[asterisk-users] How to route SIP provider without DID
Hi, I'm struggling to separate inbound calls fro a SIP provider that does not send DID. I have tried ...sip.com/12345678 on register string different context=from-no-did Port not possible as only support 5060 Any suggestions? Thank you! HB -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sending SMS from asterisk
Hi, I have some singe port GoIP GSM gateways, good price and after numerous updates their run several weeks before radio hangs and a cold start is needed. It has a WEB based feature to send and receive SMS, it had been nice to be able to send SMS from Asterisk with it. Tips on this wanted:) Best regards HB asterisk-users-requ...@lists.digium.com wrote: Re: Sending SMS from asterisk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk as a E1/T1 FXO ?
Hi, Does anybody here have experience using Asterisk as an FXO to emulate a E1/T1/PRI line for test purpose? Gateway or PCI card? Thank you! HB -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Mediatrix 4400plus ISDN setup
Hi, Strugling to get Mediatrix 4400plus running on *, does anyone have a script or example to share:) Thank you! HB Norway -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Router with PPPoE and QOS for light Asterisk and general use
Hi, I have used Linksys, Buffalo and current Asus router with DD-WRT there are some stability issues and PPPoE and QOS does not work. I tried Tomato too but that gives strange SIP problems. I would very much appreciated tips on a good and stable router for Asterisk thats not too hard to set up, and hopefully bellow my 500 USD pain limit:) Linux/PC based router also interesting, if recommendable? Thank you! HB Norway -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Missed call when call is answered by other phone
Hi, I use follow me and have several SIP phones answering, works nice but: All phones that did not answer a call have the number in missed call list even if answered by other ext. CDR gets messy too. Difficult to see if call is answered. I was thinking of possible solution: Turn of missed call on phones and instead have a php script make a list on web or maybe if not a smarter way over SIP? Any good idea? Best regards HB -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Mirroring or other arangement to secure *
Hi, Please excuse me for addressing this Linux issue on this list, however I hope that some of you have found a solution thats matches the * use and also easy to install without very deep knowledge of Linux. My wish are a program that maintain a mirror copy of the HD. Thank you! Best regards HB -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Howto regret blind transfer?
Hi, Is it possible to regret blind transfer while its ringing (not answered)? Thank you! Best regards HB -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] iphone client app
IAXDIAL is free on app store works great on WiFi even true NATs but seem blocked for GPRS. HB Re: [asterisk-users] iphone client app From: Alex Samad a...@samad.com.au Date: Tue, 15 Dec 2009 12:08:37 +1100 To: asterisk-users@lists.digium.com To: asterisk-users@lists.digium.com Well I have a 3gs - will tell you how that goes. decided against siax - have to pay for the base model. installed fringe, but no voip over 3g, have to wait till i get home, but it registered with my asterisk server so .. I am looking for the hacked fring.ipa which allows voip over 3g, just so I can try I am in Aus, and i bought the phone outright so no lock in. A On Mon, Dec 14, 2009 at 07:25:37PM -0500, Mike Bessette wrote: I find that Siphone works great on the iTouch. Tried it with my own asterisk box as well as Callcentric and MagicJack and it was very clear and stable. Haven't played with it since the last firmware update though as the update removed support for 3rd party headsets . On 12/14/09, Alex Balashov abalas...@evaristesys.com wrote: I personally have not had much luck with these softphones because the iPhone 3G seems to be underpowered and just doesn't run them well enough to sustain good voice quality, irrespective of wifi network conditions. I could be mistaken, though. It's not going to happen over ATT's 3G network -- not down here in the southeastern US, as far as I can tell. So I can't speak to whether voice works over 3G. -- Sent from mobile device On Dec 14, 2009, at 6:57 PM, Alex Samad a...@samad.com.au wrote: On Mon, Dec 14, 2009 at 07:37:08AM +, Brian Chamberlain wrote: Fring, it's free and works perfectly with an Asterisk server.. thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Random DTMF tones generated from speech
Thank you, very interesting! As I understand the Digium card is used as a interrupt source for Asterisk? Is there a diagnostic tool available ? Anybody else experienced a simmialr problem? Thank you! HB From: cov...@ccs.covici.com Date: Sat, 12 Dec 2009 19:04:23 -0500 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com I used to have this problem with a Digium 400p -- even when not in use and a Motherboard which was inadequate in terms of the interrupts for the Digium card -- when I got a better Motherboard the problem went away. hbk fo...@online.no wrote: Hi, My Asterisk systems runs like a dream with mISDN, SIP and even and old Digium board. But have almost in every conversation some irritating DTMF being generated. The seems to be just as often from all trunks but are worse if noise load speaker in other end. Any good advices? Where to look for forgotten DTMF detection settings? Thank you! HB ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Random DTMF tones generated from speech in conversations
Hi, My Asterisk systems runs like a dream with mISDN, SIP and even and old Digium board. But have almost in every conversation some irritating DTMF being generated. The seems to be just as often from all trunks but are worse if noise load speaker in other end. Any good advices? Where to look for forgotten DTMF detection settings? Thank you! HB ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Best practice to set up 4 line phones
