[asterisk-users] Anybody used Asterfax?
Trying to install asterfax-1.1-freeb2.i386.rpm, I get following error. How can I get rid of it. Installing jreInstalling libtiffInstalling ghostscriptInstalling XvfbInstalling openoffice.orgInstalling spandspInstalling spandsp0.0.3 Spandsp did not install correctly.error: %post(asterfax-1.1-freeb2.i386) scriptlet failed, exit status 1 ___ --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] skype and SIP hardware for linux
I'm looking at the http://support.a-link.com/phonemate/IPU1.htm phone because it works with Skype (from Linux), but can do SIP, too. Not necessarily asterisk related, but possibly. My networking situation might require IAX if I'm running Linux and want to use SIP, I'm not certain (Skype works fine). Putting that unknown aside for the moment, how does this phone work under either Skype or as a SIP phone? The information I have on the driver, skypemate, is a bit sketchy. According to A-Link, the phone complies with SIP, http://www.a-link.com/us_us/IPU1.html, but the details are sketchy. No information is provided as to the interface for configuring SIP. The user manual, http://support.a-link.com/phonemate/Manual/IPU1manual_for_Linux.pdf, details using Skype but not SIP. Any user experience with this phone? For instance, has anyone used it with gizmo project or free world dialup, or even Skype? thanks, Thufir ___ --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 FXO Digium Card for Analog line
Hi For add a analog line to my asterisk, i want add a Dgium Fxo card. but i want know a small information: The quality of the call are good or not with this type of card ? Thanks for your returns ___ --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] skype and SIP hardware for linux
It''s a USB Sound card / keypad / display, not a phone. It contols a softphone on the PC it's plugged into - they say it works with XLite - the SIP setup will be done in Xlite, not the 'phone'. Peter On 05/11/06, Thufir [EMAIL PROTECTED] wrote: I'm looking at the http://support.a-link.com/phonemate/IPU1.htm phone because it works with Skype (from Linux), but can do SIP, too. Not necessarily asterisk related, but possibly. My networking situation might require IAX if I'm running Linux and want to use SIP, I'm not certain (Skype works fine). Putting that unknown aside for the moment, how does this phone work under either Skype or as a SIP phone? The information I have on the driver, skypemate, is a bit sketchy. According to A-Link, the phone complies with SIP, http://www.a-link.com/us_us/IPU1.html, but the details are sketchy. No information is provided as to the interface for configuring SIP. The user manual, http://support.a-link.com/phonemate/Manual/IPU1manual_for_Linux.pdf, details using Skype but not SIP. Any user experience with this phone? For instance, has anyone used it with gizmo project or free world dialup, or even Skype? thanks, Thufir ___ --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 -- Peter Bowyer Email: [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] Re: Audio goes one way during the call for a few seconds. Is it RTP, NAT, dyndns, or what it is?
Tried inspecting packet dumps with an analyser like Wireshark (ex Ethereal)? They can prove very useful when troubleshooting issues like these. On 11/5/06, Zeeshan Zakaria [EMAIL PROTECTED] wrote: Seems likes I am the only person in Asterisk world with this problem, everybody else is fine with audio. ___ --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] Reading Voicemail Config from MySQL
Hi all, I have been trying to get my asterisk (v1.2.10) to lookup voicemail config data from my mysql database as opposed to voicemail.conf + sip.conf for my users. Users register with SER and get passed through to asterisk when they dial out. I followed the instructions as per http://www.voip-info.org/wiki/view/Asterisk+voicemail+database so basically I have 1) Build asterisk-addons-1.2.5 and added the USE_MYSQL_VM_INTERFACE=1 to the asterisk/apps folder and built asterisk again 2) I configured my voicemail.conf appropriately: dbuser=username dbpass=password dbhost=localhost dbname=asterisk_vm and have a database called asterisk_vm with a table called users with the fields needed by asterisk 3) Populated my database with some values When I try to leave a voicemail, I get the following error in the CLI: -- Called [EMAIL PROTECTED] -- SIP/myserver-08c5ef80 is ringing -- Nobody picked up in 2 ms -- Executing VoiceMail(SIP/myserver-08c731a0, u7521) in new stack Nov 5 13:30:55 WARNING[18146]: app_voicemail.c:2412 leave_voicemail: No entry in voicemail config file for '7521' I was able (thanks to some guys on this list) to get my Prepaid application to read from the database but voicemail won't. Please help me with this error. 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] call transfer problem
