Re: [asterisk-users] no audio while call forwarding, yes audio with followme
Hi. Thank you. You mean do each call separately? That works without a glitch, nothing peculiar. Thx, BC On 09/25/12 23:28, Danny Nicholas wrote: Do the call both ways again and check(post) the CLI output. *From:*asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Bart Coninckx *Sent:* Tuesday, September 25, 2012 4:23 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] no audio while call forwarding, yes audio with followme Hi all, the subject says it all. Technical details: - Asterisk 1.8.7.1 - Behind NAT - Using external SIP provider The call forwarding is tested both with this functionality on the phone and with configuration in the dialplan. In the latter case a database variable is set to the external number, if set a Dial command calls this number. So really nothing fancy (actually I followed the example on http://www.voip-info.org/wiki/view/Asterisk+call+forwarding ). sip.conf has nat=yes, externip= ... and I tried every setting of directmedia in the providers configuration part. Followme works flawlessly, so I'm really wondering if this is a NAT issue. Can anyone point me into a certain direction? Thx BC -- _ -- 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 -- _ -- 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] no audio while call forwarding, yes audio with followme
Hi all, the subject says it all. Technical details: - Asterisk 1.8.7.1 - Behind NAT - Using external SIP provider The call forwarding is tested both with this functionality on the phone and with configuration in the dialplan. In the latter case a database variable is set to the external number, if set a Dial command calls this number. So really nothing fancy (actually I followed the example on http://www.voip-info.org/wiki/view/Asterisk+call+forwarding ). sip.conf has nat=yes, externip= ... and I tried every setting of directmedia in the providers configuration part. Followme works flawlessly, so I'm really wondering if this is a NAT issue. Can anyone point me into a certain direction? Thx BC -- _ -- 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] Belgian BRI (euroisdn): what to use for a B410P
On 05/10/12 00:09, Richard Mudgett wrote: Please just reply to the mailing list. oops, that was my intention, my bad. Are you able to make calls when in PTP mode? I just tested: yes it seams so! The warning message is just complaining about receiving unexpected TEI management messages because the span is in PTP mode. It is otherwise benign if the line is really PTP. If you can make calls, please create a JIRA issue on the PRI project so the message level can be reduced. Please attach an intense pri debug output showing the received MDL messages. pri set debug 2 span 4 https://issues.asterisk.org/jira Richard Will do. I suppose there is no way to make them disappear already, except for turning of WARNING messages. BC -- _ -- 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] looking for solid state like PC suitable for Asterisk
Hi all, for smaller (or maybe even bigger) sites I'm looking for a smaller, appliance-type like PC, preferably solid state and fanless PC. Since it's only going to run Asterisk for a couple of extensions I don't think CPU and RAM need to be maxed out. Does anyone have inspiration/experience for/about such a model? thx!! BC -- _ -- 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] looking for solid state like PC suitable for Asterisk
On 05/10/12 13:49, A J Stiles wrote: On Thursday 10 May 2012, Bart Coninckx wrote: I'm looking for a smaller, appliance-type like PC, preferably solid state and fanless PC. Since it's only going to run Asterisk for a couple of extensions I don't think CPU and RAM need to be maxed out. Does anyone have inspiration/experience for/about such a model? Raspberry Pi would be the obvious choice, surely? I'm in the waiting queue for one, but they still seem to be needing to sell one per person, while I need many. Not a bad idea though, BC -- _ -- 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] looking for solid state like PC suitable for Asterisk
That's Soekris I suppose. Never heard of them, but it looks mighty interesting. Cheers, BC On 05/10/12 13:35, John Novack wrote: I use HP Thin Clients with AstLinux installed. HP 5720's are available on eBay for not much money, or there are many small boards available new if you don't or can't use used. 10 watts, no fan, no HD Not sure what might be available in your part of the world, but there are Sockris and ALIX flash based boards. AstLinux has special configurations for these. I have 20-30 AstLinux on thin clients working without a belch on a private collectors network John Novack Bart Coninckx wrote: Hi all, for smaller (or maybe even bigger) sites I'm looking for a smaller, appliance-type like PC, preferably solid state and fanless PC. Since it's only going to run Asterisk for a couple of extensions I don't think CPU and RAM need to be maxed out. Does anyone have inspiration/experience for/about such a model? thx!! BC -- _ -- 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 -- _ -- 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] looking for solid state like PC suitable for Asterisk