Hi, I would like some advice from you on how to configure a multi line phone the best way! So far I have given the phone 4 sip accounts one for each line, this is a lot of work and gets messy. Is it a better way to do this? Thank you! Best regards Helge-Bjørn ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Skyp SIP? - what is free for a home *
Hi, I get confused about all solutions for Skype! I want to connect as simple as possible out home * to be able to at least answer Skype calls. Now I use a PC USB box and a FXO, works ok both call directions but uses a PC. Any good and free idea ? Thank you! HB ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SNOM 870
Hi, I have played with the 820 for some weeks, mostly love it excellent speech quality. Even got the mini browser running showing my favorite webcam, this could be put to real use too:) Issues so far: Some embarrassing crashes while speaking, was able to speak but all freezed. Still a little fresh firmware I guess. Error 404 after showing webcam picture, but it works! Have to use *1 to start recording, record soft button does not seem to work with *. Still I recommend it, best IP phone I have tried! Not sure 870 is worth the extra money, not tested that yet. Best regards HB Norway Subject: [asterisk-users] SNOM 870 From: --[ UxBoD ]-- ux...@splatnix.net Date: Fri, 30 Oct 2009 17:22:30 + (GMT) To: asterisk-users@lists.digium.com To: asterisk-users@lists.digium.com Hi, Are any of you using them yet ? How did you get on ? Have you tried the streaming capabilities in the XML browser ? Any issues with Asterisk ? Seems very clever technology :) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Best afordable router with QOS for *
Hi, I need to get a new router for private/SOHO use of *, especially when the kids are on internet:( Any a good advice? Thank you! Best regards HB ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] www.chan_mobile.org seems dead ?
Any tip on where to find install doc on chan_mobile ? http://www.chan_mobile.org Thnak you! HB ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] mISDN NT mode config setting
Hi, I am struggling to get plain Cologne chip cards to run in NT mode, runs nice in TE mode despite the error message: login as: root r...@192.168.2.22's password: Last login: Tue Sep 1 23:09:24 2009 from 192.168.2.50 Welcome to Elastix misdnportinfo Port 1: TE-mode BRI S/T interface line (for phone lines) - Protocol: DSS1 (Euro ISDN) - Layer 4 protocol 0x0401 is detected, but not allowed for TE lib. - childcnt: 2 * Port NOT useable for PBX (maybe there is already a PBX running?) Port 2: TE-mode BRI S/T interface line (for phone lines) - Protocol: DSS1 (Euro ISDN) - Layer 4 protocol 0x0401 is detected, but not allowed for TE lib. - childcnt: 2 * Port NOT useable for PBX (maybe there is already a PBX running?) Port 3: NT-mode BRI S/T interface port (for phones) - Layer 2 protocol 0x0202 is detected, but not allowed for NT lib. * Port NOT useable for PBX (maybe there is already a PBX running?) Port 4: NT-mode BRI S/T interface port (for phones) - Layer 2 protocol 0x0202 is detected, but not allowed for NT lib. * Port NOT useable for PBX (maybe there is already a PBX running?) I do: etc/misdn-init.conf card=1,hfcpci card=2,hfcpci card=3,hfcpci card=4,hfcpci te_ptmp=1,2 nt_ptp=3,4 --- etc/misdn.conf mISDNconf module poll=128 debug=0 timer=nohfcmulti/module module debug=0 options=0mISDN_dsp/module devnode user=asterisk group=asterisk mode=660mISDN/devnode card type=hfcpci port mode=te link=ptmp1/port /card card type=hfcpci port mode=te link=ptmp1/port /card card type=hfcpci dtmf=yes crystalclock=yes port mode=nt link=ptp1/port /card card type=hfcpci dtmf=yes crystalclock=yes port mode=nt link=ptp1/port /card /mISDNconf -- etc/asterisk/misdn.conf ;Tried with both config's no change misdn_init=/etc/misdn-init.conf ;misdn_init=/etc/mISDN.conf [trunks] ports=1,2 context=from-trunk msns=* [NTports] context=from-internal ports=3,4 msns=* --- I am using Elastix 1.5 not updated Any tips? Where can I find documentation? Thank you! HB ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] KEY SYSTEM, Intercom
Hi, I am still dreaming of replacing our now 16 year old Panasonic with *! I have played with * for many years but are uncertain how to get Panasonic functions like: KEY system buttons/LED or LCD soft keys indicating outside line usage. Intercom function that allows one way talking true speaker on all phones in a group. Intercom function that allows two way direct speaker phone conversation from approved phones without receiver having to touch his phone. Any suggestion on Phone and how to implement? Thank you! HB Norway ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Secure WAN connection too * extention
Hi, Is there an easy way to encrypt connection to an * extension ? To soft phone or preferably a HW phone? Thank you! Best regards HB Norway ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Door phone
Hi, Is there an affordable HW solution to do a door phone on *? I do not mind using the solder iron to modify an existing door box. Thank you! Best regards HB Norway ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk = SKYPE
Hi Any solution for connecting Asterisk to Skype without using fsx/fxo hardware ? Best Regards HB Norway ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: EuroISDN BRI 2 or 4 wires? (Remco Barende)
Hi ISDN wire: From phone company you receive on two wire, this is called U interface on this you can connect only one device, normaly the NT1 box. On the NT1 there is a S/T bus that allows several devices (phones) connected (in TE mode)! Yes S/T is four wire ! HB ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk@home ISO install of ISDN card with HFC ?