Can anyone help with the following problem please? 1) On a receptionist's phone (Snom 360 latest firmware), a call is answered. 2) While on this call a second call comes to the phone but she does not answer it. 3) The receptionist makes an attended transfer placing the first caller on hold and dialing an extension internally, but the internal party is not willing to pick up the call so she hangs up the internal call. The second call remains unanswered. 4) The receptionist now has two blinking lights on the phone for the original call and the new call is still unanswered. 5) If either button is pressed, the call that is picked up is the second call and the first call remains on hold ... anyone know why this is? The funny thing is if a blind transfer or an attended transfer that is accepted by the internal party is performed, the functions work correctly. Regards, Colin ___ --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 FXO Digium Card for Analog line
Yes. You can use the TDM400P. It should do the trick. Make sure to look in to echo cancelation. - Original Message - From: Noc Phibee [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Sunday, November 05, 2006 11:51 AM Subject: [asterisk-users] Asterisk and FXO Digium Card for Analog line Hi For add a analog line to my asterisk, i want add a Dgium Fxo card. but i want know a small information: The quality of the call are good or not with this type of card ? Thanks for your returns ___ --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] Only one out of 10 remote extensions expiring registry
I experimented with my router, and setup DHCP Lease time to expire every minute. After doing this, my phone started to register every hour. But in the above example, on same phone, one account registers every minute and other account every other minute. This is how it is setup in the phone. But CLI doesn't have to show this everytime. There is something else going on with that network, which is changng ports on every registry, which is causing the messages to appear on the CLI. ___ --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] Re: skype and SIP hardware for linux
On Sun, 05 Nov 2006 09:53:52 +, Peter Bowyer wrote: It''s a USB Sound card / keypad / display, not a phone. It contols a softphone on the PC it's plugged into - they say it works with XLite - the SIP setup will be done in Xlite, not the 'phone'. Peter On 05/11/06, Thufir [EMAIL PROTECTED] wrote: I'm looking at the http://support.a-link.com/phonemate/IPU1.htm phone because it works with Skype (from Linux), but can do SIP, too. [...] Did I miss that info on Xlite? Sounds like this might work under linux, at least for xlite...? thanks, Thufir ___ --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] Only one out of 10 remote extensions expiring registry
Sorry, just a correction. DHCP lease time setup to expire every hour, not every minute. On 11/5/06, Zeeshan Zakaria [EMAIL PROTECTED] wrote: I experimented with my router, and setup DHCP Lease time to expire every minute. After doing this, my phone started to register every hour. But in the above example, on same phone, one account registers every minute and other account every other minute. This is how it is setup in the phone. But CLI doesn't have to show this everytime. There is something else going on with that network, which is changng ports on every registry, which is causing the messages to appear on the CLI. -- Zeeshan A Zakaria ___ --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] Re: skype and SIP hardware for linux
It seems that xlite doesn't support IAX? Too bad. While xlite does, apparently, run under linux it's not clear to me whether or not the a-link device will work with the linux version of xlite. -Thufir ___ --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] Re: skype and SIP hardware for linux
On Sun, 05 Nov 2006 09:53:52 +, Peter Bowyer wrote: It''s a USB Sound card / keypad / display, not a phone. It contols a softphone on the PC it's plugged into - they say it works with XLite - the SIP setup will be done in Xlite, not the 'phone'. [...] Heh, I did miss it. Yes, for windows, it specifies X-Lite software. That x-Lite isn't mentioned for Linux implies that it'll only work for windows. Curious, but not unusual, state of affairs. In any event, x-Lite doesn't support IAX, which I require. -Thufir ___ --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] Re: skype and SIP hardware for linux
I downloaded a softphone called kiax last night. Its working great. I was real tired then so I dont remember where I got it from. Hope that helps. (and its open source as well as they give you the source files for it :) ) - Original Message - From: Thufir [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Sunday, November 05, 2006 3:09 PM Subject: [asterisk-users] Re: skype and SIP hardware for linux On Sun, 05 Nov 2006 09:53:52 +, Peter Bowyer wrote: It''s a USB Sound card / keypad / display, not a phone. It contols a softphone on the PC it's plugged into - they say it works with XLite - the SIP setup will be done in Xlite, not the 'phone'. [...] Heh, I did miss it. Yes, for windows, it specifies X-Lite software. That x-Lite isn't mentioned for Linux implies that it'll only work for windows. Curious, but not unusual, state of affairs. In any event, x-Lite doesn't support IAX, which I require. -Thufir ___ --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] Audio goes one way during the call for a few seconds. Is it RTP, NAT, dyndns, or what it is?
