Re: [asterisk-users] Record of all calls
use Asterisk CDR (Call Detail Record) ref: http://areski.net/asterisk-stat-v2/about.php Greetings! Prompt how to make that the asterisk wrote down all calls automatically ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Bad FCS hangup
Hello, All! It is a Piece of my log At the moment of a call: Jan 9 09:50:21 NOTICE[10902]: chan_zap.c:8194 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 3 -- Accepting call from '' to '0033444' on channel 1/1, span 1 -- Executing Dial(Zap/1-1, SIP/3444|60|rgT) in new stack -- Called 3444 -- SIP/3444-0823e6b0 is ringing -- Channel 1/1, span 1 got hangup request -- Hungup 'Zap/1-1' -- Accepting call from '80577591759' to '786' on channel 3/1, span 3 -- Executing Dial(Zap/63-1, SIP/3444|60|rgT) in new stack -- Called 3444 -- Channel 3/1, span 3 got hangup -- Hungup 'Zap/63-1' The description of actions: I call from number 80577591759 on number 786 Then the asterisk should translate a call on internal SIP 3444 Translates and does hangup :-( And messages of type of it emerge constantly: Jan 9 09:50:21 NOTICE[10902]: chan_zap.c:8194 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 3 Please Help! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT:spa942 provisioning
On Mon, 8 Jan 2007 20:03:50 -0500 Andrew Joakimsen [EMAIL PROTECTED] wrote: Good luck dealing with Linksys on that http://voxilla.com/tools/device-configuration-wizard/certificate-authority-service-for-linksys-analog-voip-adaptors-808.html Hi Andrew! Thanks for the response, unfortunately this is about encrypting voice, not about provisioning. What i want to do is to configure a spa942 to fetch it's configuration files via http, since i don't want anyone to get the configuration files, the client that connects to the webserver has to verify it's identity with a client side ssl-certificate(which is already preinstalled on the phone), this cert will be verified by a CA that is installed on the webserver. However, i have no idea where i can find the CA resp. how the client cert looks like... regards christian On 1/8/07, Benko [EMAIL PROTECTED] wrote: Hello! Sorry for the OT-thread, but i don't know where else too ask... Has anyone done http-provisioning of a Linksys SPA942 with client side ssl-authentication? Where do i get the CA from? I'm aware of the Sipura mass deployment howto on voip-info.org, but it doesn't cover the authentification part. Thanks Christian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with polycom video conference
I have success register polycom in to asterisk and it can called by other extension. But why it can't calling other extension ? and i have warning from asterisk chan.sip.c:3602 process_sdp: Unknown SDP media type in offer: application 49200 RTP/AVP 100 anyone undertand this warning ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] WIFI SIP- The Best phone
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of mitcheloc Sent: Sunday, December 31, 2006 8:52 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] WIFI SIP- The Best phone Those wifi phones are neat but I'd rather not carry around two devices, does anyone know of any good dual-mode GSM/SIP phones? I'm using a T-Mobile MDA right now and it is way too slow. Apparently the Nokia e61 has a built in SIP client, but there might be a new model around the corner (worth the wait?) Suggestions? I've had a play with a Nokia E70 - the 'bar' version of the E61 and gave up! Menu navigation is dire - I went through hoops trying to get SIP working - I know from others it can be done, but I bailed out when I realised that to put these phones in the hands of inexperienced users would be a recipe for a lot of frustration and support calls. I'd expect the 'PDA style' E61 might be easier to use. I have an HTC Hermes phone (Vodafone V1605 in the UK) running Windows Mobile 5. I have fired up the beta version of SJPhone on it and it was just about useable, but not 'production ready'. I hope that there will be some decent WM5 software in the near future but am wondering what sort of battery life can be expected. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk and 3PCC
Hi all, I have an Asterisk server running, and some hardware phones, and I want to do 3PCC : third party call control. The third party is a software running on the asterisk box, which can for example ask a hard SIP phone to put a call on hold. To do that, this software has to send a SIP message to this phone. How can I do that ? Greg ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and 3PCC
Gregory, I know there is something called SIP CTI TR87. It's used by Nortel to integrate with Microsoft's Live Communication Server. Don't know if something similar exists for Asterisk. This links could be helpfull: http://www.ecma-international.org/publications/techreports/E-TR-087.htm Regards, Koen On 1/9/07, Gregory Duchatelet [EMAIL PROTECTED] wrote: Hi all, I have an Asterisk server running, and some hardware phones, and I want to do 3PCC : third party call control. The third party is a software running on the asterisk box, which can for example ask a hard SIP phone to put a call on hold. To do that, this software has to send a SIP message to this phone. How can I do that ? Greg ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk and 3PCC
Seems that this has to be implemented by the phones, or by a B2BUA I think that a B2BUA could be used for 3PCC, but dont know if an open-source B2BUA exists and works with Asterisk Greg _ De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Koen Van Impe Envoyé : mardi 9 janvier 2007 10:33 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [asterisk-users] Asterisk and 3PCC Gregory, I know there is something called SIP CTI TR87. It's used by Nortel to integrate with Microsoft's Live Communication Server. Don't know if something similar exists for Asterisk. This links could be helpfull: http://www.ecma-international.org/publications/techreports/E-TR-087.htm Regards, Koen On 1/9/07, Gregory Duchatelet [EMAIL PROTECTED] wrote: Hi all, I have an Asterisk server running, and some hardware phones, and I want to do 3PCC : third party call control. The third party is a software running on the asterisk box, which can for example ask a hard SIP phone to put a call on hold. To do that, this software has to send a SIP message to this phone. How can I do that ? Greg ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/ -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk and 3PCC
uhm... On Tue, 2007-01-09 at 12:28 +0100, Gregory Duchatelet wrote: Seems that this has to be implemented by the phones, or by a B2BUA… I think that a B2BUA could be used for 3PCC, but don’t know if an open-source B2BUA exists and works with Asterisk … asterisk IS a B2BUA just my 2cents. Matteo. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: RE: [asterisk-users] WIFI SIP- The Best phone
On 09/01/07, Nigel Kendrick [EMAIL PROTECTED] wrote: I've had a play with a Nokia E70 - the 'bar' version of the E61 and gave up! Menu navigation is dire - I went through hoops trying to get SIP working - I know from others it can be done, but I bailed out when I realised that to put these phones in the hands of inexperienced users would be a recipe for a lot of frustration and support calls. Ironically I was going to recommend the E70. It is true that the menus are complex but once configured it does do what it says on the tin - provide a very effective merging of SIP over WIFI and GSM all in one unit. Regards, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk + 7910 + Skinny Reset
I have a bunch of 7910's that I managed to get registered with Asterisk 1.2.14: managed5*CLI skinny show devices Name DeviceId IP TypeId R Model NL --- -- - -- -- test7 SEP0004C1878F8E 192.168.0.226 6 Y 7910 1 The problem is that the phone resets when I attempt to make a call from it or place a call to it. If I pick up I have no dial tone and after 3-4 seconds the phone resets. When that happens, on Asterisk I see: Attempting to Clear display on Skinny [EMAIL PROTECTED] skinny_new: tmp-nativeformats=4 fmt=4 -- Starting simple switch on '[EMAIL PROTECTED]' then the phone resets. when I try to call it, it doesn't ring and Asterisk displays: Found device: test7 -- skinny_request([EMAIL PROTECTED]) -- Skinny cw: 0, dnd: 0, so: 0, sno: 0 skinny_new: tmp-nativeformats=4 fmt=4 -- skinny_call(Skinny/[EMAIL PROTECTED]) Trying to send: '' Displaying message '' Displaying Prompt Status 'Ring-In' -- Called [EMAIL PROTECTED] -- Skinny/[EMAIL PROTECTED] is ringing skinny_hangup(Skinny/[EMAIL PROTECTED]) on [EMAIL PROTECTED] then the phone resets. Registration messages are: -- Starting Skinny session from 192.168.0.226 Device SEP0004C1878F8E is attempting to register -- Device 'test7' successfuly registered Requesting capabilities Received CapabilitiesRes RECEIVED UNKNOWN MESSAGE TYPE: 2b Buttontemplate requested Sending 7910 template to [EMAIL PROTECTED] (7910) Recieved SoftKey Template Request Received SoftKeySetReq Received LineStateReq Received Time/Date Request what could be causing this ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk and 3PCC
True :) Here is an example of what i want to do : - a phone call extension 100 - asterisk enter the context, and execute Dial() to call another phone - ringing... - now I want that asterisk ask the called phone to answer : how to do that ?? Greg uhm... On Tue, 2007-01-09 at 12:28 +0100, Gregory Duchatelet wrote: Seems that this has to be implemented by the phones, or by a B2BUA. I think that a B2BUA could be used for 3PCC, but don't know if an open-source B2BUA exists and works with Asterisk . asterisk IS a B2BUA just my 2cents. Matteo. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Spam? Re: [asterisk-users] Asterisk and IM
is anyone using this to initiate a call back. . . I am epically interested in AIM, as it can can serve as a free GSM gateway.any ideas? On 1/8/07, Hall, Eric M. [EMAIL PROTECTED] wrote: Has anyone got Asterisk IM to work Using this link http://www.sipalive.com/dev/asterisk/ And a clean install of asteris 1.4.0-Beta3 I get the following error Any ideas? I have no idea what the .rej file is telling me so it maybe easy to see it here but I'm a little out of my strike zone her! patch -p0 sip_message_support.patch (Stripping trailing CRs from patch.) patching file chan_sip.c Hunk #1 FAILED at 90. Hunk #2 succeeded at 8165 (offset -112 lines). Hunk #4 succeeded at 9089 (offset -115 lines). Hunk #6 succeeded at 9222 (offset -115 lines). 1 out of 6 hunks FAILED -- saving rejects to file chan_sip.c.rej [Channels]# cat chan_sip.c.rej *** *** 90,95 #include asterisk.h ASTERISK_FILE_VERSION(__FILE__, $Revision: 48487 $) #include stdio.h --- 90,99 #include asterisk.h + /* Include this for message queuing support. Comment out if not wanted. + * You will need to link with sqlite */ + /* #include queue_chan_sip.h + ASTERISK_FILE_VERSION(__FILE__, $Revision: 48487 $) #include stdio.h -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kenneth Padgett Sent: Friday, January 05, 2007 9:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Spam? Re: [asterisk-users] Asterisk and IM I have been asked to get IM via the X-Ten softphone to work with Asterisk. Anyone have any ideas? I have looked on google and other places with no luck. Our system is as followed Linux CentOS 4.4 Asterisk 1.4.0-beta3 X-Lite v3.0 for Windows If by IM, you mean the built-in Jabber stuff in v1.4... I am having trouble with that and CentOS 4.4 myself, can't get the required libs or some such non-sense. -Kenneth ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Fwd: [asterisk-users] Some queries on g729 license.
