFY or SUBSCRIBE calls from a SIP registrar/proxy/call handler upstream. Now, if the answer is Yes, are there instructions anywhere on exactly HOW that is supposed to work, so that someone can start to code these methods into Asterisk? This is one of the missing features when people look at Asterisk as a PBX replacement - the simple task of looking at the phone to see what incoming lines are off-hook or what people are busy is lost, but this is a mandatory requirement for office phone systems. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call transfer with consultation
For example, when an input call comes through X100P, my Zap/3 extension rings. I pickup Zap/3 and I want to transfer the call to Zap/4, but before to establish the call between X100P and Zap/4 I need to request Zap/4 for answering the call. Currently not possible, although here is a workaround since you are using Zap interfaces: Call comes in and you answer it. Hook flash (briefly hang up and pick up the phone again) -- caller is on hold and you can dial the extension you want to transfer it to. Talk to the extension Hook flash again, and now you, the extension and the caller are in a 3-way call. Hang up -- the call is now transfered. Hope this helps. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Adtran TA750 Noise - Email found in subject
Rich, Thanks a bunch, totally understand now and that actually makes total sense. (no need for schematics). This also explains why I used an TA750 to go into a Nortel MICS system, using FXO and no buzz. Totally balanced load from the analog ports on the Nortel across the 5 feet of CAT5 to the FXO on the adtran. Now I need to get rid of some Adtrans --- Anyone lookin to buy? :) Thanks again.Greg Gregory P. Scasny Golden Technologies Inc. http://www.golden-tech.com 219-462-7200 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Thursday, April 22, 2004 6:03 PM To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Subject: [SPAM] - Re: [Asterisk-Users] Adtran TA750 Noise - Email found in subject I have an (actually 2) Adtran TA750s with 8 FXO ports. I get a terrible buzz on every FXO port. If I unplug the Adtran and put an analog phone on each incoming line, I have no buzz. I also have 2 Carrier Access Access Bank Is with 12 FXO ports. When I plug the same analog lines into either one of those, no noise or buzz whatsoever. I went so far as to move the TA750 to within 5 feet of the demark and ran a short CAT5 cable from the demark to the TA705 and still lots of buzz. I have duplicates of every part for the TA750s and swapped every component and cannot get rid of the hum on either unit. I have the power supply of the Adtran grounded. I am out of ideas L. Any assistance would be greatly appreciated. Well, you're the second one in a rather short period of time that has complained about the exact same hum/buzz when the Adtran 750 is used with FXO interfaces and with X100P cards. There was another posting earlier today in which the individual made a comment the 750 FXO interface does not support impedence matching. I thought the statement was rather strange, but since I don't own one of these units, I went to the Adtran site in search of a technical description of the FXO card. I couldn't find one, and for that matter, it appears Adtran has little reference to the 750 being used with any FXO interface. (Its almost like they know there is a problem and removed the 750 FXO options. Selling as FXS only now.) From what I'm hearing/understanding, its all beginning to make sense (believe or not). If the no-impedence-matching is true (or even if the technical words are slightly/somewhat incorrect), then its beginning to appear the Adtran FXO interface is not presenting a balanced interface to the tip ring pstn line. In other words, one side of the line must have some internal electronics hanging on it that disturbs the balance needed for pstn lines, and that imbalance is causing induced AC power (which is extremely common on most pstn lines) to be heard. This is going to be rather difficult to explain without a drawing, but I'll give it a try. The pstn line (all the way from the CO or fiber mux cabinet) is nothing more then twisted copper pairs, that have a very specific number of twists per unit of length. The twists are actually built into the cables to ensure that whatever outside electrical influence exists (such as AC Power), that outside source influences both tip and ring in exactly the same amount. At the end of that cable (whether its in your house or business) if you attached a perfectly balanced piece of equipment, it doesn't make any difference whether that outside influence (in this case, AC power) is ten volts or fifty volts, that influence is cancelled out and not heard. But, its because the attached device (usually an analog phone) presents an equal load to both the tip and ring. (That should be fairly obvious since the typical analog phone doesn't have any real way to create an imbalance since it doesn't have access to ground or AC power. For the real technical types, its the differential voltage between tip and ring that creates the sound.) If one would connect an analog phone to the pstn line that you're having the hum on, and then attach a resister from one side of the line to ground (say, from the tip to ground), you are artifically creating the imbalance that I'm talking about. The analog phone will now have the hum that you're hearing via the Adtran asterisk because of the imbalanced line. The size of the resister (whether 100 ohms or 1,000,000 ohms) will impact the loudness of the hum; the smaller the value the louder the hum. In the olden days of telephony, we use to install repeat coils to isolate the imbalanced equipment (usually customer owned stuff). (Here comes the harder part to describe in words. Really need a visual schematic for this.) Repeat coils were absolutely nothing more then a basic audio transformer with two primary windings and two secondary windings. A couple of 2 ufd capacitors and the repeat coil was all that was needed to isolate the imbalanced piece of equipment from the pstn line, pass the DC component needed for supervision, and elimate the hum. In the
Re: [Asterisk-Users] Problem With zaphfc