Sounds like a bad Internet connection messing with the IAX jitterbuffer. Try running ping plotter from your location to your host, and see if it goes 'red'/down. On 11/3/06, Zeeshan Zakaria [EMAIL PROTECTED] wrote: Hi everybody, I finally want to get rid of 1-way audio problem. Please help me here. I have 3 scenarios. 1. Audio is always one way. Caller who dialed can't listen the called party but called party can listen him. In this scenatio Asterisk is on dynamic IP with dyndns FQDN. sip.conf has externip = abc.dyndns.org and localnet = xxx.xxx.xxx.xxx entry. Trunk and extensions are SIP. Where is the voice getting lost from the called party? NAT is there but Asterisk is in DMZ. 2. Conversation is going fine when all of a sudden you realize that other parth has started saying 'hello, hello' because they can't hear you. But you are hearing them loud and clear. Now you are on static IP with dyndns FQDN. externip and localnet settings in sip.conf (do we need them for static IP?). After about 15-20 seconds, again 2-way converstaion is established again. IAX trunk, SIP extension, no NAT. 3. Conversation goes one way for 15-20 sec during the most important part of the conversation (Murphy's Law). You are on a static IP with no dyndns enrty. Trunk is ZAP on PRI, extensions SIP. NAT present but router properly configures for port forwarding. externip and localnet settings present in sip.conf Is think may be due to some reason RTP stream gets lost, routed to wrong IP. But why would this happen during a call and how to stop it from happening. Or is there some other reason behind this? Does dyndns setting have to do anything with this problem? How can I overcome this problem once and forever. -- Zeeshan A Zakaria ___ --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] Hang up on SIP calls if connected to long
Use the set absolute timeout option on all inbound calls, and then reset that time to something really high when it connects to a sip phone. On 11/5/06, Dovid B [EMAIL PROTECTED] wrote: Is there any way to run a script and or agi that looks on asterisk and looks for calls that are connected longer X amounth of time and hang up on them and or look for calls that have not been bridged with a client within X amount of time and dump the call ? 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 ___ --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] Definity Asterisk CallerID Issue
I am hoping someone could shed some light and point me in the right direction? Im attempting to get callerid work between an Avaya Definity PBX and Asterisk via TE110P connected via T1/PRI Crossover PRI. From the Definity side Ive searched endlessly and came with an example which we modeled as close as we can, but still no luck. While doing PRI intense debug span 1 in I see a couple interesting messages but have yet to come up with meaningful knowledge about them. Ive tried decoded the setup message but dont know what Im really looking at. It appears in the decode that the calling party number or name are not being sent but as I mentioned I dont know what I am really looking at. I wondering if these error messages have any thing to do with Asterisk not knowing what to do with what the Definity is sending? Feel free to contact me offlist. Any assistance is greatly appreciated. -CP !! Unknown IE 1544 (len = 6) !! Unknown IE 8 (cs6, Unknown Information Element) Progress Description: Calling equipment is non-ISDN TON: International Number xxx*CLI [ 02 01 d4 d2 08 02 0a e6 05 04 03 90 90 a2 18 03 a1 83 8b 1e 02 81 83 70 05 91 34 33 38 39 96 08 04 d0 35 30 80 ] Protocol Discriminator: Q.931 (8) len=33 Call Ref: len= 2 (reference 2790/0xAE6) (Originator) Message type: SETUP (5) [04 03 90 90 a2] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: 3.1kHz audio (16) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: u-Law (34) [18 03 a1 83 8b] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Preferred Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 11 ] [1e 02 81 83] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Progress Description: Calling equipment is non-ISDN. (3) ] [70 05 91 34 33 38 39] Called Number (len= 7) [ Ext: 1 TON: International Number (1) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '4389' ] [96] Locking Shift (len=01): Requested codeset 6 [08 04 d0 35 30 80] !! Unknown IE 1544 (len = 6) !! Unknown IE 8 (cs6, Unknown Information Element) Sending Receiver Ready (107) [ 02 01 01 d6 ] ___ --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] SPA3k wired to PAP2 for echo testing
I'm in the US and had bad echo problems with the SPA3K and the latest firmware. I was under the impression that the echo was due to my long cable run to the CO ~15000'. Changing the impedance (900 ohms) would help for a while, but after a few days the echo came back. If I rebooted the SPA3K the echo would go away for a while, but always came back. Assuming a software problem, I back-revved to an earlier F/W version. This seemed to help, but was not a cure. It did not pass muster with my wife. BTW: I did not experience echo on SIP calls through my ITSP or locally w/in my network. I've seen some chatter about a Global option helping, but never tried it. I gave up and switched to a TDM11B. There was also some talk about having the earpiece volume up too high such that the phone's microphone picked up the sidetone and caused echo. I did have better results when the phone's volume was turned down, but I the SPA3k echo problem was never cured Bob... James Harper wrote: In my seemingly endless search for the cause of echo on my SPA3000, I wired it up in the following configuration: Analogue Handset -- (FXS)SPA3000(FXO) -- PAP2 And set the Line1 dialplan on the SPA3k to '(:@gw0S0)' which means that as soon as I pick up the handset I get linked straight through to the PAP2, which gives me dialtone. Even in this configuration, with my impedance settings set to the Australian standard of 220+820||120nf, and the PSTN and PAP2 echo cancellers enabled (or not, and all combinations of) I get local echo as soon as I pick up the handset (I hear my voice bounced back to me). Surely this shouldn't be??? There is no hybrid involved at all! If anyone on this list with a SPA3k (that doesn't have any local echo problems on the PSTN port) and an ATA with a FXS port, could they please try the above setup and post the results (including SPA3k hardware and firmware versions, and the ATA used)? I wonder if there is a problem with some versions of the SPA3k where there is some sort of inbalance on the PSTN port that causes echo right there rather than further down the line? Thanks James ___ --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