C F wrote: I knew I was doing the right thing, here is the proof, enjoy when you read it, and have a good laugh. -- Forwarded message -- From: Al Bochter [EMAIL PROTECTED] Date: Jan 8, 2007 8:22 PM Subject: Re: [asterisk-users] Some queries on g729 license. To: [EMAIL PROTECTED] (C)UNT (F)UCK! THIS IS OFF THE LIST FUCK YOU ASSHOLE! GET A JOB AND STOP LIVING OFF MY TAXES YOU DON'T KNOW WHAT YOU ARE DOING TRY AND STAY ON THE POINT. YOU ARE NOW BLOCKED I AM NOT GOING TO DEAL WITH JACKASSES LIKE YOU GOOD BYE Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email C F wrote: When I first noticed that this thread has over 20 messages i was sure it is interesting. When I read it I realized that I havn't noticed that Al Bochter has posted to it. Plain old stuff, just someone making sure to put a new twist on it. On 1/8/07, Juan Jose Comellas [EMAIL PROTECTED] wrote: The Intel IPP-based G.729 codec does work with AMD processors out of the box, both with the 32 bit and 64 bit versions. On Mon January 8 2007 19:31, Zoa wrote: I did some tests a long time ago and the speed was roughly the same. ( I think digium's was slightly faster). I think the IPP version also doesn't work on AMD out of the box. It's just 10$ a channel, that's not even worth the hassle of trying something else. Joachim Al Bochter wrote: Matthew I agree. I only know what I have told by others so I do need this input I have been told that Digum G729 is a big pain the the butt to get working with Asterisk and it is very hard on the CPU Keep in mind I have never used any Ver. of G 729 So tell me what you think. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Matthew Rubenstein wrote: All of which hassle and expense can be avoided by buying a license for Digium's codec, which is tested to work well with Asterisk (and might come with some support). And is pretty cheap per simul call. I wonder whether that per call means per codec instance, which could be multiple licenses on a single conference call, where multiple (even if not all) parties are getting de/encoded simultaneously. And whether there are other tools for editing (/mixing/transforming) g729 data, in realtime (streams) or not (files), and whether they require a license. Ideally sox or equivalent would work on g729, maybe with a codec plugin. On Mon, 2007-01-08 at 13:23 -0500, Paul wrote: First point to tackle in any case involving patent, copyright or trademark infringement is whether or not the infringing party would have been qualified to buy any usage rights at all. In a case where you license the Intel source(read the terms, it's not really that free), you would be applying for a license under some plan that includes certain minimum payments. Even if you wrote new source from scratch you would be in the same boat. Last time I looked at the plans, I didn't see anything with low minimums. So even if you wrote code from scratch and never used it on more than 6 channels, you might have done something that normally requires a large upfront payment. Use $10k as an example. In such a case owner of the patent might have an attorney initiate contact. If you are willing to communicate they might allow you to pay the minimum and be licensed. If you can't do that, they might offer a settlement where you stop using the codec and pay them some lesser amount. If the patent holder can easily prove the violation you might as well try to deal with them and get things settled fast. If you sell or give away the codec it is easier for them to dig up proof. If you have unhappy employees that might be the way they hear about the violation in the first place. Important consideration: Bankruptcy law generally excludes debts created by things like malicious or criminal acts. Matthew Rubenstein wrote: As far as I know, the g729 patent requires buying a license to operate any implementation of it, whether Digium's, Intel's, or any other. Digium is set up to collect royalties (perhaps at a favorable rate) as part of their license from the patent holder. I don't know about Intel or any other. Or what the mechanics are for enforcing the patent on someone who operates a codec without a license. On Mon, 2007-01-08 at 10:51 -0500, Al Bochter wrote: What about the free open source G729 Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Matthew Rubenstein wrote: I connect to a PSTN carrier over SIP which requires me to connect with a g729 codec. I'm using them for just robocalling: Asterisk server originates calls which play a
[asterisk-users] Snom side car annoyance
Has anyone got this annoying sidecar to illuminate when users are on the phone? In my function key settings I have: Context: Active Type: Extension Number: sip:[EMAIL PROTECTED];user=phone (4000 is the extension I want to see/dial on the key). I can press the key and it will dial the extension, it just won't illuminate when the user is on the phone or on DND Since I have multiple lines on this one particular 360, I even tried: Context: [EMAIL PROTECTED];user=phone (4000 is the line I assigned to my phone) Type: Extension Number: sip:[EMAIL PROTECTED];user=phone Same. 4001 will ring if I hit the function key, but nothing is illuminated. I've tried Context: Line, still no dice. In extensions.conf I have: exten = 4000,hint,SIP/4000,name Using Asterisk 1.2.13 on FC5, Snom: Phone Type: snom360-SIP Kernel Version: snom360 linux 3.25 Application-Version: snom360-SIP 6.5.2 Rootfs-Version: snom360 jffs2 v3.36 -- J. Oquendo http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743 sil . infiltrated @ net http://www.infiltrated.net The happiness of society is the end of government. John Adams smime.p7s Description: S/MIME Cryptographic Signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Fwd: [asterisk-users] Some queries on g729 license.
Hi On Tue, Jan 09, 2007 at 06:20:04AM -0800, Derek Whitten wrote: [ unrelated message completely quoted snipped] [ signatures snipped ] [ offline message posted on-list snipped ] [ foul language snipped ] At least you didn't top-post. and have a nice day Thank you. Now could we please get back to the list's topics? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MFC/R2 problems + Orion GSM Gateway
Hi, I have made some headway with this. Let me explain a abit of the setup. I have an Orion GSM Gateway, that was connected to a Cisco AS5300 via E1. When I looked at the AS5300 config, it was talking R2 to the Orion. So I have tried to connect the Orion direclty to Asterisk (leaving out the Cisco), using Unicall. This is a problem I have with an incoming call, from Orion to Asterisk. Jan 9 16:35:29 WARNING[7262]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/2 - 0001 [1/ 1/Idle /Idle ] Jan 9 16:35:29 WARNING[7262]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/2 Detected Jan 9 16:35:29 WARNING[7262]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/2 Making a new call with CRN 32769 Jan 9 16:35:29 WARNING[7262]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/2 1101 - [2/ 2/Idle /Idle ] Jan 9 16:35:29 WARNING[7262]: chan_unicall.c:2644 handle_uc_event: Unicall/2 event Detected Jan 9 16:35:31 WARNING[7262]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/2 - 1001 [2/ 2/Seize ack /Seize ack] Jan 9 16:35:31 WARNING[7262]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/2 Far end disconnected(cause=Normal, unspecified cause [31]) - state 0x2 Jan 9 16:35:31 WARNING[7262]: chan_unicall.c:2644 handle_uc_event: Unicall/2 event Far end disconnected Jan 9 16:35:31 WARNING[7262]: chan_unicall.c:2930 handle_uc_event: CRN 32769 - far disconnected cause=Normal, unspecified cause [31] Jan 9 16:35:31 WARNING[7262]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/2 Call control(6) Jan 9 16:35:31 WARNING[7262]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/2 Drop call(cause=Normal Clearing [16]) Jan 9 16:35:31 WARNING[7262]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/2 Call disconnected(cause=Normal, unspecified cause [31]) - state 0x800 Jan 9 16:35:31 WARNING[7262]: chan_unicall.c:2644 handle_uc_event: Unicall/2 event Drop call Jan 9 16:35:31 DEBUG[7262]: chan_unicall.c:2978 handle_uc_event: CRN 32769 - Doing a release call Jan 9 16:35:31 WARNING[7262]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/2 Call control(7) Jan 9 16:35:31 WARNING[7262]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/2 Release call Jan 9 16:35:31 WARNING[7262]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/2 1001 - [1/ 1000/Clear fwd /Seize ack] The below output(in the mail) is of an outgoing call from Asterisk. Can anyone please help me to see what is wrong? yusuf wrote: Hi, if that means I should post my config, here goes: zaptel: span=1,1,3,cas,hdb3,crc4 cas=1-15:1101 cas=17-31:1101 unicall.conf: protocolvariant=id,10,10 protocolend=cpe group=1 channel = 1-15 channel = 17-31 wanpipe1.conf FE_MEDIA= E1 FE_LCODE= HDB3 FE_FRAME= CRC4 FE_LINE = 1 TE_CLOCK= NORMAL TE_REF_CLOCK= 0 TE_HIGHIMPEDANCE= NO LBO = 120OH TE_SIG_MODE = CAS FE_TXTRISTATE = NO MTU = 1500 UDPPORT = 9000 TTL = 255 IGNORE_FRONT_END = NO TDMV_SPAN = 1 TDMV_DCHAN = 16 [w1g1] ACTIVE_CH = ALL TDMV_ECHO_OFF = NO TDMV_HWEC = NO Josué Conti wrote: Hi Yusuf, how are you? It orders in the list its configurations, so that let us can help. Best Regards Josue 2007/1/8, yusuf [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]: Hi all, I have Asterisk 1.2.10, zaptel 1.2.7, spandsp-0.0.3pre22 compiled, and a Sangoma A101, and when I make a call I get this: Jan 8 13:04:06 DEBUG[12252]: chan_unicall.c:2000 unicall_exception: Exception on 19, channel 1 Jan 8 13:04:06 WARNING[12252]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 1101 [1/ 40/Seize /Idle ] Jan 8 13:04:06 WARNING[12252]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 0 on - [2/ 40/Group I /Idle ] Jan 8 13:04:11 WARNING[12252]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 R2 prot. err. [2/ 40/Group I /DNIS ] cause 32769 - T1 timed out Jan 8 13:04:11 WARNING[12252]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 0 off - [1/ 1/Idle /Idle ] Jan 8 13:04:11 WARNING[12252]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 1001 - [1/ 1/Idle /Idle ] Jan 8 13:04:11 WARNING[12252]: chan_unicall.c:2644 handle_uc_event: Unicall/1 event Protocol failure -- Unicall/1 protocol error. Cause 32769 Jan 8 13:04:11 WARNING[12252]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Channel echo cancel Jan 8 13:04:11 DEBUG[12252]: chan_unicall.c:955 unicall_disable_ec: disabled echo cancellation on channel 1 Jan 8 13:04:11 WARNING[12250]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 1001 [1/ 1/Idle /Idle ] Jan 8 13:04:11 WARNING[12250]: chan_unicall.c:627
Re: [asterisk-users] MFC/R2 problems + Orion GSM Gateway