rc19 work better for me rc20a is less stable on my configuration (driver crash / line 50% not correctly hangup) At 15:37 23/04/2004, you wrote: Yes i use this version Thank's Tiziano - Original Message - From: mailto:[EMAIL PROTECTED]Robinson Tim-W10277 To: mailto:'[EMAIL PROTECTED]''[EMAIL PROTECTED]' Sent: Friday, April 23, 2004 2:59 PM Subject: RE: [Asterisk-Users] Problem With zaphfc You don't say which version you are using, but upgrade to RC20a. There were some ISDN Layer 2 issues in earlier versions which have been fixed recently. http://ns1.jnetdns.de/jn/relaunch/asterisk/downloads/bri-stuff-0.0.2rc20a.tar.gzhttp://ns1.jnetdns.de/jn/relaunch/asterisk/downloads/bri-stuff-0.0.2rc20a.tar.gz Rgds Tim -Original Message- From: mailto:[EMAIL PROTECTED][EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tiziano Crescimbeni Sent: 23 April 2004 11:42 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Problem With zaphfc I've this error How i can find the problem? Apr 23 12:24:43 WARNING[131081]: PRI: received TEI check request for TEI = 89 Apr 23 12:24:47 WARNING[131081]: PRI: received TEI check request for TEI = 89 Apr 23 12:24:48 WARNING[131081]: PRI: !! Got a UA, but i'm in state 1 Apr 23 12:24:53 WARNING[131081]: PRI: received TEI check request for TEI = 89 Apr 23 12:25:02 WARNING[131081]: PRI: received TEI check request for TEI = 89 Apr 23 12:25:03 WARNING[131081]: PRI: !! Got a UA, but i'm in state 1 Apr 23 12:25:09 WARNING[131081]: PRI: received TEI check request for TEI = 89 Apr 23 12:25:13 WARNING[131081]: PRI: !! Got reject for frame 2, retransmitting frame 2 now, updating n_r! Apr 23 12:25:13 WARNING[131081]: PRI: !! Got reject for frame 3, but we have nothing -- resetting! Apr 23 12:25:23 WARNING[131081]: PRI: received TEI check request for TEI = 89 Apr 23 12:25:26 WARNING[131081]: PRI: received TEI check request for TEI = 89 Apr 23 12:25:39 WARNING[131081]: Ring requested on channel 1 already in use on span 1. Hanging up owner. Apr 23 12:26:22 WARNING[131081]: Ring requested on channel 2 already in use on span 1. Hanging up owner. Apr 23 12:47:33 WARNING[131081]: PRI: Double assgined TEI! Apr 23 12:47:33 WARNING[131081]: PRI: !! Got a UA, but i'm in state 1 Apr 23 12:47:42 WARNING[131081]: PRI: !! Got reject for frame 2, retransmitting frame 2 now, updating n_r! Apr 23 12:47:42 WARNING[131081]: PRI: !! Got reject for frame 2, retransmitting frame 3 now, updating n_r! Apr 23 12:47:42 WARNING[131081]: PRI: !! Got reject for frame 2, retransmitting frame 4 now, updating n_r! Apr 23 12:47:42 WARNING[131081]: PRI: !! Got reject for frame 2, retransmitting frame 5 now, updating n_r! Apr 23 12:47:42 WARNING[131081]: PRI: !! Got reject for frame 2, retransmitting frame 6 now, updating n_r! Apr 23 12:47:42 WARNING[131081]: PRI: !! Got reject for frame 2, retransmitting frame 7 now, updating n_r! Apr 23 12:47:42 WARNING[131081]: PRI: !! Got reject for frame 2, retransmitting frame 8 now, updating n_r! Apr 23 12:47:42 WARNING[131081]: PRI: !! Got reject for frame 2, retransmitting frame 9 now, updating n_r! Apr 23 12:47:42 WARNING[131081]: PRI: !! Got reject for frame 2, retransmitting frame 10 now, updating n_r! Apr 23 12:47:42 WARNING[131081]: PRI: !! Got reject for frame 2, retransmitting frame 11 now, updating n_r! Apr 23 12:47:42 WARNING[131081]: PRI: !! Got reject for frame 2, retransmitting frame 12 now, updating n_r! Apr 23 12:47:42 WARNING[131081]: PRI: !! Got reject for frame 2, retransmitting frame 13 now, updating n_r! Apr 23 12:47:42 WARNING[131081]: PRI: !! Got reject for frame 2, retransmitting frame 14 now, updating n_r! Apr 23 12:47:42 WARNING[131081]: PRI: !! Got reject for frame 2, retransmitting frame 15 now, updating n_r! Apr 23 12:47:42 WARNING[131081]: PRI: !! Got reject for frame 2, retransmitting frame 16 now, updating n_r! Apr 23 12:47:42 WARNING[131081]: PRI: !! Got reject for frame 2, retransmitting frame 17 now, updating n_r! Apr 23 12:47:42 WARNING[131081]: PRI: !! Got reject for frame 2, retransmitting frame 18 now, updating n_r! Apr 23 12:47:43 WARNING[131081]: PRI: !! Got a UA, but i'm in state 1 Apr 23 12:47:44 WARNING[131081]: PRI: ACK received outside of window, restarting Apr 23 12:48:16 WARNING[16384]: MySQL database sock file not specified. Using default Apr 23 12:48:16 WARNING[16384]: No '=' (equal sign) in line 34 of mgcp.conf Apr 23 12:48:16 WARNING[16384]: Ignoring port for now Apr 23 12:49:14 NOTICE[311316]: Unable to create channel of type 'Zap' Apr 23 12:49:24 WARNING[311316]: Timeout, but no rule 't' in context 'archimedia' Apr 23 12:49:38 NOTICE[327700]: Unable to create channel of type 'Zap' Apr 23 12:49:48 WARNING[327700]: Timeout, but no rule 't' in context 'archimedia' Apr 23 12:51:39 WARNING[16384]: MySQL database sock file not specified. Using default Apr 23 12:51:39 WARNING[16384]: No '=' (equal sign) in line 34 of mgcp.conf Apr 23 12:51:40 WARNING[16384]: Ignoring port for now
Re: [Asterisk-Users] Cisco phones
On Fri, 23 Apr 2004, Paul Tyreman wrote: What website do I have to go to in order to buy a SIP image update ? When I bought mine, I did a Google search on their part number: SW-SM-UL-7960 (Cisco SIP license for 7960 IP Phone) Also, read this message: http://lists.digium.com/pipermail/asterisk-users/2004-February/037531.html ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco phones
If the $8 service contract only gives you access to the image, but you aren't really allowed to use it, then why do Cisco offer that contact in the first place ? So are you telling me that to be legal, I need to pay$105, but could get away with $8 ? -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric WielingPosted At: 23 April 2004 14:33Posted To: Asterisk-UsersConversation: [Asterisk-Users] Cisco phonesSubject: Re: [Asterisk-Users] Cisco phones On Fri, 2004-04-23 at 03:12, Paul Tyreman wrote: Why is there such a variation in price between what the two of you have paid to get the SIP image for a 7960 phone ? $8 would be acceptable, but I don't want to have to pay $105 ! The $8 service contract gives you access to the Cisco software images, but you are NOT licensed for these images. The $105 is for buying the actual SIP license. In the summary, the $8 service contract lets you pirate the SIP image, the $105 lets you buy a SIP image license and a CD(?) with the software on it. -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 "In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss." ___Asterisk-Users mailing list[EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zaphfc