This is for an ISDN project, but Beronet has ISDN gateways with ethernet, so even that might not be an issue, cheers, BC On 05/10/12 13:43, Arstan Jusupov wrote: Another option is to get those routers that are capable of running dd-wrt firmware with USB ports(for storage) This option is rather good if you don't need any VoIP cards and if you are OK to use sip/iax2 etc trunks. I have my wifi router with dd-wrt firmware running asterisk for home use. It's cheap, small, uses less power, noiseless and :) just cool Sent from my iPhone On May 10, 2012, at 7:35 PM, John Novackjnov...@stromberg-carlson.org wrote: I use HP Thin Clients with AstLinux installed. HP 5720's are available on eBay for not much money, or there are many small boards available new if you don't or can't use used. 10 watts, no fan, no HD Not sure what might be available in your part of the world, but there are Sockris and ALIX flash based boards. AstLinux has special configurations for these. I have 20-30 AstLinux on thin clients working without a belch on a private collectors network John Novack Bart Coninckx wrote: Hi all, for smaller (or maybe even bigger) sites I'm looking for a smaller, appliance-type like PC, preferably solid state and fanless PC. Since it's only going to run Asterisk for a couple of extensions I don't think CPU and RAM need to be maxed out. Does anyone have inspiration/experience for/about such a model? thx!! BC -- _ -- 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 -- Dog is my Co-pilot -- _ -- 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 -- _ -- 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 -- _ -- 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] looking for solid state like PC suitable for Asterisk
There seems to even be a 1.6 Ghz Intel Atom device. One site I'm looking to use this for has about 40 SIP phones and three BRIs. It's always a guessing game whether devices like this are up for that. If they do have some processing power, I might even consider combining them as a highly available Asterisk cluster (using DRBD and Pacemaker). Anyone 2 cents about that? BC On 05/10/12 14:28, John Novack wrote: Correct. I have never been accused of being a good speller! JN Bart Coninckx wrote: That's Soekris I suppose. Never heard of them, but it looks mighty interesting. Cheers, BC On 05/10/12 13:35, John Novack wrote: I use HP Thin Clients with AstLinux installed. HP 5720's are available on eBay for not much money, or there are many small boards available new if you don't or can't use used. 10 watts, no fan, no HD Not sure what might be available in your part of the world, but there are Sockris and ALIX flash based boards. AstLinux has special configurations for these. I have 20-30 AstLinux on thin clients working without a belch on a private collectors network John Novack -- _ -- 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] looking for solid state like PC suitable for Asterisk
These prices are simply stunning ... Little can go wrong with the CPU's speed. awesome, BC On 05/10/12 14:32, Terry Brummell wrote: This thread may interest you. Add a SSD and RAM and you're good to go! http://pbxinaflash.com/community/index.php?threads/diy-piaf2-server-200. 12460/ -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Novack Sent: Thursday, May 10, 2012 8:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] looking for solid state like PC suitable for Asterisk Correct. I have never been accused of being a good speller! JN Bart Coninckx wrote: That's Soekris I suppose. Never heard of them, but it looks mighty interesting. Cheers, BC On 05/10/12 13:35, John Novack wrote: I use HP Thin Clients with AstLinux installed. HP 5720's are available on eBay for not much money, or there are many small boards available new if you don't or can't use used. 10 watts, no fan, no HD Not sure what might be available in your part of the world, but there are Sockris and ALIX flash based boards. AstLinux has special configurations for these. I have 20-30 AstLinux on thin clients working without a belch on a private collectors network John Novack Bart Coninckx wrote: Hi all, for smaller (or maybe even bigger) sites I'm looking for a smaller, appliance-type like PC, preferably solid state and fanless PC. Since it's only going to run Asterisk for a couple of extensions I don't think CPU and RAM need to be maxed out. Does anyone have inspiration/experience for/about such a model? thx!! BC -- _ -- 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 -- _ -- 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 -- _ -- 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] looking for solid state like PC suitable for Asterisk