Yusuf, there are several things can be wrong. Make sure you have configured the correct protocol variant, DNIS and CID. Also check you really need CRC4 checking. I wrote a document to help debugging this stuff, you can find it here: http://moy.ivsol.net/unicall/mfcr2-asterisk-unicall-0.2-english.pdf Good Look Kind Regards On 1/9/07, yusuf [EMAIL PROTECTED] wrote: Hi, I have made some headway with this. Let me explain a abit of the setup. I have an Orion GSM Gateway, that was connected to a Cisco AS5300 via E1. When I looked at the AS5300 config, it was talking R2 to the Orion. So I have tried to connect the Orion direclty to Asterisk (leaving out the Cisco), using Unicall. This is a problem I have with an incoming call, from Orion to Asterisk. Jan 9 16:35:29 WARNING[7262]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/2 - 0001 [1/ 1/Idle /Idle ] Jan 9 16:35:29 WARNING[7262]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/2 Detected Jan 9 16:35:29 WARNING[7262]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/2 Making a new call with CRN 32769 Jan 9 16:35:29 WARNING[7262]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/2 1101 - [2/ 2/Idle /Idle ] Jan 9 16:35:29 WARNING[7262]: chan_unicall.c:2644 handle_uc_event: Unicall/2 event Detected Jan 9 16:35:31 WARNING[7262]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/2 - 1001 [2/ 2/Seize ack /Seize ack] Jan 9 16:35:31 WARNING[7262]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/2 Far end disconnected(cause=Normal, unspecified cause [31]) - state 0x2 Jan 9 16:35:31 WARNING[7262]: chan_unicall.c:2644 handle_uc_event: Unicall/2 event Far end disconnected Jan 9 16:35:31 WARNING[7262]: chan_unicall.c:2930 handle_uc_event: CRN 32769 - far disconnected cause=Normal, unspecified cause [31] Jan 9 16:35:31 WARNING[7262]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/2 Call control(6) Jan 9 16:35:31 WARNING[7262]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/2 Drop call(cause=Normal Clearing [16]) Jan 9 16:35:31 WARNING[7262]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/2 Call disconnected(cause=Normal, unspecified cause [31]) - state 0x800 Jan 9 16:35:31 WARNING[7262]: chan_unicall.c:2644 handle_uc_event: Unicall/2 event Drop call Jan 9 16:35:31 DEBUG[7262]: chan_unicall.c:2978 handle_uc_event: CRN 32769 - Doing a release call Jan 9 16:35:31 WARNING[7262]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/2 Call control(7) Jan 9 16:35:31 WARNING[7262]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/2 Release call Jan 9 16:35:31 WARNING[7262]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/2 1001 - [1/ 1000/Clear fwd /Seize ack] The below output(in the mail) is of an outgoing call from Asterisk. Can anyone please help me to see what is wrong? yusuf wrote: Hi, if that means I should post my config, here goes: zaptel: span=1,1,3,cas,hdb3,crc4 cas=1-15:1101 cas=17-31:1101 unicall.conf: protocolvariant=id,10,10 protocolend=cpe group=1 channel = 1-15 channel = 17-31 wanpipe1.conf FE_MEDIA= E1 FE_LCODE= HDB3 FE_FRAME= CRC4 FE_LINE = 1 TE_CLOCK= NORMAL TE_REF_CLOCK= 0 TE_HIGHIMPEDANCE= NO LBO = 120OH TE_SIG_MODE = CAS FE_TXTRISTATE = NO MTU = 1500 UDPPORT = 9000 TTL = 255 IGNORE_FRONT_END = NO TDMV_SPAN = 1 TDMV_DCHAN = 16 [w1g1] ACTIVE_CH = ALL TDMV_ECHO_OFF = NO TDMV_HWEC = NO Josué Conti wrote: Hi Yusuf, how are you? It orders in the list its configurations, so that let us can help. Best Regards Josue 2007/1/8, yusuf [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]: Hi all, I have Asterisk 1.2.10, zaptel 1.2.7, spandsp-0.0.3pre22 compiled, and a Sangoma A101, and when I make a call I get this: Jan 8 13:04:06 DEBUG[12252]: chan_unicall.c:2000 unicall_exception: Exception on 19, channel 1 Jan 8 13:04:06 WARNING[12252]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 1101 [1/ 40/Seize /Idle ] Jan 8 13:04:06 WARNING[12252]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 0 on - [2/ 40/Group I /Idle ] Jan 8 13:04:11 WARNING[12252]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 R2 prot. err. [2/ 40/Group I /DNIS ] cause 32769 - T1 timed out Jan 8 13:04:11 WARNING[12252]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 0 off - [1/ 1/Idle /Idle ] Jan 8 13:04:11 WARNING[12252]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 1001 - [1/ 1/Idle /Idle ] Jan 8 13:04:11 WARNING[12252]: chan_unicall.c:2644 handle_uc_event: Unicall/1 event Protocol failure -- Unicall/1 protocol error. Cause 32769
[asterisk-users] Strange queue behaviour
Hello, I have just installed Asterisk 1.4 and I am playing with it. I've created some sip accounts and some queues. When I start asterisk I see many queues like this: all-phones-r has 0 calls (max unlimited) in 'rrmemory' strategy (0s holdtime), W:0, C:0, A:0, SL:0.0% within 0s Members: SIP/03 (Invalid) has taken no calls yet SIP/02 (Invalid) has taken no calls yet No Callers that is, the members are in invalid state. What does that mean ? If I try to call the queue I get this messages: [Jan 8 17:48:20] WARNING[1681]: app_queue.c:3523 queue_exec: Unable to join queue 'all-phones' and the most weird thing happens when I make one of the members of the queue call the queue. I will get the same unable to join message until something happens and the call gets thru. (I've tried many, many calls with an external phone with no luck). When that happens, queues are now like this: all-phones-r has 0 calls (max unlimited) in 'rrmemory' strategy (0s holdtime), W:0, C:0, A:0, SL:0.0% within 0s Members: SIP/03 (Not in use) has taken no calls yet SIP/02 (Not in use) has taken no calls yet No Callers or all-phones has 0 calls (max unlimited) in 'ringall' strategy (0s holdtime), W:0, C:1, A:0, SL:0.0% within 0s Members: SIP/03 (Not in use) has taken no calls yet SIP/02 (Not in use) has taken 1 calls (last was 38 secs ago) and now it just works, even for external calls or anything. Any ideas what's going on here ? Thank you. -- José Pablo Fernández [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cell phone dock/switch as Asterisk FXO source
There has been talk about it before and I think people have done it. Paging Sam Tam - Original Message - From: Joao Pereira [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com; [EMAIL PROTECTED] Sent: Tuesday, January 02, 2007 4:56 PM Subject: Re: [Asterisk-Users] Cell phone dock/switch as Asterisk FXO source Do you know If its possible to do the same with Dock and Talk and an ATA GrandStream HandyTone 386? Thanks Joao Pereira Jonathan Attwood wrote: I use a Dock-n-Talk in conjuction with a Sipura SPA3000 Asterisk. Because I'm using Asterisk, I cannot use voice dialling, however inbound outbound calls work extremely well. I have Asterisk outbound routes set up to make a calls to cell phones go through the Dock-n-Talk. On 1/1/06, *Brian McEntire* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Is anyone familiar with cell phone switches that allow routing cell phone calls through in-home wiring? One example of these devices is the Phone Labs Dock-N-Talk. It says it keeps your cell charged when you are home and connects your cell (for incoming and outgoing calls) to your home wiring or cordless phones. But it also has features such as allowing speed dialing and voice dialing from extensions if your cell phone has those features. So I'm not sure if the device offers a fully compatible FXO signalling. I'm currently running Asterisk with 1 POTS and 1 VOIP (via Sipura 3000) lines coming into Zaptel FXS modules, and then I have two FXO modules for two extensions. I'm thinking of doing away with the land line. Should something like the Dock-N-Talk allow substituting a cell phone line for the POTS line? ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/ -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fax through Sangoma A102
Hello, in our company we are trying to do this: Fax -- Traditional PBX -- Asterisk -- PSTN In practice, we have put an Asterisk equipped with a Sangoma A102 (2 PRI ports) between our PBX (Siemens HiCom) and the PSTN in order to have a VoIP network along the traditional telephony network. The problem is with the fax. We just want to send and receive faxes from/to our fax machine connected to the Siemens (without needing any interaction with our VoIP network, the faxes are sent to/received from PSTN). Unfortunately we are experiencing a lot of problems: the faxes not always work and when they work, it's likely to have incomplete pages. I know that faxing with VoIP is very troublesome, but maybe someone else is using a similar configuration and he found a good configuration or maybe has some hints to improve the results. We are using Asterisk 1.2.13. Thanks, Jeremi ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cell phone dock/switch as Asterisk FXO source
Does somebody know a similar device that does the same for GSM networks ? Zoa Dovid B wrote: There has been talk about it before and I think people have done it. Paging Sam Tam - Original Message - From: Joao Pereira [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com; [EMAIL PROTECTED] Sent: Tuesday, January 02, 2007 4:56 PM Subject: Re: [Asterisk-Users] Cell phone dock/switch as Asterisk FXO source Do you know If its possible to do the same with Dock and Talk and an ATA GrandStream HandyTone 386? Thanks Joao Pereira Jonathan Attwood wrote: I use a Dock-n-Talk in conjuction with a Sipura SPA3000 Asterisk. Because I'm using Asterisk, I cannot use voice dialling, however inbound outbound calls work extremely well. I have Asterisk outbound routes set up to make a calls to cell phones go through the Dock-n-Talk. On 1/1/06, *Brian McEntire* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Is anyone familiar with cell phone switches that allow routing cell phone calls through in-home wiring? One example of these devices is the Phone Labs Dock-N-Talk. It says it keeps your cell charged when you are home and connects your cell (for incoming and outgoing calls) to your home wiring or cordless phones. But it also has features such as allowing speed dialing and voice dialing from extensions if your cell phone has those features. So I'm not sure if the device offers a fully compatible FXO signalling. I'm currently running Asterisk with 1 POTS and 1 VOIP (via Sipura 3000) lines coming into Zaptel FXS modules, and then I have two FXO modules for two extensions. I'm thinking of doing away with the land line. Should something like the Dock-N-Talk allow substituting a cell phone line for the POTS line? ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/ -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with polycom video conference
Do you have the videosupport=yes in your sip.conf for that device? You might try adding: videosupport=yes allow=h263 ; H.263 is our video codec allow=h263p ; H.263p is the enhanced video codec On 1/9/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: I have success register polycom in to asterisk and it can called by other extension. But why it can't calling other extension ? and i have warning from asterisk chan.sip.c:3602 process_sdp: Unknown SDP media type in offer: application 49200 RTP/AVP 100 anyone undertand this warning ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk build for Suse 10.1
Has anyone heard of a build or instructions for installing Asterisk on a Suse 10.1 system? Bob Rawlinson ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk and Avaya IP Office
I would like to connect an Asterik server to an Avaya IP Office IP406 and use the * as an VoIP Gateway. The IP Office has two Analog extensions available. I thought connecting this analog extensions to 2 FXO ports in the * to interconnect the PBXs. Is this possible? Does any one have experience with such a configuration? Thanks in advance for all recommandations and suggestions.. Housi Mueller __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Fwd: [asterisk-users] Some queries on g729 license.