Try with : channel = 1-2 Regards, At 11:40 20/04/2004, you wrote: Hello, Here it goes: zaptel.conf: --- span=1,1,3,ccs,ami bchan=1-2 dchan=3 --- zapata.conf --- switchtype = euroisdn signalling = bri_net_ptmp pridialplan=local echocancel=yes immediate=yes group = 1 context=local channel = 1 - Thanks, --- Paulo Loureiro. On Mon, 2004-04-19 at 21:27, Arnaud Pignard wrote: Hello, Can you post zapata.conf and zaptel.conf ? It's seems a config file problem. At 19:32 19/04/2004, you wrote: Hello list, I'm trying to use zaphfc, the module loads ok, and it identifies the hfc boards in the machine. The problem is: whenever i try to ztcfg -vv I get the following: 8x--- Zaptel Configuration == SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1) Channel map: Channel 01: Individual Clear channel (Default) (Slaves: 01) Channel 02: Individual Clear channel (Default) (Slaves: 02) Channel 03: D-channel (Default) (Slaves: 03) 3 channels configured. ZT_SPANCONFIG failed on span 1: Invalid argument (22) 8x-- when I try to start * it bails out with: == Parsing '/etc/asterisk/zapata.conf': Found Apr 19 17:27:34 WARNING[16384]: chan_zap.c:671 zt_open: Unable to specify channel 1: No such device or address Apr 19 17:27:34 ERROR[16384]: chan_zap.c:5338 mkintf: Unable to open channel 1: No such device or address here = 0, tmp-channel = 1, channel = 1 Apr 19 17:27:34 ERROR[16384]: chan_zap.c:7490 setup_zap: Unable to register channel '1' Apr 19 17:27:34 WARNING[16384]: loader.c:313 ast_load_resource: chan_zap.so: load_module failed, returning -1 == Unregistered channel type 'Tor' == Unregistered channel type 'Zap' -- Unregistered channel 1 Apr 19 17:27:34 WARNING[16384]: loader.c:408 load_modules: Loading module chan_zap.so failed! Junk at the beginning 49443303 Can anyone out there using zaphfc, help me on this? Thanks in advance, --- Paulo Loureiro. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] smallest phone
On Fri, Apr 23, 2004 at 08:37:42AM -0500, Eric Wieling wrote: On Fri, 2004-04-23 at 00:39, James H. Thompson wrote: A standard butt set (e.g. http://www.sandman.com/pdf/page81.pdf) combined with a Grandstream (very small) or Sipura ATA would make a pretty small combination and be useful for analog PSTN POTS line testing too. We use a buttset and the beta release of the IAXy (see the mailing list archives if you don't know what an IAXy is) I know what the IAXy is, but I've love to know where to get one! Tim -- Tim Sailer Coastal Internet, Inc. Network and Systems Operations PO Box 726 http://www.buoy.comMoriches, NY 11955 [EMAIL PROTECTED] (631) 399-2910 IAX 17003992910 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P Answer
in the [context] set in zaptel.conf ; exten = 6165551212,1,NoOp exten = 6165551212,2,Wait,2; seconds to wait before pickup exten = 6165551212,3,Answer ; On Fri, 23 Apr 2004, Mark Olliver wrote: Hi, I seam to have a problem working out how to get my X100P to answer after 1 ring. Currently it is working fine and connects to the switchboard menu correctly but just does it after 4 rings, which I would prefer if we could reduce. Thanks Mark -- Mark Olliver Thermeon Europe Ltd. e-Card: http://www.thermeoneurope.com/e-Card/mpo Email [EMAIL PROTECTED] Web www.thermeoneurope.com Support 0906 515 0908 Int. Support +44 1293 864 341 Support Email [EMAIL PROTECTED] Sales +44 1293 864 334 Sales Email [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -Mike == Network Engineer Pathway Internet Services 616.774.3131 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco phones
On Fri, 2004-04-23 at 09:11, Paul Tyreman wrote: If the $8 service contract only gives you access to the image, but you aren't really allowed to use it, then why do Cisco offer that contact in the first place ? Support contracts give you access to all Cisco firmware. So are you telling me that to be legal, I need to pay $105, but could get away with $8 ? Correct. -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco phones
On Fri, 23 Apr 2004, Paul Tyreman wrote: So are you telling me that to be legal, I need to pay $105, but could get away with $8 ? *IF* your phone qualifies for service contract (which is US$ 8), yes. You still will have an illegal copy, and you can also be charged later for all the software you downloaded (read the fineprint for your service contract agreement... if you donwload something you are not entitled for, you will be charged for it) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_capi
Andrea,
Here is a little patch for compiling chan_capi.0.3.1 with latest
asterisk CVS.
I could read in the lists that a new chan_capi.0.3.2 will soon arrive.
In the wait time you can use this patch.
put the patch in the chan_capi directory and tip:
# patch -p1 patch.chan_capi-against-0.3.1.diff
It should compile now.
Have fun !
On Tue, 2004-04-20 at 17:32, Andreas Anderson wrote:
Hi Guys,
does anyone know how to fix chan_capi to work with the current CVS HEAD?
It's no
longer possible to compile after the recent changes in the locking...
Regards,
Andreas
_
Theres never been a better time to get Xtra JetStream @
http://xtra.co.nz/jetstream
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--- chan_capi.bad/chan_capi.c 2004-04-22 20:18:03.0 +0200
+++ chan_capi/chan_capi.c 2004-04-22 20:18:35.0 +0200
@@ -1184,7 +1184,7 @@
tv.tv_sec = 0;
tv.tv_usec = 10;
if ((f-frametype == AST_FRAME_VOICE) (p-i-doDTMF == 1) (p-i-vad !=
NULL)) {
- f = ast_dsp_process(p-c,p-i-vad,f,0);
+ f = ast_dsp_process(p-c,p-i-vad,f);
if (f-frametype == AST_FRAME_NULL) {
return 0;
}
Re: [Asterisk-Users] X100P Answer
On Fri, 23 Apr 2004, Mark Olliver wrote:
I seam to have a problem working out how to get my X100P to answer after
1 ring. Currently it is working fine and connects to the switchboard
menu correctly but just does it after 4 rings, which I would prefer if
we could reduce.
Try this:
zapata.conf:
...
immediate=yes
usecallerid=no
...
=
Check all your rules for the context you included into zapata.conf for
that line and get rid of any wait(x) line.
My extensions.conf:
[inbound]
exten = s,1,Dial(${RECEPTION},25,tr)
exten = s,2,Hangup
exten = h,1,Hangup
=
I'm getting an immediate ring with these settings...