Tim, looked at these briefly, they all seemed pre-installed, correct? Is reinstallation with, let's say, CentOS possible? thx, BC On 05/10/12 14:39, Tim Nelson wrote: Bart Coninckx wrote: Hi all, for smaller (or maybe even bigger) sites I'm looking for a smaller, appliance-type like PC, preferably solid state and fanless PC. Since it's only going to run Asterisk for a couple of extensions I don't think CPU and RAM need to be maxed out. Does anyone have inspiration/experience for/about such a model? Have a look at the Blackbochs SBC***. It is small, low power, plenty of storage options, built in analog telephony ports, etc: http://www.rockbochs.com/products/blackbochs-sbc --Tim ***Yes, I'm affiliated with the product/company, but it is on topic for this discussion. My apologies if this offends anyone. -- _ -- 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 -- _ -- 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] looking for solid state like PC suitable for Asterisk
On 05/10/12 18:38, Kevin P. Fleming wrote: On 05/10/2012 03:49 AM, Bart Coninckx wrote: Hi all, for smaller (or maybe even bigger) sites I'm looking for a smaller, appliance-type like PC, preferably solid state and fanless PC. Since it's only going to run Asterisk for a couple of extensions I don't think CPU and RAM need to be maxed out. Just a small comment here... I really find it quite humorous that people use 'solid state' to mean 'no moving parts'. All of the parts of my computers that move are still composed of solid materials, and the electrical currents involved in them still move through solid materials :-) Yeah, well, have you seen crawling any bugs in software lately? Still they are called bugs ... :-s -- _ -- 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] Belgian BRI (euroisdn): what to use for a B410P
Hi, I'm experiencing difficulties to get a B410P running with Asterisk 10.3.1 and DAHDI 2.6.1. Am I supposed to use DAHDI for this card and ISDN BRI for my country (Belgium)? thx, BC -- _ -- 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] Belgian BRI (euroisdn): what to use for a B410P
Hi Khalid, my setup is almost identical except for loadzone = be defaultzone = be (obviously) and in chan_dahdi.conf: [isdn4] signaling = bri_cpe_ptmp switchtype = euroisdn group = 2 context = isdn dahdichan = 10,11 this results into: ERROR[1021]: chan_dahdi.c:14182 dahdi_pri_error: PRI Span: 4 Unable to receive TEI from network in state 2(Assign awaiting TEI)! and after trying to call: May 9 21:25:51] NOTICE[1021]: chan_dahdi.c:3136 my_handle_dchan_exception: PRI got event: Alarm (4) on D-channel of span 4 [May 9 21:25:51] WARNING[1022]: chan_dahdi.c:7895 handle_alarms: Detected alarm on channel 10: Red Alarm [May 9 21:25:51] WARNING[1022]: chan_dahdi.c:7895 handle_alarms: Detected alarm on channel 11: Red Alarm [May 9 21:25:53] ERROR[1021]: chan_dahdi.c:14182 dahdi_pri_error: PRI Span: 4 Unable to receive TEI from network in state 3(Establish awaiting TEI)! [May 9 21:25:53] NOTICE[1021]: chan_dahdi.c:3136 my_handle_dchan_exception: PRI got event: No more alarm (5) on D-channel of span 4 [May 9 21:25:53] NOTICE[1022]: chan_dahdi.c:3826 handle_clear_alarms: Alarm cleared on channel 10 [May 9 21:25:53] NOTICE[1022]: chan_dahdi.c:3826 handle_clear_alarms: Alarm cleared on channel 11 An old Asterisk 1.4 installation with mISDN connected to the same line used ptp, not ptmp. When I used bri_cpe_ptp, I get even more problems on the console however (every second). This is why I wondered if DAHDI is supposed to work over here, cheers, BC On 05/09/12 21:14, khalid touati wrote: Hi Bart, here is a working configuration in Netherlands: /etc/dahdi/system.conf: span = 1,1,0,ccs,ami bchan = 1,2 hardhdlc = 3 span = 2,1,0,ccs,ami bchan = 4,5 hardhdlc = 6 span = 3,1,0,ccs,ami bchan = 7,8 hardhdlc = 9 span = 4,1,0,ccs,ami bchan = 10,11 hardhdlc = 12 loadzone= nl defaultzone= nl (of course change those to your country initials) /etc/asterisk/chan_dahdi.conf: group = 1 signalling = bri_cpe_ptmp switchtype = euroisdn context = mainmenu echocancel = yes channel = 1,2,4,5,7,8,10,11 I am not using dahdi-channels, hope it helps! On Wed, May 9, 2012 at 1:59 PM, Bart Coninckx bart.conin...@telenet.be mailto:bart.conin...@telenet.be wrote: Hi, I'm experiencing difficulties to get a B410P running with Asterisk 10.3.1 and DAHDI 2.6.1. Am I supposed to use DAHDI for this card and ISDN BRI for my country (Belgium)? thx, BC -- _ -- 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 -- Khalid Touati Network Administrator at Endosoft, LLC CCNA -- _ -- 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 -- _ -- 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] Belgian BRI (euroisdn): what to use for a B410P