This is by far the most volotile list I have ever been on. I'm not sure that's exactly the reputation Digium/Asterisk is shooting for, but even so it does provide some much needed comedy relief. After seeing the G.729 pricing direct from SIPRO, I now take the shut-up and be thankful position. I think Digium has done us a great service by working out favorable pricing with SIPRO. Regards, David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Fwd: [asterisk-users] Some queries on g729 license.
David Thomas wrote: This is by far the most volotile list I have ever been on. I'm not sure that's exactly the reputation Digium/Asterisk is shooting for, but even so it does provide some much needed comedy relief. After seeing the G.729 pricing direct from SIPRO, I now take the shut-up and be thankful position. I think Digium has done us a great service by working out favorable pricing with SIPRO. Also, I remember reading some SIPRO announcements about changes in licensing. My intepretation is that digium is grandfathered into a license with some flexibility that SIPRO might be reluctant to allow now. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax through Sangoma A102
jeremij jerome wrote: The problem is with the fax. We just want to send and receive faxes from/to our fax machine connected to the Siemens (without needing any interaction with our VoIP network, the faxes are sent to/received from PSTN). Unfortunately we are experiencing a lot of problems: the faxes not always work and when they work, it's likely to have incomplete pages. What are you using to fax? Fax machines connected to ATAs? txfax/rxfax? IAXmodem and HylaFAX? If you are using IAXmodem and HylaFAX a fax session log (HylaFAX) would be quite revealing. Lee. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Is there any Asterisk controllable thermostat?
My garage door is... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Crompton Sent: Monday, December 04, 2006 11:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Is there any Asterisk controllable thermostat? I remembered I had an x10 bottlerocket in my X10 junkbox so I connected it to a spare serial port on my linux server (asterisk resides there) and implemented with some mods the code mentioned earlier http://lorance.freeshell.org/asterisk/#asterisk-can-control-the-world and it works great. Now I have one more way to control X10 devices. I can even call my VM on the way home and turn on my lights or whatever before I get home. Doug ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Snom side car annoyance
you are asking about Shared line apperance or hints. Look at this http://www.voip-info.org/wiki/view/snom+360 On 1/9/07, J. Oquendo [EMAIL PROTECTED] wrote: Has anyone got this annoying sidecar to illuminate when users are on the phone? In my function key settings I have: Context: Active Type: Extension Number: sip:[EMAIL PROTECTED];user=phone (4000 is the extension I want to see/dial on the key). I can press the key and it will dial the extension, it just won't illuminate when the user is on the phone or on DND Since I have multiple lines on this one particular 360, I even tried: Context: [EMAIL PROTECTED];user=phone (4000 is the line I assigned to my phone) Type: Extension Number: sip:[EMAIL PROTECTED];user=phone Same. 4001 will ring if I hit the function key, but nothing is illuminated. I've tried Context: Line, still no dice. In extensions.conf I have: exten = 4000,hint,SIP/4000,name Using Asterisk 1.2.13 on FC5, Snom: Phone Type: snom360-SIP Kernel Version: snom360 linux 3.25 Application-Version: snom360-SIP 6.5.2 Rootfs-Version: snom360 jffs2 v3.36 -- J. Oquendo http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743 sil . infiltrated @ net http://www.infiltrated.net The happiness of society is the end of government. John Adams ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! Hind sight is most always 20/20 or better. --- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ooh323c calls
Hi, I have two asterisk servers where softphone A is connected to asterisk A. On those two asterisk servers, ooh323c is installed. I tried to call a test context on asterisk B from softphone A. But I always fall into context default of asterisk B. ( I don't know how to tell asterisk A extensions.conf to call asterisk B test context) Here are conf files on asterisk A : ooh323.conf [softA] ; softphone A uses this channel type=user context=test ip=10.0.0.1 port=1720 disallow=all allow=gsm allow=ulaw [mypeer1] type=peer ip=10.0.0.2 port=1720 extensions.conf [test] exten = 15,1,Answer() exten = 15,n,Playback(vm-hello) exten = 15,n,Dial(OOH323/150/mypeer1);or exten = 15,n,Dial(OOH323/[EMAIL PROTECTED]) exten = 15,n,Hangup() May I use a gatekeeper? I learnt that ooh323c can act as gatekeeper, but I didn't success to configure it (I have gatekeeper is not responding error!). Can one of my server acts as gatekeeper and gateway? Do anyone success to configure gatekeeper with ooh323c ? Thanks you for you help! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk build for Suse 10.1
When you say build do you mean a plug and play binary? I use SUSE 7.3 here and it is easy to get the source files and compile it. It should just work. The instructions would be in the README or INSTALL file in the source. 1. Get the source at digium (the 1.2.x version might be better to start with. 2. tar -xvzf version_name 3. cd to the directory tree made by step 2 4. Read the README and/or INSTALL text files for info on how to proceed 5. make 6. make install These steps would vary depending on wether you need zap or other drivers which would be compiled first. you would also want to download and install the sound files. There might be an easier install for a specific O/S version but I prefer to do things manually here. Of course I still do most everything at the command prompt also. I do not use a windowed system for my server. I don't see any reason why 10.1 would be any different then my 7.3 in the installation procedure. Doug On Tue, 9 Jan 2007, Robert A. Rawlinson wrote: Has anyone heard of a build or instructions for installing Asterisk on a Suse 10.1 system? Bob Rawlinson ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Caller Id problem
Dear List,
My problem is that the incoming Caller Id is not displayed on the local analog
phones (connected to a TDM400 card).
I receive the CID correctly from my telco, but when I place the call to the
internal analog line, the CID is not propagated.
An interesting point: when I try to place a new call to an already bridged line,
I see the second call with the CID on the analog phone. The second call is
placed exactly with the same command/config as the first one.
In the debug log I see (for the second call):
-- Launched AGI Script /usr/share/asterisk/agi-bin/incoming.pl
-- AGI Script Executing Application: (Dial) Options: (Zap/2||otw)
-- CPE supports Call Waiting Caller*ID. Sending '/066332XX'
In other words, the CID is transmitted during a Call Waiting, but not during a
normal call. It looks like Asterisk does not send the CID (or send it too soon /
too late) during the first (normal) call.
Any idea is welcome.
Thanks!
AF.
--
*zapata.conf*
usecallerid=yes
usecallingpres=yes
callerid=asreceived
hidecallerid=no
callwaiting=yes
callwaitingcallerid=yes
treewaycalling=yes
transfer=yes
useincomingcalleridonzaptransfer=yes
...
context=home
signalling=fxo_ks
channel = 1
context=office
signalling=fxo_ks
channel = 2
context=freebox
signalling=fxs_ks
callerid=asreceived
channel = 3
context=francetelecom
signalling=fxs_ks
callerid=asreceived
channel = 4
*extensions.conf*
exten = s,1,Dial(${HOME},,otw)
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Re: Fwd: [asterisk-users] Some queries on g729 license.