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Off Topic: RE: [Asterisk-Users] :)
On Fri, Apr 23, 2004 at 09:11:48AM +0200, Dave Cotton said: On Fri, 2004-04-23 at 08:43 +0200, Florian Overkamp wrote: Thanks to this message where a virus chose to use my from-address to send its crap from I am now being harassed with many many virus warning messages. A call to anyone operating virusscanners (as I am too): I think we can all do without these reports - over 90% of all virusses using email has faked from-headers anyway :-P Florian, don't forget that the vast majority of virus scanners have been set up by people only used to using the very OS that's caused the problem in the first place. They have no idea whatsoever of how to configure something, they just click the Install button. The scanner writers are the problem, they've seen a wonderful way of spamming and then claiming not me guv, honest. Perhaps Gates can use some of the vast profit he's just announced to sort his crap out. Pigs might fly. Thanks to Exim and Exiscan, Most of these stupid virus reports are rejected at SMTP time with a cluebat message... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco phones
Does anyone have a part number or know of anywhere in the UK that resells the image or the license or both? On Fri, 2004-04-23 at 03:12, Paul Tyreman wrote: Why is there such a variation in price between what the two of you have paid to get the SIP image for a 7960 phone ? $8 would be acceptable, but I don't want to have to pay $105 ! The $8 service contract gives you access to the Cisco software images, but you are NOT licensed for these images. The $105 is for buying the actual SIP license. In the summary, the $8 service contract lets you pirate the SIP image, the $105 lets you buy a SIP image license and a CD(?) with the software on it. -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 3com SIP phone working with asterisk
Hello everyone, I just like to let you know that I tested Asterisk with 3COM SIP phones and it worked fine. The 3Com phones are old ones with the same look of NBX 2102 phone but different product number: P/N: 655005001 Rev B There is no special set up except that I have to specifically put allow=ulaw in sip.conf. Otherwise, there is codec unrecognized error. [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) allow=ulaw; Allow all codecs Lisa ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] smallest phone
why not wisip? its size its like a regular cellphone and it uses wifi Miguel Cavazos On Fri, 2004-04-23 at 08:00, Chris Hirsch wrote: Tim Sailer wrote: Folks, I'm looking for a SIP or IAX phone for field techs to take with them when out on service calls. The regular desktop phones are just way too big. Is there anything like the size of a full-sized cell phone? Or smaller, not I doubt that... If a softphone is acceptable what about something like http://www.kauss.org/Stephan/ziaxphone/ Can't get much smaller than that :-) -- The older you get, the better you realize you were. http://ccicolorado.org Exceptional Dogs for Exceptional People - Help Out Today! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CLI command
Hi all, Here is a simple question. How can I know if a call is in pass-thru mode, i.e. * is not in the media path??? Thanks. Ben
[Asterisk-Users] Info abaut zaphfc
I'm trying to correct the cid for italy because when arrive a call cid display the number without the initial 0 and when i want to redial the missed call i can't because the number is wrong Thank's Tiziano
[Asterisk-Users] SIP to H323 with no joy
Greetings and salutations to all... I'm having a bit of a problem getting a SIP phone (Xten) to call an H323 Cisco ATA-186. Both devices can call into the * and get the demo, voicemail, etc... I'm pretty sure my problem is in my configs as it feels like a stupid error and to prove this to myself I set tcpdump on the * box to capture all UDP traffic going to and from the ATA-186. If I call the * box from the ATA tcpdump sees all. When I try to call the ATA from the SIP phone tcpdump sees nothing at all and my SIP phone times out. I also ran a port scan on the ATA to make sure everything on it is as it should be. Believe me, I wish I could use SIP for everything but I have no choice in the matter and upper management doesn't listen to reason (imagine that) so this is what I'm forced to deal with. I'm sure it's a stupid mistake on my part. I just need someone to hit me with the cluebat and open my eyes a little. A little about the * box OS = debian (woody) Asterisk CVS-04/08/04-09:04:44 no firewalls, just on a LAN test seg.. And now for the configs /etc/asterisk/h323.conf [general] port = 1720 bindaddr = 0.0.0.0 ; allow=gsm dtmfmode=rfc2833 context=default ; [2001] type=friend host=192.168.1.51 context=from-h323 ;incominglimit=4 /etc/asterisk/sip.conf [general] port = 5060 bindaddr = 0.0.0.0 context = default ; disallow=all allow=gsm ; [2000] type=friend username=2000 secret=blah host=192.168.1.50 nat=1 context=from-sip /etc/asterisk/extensions.conf [general] ; static=yes writeprotect=yes ; [globals] CONSOLE=Console/dsp ; [default] include = from-sip include = from-h323 ;--- [from-sip] exten = 2001,1,Answer exten = 2001,2,Dial(H323/198.135.222.192|30|r) exten = 2001,102,Playback(away-naughty-boy) exten = 2001,103,Hangup ; testing :) exten = 555,1,Answer exten = 555,2,Wait,2 exten = 555,3,Playback(wrong-try-again-smarty) exten = 555,4,Hangup ;--- [from-h323] exten = 2000,1,Answer exten = 2000,2,Wait,2 exten = 2000,3,Dial(SIP/2000,20) exten = 2000,105,Playback(away-naughty-boy) exten = 2000,106,Hangup Anybody here wanna beat me to death with the cluebat please? I would appreciate it! :) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P Answer
On Fri, 23 Apr 2004 14:55:52 +0100, Mark Olliver wrote Hi, I seam to have a problem working out how to get my X100P to answer after 1 ring. Currently it is working fine and connects to the switchboard menu correctly but just does it after 4 rings, which I would prefer if we could reduce. I found recently that my X100P was getting two rings before answer. That´s because the way Caller ID works in US; it sends the info after the first ring and my board was waiting for it. I disabled caller ID on zapata.conf using usecallerid=no and now it aswers on first ring. Gelson ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco phones
I have three questions to ask about this: 1) How do I know if my phone qualifies for a service contrct ? 2) Where do I buy a service contract from ? 3) How will Cisco know that I have downloaded a image that I don't have a licence for ? Thanks, Paul. -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hermann WeckePosted At: 23 April 2004 15:37Posted To: Asterisk-UsersConversation: [Asterisk-Users] Cisco phonesSubject: Re: [Asterisk-Users] Cisco phones On Fri, 23 Apr 2004, Paul Tyreman wrote: So are you telling me that to be legal, I need to pay $105, but could get away with $8 ? *IF* your phone qualifies for service contract (which is US$ 8), yes. You still will have an illegal copy, and you can also be charged later for all the software you downloaded (read the fineprint for your service contract agreement... if you donwload something you are not entitled for, you will be charged for it) ___Asterisk-Users mailing list[EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 3com SIP phone working with asterisk
interesting... did you tried all the function? ie, can you put a call on hold, and more important do blind supervised transfer? what about the prices? more or less, just to have an idea... tnx, Matteo Il ven, 2004-04-23 alle 17:08, Lisa Xie ha scritto: Hello everyone, I just like to let you know that I tested Asterisk with 3COM SIP phones and it worked fine. The 3Com phones are old ones with the same look of NBX 2102 phone but different product number: P/N: 655005001 Rev B There is no special set up except that I have to specifically put allow=ulaw in sip.conf. Otherwise, there is codec unrecognized error. [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) allow=ulaw; Allow all codecs Lisa ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] smallest phone
why not wisip? its size its like a regular cellphone and it uses wifi Because it sucks ass? Check the archives for some very valid gripes about the device. -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MP3 encoding of Monitor files