Right you are, but when using bri_cpe I get: [May 9 23:08:45] WARNING[2775]: sig_pri.c:6969 pri_dchannel: PRI Error on span 4: Received MDL/TEI managemement message, but configured for mode other than PTMP! This repeats itself every second. The bri_cpe_ptmp settings seems to give the least troubles, but no calling possible, BC On 05/09/12 22:10, Richard Mudgett wrote: Hi Khalid, my setup is almost identical except for loadzone = be defaultzone = be (obviously) and in chan_dahdi.conf: [isdn4] signaling = bri_cpe_ptmp switchtype = euroisdn group = 2 context = isdn dahdichan = 10,11 this results into: ERROR[1021]: chan_dahdi.c:14182 dahdi_pri_error: PRI Span: 4 Unable to receive TEI from network in state 2(Assign awaiting TEI)! and after trying to call: May 9 21:25:51] NOTICE[1021]: chan_dahdi.c:3136 my_handle_dchan_exception: PRI got event: Alarm (4) on D-channel of span 4 [May 9 21:25:51] WARNING[1022]: chan_dahdi.c:7895 handle_alarms: Detected alarm on channel 10: Red Alarm [May 9 21:25:51] WARNING[1022]: chan_dahdi.c:7895 handle_alarms: Detected alarm on channel 11: Red Alarm [May 9 21:25:53] ERROR[1021]: chan_dahdi.c:14182 dahdi_pri_error: PRI Span: 4 Unable to receive TEI from network in state 3(Establish awaiting TEI)! [May 9 21:25:53] NOTICE[1021]: chan_dahdi.c:3136 my_handle_dchan_exception: PRI got event: No more alarm (5) on D-channel of span 4 [May 9 21:25:53] NOTICE[1022]: chan_dahdi.c:3826 handle_clear_alarms: Alarm cleared on channel 10 [May 9 21:25:53] NOTICE[1022]: chan_dahdi.c:3826 handle_clear_alarms: Alarm cleared on channel 11 An old Asterisk 1.4 installation with mISDN connected to the same line used ptp, not ptmp. When I used bri_cpe_ptp, I get even more problems on the console however (every second). bri_cpe_ptp is not a valid value for the signaling parameter. From chan_dahdi.conf.sample: ; bri_cpe:BRI PTP signalling, CPE side ; bri_net:BRI PTP signalling, Network side ; bri_cpe_ptmp: BRI PTMP signalling, CPE side ; bri_net_ptmp: BRI PTMP signalling, Network side Richard -- _ -- 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 -- _ -- 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] using Wifi smartphones as SIP clients
All, has anyone any experience in using Wifi smartphones as SIP clients? Does this work properly? What models/brands are optimal for this (in terms of ease of use, battery life etc)? Thx!! B. -- _ -- 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] using Wifi smartphones as SIP clients
Hi, thx! The Snom M9 does not look like a Wifi phone however, nor is it a smarthpone. In order to use that I would have to use access points that can handle both Wifi and DECT. Astraa seems to have (an expensive) one. SIPDroid crossed my mind. You say they work OK but the latency is problematic? Is the experience as a whole then not poblematic? thx! B. On 05/07/12 10:29, Mitul Limbani wrote: Used the Snom M9 Wifi DECT phones, they work like charm. SIPDroid on Android phones work good too, however latency is going to be nightmare for u in softphone n wifi kinda scenario. Use good quality Access Points like Ruckus Wireless. Mitul On May 7, 2012 1:55 PM, Bart Coninckx bart.conin...@telenet.be mailto:bart.conin...@telenet.be wrote: All, has anyone any experience in using Wifi smartphones as SIP clients? Does this work properly? What models/brands are optimal for this (in terms of ease of use, battery life etc)? Thx!! B. -- _ -- 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 -- _ -- 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 -- _ -- 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] using Wifi smartphones as SIP clients