Free PBX, was - Re: [asterisk-users] best gui
Tom Vile[EMAIL PROTECTED] Wrote on: 11/4/2006 8:45 PM: He is not talking about Trixbox but FreePBX and his assumption is correct. Just load Asterisk and then FreePBX later. Thanks. I see that 2.2.x is spoken about, but 2.1.3 is the latest that sourceforge offers. Is 2.2.x out or still in a closed beta state? If in a release state, where can I get it? Or should one wait? If there was an obvious way to get it, it was not obvious enough for me. joe a ___ --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] Voicemail.conf multi languages
Hello, Im a student of the school of engineer of Yverdon Switzerland and Im working for my project of diploma (VoIP-Asterisk) Im wondering if it is possible to have multi languages email with the voicemail.conf. I wish to set the emailbody/emailsubject relatively to the user language of the mailbox? Any advice or idea will be appreciated! Thanks Salim ___ --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] Newbie questions about Voice mail
Dean Thanks for responding. I have added more info in your reply. Right now we do not operate our own PBX or voice mail system. All of the service is provided by the telco. As a start I was wondering if I could simply put in asterisk to do just voicemail. I am assuming the telco can configure all the phone to automatically call forward to asterisk on no answer. If asterisk can handle this I am assuming that a user would just call some number to retiev voice mail. They would lose the call waiting light on their phone so the email notification of a voice mail would be necessary. ..Brian On Sun, 5 Nov 2006, Dean Collins wrote: Date: Sun, 5 Nov 2006 00:04:36 -0500 From: Dean Collins [EMAIL PROTECTED] To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: RE: [asterisk-users] Newbie questions about Voice mail Hi Brian, I'm sure some other people will give you better answers but quick answers are; 1/ Depends on volume of message leaving/collection, is it in a single location? Multiple locations with multiple time zones? Two locations, one time zone. Could be two different systems since they are in two different cites connected by a 1G connection. Estimate the number of voicemails left per hour and reply with this. There are about 3000 phones. Some are busier than others os lets say 2 messages per phone per day. An they are mostly in the peak work day so lets say 500 per hour and the average length is 30 seconds. 2/ retrieve either via deliver to email or dial in to a number to collect voicemail via phone (or collect and play via a website) What does the conversion and how does one handle bulk updates? to users? How much control does the user have? How are the retrieving their voicemail now? Do you want to replicate this for ease of replacement as near as possible? Right now we are using the voice mail service provided by the teclo and are spending $0.06 per minute. The user connects to the voice mail by dialing *99 and entering a password on their office set or remoetely by dialing 123-MAIL on any phone (123 is the three digit prefix of their phone number) and then entering their password. They do not have any voice to email service today. If possible I would like to ease the transition if it can be done. Lots of stepswill follow discovery if it can be done. 3/ Not sure what you mean by tie in? How do you match a voice mail box to an email address? Can there be multiple email addresses for one voice mail box? 4/ Sure, how do you have this configured at the moment? Why not replicate voicemail group delivery in the same format? Talkmail is a service provided by the telco where you group a bunch of numbers together so you can send the same message to all of them at the same time. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Saturday, 4 November 2006 11:54 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Newbie questions about Voice mail I am totally ignorant about actually using asterisk for any purpose. I have read some of the docs but not all. I am currently doing a telephone audit for my company and one of the issues is voice mail. We are spending quit a bit of money with our telco for voice mail services and I was wondering about using asterisk as just a voice mail system. We are not quite ready to move to a full VOIP system yet but if I can get this system in place the VOIP will follow. Could I get all 3000 phones (on 2 sites) or a large subset set to have a call forward no-answer feature set to call a number that would be answered by asterisk's voice mail. If so: 1. what hardware do I need to handle 3000 phones? 2. how would users retrieve their voice mail? 3. how does one tie voice mail into an e-mail address? Are their ways to do bulk updates for several thousand new users every year? 4. is there a feature what we call talk mail where you set up a group of phone numbers and send the same message to all of them? Any help would be greatly appreciated. .TIA Brian Kaye ...UNB ___ --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] SPA3k wired to PAP2 for echo testing
On 05/11/06, James Harper [EMAIL PROTECTED] wrote: Even in this configuration, with my impedance settings set to the Australian standard of 220+820||120nf, and the PSTN and PAP2 echo cancellers enabled (or not, and all combinations of) I get local echo as soon as I pick up the handset (I hear my voice bounced back to me). Surely this shouldn't be??? There is no hybrid involved at all! 'course there is. The telephone interface on the one end and the line interface on the other are both 2 wire. Did you have a phone line connected to the other side. Running into an unconnected FXO port is likely to make echo because of the unbalanced impedance. 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