Hi Have anybody successfully installed ISDN with HFC chips on [EMAIL PROTECTED] ISO ? Please tell me how you did it ? Thank you ! HB Norway ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Bri-stuff + TDM 2-Port FXS 2 Port FXO Card
Hi Mary Christmas to y'all ! I am trying to configure * with one TDM and one ISDN and Bri-stuff, all is ok with only TDM but when add ISDN I get following error: I have tried to switch fxs/fxo but then I get error also without ISDN, card is configured from Digium with FXS(green) modules are closest to the bracket. ZT_CHANCONFIG failed on channel 1: Invalid argument (22) Did you forget that FXS interfaces are configured with FXO signalling and that FXO interfaces use FXS signalling? loadzone=no defaultzone=no fxoks=1-2 # Make sure that the FXS(green) modules are closest to the bracket fxsks=3-4 # This is for the FXO module(s) because it uses FXS span=2,1,3,ccs,ami bchan=5-6 # Old value: 1-2 dchan=7# Old value: 3 Thank you ! HB Norway ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ISDN HiSax: unauthorized source code changes
Hi After modprobe hisax type=35 (Billion HFC PCI) on a Xorcom Rapid ISO I get: HISAX Dec 12 16:25:35 localhost kernel: HiSax: Linux Driver for passive ISDN cards Dec 12 16:25:35 localhost kernel: HiSax: Version 3.5 (module) Dec 12 16:25:35 localhost kernel: HiSax: Layer1 Revision 1.1.4.1 Dec 12 16:25:35 localhost kernel: HiSax: Layer2 Revision 1.1.4.1 Dec 12 16:25:35 localhost kernel: HiSax: TeiMgr Revision 1.1.4.1 Dec 12 16:25:35 localhost kernel: HiSax: Layer3 Revision 1.1.4.1 Dec 12 16:25:35 localhost kernel: HiSax: LinkLayer Revision 1.1.4.1 Dec 12 16:25:35 localhost kernel: HiSax: Approval certification failed because of Dec 12 16:25:35 localhost kernel: HiSax: unauthorized source code changes I found this on: http://info.ccone.at/INFO/ISDN4Linux/i4lfaq-7.html#ss7.26 What to do, where to find a rpm or ?? to fix ?? Thank you! HB Norway ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Recomended ISDN on Asterisk@home ?
Hi I have installed the http://asteriskathome.sourceforge.net/ with a Digium card with no problems, very good ! Now I want to install my Billion PCI ISDN card (HFC based) in TE mode. I get a little confused with Isdn4Linux, ZapHFC HIAX and the need to install Capi ! Please suggest best and easiest approach ? Thank you ! HB Norway ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Recomended ISDN for Asterisk ?
Hi I have installed the http://asteriskathome.sourceforge.net/ with a Digium card with no problems, very good ! Now I want to install my Billion PCI ISDN card (HFC based) in TE mode. I get a little confused with Isdn4Linux, ZapHFC HIAX and the need to install Capi ! Please suggest best and easiest approach ? Thank you ! HB Norway ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and ISDN HFC-S card (Biilion) instead of Fritz Capi ?
Hi I am trying the fine iso at http://www.asterisk.de.ms/ but are having problems with Capi probably due to having to old Fritz PCI card. Trying with both non version marked version and version marked V 2.0. I get following error when booting Astrisk on Debian: Oct 31 02:26:10 asterisk kernel: kcapi: capi20 attached Oct 31 02:26:10 asterisk kernel: capi20: Rev 1.1.4.2: started up with major 68 ( middleware+capifs) Oct 31 02:26:10 asterisk kernel: fcpci: AVM FRITZ!Card PCI driver, revision 0.2 Oct 31 02:26:10 asterisk kernel: fcpci: (fcpci built on Jun 3 2004 at 12:02:52) Oct 31 02:26:10 asterisk kernel: fcpci: Loading... Oct 31 02:26:10 asterisk kernel: fcpci: Driver 'fcpci' attached to stack Oct 31 02:26:10 asterisk kernel: kcapi: driver fcpci attached Oct 31 02:26:10 asterisk kernel: fcpci: Auto-attaching... Oct 31 02:26:10 asterisk kernel: PCI: Found IRQ 11 for device 00:0c.0 Oct 31 02:26:10 asterisk kernel: fcpci: Error: Invalid parameters (base=0xe800, irq=11) Oct 31 02:26:10 asterisk kernel: fcpci: Not loaded. Oct 31 02:26:10 asterisk kernel: kcapi: driver fcpci detached Can anbody on the list give a clue on this problem ? Or can you please direct me to a guide on how to install driver for HFC-S based ISDN card on Asterisk/Debian ? Thank you ! HB Norway ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