Benny, very useful, thank you. So, in short, at this stage it's best to go DECT for wireless and if DECT and Wifi need to be combined (because both types of devices exist in the organization), it's preferable to go to access points that offer both networks. Correct? thx, BC On 05/07/12 10:57, Benny Amorsen wrote: Bart Coninckxbart.conin...@telenet.be writes: has anyone any experience in using Wifi smartphones as SIP clients? Yes... Does this work properly? It works nicely for home use for power users who can accept the odd lost call and know how to restart the app or the phone when something goes wrong. Unfortunately I haven't found anything so far which works for business use. The largest problem is that smartphones can't afford (battery-wise) to check for wifi connectivity all the time. If the phone loses connection to the wifi, it often takes more than a minute before it is ready to receive calls again. What models/brands are optimal for this (in terms of ease of use, battery life etc)? iPhone, Android, and Symbian are about equally troublesome. /Benny -- _ -- 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 -- _ -- 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] using Wifi smartphones as SIP clients
What about phones like the Unidata WPU-7800 ( http://www.udcsystems.com/product/wpu7800.php) ? Does anyone have experience with those? Would these also suffer from connection losses? thx, BC On 05/07/12 10:57, Benny Amorsen wrote: Bart Coninckxbart.conin...@telenet.be writes: has anyone any experience in using Wifi smartphones as SIP clients? Yes... Does this work properly? It works nicely for home use for power users who can accept the odd lost call and know how to restart the app or the phone when something goes wrong. Unfortunately I haven't found anything so far which works for business use. The largest problem is that smartphones can't afford (battery-wise) to check for wifi connectivity all the time. If the phone loses connection to the wifi, it often takes more than a minute before it is ready to receive calls again. What models/brands are optimal for this (in terms of ease of use, battery life etc)? iPhone, Android, and Symbian are about equally troublesome. /Benny -- _ -- 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 -- _ -- 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] using Wifi smartphones as SIP clients
The phone I pointed to is a WiFi phone, the M9 is a DECT phone. Different animal. The question is regarding using WiFi as the WiFi infrastructure is already in place. I understand smartphones is not a good option, but what about these WiFi SIP phones? thx! B. On 05/07/12 12:21, Mitul Limbani wrote: Well in that case, you might seriously want to look @ M9 they are cost effective n definitely work. Most of these cell phone type looking phones have a serious battery drainage problem. Smartphones really have a long way to understand how to preserve battery and deliver one thing (i.e. calls) very effectively, infact a Multi Function device (MFD) like smart phone has this issue. Regards, Mitul Limbani, Chief Architech Founder, Enterux Solutions Pvt. Ltd. 110 Reena Complex, Opp. Nathani Steel, Vidyavihar (W), Mumbai - 400 086. India http://www.enterux.com/ http://www.entvoice.com/ email: mi...@enterux.in mailto:mi...@enterux.in DID: +91-22-61447605 Cell: +91-9820332422 On Mon, May 7, 2012 at 3:33 PM, Bart Coninckx bart.conin...@telenet.be mailto:bart.conin...@telenet.be wrote: What about phones like the Unidata WPU-7800 ( http://www.udcsystems.com/product/wpu7800.php) ? Does anyone have experience with those? Would these also suffer from connection losses? thx, BC On 05/07/12 10:57, Benny Amorsen wrote: Bart Coninckxbart.conin...@telenet.be mailto:bart.conin...@telenet.be writes: has anyone any experience in using Wifi smartphones as SIP clients? Yes... Does this work properly? It works nicely for home use for power users who can accept the odd lost call and know how to restart the app or the phone when something goes wrong. Unfortunately I haven't found anything so far which works for business use. The largest problem is that smartphones can't afford (battery-wise) to check for wifi connectivity all the time. If the phone loses connection to the wifi, it often takes more than a minute before it is ready to receive calls again. What models/brands are optimal for this (in terms of ease of use, battery life etc)? iPhone, Android, and Symbian are about equally troublesome. /Benny -- _ -- Bandwidth and Colocation Provided byhttp://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 -- _ -- 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 -- _ -- 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 -- _ -- 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] using Wifi smartphones as SIP clients