RE: [asterisk-users] Newbie questions about Voice mail
Hi Brian, Uhmmm as it appears you are using a centrex service from your telco (your comment about not having any pabx) I need to ask this question..are you sure that under your current commercial arrangements you are actually allowed to continue to use the telco as your centrex provider but not use them for your voicemail? Also if you decided to use a separate asterisk server for your voicemail service how would calls be transferred to this number? Would the carrier allow you to host and asterisk service off some of your existing centrex extensions? Would this incur a cost or similar. I think for your bosses 'discovery' report the answer would be Yes to can asterisk be used as just a voicemail server Yes to people can operate with the same methods of retrival they currently do Yes to people can also retrieve via additional methods such as web or email And finally yes this will save us money in the longer term at 6c per minute currently. The next step should be 1a/ You boss decides You or someone in your team skill up in asterisk Or Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Sunday, 5 November 2006 3:02 PM To: Dean Collins Cc: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Newbie questions about Voice mail Dean Thanks for responding. I have added more info in your reply. Right now we do not operate our own PBX or voice mail system. All of the service is provided by the telco. As a start I was wondering if I could simply put in asterisk to do just voicemail. I am assuming the telco can configure all the phone to automatically call forward to asterisk on no answer. If asterisk can handle this I am assuming that a user would just call some number to retiev voice mail. They would lose the call waiting light on their phone so the email notification of a voice mail would be necessary. ..Brian On Sun, 5 Nov 2006, Dean Collins wrote: Date: Sun, 5 Nov 2006 00:04:36 -0500 From: Dean Collins [EMAIL PROTECTED] To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: RE: [asterisk-users] Newbie questions about Voice mail Hi Brian, I'm sure some other people will give you better answers but quick answers are; 1/ Depends on volume of message leaving/collection, is it in a single location? Multiple locations with multiple time zones? Two locations, one time zone. Could be two different systems since they are in two different cites connected by a 1G connection. Estimate the number of voicemails left per hour and reply with this. There are about 3000 phones. Some are busier than others os lets say 2 messages per phone per day. An they are mostly in the peak work day so lets say 500 per hour and the average length is 30 seconds. 2/ retrieve either via deliver to email or dial in to a number to collect voicemail via phone (or collect and play via a website) What does the conversion and how does one handle bulk updates? to users? How much control does the user have? How are the retrieving their voicemail now? Do you want to replicate this for ease of replacement as near as possible? Right now we are using the voice mail service provided by the teclo and are spending $0.06 per minute. The user connects to the voice mail by dialing *99 and entering a password on their office set or remoetely by dialing 123-MAIL on any phone (123 is the three digit prefix of their phone number) and then entering their password. They do not have any voice to email service today. If possible I would like to ease the transition if it can be done. Lots of stepswill follow discovery if it can be done. 3/ Not sure what you mean by tie in? How do you match a voice mail box to an email address? Can there be multiple email addresses for one voice mail box? 4/ Sure, how do you have this configured at the moment? Why not replicate voicemail group delivery in the same format? Talkmail is a service provided by the telco where you group a bunch of numbers together so you can send the same message to all of them at the same time. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Saturday, 4 November 2006 11:54 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Newbie questions about Voice mail I am totally ignorant about actually using asterisk for any purpose. I have read some of the docs but not all. I am currently doing a telephone audit for my company and one of the issues is voice mail. We are spending quit a bit of money with our telco for voice mail services and I was wondering about using asterisk as just a
Re: RE: [asterisk-users] SIP v IAX2
On 26/10/06, Guillermo Salas M. [EMAIL PROTECTED] wrote: What about the bandwidth used for both protocols? Is IAX using less or more bandwidth than SIP? I'll give you an actual measured result. A trunked IAX2 link, carrying 30 simultaneous calls using variable-bit-rate Speex - we saw 7 kilobits / call / second. That's INCLUDING all IAX2, UDP, IP overheads. That's the magic of Speex VBR and trunking. Its much much much less than you can do with SIP. Better even than any of the proprietary boxes with packet-saver technology and the like when using a codec with quality comparable to Speex. 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
RE: [asterisk-users] Newbie questions about Voice mail