I have having problems trying to take a file recorded with Monitor and convert it to MP3. When I use 'play' to play the .wav file, it sounds fine. After bladenc'ing it, it plays at lightening speed, and the voices are all high pitch. I tried using sox to resample to 32000 before encoding, but that didnt work either. Do any of you convert your .wav files to mp3? Monitor call: Monitor(wav|test) 'file' output: test.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz Sox resample: sox test.wav -r 32000 newtest.wav Bladeenc call: bladeenc newtest.wav newtest.mp3 mpg123 newtest.mp3 # sounds like Im listening in fast-forward mode... Any suggestions on how I can get mp3 versions of files produced by Monitor? On Thu, 2004-04-22 at 15:49, Roscinante wrote: On Thu, 22 Apr 2004, Dennis Sorge wrote: Any recommendations for ripping my .wavs to MP3's? I'm running Mandrake 9.2 for a potential music server. Thank you in advance for your suggestions. I use bladeenc, I imagine there is some spiffy front end for it out there somewhere.. ___ Lug-nuts mailing list [EMAIL PROTECTED] http://felix.mikesoffice.org/mailman/listinfo/lug-nuts ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco phones
Does anyone have a part number or know of anywhere in the UK that resells the image or the license or both? Matt, I have tried www.cisilion.com/ for a price on the license, but so far have not had a reply. These are the only place I have found to sell the license. The support contracts are a little easier to find, try www.microwarehouse.co.uk and http://uk.insight.com - though I certainly cannot find anything as cheap as $8! (Obviously UK equiv) Paul Any use of, or any action relying upon, information in an email by persons other than the intended recipient is prohibited. Although this e-mail and its attachments have been scanned and are believed to be free from any virus, we cannot guarantee a communication to be free of all viruses nor accept any responsibility for viruses. It is the responsibility of the recipient to ensure that they are virus free. The views expressed by the sender (Paul A. Nichols) of this message do not necessarily reflect those of NT-2000. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] smallest phone
If you do that, you'll have to carry around a wireless access point as well. Nathan - Original Message - From: Miguel Cavazos [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, April 23, 2004 10:08 AM Subject: Re: [Asterisk-Users] smallest phone why not wisip? its size its like a regular cellphone and it uses wifi Miguel Cavazos On Fri, 2004-04-23 at 08:00, Chris Hirsch wrote: Tim Sailer wrote: Folks, I'm looking for a SIP or IAX phone for field techs to take with them when out on service calls. The regular desktop phones are just way too big. Is there anything like the size of a full-sized cell phone? Or smaller, not I doubt that... If a softphone is acceptable what about something like http://www.kauss.org/Stephan/ziaxphone/ Can't get much smaller than that :-) -- The older you get, the better you realize you were. http://ccicolorado.org Exceptional Dogs for Exceptional People - Help Out Today! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7940/7960 SIP functionality questions
Hi, Would it be possible for you to provide some more info on this. I have just bought a Cisco 7960 on eBay, but only now has the reality of needing a login to upgrade to SIP become clear. Can you tell me how you managed to get your phone going on Asterisk without the image change ? Thanks, Paul, -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Louis van DompselaarPosted At: 22 April 2004 07:39Posted To: Asterisk-UsersConversation: [Asterisk-Users] Cisco 7940/7960 SIP functionality questionsSubject: Re: [Asterisk-Users] Cisco 7940/7960 SIP functionality questions There are two SCCP modules, but I haven't heard about anyone using 7940/60s with SCCP and Asterisk. Let me be the first to say that I do, then. Three 7940g over Skinny without any problems.Some initial setup trouble as this setup isn't really documented anywhere. But no Cisco login,so no SIP and no choice... ___Asterisk-Users mailing list[EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Extension buttons
On 06:36 AM 4/23/2004, John Todd wrote: Is it possible (ignoring Asterisk for the minute) for Polycom phones to indicate visually (on the LCD or on a lighted extension button or something) that a particular line is in use? I would expect this method to be via NOTIFY or SUBSCRIBE calls from a SIP registrar/proxy/call handler upstream. It appears the answer to this is yes. In the Polycom config, you can define a line as private or shared. If you define the line as shared, then the Polycom issues a SUBSCRIBE for the line so configured. Note, I only tested this against Asterisk to see if the line would function as a private line with asterisk while set to shared (basically expecting asterisk to ignore the SUBSCRIBE). However this wasn't the case, the line was non-functional with Asterisk when configured as shared. When I had the phone in my hands, unfortunately I didn't have time to sniff the SIP traffic and see what exactly it was doing. Now, if the answer is Yes, are there instructions anywhere on exactly HOW that is supposed to work, so that someone can start to code these methods into Asterisk? This is one of the missing features when people look at Asterisk as a PBX replacement - the simple task of looking at the phone to see what incoming lines are off-hook or what people are busy is lost, but this is a mandatory requirement for office phone systems. I would suspect that it follows RFC3265 definition of SUBSCRIBE/NOTIFY, but thats merely a guess. The SNOM 200 also issues a SUBSCRIBE message when configured to do so, however it seems to still function as a normal line even though it's configured for a shared style line. Currently there is no way to add multiple sip UA entries for the same line in asterisk's sip.conf. Internally, if Asterisk uses the SIP extension as defined in sip.conf as a unique identifier for the line, then the changes look to be quite significant. I would think the first step would be to modify Asterisk to support shared lines in general using the SUBSCRIBE/NOTIFY method as described in the RFC. Unfortunately my c skills border on non-existant and a hack so any chance of me doing this is out the window. -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] * INSTRUCTIONS
On Fri, 23 Apr 2004 03:55:57 -0400, tmpm [EMAIL PROTECTED] wrote: Might I humbly request someone, somewhere in the community establish a dummies guide to asterisk kind of site, that explains in detail what the cryptic scripts actually do, line by line. The Wiki is helpful, but unless you were in on the movie from the first part, the scene discussions are moot. If you haven't seen the movie yet, the hardest part will be understanding the dial plan. http://www.asteriskdocs.org/stable/docs-html/c511.html gives some insight, but http://www.loligo.com/asterisk/current/ is a working example. I finally figured out what was going on by poring over his extensions.conf ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] H323 error
While calling to H323 peer *CLI 1:22:59.944 H225 Caller:81e5c48 assert.cxx(105) PWLib Assertion fail: Invalid array element, file /root/pwlib/include/ptlib/array.h, line 1183, Error=115 Abort, Core dump, Ignore?*CLI *CLI show versionAsterisk CVS-04/22/04-23:56:01 built by [EMAIL PROTECTED] on a i686 running Linux*CLI
Re: [Asterisk-Users] Cisco phones
On Fri, 23 Apr 2004, Paul Tyreman wrote: I have three questions to ask about this: 1) How do I know if my phone qualifies for a service contrct ? When you (try to) buy your service contract, you will need to give the model and serial number of the item you are trying to include into your contract. If the item qualifies, then you are approved. I had some problems adding mine. For some reason the contract was made under the previous owner name. This was promptly solved, BTW! 2) Where do I buy a service contract from ? Any Cisco partner around the world: http://www.cisco.com/pcgi-bin/cpn/cpn_pub_bassrch.pl (find a Service Provider) 3) How will Cisco know that I have downloaded a image that I don't have a licence for ? Every time you want to download an image, they will present you a copyright notice and an agreement to be accepted. If you choose to download that image, it is registered into their database and match against your service contract. If you have a service contract for a 7960 SCCP and is downloading an image for a 12416 router you *MAY* get charged for that... *MAY* does not mean that you *will*. But don't complain if you receive an invoice or a credit card charge billing for that download. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MP3 encoding of Monitor files