On 05/07/12 13:04, Benny Amorsen wrote: man, 07 05 2012 kl. 12:03 +0200, skrev Bart Coninckx: What about phones like the Unidata WPU-7800 ( http://www.udcsystems.com/product/wpu7800.php) ? Does anyone have experience with those? Would these also suffer from connection losses? I don't know that particular phone, but dedicated wifi phones definitely CAN work for professional use. E.g. ASCOM phones work absolutely great, they are just expensive. /Benny Does anyone know how things have evolved regarding sessions/call handover in wifi? I remember reading this document in 2008: http://www.abpsec.com/blog/the-dect-versus-wlan-wifi-debate/ http://www.abpsec.com/blog/the-dect-versus-wlan-wifi-debate/ which basically states that it's a bad idea as interruptions of 70 ms are involved. Is this still a challenge? Rgds, BC http://www.abpsec.com/blog/the-dect-versus-wlan-wifi-debate/ -- _ -- 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] using Wifi smartphones as SIP clients
On 05/07/12 13:50, giovanni.v wrote: I thing smartfones simply lack a business grade softhone implementation. WiFi SIP phones share some problem with smartphones: battery runtime and, most important, roaming and handover. In WiFi network handling roaming/handover is up to the client, this is the same kind of problem that arises in mobile WiFi POS systems. Compared to DECT standard which implements roaming/handover this is a major drawback. Implementing a proper roaming/handover in WiFi networks, using wireless controllers and suitable access points, is very expensive. Where such expensive WiFi infrastructure have to be built only to properly serve wireless phones a DECT multicell system is definitely a winner considering coverage, features and battery runtime. -- _ -- 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 Giovanni, I think you're completely right. Even to this day, it seems that Wifi is not ready for voice, except while investing a lot of time/money. DECT it is, BC -- _ -- 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] medooze MCU versus confbridge
Hi, Looking to do video conferencing with Asterisk and after some research I noticed there's mainly the new confbridge application in Asterisk 10 or there's the Medooze MCU software. I'm not sure as how they compare feature-wise. I get the impression the video support in confbridge is rather limited at this point, seeing you can only see one party at the time (right?). Would Medooze MCU be better at this? Cheers, BC -- _ -- 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] elegant way to change codec whn failing over to another line
Hi all, After having done some successful tests with 3G in combination with the G729a codec, I plan to use this as a failover path for when the main internet connection goes down. However, on this usual connection, G711a is used. I could have the script that monitors the main line also sed the sip.conf for changing to another codec when a failover to 3G is done (and have it do a asterisk -rx 'reload'), but I'm wondering if there's a more elegant way to change the codec sort of on the fly. Thank you! B. -- _ -- 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] Blind transfer issue on Asterisk 1.8.4.2
Hi all, when doing a blind transfer using the keys defined in features.conf, we hear a confirmation of the attempt to blindly transfer, followed by an invalid extension message. The console says this: [Jun 4 22:30:31] VERBOSE[11301] res_musiconhold.c: -- Started music on hold, class 'default', on SIP/570-0006 [Jun 4 22:30:31] VERBOSE[11301] file.c: -- SIP/518-0005 Playing 'pbx-transfer.gsm' (language 'nl') [Jun 4 22:30:32] WARNING[11301] features.c: Extension '53' does not exist in context 'transfer_context,570,1' [Jun 4 22:30:32] VERBOSE[11301] file.c: -- SIP/518-0005 Playing 'pbx-invalid.gsm' (language 'nl') [Jun 4 22:30:34] VERBOSE[11301] res_musiconhold.c: -- Stopped music on hold on SIP/570-0006 Mind you, the entered extension is 531, so it seems part of the entry is cut off. Sometimes it shows just 5. It is as if featuredigittimeout (set to 2000) is not taken into account. Thx!! B. -- _ -- 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] calls dropped after 20 seconds in a non NAT situation
Hi all, I have a rather old 1.4 installation that recently was connected to a new network via an IPSEC tunnel. No NAT-ing is involved anywhere (I've seen posts about the same phenomenon but with NAT). It first the phones on the PBX network did not get the audio of the phones on the remote network, but that was fixed by removing an externip entry out of sip.conf. What I have now, is that all calls are cut after 20 seconds. The log files say: Maximum retries exceeded on transmission 629be818-f3721...@192.168.10.104 for seqno 101 (Critical Response) It seems Asterisk gives up after 20 seconds to wait for some answer from the phone. What do you guys think is the way to proceed here: change the configs somewhere or upgrade Asterisk? Thank you, Bart -- _ -- 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] Where are phone registrations kept?