On Sun, 5 Nov 2006, Dean Collins wrote: Date: Sun, 5 Nov 2006 15:21:19 -0500 From: Dean Collins [EMAIL PROTECTED] To: [EMAIL PROTECTED] Cc: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: RE: [asterisk-users] Newbie questions about Voice mail Hi Brian, Uhmmm as it appears you are using a centrex service from your telco (your comment about not having any pabx) yes. I need to ask this question..are you sure that under your current commercial arrangements you are actually allowed to continue to use the telco as your centrex provider but not use them for your voicemail? Voice mail is a separately billed service that some lines have and some don't. We pay $0.06 per minute to use it. Its cash cow for the telco and a big bill for us. Also if you decided to use a separate asterisk server for your voicemail service how would calls be transferred to this number? I am assuming there is a feature to transfer a call when the phoen does not ring after a certain number of rings. But I don't thing I know to handle getting voice mail if the line is busy. Would the carrier allow you to host and asterisk service off some of your existing centrex extensions? Would this incur a cost or similar. I am sure there would be a cost whatever we had them do. I was hoping to go a little further if possible to install a server and a t1 circuit with enough capacity to handle the load. I think for your bosses 'discovery' report the answer would be Yes to can asterisk be used as just a voicemail server Yes to people can operate with the same methods of retrieval they currently do Yes to people can also retrieve via additional methods such as web or email And finally yes this will save us money in the longer term at 6c per minute currently. The next step should be 1a/ You boss decides You or someone in your team skill up in asterisk Or Does the asterisk communitty have a presence at any of the IP telephony conference? ..Brian Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Sunday, 5 November 2006 3:02 PM To: Dean Collins Cc: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Newbie questions about Voice mail Dean Thanks for responding. I have added more info in your reply. Right now we do not operate our own PBX or voice mail system. All of the service is provided by the telco. As a start I was wondering if I could simply put in asterisk to do just voicemail. I am assuming the telco can configure all the phone to automatically call forward to asterisk on no answer. If asterisk can handle this I am assuming that a user would just call some number to retiev voice mail. They would lose the call waiting light on their phone so the email notification of a voice mail would be necessary. ..Brian On Sun, 5 Nov 2006, Dean Collins wrote: Date: Sun, 5 Nov 2006 00:04:36 -0500 From: Dean Collins [EMAIL PROTECTED] To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: RE: [asterisk-users] Newbie questions about Voice mail Hi Brian, I'm sure some other people will give you better answers but quick answers are; 1/ Depends on volume of message leaving/collection, is it in a single location? Multiple locations with multiple time zones? Two locations, one time zone. Could be two different systems since they are in two different cites connected by a 1G connection. Estimate the number of voicemails left per hour and reply with this. There are about 3000 phones. Some are busier than others os lets say 2 messages per phone per day. An they are mostly in the peak work day so lets say 500 per hour and the average length is 30 seconds. 2/ retrieve either via deliver to email or dial in to a number to collect voicemail via phone (or collect and play via a website) What does the conversion and how does one handle bulk updates? to users? How much control does the user have? How are the retrieving their voicemail now? Do you want to replicate this for ease of replacement as near as possible? Right now we are using the voice mail service provided by the teclo and are spending $0.06 per minute. The user connects to the voice mail by dialing *99 and entering a password on their office set or remoetely by dialing 123-MAIL on any phone (123 is the three digit prefix of their phone number) and then entering their password. They do not have any voice to email service today. If possible I would like to ease the transition if it can be done. Lots of stepswill follow discovery if it can be done. 3/ Not sure what you mean by tie in? How do you match a voice mail box to an email address? Can there be multiple email addresses for one voice mail box? 4/ Sure, how do you have
[asterisk-users] Call Quality Issues with IAX?
Hey all, I recently got a message from my provider about IAX: We do not recommend the use of IAX. It is a lossy protocol that is known to cause crackling, loss of audio and other issues. You can use IAX if you want, but we will not assist with any issues you may encounter. Does anyone else know about these known problems? I'm not sure where this provided got this information, but it sounds like a crock. I've never experienced any of the above issues with IAX. I am concered about the reference to a lossy protocol. How is a protocol lossy? I've heard of lossy compression, which has nothing to do with the protocol used to trasmit compressed data...but I've never heard of a lossy protocol. Thoughts? Thanks, Aaron -- http://www.aaronjangel.us/ ___ --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] Definity Asterisk CallerID Issue
cp wrote: I am hoping someone could shed some light and point me in the right direction? I’m attempting to get callerid work between an Avaya Definity PBX and Asterisk via TE110P connected via T1/PRI Crossover PRI. From the Definity side I’ve searched endlessly and came with an example which we modeled as close as we can, but still no luck. While doing PRI intense debug span 1 in I see a couple interesting messages but have yet to come up with meaningful knowledge about them. I’ve tried decoded the setup message but don’t know what I’m really looking at. It appears in the decode that the calling party number or name are not being sent but as I mentioned I don’t know what I am really looking at. I wondering if these error messages have any thing to do with Asterisk not knowing what to do with what the Definity is sending? Feel free to contact me offlist. Any assistance is greatly appreciated. -CP !! Unknown IE 1544 (len = 6) !! Unknown IE 8 (cs6, Unknown Information Element) Progress Description: Calling equipment is non-ISDN TON: International Number xxx*CLI [ 02 01 d4 d2 08 02 0a e6 05 04 03 90 90 a2 18 03 a1 83 8b 1e 02 81 83 70 05 91 34 33 38 39 96 08 04 d0 35 30 80 ] Protocol Discriminator: Q.931 (8) len=33 Call Ref: len= 2 (reference 2790/0xAE6) (Originator) Message type: SETUP (5) [04 03 90 90 a2] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: 3.1kHz audio (16) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: u-Law (34) [18 03 a1 83 8b] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Preferred Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 11 ] [1e 02 81 83] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Progress Description: Calling equipment is non-ISDN. (3) ] [70 05 91 34 33 38 39] Called Number (len= 7) [ Ext: 1 TON: International Number (1) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '4389' ] [96] Locking Shift (len=01): Requested codeset 6 [08 04 d0 35 30 80] !! Unknown IE 1544 (len = 6) !! Unknown IE 8 (cs6, Unknown Information Element) Sending Receiver Ready (107) [ 02 01 01 d6 ] Why type of card do you have on the Definity Side? ___ --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] Re: Re: skype and SIP hardware for linux