(C)harlie (F)oxtrot Incredible! I didn't see it until just now. I use postini and it was in the quarantine with a XXX icon as the reason it got filtered. Let's set up a betting pool on him. :) C F wrote: I knew I was doing the right thing, here is the proof, enjoy when you read it, and have a good laugh. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] getting tones during conversation
after the Dial has connected, I want the caller (on a SIP phone) to be able to press keys in order to record call status. is this possible? __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Fax through Sangoma A102
Incoming faxes, the Sangoma will detect the tones and disable echo cancel. To send outbound, you will have to add another trunk group, of one or more channels and disable echo cancellation and use that to dial out. Example (/etc/asterisk/zapata.conf) blah blah echocancel=yes blah blah group = 1 channel =1-20 blah blah echo cancel=no group = 2 channel=22-23 So you would use for faxing specifically Zap/g2/number and it will use channel 22 or 23 but with echo turned off. Use Zap/g1 and it will use the first group of channels with echo cancel on (or whatever other parameters come before the group command) Bill From: [EMAIL PROTECTED] on behalf of jeremij jerome Sent: Tue 1/9/2007 10:52 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Fax through Sangoma A102 Hello, in our company we are trying to do this: Fax -- Traditional PBX -- Asterisk -- PSTN In practice, we have put an Asterisk equipped with a Sangoma A102 (2 PRI ports) between our PBX (Siemens HiCom) and the PSTN in order to have a VoIP network along the traditional telephony network. The problem is with the fax. We just want to send and receive faxes from/to our fax machine connected to the Siemens (without needing any interaction with our VoIP network, the faxes are sent to/received from PSTN). Unfortunately we are experiencing a lot of problems: the faxes not always work and when they work, it's likely to have incomplete pages. I know that faxing with VoIP is very troublesome, but maybe someone else is using a similar configuration and he found a good configuration or maybe has some hints to improve the results. We are using Asterisk 1.2.13. Thanks, Jeremi ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: RE: [asterisk-users] WIFI SIP- The Best phone
I've had the E70 for about a month. The first few days were not fun. But now that I've learned the gotchas and the workarounds, it is GREAT. You -can- configure it, and asterisk, to work perfectly together, every time. With automatic failover to conventional GSM phone behaviour if not in 802.11 land. I would be happy to give these to blinking 12:00 users, if -I- preconfigured them. This thing is great. Far exceeds expectation. Battery life with 802.11 is great, not discernibly worse than without it. _ On Tue, 9 Jan 2007, Stephen Davies wrote: On 09/01/07, Nigel Kendrick [EMAIL PROTECTED] wrote: I've had a play with a Nokia E70 - the 'bar' version of the E61 and gave up! Menu navigation is dire - I went through hoops trying to get SIP working - I know from others it can be done, but I bailed out when I realised that to put these phones in the hands of inexperienced users would be a recipe for a lot of frustration and support calls. Ironically I was going to recommend the E70. It is true that the menus are complex but once configured it does do what it says on the tin - provide a very effective merging of SIP over WIFI and GSM all in one unit. Regards, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Avaya IP Office
housi mueller wrote: I would like to connect an Asterik server to an Avaya IP Office IP406 and use the * as an VoIP Gateway. The IP Office has two Analog extensions available. I thought connecting this analog extensions to 2 FXO ports in the * to interconnect the PBX’s. What sort of interaction are you after? It may be a better idea to try to intercept the line card with asterisk, or if the IP406 has a VCM card then to talk to it through the ethernet interface. Is this possible? Does any one have experience with such a configuration? Thanks in advance for all recommandations and suggestions.. Housi Mueller __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] VOIP provider reliability
Anyone out there using VOIP for business class inbound/outbound services? I've found my VOIP provider to be less than reliable, SIP registrations timeout, calls drop, they claim IAX2 is too buggy (I find that hard to believe), and pretty much blame all problems on other circumstances and don't actually fix their problems. Their auto rollover doesn't even work right, so if my PBX is down, I just miss calls half the time. This isn't my first provider either, I've tried a couple now. Controversy, I've never had problems with my Vonage line, and I've had them for 3 years now (I know, unfair comparison). Anyone out there with better experiences? I'm mainly interested in hearing if people are replacing land lines with VOIP in a business setting and optionally what setups they've implemented to minimize or eliminate downtime (backup system, call routing, multiple SIP providers, etc). I realize this isn't the business list, so if you're going to recommend a provider, email me off list so as not to anger the natives. Thanks! -Kenneth ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Console\DSP
I am using a extension to dial the console which has autoanswer enabled. I am getting a strange warning, has anyone seen this before? Nothing on Google, or Voip-Info [Jan 9 13:50:05] WARNING[5009]: chan_oss.c:1048 oss_request: oss_request ty console data 0x0xb7851e00 dsp Call to device 'dsp' dnid '(null)' rdnis '(null)' on console from 'XX' XX Auto-answered -- Called dsp -- OSS/dsp answered SIP/mspri-usasterisk-0a119be0 Hangup on console == Spawn extension (system1, 6, 1) exited non-zero on 'SIP/mspri-usasterisk-0a119be0' It works fine, I am just concerned what the warning is for. the extension is simple exten = 6,1,Dial(console\dsp) BTW.. I am using chan_oss not alsa. Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to test VOIP quality?
I did a search: http://www.google.com/search?q=voip+quality+%28test+OR+testing%29+asterisk-users+site%3Amail-archive.com and found this: http://www.testyourvoip.com/ This seems to have quite a bit of detail. Does anyone have a better solution for testing VOIP quality? Comments? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Avaya IP Office
The main goal is that any extension from the Avaya PBX can make long distance calls using the asterisk server as VoIP gateway (using a SIP Provider). It would be also great if from a remote IP Phone (in an other location), a user could use the Asterisk server to dial in and the * forwards the call to an Avaya extension. The Avaya has an VCM card an IP Phones (5610) as extensions. First I thought to connect the * to the Avaya through the ethernet interface but then I was reading in forums that there are for Avaya third party IP phone licence needed and that the communication with oh323 is not stable. I thought also putting the Asterisk in front of the Avaya. Telco T1 - Asterisk - T1 - Avaya PBX This could be a solution for later one. Right know for testing it would be to expensive. That's why I thought about the Avaya analog Asterisk FXO interconnection. Any suggestions..? Thomas Kenyon [EMAIL PROTECTED] wrote: housi mueller wrote: I would like to connect an Asterik server to an Avaya IP Office IP406 and use the * as an VoIP Gateway. The IP Office has two Analog extensions available. I thought connecting this analog extensions to 2 FXO ports in the * to interconnect the PBXs. What sort of interaction are you after? It may be a better idea to try to intercept the line card with asterisk, or if the IP406 has a VCM card then to talk to it through the ethernet interface. Is this possible? Does any one have experience with such a configuration? Thanks in advance for all recommandations and suggestions.. Housi Mueller __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] postgres and asterisk
Hi O.Youssef, if you asterisk version is 1.2.X edit apps/Makefile and discomment the line that contain 'app_sql_postgres.so': # # Obsolete things... # APPS+=app_sql_postgres.so #APPS+=app_sql_odbc.so save if you use debian: aptitude install libpq-dev and compile again I hope this be helpfull ;p -- Humberto Figuera - Using Linux 2.6.18 Usuario GNU/Linux 369709 Caracas - Venezuela GPG Key Fingerprint = 5AAC DF0C 00F4 2834 28BA 37AD 3364 01D1 74CA 0603 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Snom side car annoyance
Andrew Latham wrote: you are asking about Shared line apperance or hints. Look at this http://www.voip-info.org/wiki/view/snom+360 Been there done that page. Nothing worth noting in there. -- J. Oquendo http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743 sil . infiltrated @ net http://www.infiltrated.net The happiness of society is the end of government. John Adams smime.p7s Description: S/MIME Cryptographic Signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Snom side car annoyance
J. Oquendo wrote: Andrew Latham wrote: you are asking about Shared line apperance or hints. Look at this http://www.voip-info.org/wiki/view/snom+360 Been there done that page. Nothing worth noting in there. Do the line appearances work on the 12 non-sidecar buttons? - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Avaya IP Office
Just done this for a client using an E1 Pri card in the avaya box and a sangoma a102, using qsig , works fine, I wouls recommend this to any oneits been up and stable for two months now Regards Robb housi mueller wrote: The main goal is that any extension from the Avaya PBX can make long distance calls using the asterisk server as VoIP gateway (using a SIP Provider). It would be also great if from a remote IP Phone (in an other location), a user could use the Asterisk server to dial in and the * forward’s the call to an Avaya extension. The Avaya has an VCM card an IP Phones (5610) as extensions. First I thought to connect the * to the Avaya through the ethernet interface but then I was reading in forums that there are for Avaya third party IP phone licence needed and that the communication with oh323 is not stable. I thought also putting the Asterisk in front of the Avaya. Telco T1 - Asterisk - T1 - Avaya PBX This could be a solution for later one. Right know for testing it would be to expensive. That's why I thought about the Avaya analog Asterisk FXO interconnection. Any suggestions..? */Thomas Kenyon [EMAIL PROTECTED]/* wrote: housi mueller wrote: I would like to connect an Asterik server to an Avaya IP Office IP406 and use the * as an VoIP Gateway. The IP Office has two Analog extensions available. I thought connecting this analog extensions to 2 FXO ports in the * to interconnect the PBX’s. What sort of interaction are you after? It may be a better idea to try to intercept the line card with asterisk, or if the IP406 has a VCM card then to talk to it through the ethernet interface. Is this possible? Does any one have experience with such a configuration? Thanks in advance for all recommandations and suggestions.. Housi Mueller __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Snom side car annoyance
Dr. Michael J. Chudobiak wrote: J. Oquendo wrote: Andrew Latham wrote: you are asking about Shared line apperance or hints. Look at this http://www.voip-info.org/wiki/view/snom+360 Been there done that page. Nothing worth noting in there. Do the line appearances work on the 12 non-sidecar buttons? - Mike Those I can get to work fine. Just when the side car comes into play The 12 on the Snom stop working as well -- J. Oquendo http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743 sil . infiltrated @ net http://www.infiltrated.net The happiness of society is the end of government. John Adams smime.p7s Description: S/MIME Cryptographic Signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Caller Id problem
always include a wait before a dial
give the callerid time to get into * before dialing, it arrives
between the first and second ring, if you have * dial after the first
ring it will not be there yet to pass along
On Jan 9, 2007, at 12:16 PM, Anton Frolov wrote:
Dear List,
My problem is that the incoming Caller Id is not displayed on the
local analog
phones (connected to a TDM400 card).
I receive the CID correctly from my telco, but when I place the
call to the
internal analog line, the CID is not propagated.
An interesting point: when I try to place a new call to an already
bridged line,
I see the second call with the CID on the analog phone. The second
call is
placed exactly with the same command/config as the first one.
In the debug log I see (for the second call):
-- Launched AGI Script /usr/share/asterisk/agi-bin/incoming.pl
-- AGI Script Executing Application: (Dial) Options: (Zap/2||otw)
-- CPE supports Call Waiting Caller*ID. Sending '/066332XX'
In other words, the CID is transmitted during a Call Waiting, but
not during a
normal call. It looks like Asterisk does not send the CID (or send
it too soon /
too late) during the first (normal) call.
Any idea is welcome.
Thanks!
AF.