On Fri, 2004-04-23 at 10:33, Mike Machado wrote: I have having problems trying to take a file recorded with Monitor and convert it to MP3. When I use 'play' to play the .wav file, it sounds fine. After bladenc'ing it, it plays at lightening speed, and the voices are all high pitch. I tried using sox to resample to 32000 before encoding, but that didnt work either. Do any of you convert your .wav files to mp3? Why would this matter here? Also some more recent sox builds are capable of encoding to mp3. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MP3 encoding of Monitor files
use lame Il ven, 2004-04-23 alle 17:33, Mike Machado ha scritto: I have having problems trying to take a file recorded with Monitor and convert it to MP3. When I use 'play' to play the .wav file, it sounds fine. After bladenc'ing it, it plays at lightening speed, and the voices are all high pitch. I tried using sox to resample to 32000 before encoding, but that didnt work either. Do any of you convert your .wav files to mp3? Monitor call: Monitor(wav|test) 'file' output: test.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz Sox resample: sox test.wav -r 32000 newtest.wav Bladeenc call: bladeenc newtest.wav newtest.mp3 mpg123 newtest.mp3 # sounds like Im listening in fast-forward mode... Any suggestions on how I can get mp3 versions of files produced by Monitor? On Thu, 2004-04-22 at 15:49, Roscinante wrote: On Thu, 22 Apr 2004, Dennis Sorge wrote: Any recommendations for ripping my .wavs to MP3's? I'm running Mandrake 9.2 for a potential music server. Thank you in advance for your suggestions. I use bladeenc, I imagine there is some spiffy front end for it out there somewhere.. ___ Lug-nuts mailing list [EMAIL PROTECTED] http://felix.mikesoffice.org/mailman/listinfo/lug-nuts ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P Answer
You've probably got callerID enabled in zapata.conf. That will cause a wait of several rings whilst * looks for the caller ID info. Since this only works in the US (or pkaces with similar phone systems), disabling it in other territories saves the ring delay. Make sure you have this in zapata.conf usecallerid=no IAin --On Friday, April 23, 2004 2:55 pm +0100 Mark Olliver [EMAIL PROTECTED] wrote: Hi, I seam to have a problem working out how to get my X100P to answer after 1 ring. Currently it is working fine and connects to the switchboard menu correctly but just does it after 4 rings, which I would prefer if we could reduce. Thanks Mark -- Mark Olliver Thermeon Europe Ltd. e-Card: http://www.thermeoneurope.com/e-Card/mpo Email [EMAIL PROTECTED] Web www.thermeoneurope.com Support 0906 515 0908 Int. Support +44 1293 864 341 Support Email [EMAIL PROTECTED] Sales +44 1293 864 334 Sales Email [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco phones
Paul Tyreman wrote: I have bough a cisco phone on eBay to use with Asterisk, but according to that website, you need a contract with Cisco systems to upgrade the phone to work with SIP. I am guessing the phone that I get won't come with that as it was used with the cisco call manager software in the past. Can I still use this phone with Asterisk, or have I waited my money ? I believe that s correct - the phone comes using SCCP - you have to get a SIP image. The service contract on the phone costs 8 bucks a year. One you upgrade the phone you'll know its SIP capable by the SIP in the upper right hand corner of the display. -- Rock River Internet Roger Grunkemeyer 202 W. State St, 8th Floor[EMAIL PROTECTED] Rockford, IL 61101 815-968-9888 x101 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco phones
All I can find on that Cisco website is this: http://www.cisco.com/pcgi-bin/cpn/cpn_match_result.pl?CurPosition=0Direction=ResultType=ECsearch_id=156576tab_name=findspcountry_id=GB I can't see the likes of BT, O2, Vodaphone etc wanting to deal with me ! -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hermann WeckePosted At: 23 April 2004 17:23Posted To: Asterisk-UsersConversation: [Asterisk-Users] Cisco phonesSubject: Re: [Asterisk-Users] Cisco phones On Fri, 23 Apr 2004, Paul Tyreman wrote: I have three questions to ask about this: 1) How do I know if my phone qualifies for a service contrct ? When you (try to) buy your service contract, you will need to give the model and serial number of the item you are trying to include into your contract. If the item qualifies, then you are "approved". I had some problems adding mine. For some reason the contract was made under the previous owner name. This was promptly solved, BTW! 2) Where do I buy a service contract from ? Any Cisco partner around the world: http://www.cisco.com/pcgi-bin/cpn/cpn_pub_bassrch.pl(find a Service Provider) 3) How will Cisco know that I have downloaded a image that I don't have a licence for ? Every time you want to download an image, they will present you a copyright notice and an agreement to be accepted. If you choose to download that image, it is registered into their database and match against your service contract. If you have a service contract for a 7960 SCCP and is downloading an image for a 12416 router you *MAY* get charged for that...*MAY* does not mean that you *will*. But don't complain if you receive an invoice or a credit card charge billing for that download. ___Asterisk-Users mailing list[EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zaphfc
How i can obtain a complete caller ID from ISDN zaphfc in italy because i obtain a caller id without a initial 0 (for example cid=305001010 the correct number is 0305001010) Thank's Tiziano
[Asterisk-Users] Planning Asterisk
Hello, I'm planning to convert my phone system to Asterisk, as I've outgrown my TalkSwitch system. I have a few questions for experienced * users, most of which can be answered yes/no. Current Setup: - Talkswitch 48NLS (4CO/8Ext) phone system. - One CO line, two Vonage lines, one Voicepulse line connected to phone system - A third Vonage line directly connected to a fax machine - A sipgate.de line connected through Port#2 of VoicePulse's Sipura to a stand-alone phone. Getting VoicePulse (recently) and finding sipgate.de pushed me over the 4-line limit of the Talkswitch PBX, plus there are some shortcomings to Talkswitch which I could, but don't want to live with. To get my CO and SIP lines connected I can: 1. Use a voice-modem as FXO? 2. Use a Digium X100P? What's the advantage over using a voice-modem? 3. Set up * as a SIP client for VoicePulse and sipgate.de. I could add lines via broadvoice.com and FWD? And if I'm REALLY lucky, I could even convince Vonage to allow open access and connect directly? 4. Is anyone running an ATA186 into an FXO device? Sound-quality? Are there multi-FXO cards, because I'm afraid I'll be running out of PCI slots. To get my extensions connected, I can: 1. Use a Digium TDM400P? 2. Use one ore more Sipuras? 3. Use any Software IP Phone? 4. Use any Hardware IP Phone? TDM400P cost $75/port, while the Sipuras are only $50/port. Is there an advantage to using the Digium? Now once everything connected, it'll probably take me a while to get things configured. I assume that I can do pretty much anything I want, just as long as I have access to the sources. Can I: 1. Set up auto-attendants based on the incoming phone line? Based on id of the caller? 2. Set up least-cost call routing? 3. Have integrated dialing plans, such as -- 1 xxx yyy = call outside line 011 = call internationally * xxx = call extension xxx # 4 xxx yyy = call using outside line #4 911 = call 911 using actual landline (My wife needs to be able to use this too) From some of your who have set up and are maintaining * PBXs, how difficult is it to get started for someone who doesn't do linux 8 hours a day (I'm a PC guy, but am maintaining a dedicated linux server for webhosting). What's the preferred linux distro for running Asterisk? I have RH8 and RH9 here. I think that's all -- thanks in advance for your help answers! -- Jay ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zaphfc
You can do something like :
[incoming]
exten = s,1,Answer
exten = s,2,SetCallerID(0${CALLERID})
enten = s,3,
There is maybe a better way to do the samething.