On Sun, 05 Nov 2006 15:21:24 +0200, Dovid B wrote: I downloaded a softphone called kiax last night. Its working great. I was real tired then so I dont remember where I got it from. Hope that helps. (and its open source as well as they give you the source files for it :) ) [...] http://www.kiax.org/screenshots/ looks good. I'm looking at http://www.nslu2-linux.org/wiki/HowTo/ConnectUSBPhone, which _appears_ to describe the same phone. If so, this brings me full circle to asterisk as a solution. I'd definitely need the IAX, which kiax supports. Are the nslu2 folks describing hacking the http://www.yealink.com/english/prodetail_p1k.htm phone, or using that phone _with_ a slug? If I can run asterisk on my computer, and not hack any hardware, that'd be preferable. thanks, Thufir ___ --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 upgrade from 1.0.9 to 1.2.6 not working
Matt wrote: Hi, I am trying to upgrade my system (running 2.4 kernel) from 1.0.9 to 1.2.6, everything upgraded fine, however asterisk is not seeing any zap/sip/iax2 channels. I compiled in this order: libpri/zaptel/asterisk. Zaptel comes up fine... ztcfg -vv shows all of my channels, however asterisk lacks the 'zap show' 'sip show' or 'iax2 show' commands, further, if I try to force the chan_zap.so to load in modules it says: Nov 4 04:12:36 VERBOSE[24761] logger.c: [chan_zap.so]Nov 4 04:12:36 WARNING[24761] loader.c: /usr/lib/asterisk/modules/chan_zap.so: undefined symbol: ast_pickup_call Nov 4 04:12:36 WARNING[24761] loader.c: Loading module chan_zap.so failed! There is a 1.2.6 version? I would try with 1.2.13. ___ --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] best gui
On Sat, 4 Nov 2006 06:36:06 -0500 Zeeshan Zakaria [EMAIL PROTECTED] wrote: Trixbox www.trixbox.org ___ --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] Definity Asterisk CallerID Issue
Steve Totaro wrote: cp wrote: I am hoping someone could shed some light and point me in the right direction? I’m attempting to get callerid work between an Avaya Definity PBX and Asterisk via TE110P connected via T1/PRI Crossover PRI. From the Definity side I’ve searched endlessly and came with an example which we modeled as close as we can, but still no luck. While doing PRI intense debug span 1 in I see a couple interesting messages but have yet to come up with meaningful knowledge about them. I’ve tried decoded the setup message but don’t know what I’m really looking at. It appears in the decode that the calling party number or name are not being sent but as I mentioned I don’t know what I am really looking at. I wondering if these error messages have any thing to do with Asterisk not knowing what to do with what the Definity is sending? Feel free to contact me offlist. Any assistance is greatly appreciated. -CP !! Unknown IE 1544 (len = 6) !! Unknown IE 8 (cs6, Unknown Information Element) Progress Description: Calling equipment is non-ISDN TON: International Number xxx*CLI [ 02 01 d4 d2 08 02 0a e6 05 04 03 90 90 a2 18 03 a1 83 8b 1e 02 81 83 70 05 91 34 33 38 39 96 08 04 d0 35 30 80 ] Protocol Discriminator: Q.931 (8) len=33 Call Ref: len= 2 (reference 2790/0xAE6) (Originator) Message type: SETUP (5) [04 03 90 90 a2] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: 3.1kHz audio (16) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: u-Law (34) [18 03 a1 83 8b] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Preferred Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 11 ] [1e 02 81 83] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Progress Description: Calling equipment is non-ISDN. (3) ] [70 05 91 34 33 38 39] Called Number (len= 7) [ Ext: 1 TON: International Number (1) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '4389' ] [96] Locking Shift (len=01): Requested codeset 6 [08 04 d0 35 30 80] !! Unknown IE 1544 (len = 6) !! Unknown IE 8 (cs6, Unknown Information Element) Sending Receiver Ready (107) [ 02 01 01 d6 ] Why type of card do you have on the Definity Side? Also, what is your signalling on the Definity and Asterisk? Thanks, Steve Totaro ___ --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] Problems Overwriting CallerID with True ANI
Thanks for the reply but I got it worked out a few moments after I sent
the email.
BTW, exten = _*NXXNXX*NXXNXX*,8,NoOP(${CALLERID}) works just
fine. My only problem was the double underscores before setting the
callerID.
Thanks,
Steve Totaro
Kevin Bockman wrote:
Response inline.
Steve Totaro wrote:
I receive calls over a T1 with callerid and then *ani*dnis*. I am
able to strip out the ani and the dnis in the dialplan but when I try
to set the caller ID to be the ani, it looks ok but then if I do a
NoOp callerid on the next line, I get unknown.
Here is the section of my dialplan:
exten = _*NXXNXX*NXXNXX*,1,Set(ANI=${EXTEN})
exten = _*NXXNXX*NXXNXX*,2,Set(__ANI=${CUT(ANI,*,2)})
Why don't you just do that in one line?
exten = _*NXXNXX*NXXNXX*,3,Set(DNIS=${EXTEN})
exten = _*NXXNXX*NXXNXX*,4,Set(__DNIS=${CUT(EXTEN,*,3)})
And this?
exten = _*NXXNXX*NXXNXX*,5,SetVar(__TransferToExt=6101)
Use Set.
exten = _*NXXNXX*NXXNXX*,6,Set(__CALLERID(name)=${ANI})
exten = _*NXXNXX*NXXNXX*,7,Set(__CALLERID(number)=${ANI})
I'm not sure what would actually be getting set here. CALLERID() is a
function, not a variable (which you use the _s for).