--
*zapata.conf*
usecallerid=yes
usecallingpres=yes
callerid=asreceived
hidecallerid=no
callwaiting=yes
callwaitingcallerid=yes
treewaycalling=yes
transfer=yes
useincomingcalleridonzaptransfer=yes
...
context=home
signalling=fxo_ks
channel = 1
context=office
signalling=fxo_ks
channel = 2
context=freebox
signalling=fxs_ks
callerid=asreceived
channel = 3
context=francetelecom
signalling=fxs_ks
callerid=asreceived
channel = 4
*extensions.conf*
exten = s,1,Dial(${HOME},,otw)
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Re: [asterisk-users] Snom side car annoyance
I am siting in a building with 30 Snom 360s and 25 sidecars, I can assure you that it can work. Check you Snom Firmware, settings on the extra lines (you set them as shared). I should update the wiki someday, been a while... On 1/9/07, J. Oquendo [EMAIL PROTECTED] wrote: Dr. Michael J. Chudobiak wrote: J. Oquendo wrote: Andrew Latham wrote: you are asking about Shared line apperance or hints. Look at this http://www.voip-info.org/wiki/view/snom+360 Been there done that page. Nothing worth noting in there. Do the line appearances work on the 12 non-sidecar buttons? - Mike Those I can get to work fine. Just when the side car comes into play The 12 on the Snom stop working as well -- J. Oquendo http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743 sil . infiltrated @ net http://www.infiltrated.net The happiness of society is the end of government. John Adams ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! Hind sight is most always 20/20 or better. --- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.2.11 - ResponseTimeout being ignored
All - this is probably a simple problem, but I've been pulling my hair
out trying to figure out what I'm doing wrong. I'm building a
*simple* IVR menu. Here it is:
[main-menu]
exten = s,1,Answer
exten = s,2,SetMusicOnHold(default)
exten = s,3,DigitTimeout(5)
exten = s,4,ResponseTimeout(30)
exten = s,5,Background(logic-main)
exten = _4XX,1,Macro(stdexten,SIP/${EXTEN})
exten = 0,1,VoiceMail([EMAIL PROTECTED])
exten = 2,1,Directory(default|logic-boston]
exten = 2,2,Goto(main-menu,s,5)
exten = 3,1,Playback(logic-directions)
exten = 3,2,Goto(main-menu,s,5)
exten = t,1,GoTo(main-menu,s,5)
Everything is working fine except the ResponseTimeout(). My
understanding is that, as configured above, asterisk will wait for 30
seconds...if, after that amount of time, it hasn't received valid
digits, it'll jump to the t extension. That's not happening.
Immediately after the Background() sound file completes, I get this:
-- Playing 'logic-main' (language 'en')
== Auto fallthrough, channel 'SIP/445-0815e1d0' status is 'UNKNOWN'
Any ideas? This seemed like it should be simple, but it's getting the
best of me.
Thanks-
Erik
--
Erik Anderson
http://andersonfam.org
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Re: [asterisk-users] Handling SIP 482 condition
Eric ManxPower Wieling wrote: Chris Miller wrote: I would tend to agree, but the context that holds these number is an inbound context which includes additional logic that would fail normal calls. Yes, I can add the DIDs to the outbound context, but the point here is not to have a bloated dialplan with parallel data in multiple contexts. If I must have parallel data, I'd rather do a lookup in an external table using AstDB or an application similar to DUNDILookup() or ENUMLookup(). Another route I tried was to setup a local SIP trunk to catch the loops and send them down the inbound context. This fails because there are no SIP headers and the unknown peer is effectively NULL and will never match this trunk. As I said, they just get routed to from-sip-external. Put the DIDs in a context by themselves. include = that context in both your incoming context and your phones context. Thanks for the reply. I know this will work and am already doing this as a temporary workaround, but this doesn't really scale with hundreds/thousands of DIDs. I'm trying to avoid a bloated dialplan and the DIDs are already listed in another context, taking up space. What I'm looking for is some way to catch 482 loops and treat them as inbound calls without resorting to a parallel context. Failing that, I'd like to perform efficient lookups in an external DB, perhaps killing two birds with one stone (all DIDs can just exist in the DB). Chris ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Caller Id problem
From: Jerry Jones [EMAIL PROTECTED]
always include a wait before a dial
give the callerid time to get into * before dialing, it arrives between
the first and second ring, if you have * dial after the first ring it will
not be there yet to pass along
Is there a way to count number of rings?
Yuan Liu
On Jan 9, 2007, at 12:16 PM, Anton Frolov wrote:
Dear List,
My problem is that the incoming Caller Id is not displayed on the local
analog
phones (connected to a TDM400 card).
I receive the CID correctly from my telco, but when I place the call to
the
internal analog line, the CID is not propagated.
An interesting point: when I try to place a new call to an already
bridged line,
I see the second call with the CID on the analog phone. The second call
is
placed exactly with the same command/config as the first one.
In the debug log I see (for the second call):
-- Launched AGI Script /usr/share/asterisk/agi-bin/incoming.pl
-- AGI Script Executing Application: (Dial) Options: (Zap/2||otw)
-- CPE supports Call Waiting Caller*ID. Sending '/066332XX'
In other words, the CID is transmitted during a Call Waiting, but not
during a
normal call. It looks like Asterisk does not send the CID (or send it too
soon /
too late) during the first (normal) call.
Any idea is welcome.
Thanks!
AF.
--
*zapata.conf*
usecallerid=yes
usecallingpres=yes
callerid=asreceived
hidecallerid=no
callwaiting=yes
callwaitingcallerid=yes
treewaycalling=yes
transfer=yes
useincomingcalleridonzaptransfer=yes
...
context=home
signalling=fxo_ks
channel = 1
context=office
signalling=fxo_ks
channel = 2
context=freebox
signalling=fxs_ks
callerid=asreceived
channel = 3
context=francetelecom
signalling=fxs_ks
callerid=asreceived
channel = 4
*extensions.conf*
exten = s,1,Dial(${HOME},,otw)
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[asterisk-users] Is there a low cost cell phone base station for asterisk ?
I don't really know the name of what I want to look for but maybe someone could tell me if it would be available. I have a number of old analogue cell phones laying about here and I was thinking it would be useful if I could set up a short range base station for them that would cover maybe an acre or so. What I would like to be able to do is use it to connect into Asterisk and this way have a useful wireless extension-phone range. I do know that there are WiFi IP phones available but based on the connection range to our WiFi access points it seems limited as is our existing wireless handset (POTS). Any thoughts, suggestions ? Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there a low cost cell phone base station for asterisk ?
On Tue, Jan 09, 2007 at 05:11:55PM -0500, M.Hockings wrote: I don't really know the name of what I want to look for but maybe someone could tell me if it would be available. I have a number of old analogue cell phones laying about here and I was thinking it would be useful if I could set up a short range base station for them that would cover maybe an acre or so. What I would like to be able to do is use it to connect into Asterisk and this way have a useful wireless extension-phone range. Where are you. Generally you cant do that sort of thing as you don't have a license to operate in those frequencies. In the UK you definately don't (each cellphone has a license attached to it, it's just the operator pays the license fee). You cant get a license to operate a base station. Even if you could, running a basestation tends to need a hell of a lot of infrastructure behind it: - Basestation or BTS BSC (basesite controller) - generally can control up to about 100 BTSs. MSC (mobile switch centre) - like a telephony switch, connects BSCs and PSTN. HLR (home location register) - database of registered phones. Might need a VLR if allowing roaming. SMSC (short message service centre) handles SMS. Lots of glue ... Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Caller Id problem
thanks, Jerry but I don't thinks it's a problem, since I correctly get the CID from external line (moreover, I do some lookup of the received number in my LDAP database and making some decisions based on it). So when I call the Dial function, the CID is present in asterisk for sure. AF. Jerry Jones wrote: always include a wait before a dial give the callerid time to get into * before dialing, it arrives between the first and second ring, if you have * dial after the first ring it will not be there yet to pass along ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MINNESOTA: TwinCities Asterisk Users Group - Saturday January 13th 2007 - 11:30am
This is a reminder that the Twin Cities Asterisk Users Group will be meeting this Saturday, January 13 at 11:30am. - This month's meeting will focused on IP Telephony (VoIP) and network security, threats, defenses and countermeasures you can use to strengthen your Asterisk system. Meetings are held monthly on the second Saturday of each month, excluding July and December. The Agenda is posted online http://www.voip-info.org/wiki/index.php?page=Twin+Cities+Asterisk+User+Group+Agenda This meeting will be held at Atacomm Corporation Headquarters... -= 7365 Kirkwood Court N., Suite 350, Maple Grove, Minnesota USA 55369 =- http://maps.google.com/maps?f=qhl=enq=7365+kirkwood+court+n.+55369ie=UTF8z=15ll=45.089248,-93.433356spn=0.014392,0.039611om=1iwloc=addr Come to a meeting to meet other asterisk users, see asterisk solutions, win a door prize, eat food, or for the good company, to look for work, if your looking for employees, to go out for a drive, to get out of your house, whatever, JUST COME TO THE MEETING! Don't miss this meeting! DOOR PRIZES: 10 Snom 300's will be raffled at this meeting. These provide are a wonderful addition to your Asterisk hardware collection. Some members have been known to swap hardware at the meetings. Have extra VoIP gear, looking for VoIP gear? There's plenty of hardware to see. Have you been to a meeting recently? Please let us know if you can make it so we can plan accordingly. Come and share your own ideas and learn from others. As always, free food. We are always looking for help with meeting topics. If you feel like taking the lead, please do and simply let me know if you need anything. Meeting starts at 11:30am and parking is available everywhere. Meetings run about 2 hours. Look forward to seeing you there. http://www.voip-info.org/tiki-index.php?page=Asterisk%20User%20Group%20TwinCities%20Minnesota%20USA If you have a product or service you'd like to introduce to our members, send a private message to ejo1(at)soundchoicecomm.com and we'll see if we can't get you listed as next month's sponsor. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: RE: [asterisk-users] WIFI SIP- The Best phone
Wait for the iPhone...seriously. On 1/9/07, Jerry Glomph Black [EMAIL PROTECTED] wrote: I've had the E70 for about a month. The first few days were not fun. But now that I've learned the gotchas and the workarounds, it is GREAT. You -can- configure it, and asterisk, to work perfectly together, every time. With automatic failover to conventional GSM phone behaviour if not in 802.11 land. I would be happy to give these to blinking 12:00 users, if -I- preconfigured them. This thing is great. Far exceeds expectation. Battery life with 802.11 is great, not discernibly worse than without it. _ On Tue, 9 Jan 2007, Stephen Davies wrote: On 09/01/07, Nigel Kendrick [EMAIL PROTECTED] wrote: I've had a play with a Nokia E70 - the 'bar' version of the E61 and gave up! Menu navigation is dire - I went through hoops trying to get SIP working - I know from others it can be done, but I bailed out when I realised that to put these phones in the hands of inexperienced users would be a recipe for a lot of frustration and support calls. Ironically I was going to recommend the E70. It is true that the menus are complex but once configured it does do what it says on the tin - provide a very effective merging of SIP over WIFI and GSM all in one unit. Regards, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mitchel Constantin Snap - A desktop user interface for Asterisk www.snapanumber.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2.11 - ResponseTimeout being ignored
You need a 'waitexten()' after the background command.