At 18:40 23/04/2004, you wrote:
How i can obtain a complete caller ID from ISDN zaphfc in italy
because i obtain a caller id without a initial 0 (for example
cid=305001010 the correct number is 0305001010)
Thank's Tiziano
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with instalation T100P
When I do modeprobe wct1xxp I get it : modprobe wct1xxp /lib/modules/2.4.18-386/misc/zaptel.o: unresolved symbol create_proc_entry_R1b235e62 /lib/modules/2.4.18-386/misc/zaptel.o: unresolved symbol devfs_unregister_Re139a4b3 /lib/modules/2.4.18-386/misc/zaptel.o: unresolved symbol __pollwait_Rdead6af1 /lib/modules/2.4.18-386/misc/zaptel.o: unresolved symbol proc_mkdir_Rbf18a3b5 /lib/modules/2.4.18-386/misc/zaptel.o: unresolved symbol devfs_register_R29137d26 /lib/modules/2.4.18-386/misc/zaptel.o: unresolved symbol remove_wait_queue_R323c1df1 /lib/modules/2.4.18-386/misc/zaptel.o: unresolved symbol devfs_generate_path_Rd13d5c75 /lib/modules/2.4.18-386/misc/zaptel.o: unresolved symbol devfs_mk_symlink_R8d0baa62 /lib/modules/2.4.18-386/misc/zaptel.o: unresolved symbol remove_proc_entry_R68edbe93 /lib/modules/2.4.18-386/misc/zaptel.o: unresolved symbol add_wait_queue_R1278859d /lib/modules/2.4.18-386/misc/zaptel.o: unresolved symbol devfs_mk_dir_Re94ca1dd /lib/modules/2.4.18-386/misc/zaptel.o: unresolved symbol devfs_register_chrdev_R982a9871 /lib/modules/2.4.18-386/misc/zaptel.o: insmod /lib/modules/2.4.18-386/misc/zaptel.o failed /lib/modules/2.4.18-386/misc/zaptel.o: insmod wct1xxp failed Can somebody tell what does it mean and how to fix it ? Bartek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MP3 encoding of Monitor files
lame did the same thing. The reason I ask this on the asterisk list is that .wav files I record from other sources encode just fine. I think the hitch is the sample rates produced by asterisk. File recorded by gnome sound recorder (lame/bladeenc encode just fine): RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, stereo 44100 Hz vs File recorded with Monitor: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz I will give the newer version of sox a try. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Exception flag warnings
I keep seeing the following errors in my asterisk logs: Apr 23 12:13:36 WARNING[1226062640]: Exception flag set on 'SIP/Phone1-c016', but no exception handler Apr 23 12:23:37 WARNING[1268026160]: You might not have the soxmix installed and available in the path, please check. The soxmix one is more of a mystery, as soxmix is in the path, and asterisk always muxes the -in.wav -out.wav without any problems.. It looks like this one may be the way the return code is checked for error in res/res_monitor.c -Mike ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] oh323 goes silent after 5 seconds
I have this problem trying to talk to an ADDPAC gateway using oh323, when I call the sound is great for the first 5 seconds then it goes almost silent... all you can hear are some clicks every once in a while. Anybody seen this can point me to some config settings to change? Regards, Victor Perez
[Asterisk-Users] Festival problems
After patching and installing Festival, I am unable to get it to do
anything useful. I get the following error message on the * console when I
dial the test extension:
Parsing '/etc/asterisk/festival.conf': Found
Apr 23 13:43:06 WARNING[1226062640]: app_festival.c:382 festival_exec:
Strings do not match
My /etc/asterisk/festival.conf looks like this:
[general]
host=localhost
port=1314
usecache=yes
cachedir=/var/lib/asterisk/festivalcache/
festivalcommand=(tts_textasterisk %s 'file)(quit)\n
If it helps, I just tried turning usecache off and I didn't get an error on
the console, but still no speech.
My extension looks like this:
exten = 603,1,Answer()
exten = 603,2,Festival('this is a test testing 1 2 3')
exten = 603,3,Wait(2)
exten = 603,4,Goto(s,6)
The Goto never gets executed either.
The festival_server.log is showing that the server is accepting the
connection.
Any ideas?
-J
--
Jeff Workman | [EMAIL PROTECTED] | http://www.pimpworks.org
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fax problem
Hi, We have a machine with an *'s with Digium TDM400P and connected wit other machine with *'s an TDM400P too. Well, I have a fax connected to each machine, and the protocol in the middle is IAX2 alaw. The fax between two fax, on in each machine, not work. The fax answer, but error in comm. Which can be the problem ?. What can I do to find the problem ? Thanks, in advance, Pedro ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with instalation T100P
On Fri, 2004-04-23 at 12:35, Bartosz Jozwiak wrote: When I do modeprobe wct1xxp I get it : modprobe wct1xxp /lib/modules/2.4.18-386/misc/zaptel.o: unresolved symbol create_proc_entry_R1b235e62 snip /lib/modules/2.4.18-386/misc/zaptel.o: insmod /lib/modules/2.4.18-386/misc/zaptel.o failed /lib/modules/2.4.18-386/misc/zaptel.o: insmod wct1xxp failed Can somebody tell what does it mean and how to fix it ? A search through recent archives, would show at least 1 instance of this, and any broader searching will show many instances. Your problem is related to kernel module versions. Happy searching. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with instalation T100P
On Fri, 2004-04-23 at 12:35, Bartosz Jozwiak wrote: When I do modeprobe wct1xxp I get it : modprobe wct1xxp /lib/modules/2.4.18-386/misc/zaptel.o: unresolved symbol create_proc_entry_R1b235e62 snip /lib/modules/2.4.18-386/misc/zaptel.o: insmod /lib/modules/2.4.18-386/misc/zaptel.o failed /lib/modules/2.4.18-386/misc/zaptel.o: insmod wct1xxp failed Can somebody tell what does it mean and how to fix it ? A search through recent archives, would show at least 1 instance of this, and any broader searching will show many instances. Your problem is related to kernel module versions. Happy searching. -- Steven Critchfield [EMAIL PROTECTED] Well I've been searching on Google but did not find any helpful information :( bartek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom registration