Hi, I've built an Asterisk HA cluster by means of heartbeat and drbd. The following folders are stored on shared storage and referred to by means of symbolic links: /etc/asterisk /var/lib/asterisk /usr/lib/asterisk /var/spool/asterisk /var/log/asterisk I was under the impression that phone registrations were stored in /var/lib/asterisk/astdb and as such preserved when failing over. But when failing over I need to restart the phones in order to have them work with the newly actived asterisk node. This seems to point to the fact that phone registrations are stored elsewhere or are forgotten when Asterisk is restarted, but the latter seems not really true anyway. So, what is going wrong here? Were are the registrations stored? Or should I build in something to have the phones rebooted when I failover? Thank you, Bart ___ -- 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] Outgoing CallerID for KPN in Belgium
Hi,
I'm using a ISDN-30 E1 line from KPN Belgium.
The challenge is to get a correct CallerID on outgoing lines.
When I put this in my dialplan:
exten = _0.,1,Set(TEMPVAR=${CALLERID(num):1})
exten = _0.,2,Set(CALLERID(num)=144622${TEMPVAR})
exten = _0.,3,NoOp(${CALLERID(num)})
exten = _0.,4,Dial(Zap/g1/${EXTEN:1},,)
The resulting CallerID is accepted by the telco, but on phones it shows for
instance as:
+14462241, whereas it should be +3214462241. So it seems the telco adds a +.
I've tried to then use:
exten = _0.,2,Set(CALLERID(num)=32144622${TEMPVAR})
but the telco seems not to accept this since it sends the general CallerID out.
Any clues on what I need to change to get this working? Is it something in
zapata.conf? Is it related to nationalprefix and internationalprefix?
Thank you!
B.
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[asterisk-users] Consequent Dial commands not ringing
Hi, when I have this in my dialplan in order to get a cascade for incomming calls exten = 611,1,Dial(SIP/611,10) exten = 611,2,Dial(SIP/607,60) exten = 611,3,Dial(SIP/620,60) exten = 611,4,Hangup() I get a ringing tone for the first Dial command, but the others produce silence, even when I use the r option. I fixed this temporarely by adding music on hold (m option), but I'd like to know why this doesn't work. Anyone any clue? Thank you! B. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] transfers only work when voicemail enabled
Hi all, when enabling blind and attended transfers in features.conf, these only seem to work when I enable voicemail for a particular user. How can this be? Can I have transferrring without voicemail? Using Asterisk 1.4 by the way. Thank you! Bart ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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
Re: [asterisk-users] transfers only work when voicemail enabled
Hi all, when enabling blind and attended transfers in features.conf, these only seem to work when I enable voicemail for a particular user. How can this be? Can I have transferrring without voicemail? Using Asterisk 1.4 by the way. Thank you! Bart I think some clarification is necessary here. What do you mean by enable voicemail? Do you mean that you add a Voicemail() application call to the Dialplan? I don't see how that could make a difference regarding whether transfers are allowed. Transferring should be allowable just by adding either the 't' or 'T' flags to the options for Dial(). Mark Michelson Hi Mark, yes, I'm sorry, I should have been more clear about this: I'm referring to the hasvoicemail setting in the users.conf file. When this is set to no, transferring does not work. When set to yes, it does. Both t and T are added to my Dial commands, thank you, Bart ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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