On Tue, 9 Jan 2007, Erik Anderson wrote:
All - this is probably a simple problem, but I've been pulling my hair
out trying to figure out what I'm doing wrong. I'm building a
*simple* IVR menu. Here it is:
[main-menu]
exten = s,1,Answer
exten = s,2,SetMusicOnHold(default)
exten = s,3,DigitTimeout(5)
exten = s,4,ResponseTimeout(30)
exten = s,5,Background(logic-main)
exten = _4XX,1,Macro(stdexten,SIP/${EXTEN})
exten = 0,1,VoiceMail([EMAIL PROTECTED])
exten = 2,1,Directory(default|logic-boston]
exten = 2,2,Goto(main-menu,s,5)
exten = 3,1,Playback(logic-directions)
exten = 3,2,Goto(main-menu,s,5)
exten = t,1,GoTo(main-menu,s,5)
Everything is working fine except the ResponseTimeout(). My
understanding is that, as configured above, asterisk will wait for 30
seconds...if, after that amount of time, it hasn't received valid
digits, it'll jump to the t extension. That's not happening.
Immediately after the Background() sound file completes, I get this:
-- Playing 'logic-main' (language 'en')
== Auto fallthrough, channel 'SIP/445-0815e1d0' status is 'UNKNOWN'
Any ideas? This seemed like it should be simple, but it's getting the
best of me.
Thanks-
Erik
--
Erik Anderson
http://andersonfam.org
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Those that sacrifice essential liberty to obtain a little temporary safety
deserve neither liberty nor safety. -- Ben Franklin (1759)
* Doug Crompton *
* Richboro, PA 18954 *
* 215-431-6307*
* *
* [EMAIL PROTECTED]*
* http://www.crompton.com *
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Re: Fwd: [asterisk-users] Some queries on g729 license.
Derek Whitten Messages like this SHOULD NOT be posted to the list I have been trying to block you from my servers do to your abuse I will add this email address to the list also and contract your service provider. You are not doing the right thing you are acting like a child. I think you are abusing the list to send SPAM. And it is getting old blocking your email addresses And it getting old that you spoof my mail server and sending email with that look like it is coming from my servers. Derek if you keep this up I will press charges on you. I do track IP address on all email to my servers so yes I have all the proof I need from you. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Derek Whitten wrote: C F wrote: I knew I was doing the right thing, here is the proof, enjoy when you read it, and have a good laugh. -- Forwarded message -- From: Al Bochter [EMAIL PROTECTED] Date: Jan 8, 2007 8:22 PM Subject: Re: [asterisk-users] Some queries on g729 license. To: [EMAIL PROTECTED] (C)UNT (F)UCK! THIS IS OFF THE LIST FUCK YOU ASSHOLE! GET A JOB AND STOP LIVING OFF MY TAXES YOU DON'T KNOW WHAT YOU ARE DOING TRY AND STAY ON THE POINT. YOU ARE NOW BLOCKED I AM NOT GOING TO DEAL WITH JACKASSES LIKE YOU GOOD BYE Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email C F wrote: When I first noticed that this thread has over 20 messages i was sure it is interesting. When I read it I realized that I havn't noticed that Al Bochter has posted to it. Plain old stuff, just someone making sure to put a new twist on it. On 1/8/07, Juan Jose Comellas [EMAIL PROTECTED] wrote: The Intel IPP-based G.729 codec does work with AMD processors out of the box, both with the 32 bit and 64 bit versions. On Mon January 8 2007 19:31, Zoa wrote: I did some tests a long time ago and the speed was roughly the same. ( I think digium's was slightly faster). I think the IPP version also doesn't work on AMD out of the box. It's just 10$ a channel, that's not even worth the hassle of trying something else. Joachim Al Bochter wrote: Matthew I agree. I only know what I have told by others so I do need this input I have been told that Digum G729 is a big pain the the butt to get working with Asterisk and it is very hard on the CPU Keep in mind I have never used any Ver. of G 729 So tell me what you think. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Matthew Rubenstein wrote: All of which hassle and expense can be avoided by buying a license for Digium's codec, which is tested to work well with Asterisk (and might come with some support). And is pretty cheap per simul call. I wonder whether that per call means per codec instance, which could be multiple licenses on a single conference call, where multiple (even if not all) parties are getting de/encoded simultaneously. And whether there are other tools for editing (/mixing/transforming) g729 data, in realtime (streams) or not (files), and whether they require a license. Ideally sox or equivalent would work on g729, maybe with a codec plugin. On Mon, 2007-01-08 at 13:23 -0500, Paul wrote: First point to tackle in any case involving patent, copyright or trademark infringement is whether or not the infringing party would have been qualified to buy any usage rights at all. In a case where you license the Intel source(read the terms, it's not really that free), you would be applying for a license under some plan that includes certain minimum payments. Even if you wrote new source from scratch you would be in the same boat. Last time I looked at the plans, I didn't see anything with low minimums. So even if you wrote code from scratch and never used it on more than 6 channels, you might have done something that normally requires a large upfront payment. Use $10k as an example. In such a case owner of the patent might have an attorney initiate contact. If you are willing to communicate they might allow you to pay the minimum and be licensed. If you can't do that, they might offer a settlement where you stop using the codec and pay them some lesser amount. If the patent holder can easily prove the violation you might as well try to deal with them and get things settled fast. If you sell or give away the codec it is easier for them to dig up proof. If you have
Re: [asterisk-users] Asterisk 1.2.11 - ResponseTimeout being ignored
On 1/9/07, Doug Crompton [EMAIL PROTECTED] wrote: You need a 'waitexten()' after the background command. Gah! That worked perfectly. Thanks Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Fwd: [asterisk-users] Some queries on g729 license.
David So do you think Digum and Sipro is now one in the same code with G729 in mind? Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email David Thomas wrote: This is by far the most volotile list I have ever been on. I'm not sure that's exactly the reputation Digium/Asterisk is shooting for, but even so it does provide some much needed comedy relief. After seeing the G.729 pricing direct from SIPRO, I now take the shut-up and be thankful position. I think Digium has done us a great service by working out favorable pricing with SIPRO. Regards, David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Inbound (clean). Database: 0702-0, 01/09/2007 - 1/9/2007 5:23:48 PM ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: RE: [asterisk-users] WIFI SIP- The Best phone
On Tue, Jan 09, 2007 at 02:40:07PM -0800, mitcheloc wrote: Wait for the iPhone...seriously. I assume you mean Apple iPhone not Linksys iPhone ? It looks lovely, shame it's not available in UK until Q4. (also not FCC approved yet, but I assume that was deliberate as most phone leaks tend to come from filed FCC submissions). Steve p.s. also look at Truphone, they do WiFi/GSM/etc switching in client. -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Zap 1.4 error line 0: Unable to open master device '/dev/zap/ctl'
I've looked over EVERY resource I can find, but have run short of a solution. I'm running CentOS 4.4. Just installed Asterisk 1.4 and Zaptel 1.4 and libpri, but when I run ztcfg I get this error: line 0: Unable to open master device '/dev/zap/ctl' I realize this is a udev error (or from what I've read), but I cannot find out how to resolve this. I've reinstalled zaptel several times. I read a lot about having to read the README.udev file in the zaptel source, but I don't even have that file on my system. If anyone has any ideas I'd love to hear from them. -Chris ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Attatching VM via email for more than one user
Hi List, I am using asterisk 1.2.14 with real time and I am trying to send the email to more than one email address. In that field I put in [EMAIL PROTECTED];[EMAIL PROTECTED] When the call goes to VM I see in the CLI: uniqueid = 17 customer_id = 0 context = techmast mailbox = 14 password = 1234 fullname = Sales and Service email = [EMAIL PROTECTED] email = [EMAIL PROTECTED] tz = eastern attach = yes saycid = yes review = no operator = no envelope = no sayduration = no saydurationm = 1 sendvoicemail = no delete = no nextaftercmd = yes forcename = no forcegreetings = no hidefromdir = yes stamp = 2007-01-09 19:19:39 For some reason the VM's will only come in to the second address. And not the first. I looked in to the mail logs and I see no errors. Thanks. Dovid___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Attatching VM via email for more than one user
On 1/9/07, Dovid B [EMAIL PROTECTED] wrote: Hi List, I am using asterisk 1.2.14 with real time and I am trying to send the email to more than one email address. In that field I put in Send the email to an alias on the system and then have the alias point to the two email addresses. This may not work, though, depending on your particular situation. On a static type system, using conf files, it may be a solution, but since you are using realtime your application may not be able to handle the static nature of this configuration. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Fwd: [asterisk-users] Some queries on g729 license.
On 1/9/07, Al Bochter [EMAIL PROTECTED] wrote: So do you think Digum and Sipro is now one in the same code with G729 in mind? If saying this will make this go away, then yes. They both use the same code. The patented code is the same. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with zaptel drivers or card
I have an Asterisk box running Fedora Core 4, Asterisk 1.4, Lippri, 1.4, and Zaptel 1.4 The Digium cards installed are TDM2400 and TE110P. Everything was working fine until I upgraded to zaptel 1.2.12 from 1.2.9 Now when I run ztcfg I get the following error message: (CAS signalling on span 2 conflicts with Clear channel on channel 40) --NOTE: signaling was spelled wrong in the error message-- I have since upgraded to 1.4 with the same problem. Channel 40 is a standard bchan configuration and our provider sees no problem with the channel. When I disable the channel everything works fine. My assumption is that something is wrong with the TE110P card. Has anyone seen anything else like this? James Hawks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ooh323c calls
dear miche, pls place your number of softphone B into the context test dial plan. with best regards, osochebol - Original Message From: Michel [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, January 9, 2007 9:44:20 AM Subject: [asterisk-users] ooh323c calls Hi, I have two asterisk servers where softphone A is connected to asterisk A. On those two asterisk servers, ooh323c is installed. I tried to call a test context on asterisk B from softphone A. But I always fall into context default of asterisk B. ( I don't know how to tell asterisk A extensions.conf to call asterisk B test context) Here are conf files on asterisk A : ooh323.conf [softA] ; softphone A uses this channel type=user context=test ip=10.0.0.1 port=1720 disallow=all allow=gsm allow=ulaw [mypeer1] type=peer ip=10.0.0.2 port=1720 extensions.conf [test] exten = 15,1,Answer() exten = 15,n,Playback(vm-hello) exten = 15,n,Dial(OOH323/150/mypeer1);or exten = 15,n,Dial(OOH323/[EMAIL PROTECTED]) exten = 15,n,Hangup() May I use a gatekeeper? I learnt that ooh323c can act as gatekeeper, but I didn't success to configure it (I have gatekeeper is not responding error!). Can one of my server acts as gatekeeper and gateway? Do anyone success to configure gatekeeper with ooh323c ? Thanks you for you help! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] postgres and asterisk
Is that procedure the way to completely switch Asterisk from dependency on MySQL to dependency on Postgres instead? How about with Asterisk 1.4? And anyone have any idea whether FreePBX can be switched from MySQL to Postgres, too? On Tue, 2007-01-09 at 16:01 -0700, [EMAIL PROTECTED] wrote: Date: Tue, 9 Jan 2007 16:54:24 -0400 From: Humberto Figuera [EMAIL PROTECTED] Subject: Re: [asterisk-users] postgres and asterisk To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed Hi O.Youssef, if you asterisk version is 1.2.X edit apps/Makefile and discomment the line that contain 'app_sql_postgres.so': # # Obsolete things... # APPS+=app_sql_postgres.so #APPS+=app_sql_odbc.so save if you use debian: aptitude install libpq-dev and compile again I hope this be helpfull ;p -- Humberto Figuera - Using Linux 2.6.18 Usuario GNU/Linux 369709 Caracas - Venezuela GPG Key Fingerprint = 5AAC DF0C 00F4 2834 28BA 37AD 3364 01D1 74CA 0603 -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Fwd: [asterisk-users] Some queries on g729 license.