Try following the instructions at http://www.voip-info.org/wiki-Polycom+Phones I think you don't have your MACADDRESS.cfg file set right. I've never used the web interface. If it still doesn't work after that, write back. John P.S. Make sure you use a good xml editor when fixing up the cfg files. Olle E. Johansson wrote: Roger wrote: I have a PolyCom Soundpoint 500 sip phone. I'm tring to get the phone registered on an asterisk box but am having no luck. I get the following errors 192.168.22.196 being the phone and 22.254 being the asterisk box.. Apr 23 11:41:33 NOTICE[1133742896]: chan_sip.c:5623 handle_request: Registration from '110 sip:[EMAIL PROTECTED]' failed for '192.168.22.196' THe SIP uri looks strange. Please include a full SIP debug of a registration attempt. /O Apr 23 11:42:05 NOTICE[1133742896]: chan_sip.c:5623 handle_request: Registration from '110 sip:[EMAIL PROTECTED]' failed for '192.168.22.196' Apr 23 11:42:05 NOTICE[1133742896]: chan_sip.c:5623 handle_request: Registration from '110 sip:[EMAIL PROTECTED]' failed for '192.168.22.196' Apr 23 11:42:37 NOTICE[1133742896]: chan_sip.c:5623 handle_request: Registration from '110 sip:[EMAIL PROTECTED]' failed for '192.168.22.196' Attempting to dial out from the Polycom Phones gives a fast busy.. Below I've included my sip.conf file - I'm wanting to set phone as x110. [110] type=friend username=110 secret=test host=dynamic context=home callgroup=1 pickupgroup=1 canreinvite=yes dtmfmode=rfc2833 ;dtmfmode=inband ;[EMAIL PROTECTED]; put in for voicemail notification callerid=Polycom 110 ; put in for internal caller id only I've reset the phone to factory defaults and started from scratch but still - no dice when it comes to registering this puppy. I used the web interface to specify the username/password but still nothing. Any ideas or docs I could look at to get this Polycom phone setup? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with instalation T100P
Is this a new kernel? Did you recompile your modules under the new kernel after making it? John Bartosz Jozwiak wrote: When I do modeprobe wct1xxp I get it : modprobe wct1xxp /lib/modules/2.4.18-386/misc/zaptel.o: unresolved symbol create_proc_entry_R1b235e62 /lib/modules/2.4.18-386/misc/zaptel.o: unresolved symbol devfs_unregister_Re139a4b3 /lib/modules/2.4.18-386/misc/zaptel.o: unresolved symbol __pollwait_Rdead6af1 /lib/modules/2.4.18-386/misc/zaptel.o: unresolved symbol proc_mkdir_Rbf18a3b5 /lib/modules/2.4.18-386/misc/zaptel.o: unresolved symbol devfs_register_R29137d26 /lib/modules/2.4.18-386/misc/zaptel.o: unresolved symbol remove_wait_queue_R323c1df1 /lib/modules/2.4.18-386/misc/zaptel.o: unresolved symbol devfs_generate_path_Rd13d5c75 /lib/modules/2.4.18-386/misc/zaptel.o: unresolved symbol devfs_mk_symlink_R8d0baa62 /lib/modules/2.4.18-386/misc/zaptel.o: unresolved symbol remove_proc_entry_R68edbe93 /lib/modules/2.4.18-386/misc/zaptel.o: unresolved symbol add_wait_queue_R1278859d /lib/modules/2.4.18-386/misc/zaptel.o: unresolved symbol devfs_mk_dir_Re94ca1dd /lib/modules/2.4.18-386/misc/zaptel.o: unresolved symbol devfs_register_chrdev_R982a9871 /lib/modules/2.4.18-386/misc/zaptel.o: insmod /lib/modules/2.4.18-386/misc/zaptel.o failed /lib/modules/2.4.18-386/misc/zaptel.o: insmod wct1xxp failed Can somebody tell what does it mean and how to fix it ? Bartek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with instalation T100P
On Fri, 2004-04-23 at 13:04, Bartosz Jozwiak wrote: On Fri, 2004-04-23 at 12:35, Bartosz Jozwiak wrote: When I do modeprobe wct1xxp I get it : modprobe wct1xxp /lib/modules/2.4.18-386/misc/zaptel.o: unresolved symbol create_proc_entry_R1b235e62 snip /lib/modules/2.4.18-386/misc/zaptel.o: insmod /lib/modules/2.4.18-386/misc/zaptel.o failed /lib/modules/2.4.18-386/misc/zaptel.o: insmod wct1xxp failed Can somebody tell what does it mean and how to fix it ? A search through recent archives, would show at least 1 instance of this, and any broader searching will show many instances. Your problem is related to kernel module versions. Happy searching. Well I've been searching on Google but did not find any helpful information :( Notice the fact that I gave you new clues above. Surely you didn't spend a considerable bit of effort searching with these new clues if you replied in 2 minutes. Next clue is that you MUST become one with The Great and Powerful Google if you plan on getting anywhere in open source software. Responses like yours with no doubt you are giving up and wanting to be coddled means you are likely to be ignored or worse, severely flamed. Be aware, I have only lit the pilot light on the flame thrower, you will control the trigger. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] UK ISDN PRI Problems
Advance apologies for the length of this mail; I have an ISDN PRI supplied by NTL (ex Diamond Cable, Nottingham) which is currently working happily with an SDX Index phone system. I have to replace this phone system shortly and I've been trying to get a * system working for some weeks now. I have configured the dial plan (which works) and all my SIP extensions (which all work) along with voice mail etc. etc. - all this works perfectly as an internal PBX. My problem comes when I try to connect it to my ISDN line. I have a Digium E100p card which is configured in zaptel.conf thus; span=1,0,5,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 loadzone=uk defaultzone=uk The zapata.conf is like this; [channels] usecallerid=yes callerid=asreceived hidecallerid=no threewaycalling=yes transfer=yes cancelforward=yes callreturn=yes immediate=no callprogress=no language=en echotraining=yes echocancel=yes echocancelwhenbridged=yes rxgain=-5% txgain=+5% pridialplan=unknown overlapdial=yes signalling=pri_cpe switchtype=euroisdn context=inboundpstn group=1 channel = 1-15 callgroup=1 pickupgroup=1 Whenever I try to connect this up to the ISDN line I get a series of Red Alerts and any attempt at outgoing calls results in a no channels available message (essentially all the lines are shown in use and cannot be cleared). I have had my teleco reset the line which just results in further red alerts. NTL, bless them, came out with a test rig and plugged this in the back of my * box and we made a series of test calls which all worked fine, although the NTL chap said the attenuation was out as there was a lot of buzz on the line. He suggested we set the line build out to 6 (-15db) but we were still getting buzz on the test calls - other than that he was happy that the config was correct and as the test rig showed the * box talking properly and making outgoing calls OK we all reckoned that the next time we hooked up to the ISDN line it would be OK. I couldn't do this with him there as the system is in use during the day, so had to wait until the evening after he'd gone. However, I still get the red alerts. If I leave the * box connected to the ISDN line when the teleco attempt to reset the line it immediately trips with another series of red alerts, however it resets just fine when plugged back into the SDX Index system. The teleco says they can see it tripping, but don't know why (?). I have made sure that the card is not sharing interrupts and I've scoured the mail archives and google for any further information I can get my hands on. I just can't get past the red alerts. Does anyone on the list have any idea why this is happening (big question I know), I'm using the stable CVS tree from 15/4/04. Is there anyone else out there using * with an NTL ISDN PRI line? Many thanks for any help offered :) Chris Barnett ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