Al Bochter wrote: Derek Whitten Messages like this SHOULD NOT be posted to the list I have been trying to block you from my servers do to your abuse I will add this email address to the list also and contract your service provider. You are not doing the right thing you are acting like a child. I think you are abusing the list to send SPAM. And it is getting old blocking your email addresses And it getting old that you spoof my mail server and sending email with that look like it is coming from my servers. Derek if you keep this up I will press charges on you. I do track IP address on all email to my servers so yes I have all the proof I need from you. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email whatever dude.. you may consider seeking professional help. one message in your direction because you refuse to quit acting like an immature spoiled little brat on this mailing list does not constitute abuse. Track away asshat, I am not spoofing a dam thing. Why don't you go back to school and learn how to read email headers and source correctly. Not to mention that you should possibly take a few brush up classes on other topics as well since it is beyond obvious to me (and probably many other people here) that you don't have a clue. If you have so many issues with this mailing list, why don't you do yourself and everyone else here a favor, unsubscribe from this mailing list, and go on with your pathetic, minuscule existence. go away troll signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there a low cost cell phone base station for asterisk ?
M.Hockings wrote: I don't really know the name of what I want to look for but maybe someone could tell me if it would be available. I have a number of old analogue cell phones laying about here and I was thinking it would be useful if I could set up a short range base station for them that would cover maybe an acre or so. What I would like to be able to do is use it to connect into Asterisk and this way have a useful wireless extension-phone range. I do know that there are WiFi IP phones available but based on the connection range to our WiFi access points it seems limited as is our existing wireless handset (POTS). Any thoughts, suggestions ? You would have to talk to the FCC if you want to operate on any frequency those phones might support. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Handling SIP 482 condition
Chris Miller wrote: Eric ManxPower Wieling wrote: Chris Miller wrote: I would tend to agree, but the context that holds these number is an inbound context which includes additional logic that would fail normal calls. Yes, I can add the DIDs to the outbound context, but the point here is not to have a bloated dialplan with parallel data in multiple contexts. If I must have parallel data, I'd rather do a lookup in an external table using AstDB or an application similar to DUNDILookup() or ENUMLookup(). Another route I tried was to setup a local SIP trunk to catch the loops and send them down the inbound context. This fails because there are no SIP headers and the unknown peer is effectively NULL and will never match this trunk. As I said, they just get routed to from-sip-external. Put the DIDs in a context by themselves. include = that context in both your incoming context and your phones context. Thanks for the reply. I know this will work and am already doing this as a temporary workaround, but this doesn't really scale with hundreds/thousands of DIDs. I'm trying to avoid a bloated dialplan and the DIDs are already listed in another context, taking up space. What I'm looking for is some way to catch 482 loops and treat them as inbound calls without resorting to a parallel context. Failing that, I'd like to perform efficient lookups in an external DB, perhaps killing two birds with one stone (all DIDs can just exist in the DB). Put the DIDs in ONE context. Include them anywhere you need them. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there a low cost cell phone base station for asterisk ?
It is true what Eric and Steve have said, you do need a licensed GSM frequency to operate and sell GSM services (even for rural areas). however, this link might be of interest to you http://rfdesign.com/mag/radio_field_trials_allsoftware/ On 1/10/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: M.Hockings wrote: I don't really know the name of what I want to look for but maybe someone could tell me if it would be available. I have a number of old analogue cell phones laying about here and I was thinking it would be useful if I could set up a short range base station for them that would cover maybe an acre or so. What I would like to be able to do is use it to connect into Asterisk and this way have a useful wireless extension-phone range. I do know that there are WiFi IP phones available but based on the connection range to our WiFi access points it seems limited as is our existing wireless handset (POTS). Any thoughts, suggestions ? You would have to talk to the FCC if you want to operate on any frequency those phones might support. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zap 1.4 error line 0: Unable to open master device '/dev/zap/ctl'
On Tue, Jan 09, 2007 at 05:50:52PM -0600, Chris Bullock wrote: I've looked over EVERY resource I can find, but have run short of a solution. I'm running CentOS 4.4. Just installed Asterisk 1.4 and Zaptel 1.4 and libpri, but when I run ztcfg I get this error: line 0: Unable to open master device '/dev/zap/ctl' This is a generic error message ztcfg gives when it fails to open /dev/zap/ctl. It is followed by the error string of the error code it got (usually: no such file or no such device). No such file: the file /dev/zap/ctl is simply not there. No such device: The file is there, but there is no device to support it. If you use udev (or the older devfs) and have not created the device file yourself manually with mknod, you probably won't get the latter. I realize this is a udev error (or from what I've read), but I cannot find out how to resolve this. I've reinstalled zaptel several times. I read a lot about having to read the README.udev file in the zaptel source, but I don't even have that file on my system. If anyone has any ideas I'd love to hear from them. It may be because the module zaptel has failed to load. Do you have the directory /proc/zaptel ? lsmod | grep zaptel -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with zaptel drivers or card
On Tue, Jan 09, 2007 at 06:01:55PM -0700, Administrator wrote: I have an Asterisk box running Fedora Core 4, Asterisk 1.4, Lippri, 1.4, and Zaptel 1.4 The Digium cards installed are TDM2400 and TE110P. Everything was working fine until I upgraded to zaptel 1.2.12 from 1.2.9 Now when I run ztcfg I get the following error message: (CAS signalling on span 2 conflicts with Clear channel on channel 40) --NOTE: signaling was spelled wrong in the error message-- I have since upgraded to 1.4 with the same problem. Channel 40 is a standard bchan configuration and our provider sees no problem with the channel. When I disable the channel everything works fine. My assumption is that something is wrong with the TE110P card. Has anyone seen anything else like this? What do you get from: cat /proc/zaptel/* What do you have on /etc/zaptel.conf ? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Newbie question: How to config rtp packetization in 1.4?
Hi, any one test rtp packetization in 1.4?___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ztmonitor output while idle
Hi All, I am trying to tune out some echo on a analogue line and have run ztmonitor to get some info. When i run it, i get a RX reading when the line is idle - is this normal? eg: [EMAIL PROTECTED] zaptel-1.2.10]# ./ztmonitor 1 -vv Visual Audio Levels. Use zapata.conf file to adjust the gains if needed. ( # = Audio Level * = Max Audio Hit ) (RX) (TX) Rx: 132 ( 132) Tx: 0 (0) This happens on all 4 lines with varying rx levels Cheers, Ben ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Fwd: [asterisk-users] Some queries on g729 license.
Hi On Tue, Jan 09, 2007 at 05:59:55PM -0500, Al Bochter wrote: Derek Whitten Messages like this SHOULD NOT be posted to the list I fully agree here. However: I have been trying to block you from my servers do to your abuse If you want to blacklist someone on your own servers. However threatening so on-list is not a good idea. I will add this email address to the list also and contract your service provider. You are not doing the right thing you are acting like a child. I think you are abusing the list to send SPAM. Well, the list has an administrator. Look at the standard mailing list headers. The list administrator has authority to decide if somebody is abusing the list resources (as those are his resources). And it is getting old blocking your email addresses And it getting old that you spoof my mail server and sending email with that look like it is coming from my servers. Derek if you keep this up I will press charges on you. I do track IP address on all email to my servers so yes I have all the proof I need from you. Please stop those threats. They will not stop people from sending you hate mail. Ignore it. Now can we get back to the usual list topics? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Attatching VM via email for more than one user
On Tue, Jan 09, 2007 at 06:44:38PM -0600, Lacy Moore - Aspendora wrote: On 1/9/07, Dovid B [EMAIL PROTECTED] wrote: Hi List, I am using asterisk 1.2.14 with real time and I am trying to send the email to more than one email address. In that field I put in Send the email to an alias on the system and then have the alias point to the two email addresses. This may not work, though, depending on your particular situation. On a static type system, using conf files, it may be a solution, but since you are using realtime your application may not be able to handle the static nature of this configuration. Mail servers may also be able to read their aliases from a database. Postfix, for instance can mix its aliases table from several sources. Those may be a db/hash file, an LDAP DB, mysql DB, PGSQL DB, etc. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ztmonitor output while idle
On Wed, Jan 10, 2007 at 05:41:50PM +1100, Ben Dinnerville wrote: Hi All, I am trying to tune out some echo on a analogue line and have run ztmonitor to get some info. When i run it, i get a RX reading when the line is idle - is this normal? eg: [EMAIL PROTECTED] zaptel-1.2.10]# ./ztmonitor 1 -vv Visual Audio Levels. Use zapata.conf file to adjust the gains if needed. ( # = Audio Level * = Max Audio Hit ) (RX) (TX) Rx: 132 ( 132) Tx: 0 (0) This happens on all 4 lines with varying rx levels When the line is on-hook, its sample values are basically meaningless, I gather. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] cannot call out
hello all. i switched to * 1.4 and have now 2 problems. 1. i can't make a call out with the current branch i always have in the logfile: [Jan 9 14:45:09] NOTICE[15246] chan_sip.c: Unable to create/find SIP channel for this INVITE With the asterisk 1.4 Release it is working, 2. when i do core show hints i see the channels on idle and unavailable BUT when a channel is active it is still shown as IDLE, somebody has a solution? Regards Rene ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