Set(CALLERID(name)=${ANI})
Set(CALLERID(num)=${ANI)} num, not number.
exten = _*NXXNXX*NXXNXX*,8,NoOP(${CALLERID})
You need to do NoOp(${CALLERID(all)}) or CALLERID(num) or
CALLERID(name) or something. 'show function CALLERID'.
exten = _*NXXNXX*NXXNXX*,9,Goto(DNIS,${DNIS},1)
[DNIS]
exten = _NXXNXX,1,AGI(agi://172.16.1.135)
exten = _NXXNXX,2,Setvar(__ActiveCallID=${ActiveCallID})
Use Set instead of SetVar. You can probably get rid of this line
anyway by setting the variable in your AGI itself to __ActiveCallID.
exten = _NXXNXX,3,Goto(ext-queues,${TransferToExt},1)
exten = _NXXNXX,104,Goto(ext-queues,6004,1)
exten = h,1,DeadAGI(agi://172.16.1.135:4574)
Kevin
___
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[asterisk-users] Very high translation costs for g729
Hey gang, I'm hoping someone can help me out here. I've just noticed that on two of my five Asterisk boxes (CentOS 4.4, Asterisk 1.2.12.1), I'm getting the following translation cost for g729: asterisk*CLI show translation Server 1: g729 -26252525252426 -5336 Server 2: g729 -66656565656469 -9075 On my other three boxes, I get much saner vaules (costs anywhere from 3 to 6). Any ideas why two boxes have such high costs? All the servers run the same OS, updated to the same versions of everything, including kernel. Four of the five boxes run x86_64 kernels, with the two that are playing up both running x86_64 kernels. I've switched the entire network to using Speex instead of g729 until I find out why I'm getting such high numbers here. I suspect (but can't prove) that this may have been the cause of some audio issues between these two servers as the phones on either end use alaw, so Asterisk is transcoding to g729 across the IAX2 link. Thanks, Avi -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore StreetT: +61 (0) 3 9235 5400 Fitzroy, VIC F: +61 (0) 3 9235 5444 3065 W: http://www.squiz.net . Open Source - Own It - Squiz.net .. / ___ --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] Very high translation costs for g729
Try forcing asterisk recalculate those costs: CLI show translation recalc 20 Julian J. M. On 11/5/06, Avi Miller [EMAIL PROTECTED] wrote: Hey gang, I'm hoping someone can help me out here. I've just noticed that on two of my five Asterisk boxes (CentOS 4.4, Asterisk 1.2.12.1), I'm getting the following translation cost for g729: asterisk*CLI show translation Server 1: g729 -26252525252426 -5336 Server 2: g729 -66656565656469 -9075 On my other three boxes, I get much saner vaules (costs anywhere from 3 to 6). Any ideas why two boxes have such high costs? All the servers run the same OS, updated to the same versions of everything, including kernel. Four of the five boxes run x86_64 kernels, with the two that are playing up both running x86_64 kernels. I've switched the entire network to using Speex instead of g729 until I find out why I'm getting such high numbers here. I suspect (but can't prove) that this may have been the cause of some audio issues between these two servers as the phones on either end use alaw, so Asterisk is transcoding to g729 across the IAX2 link. Thanks, Avi -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore StreetT: +61 (0) 3 9235 5400 Fitzroy, VIC F: +61 (0) 3 9235 5444 3065 W: http://www.squiz.net . Open Source - Own It - Squiz.net .. / ___ --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] names of SIP aware firewalls
Besides ranch networks and borderware, what other SIP aware firewalls for the SOHO/medium market exists? -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] Newbie questions about Voice mail
The next step should be 1a/ You boss decides You or someone in your team skill up in asterisk Or Does the asterisk communitty have a presence at any of the IP telephony conference? ..Brian You just missed it check out www.astricon.net it was 2 weeks ago in Dallas. (but yes Digium were at VON and other events this year as well). Cheers, Dean ___ --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] Reading Voicemail Config from MySQL [+ ODBC]
Hi, After some more searching I decided to try USING unix ODBC for the connection. I have both the unixODBC and unixODBC-devel packages on my fedora box: [EMAIL PROTECTED] /]# rpm -qa | grep -i unixodbc unixODBC-2.2.11-7.1 unixODBC-devel-2.2.11-7.1 Here are my odbcinsi.ini and odbc.ini files respectively: [MySQL] Description = ODBC for MySQL Driver = /usr/lib/libmyodbc.so Setup = /usr/lib/libodbcmyS.so FileUsage = 1 --- [MYSQL-asterisk] Driver = MySQL Description = Data source for dynamic asterisk voicemail configuration Trace = Yes TraceFile = stderr SERVER = localhost USER = root PASSWORD = rootroot9 PORT = 3306 DATABASE = asterisk - Below are my res_odbc.conf and extconfig.conf files for supplying details of the DSN name and and database/table for asterisk [mysql1] enabled = yes dsn = MySQL-asterisk username = root password = *** pre-connect = yes --- [settings] voicemail = odbc,mysql1,users --- I am able to execute: [EMAIL PROTECTED] /]# isql -v MySQL-asterisk +---+ | Connected!| | | | sql-statement | | help [tablename] | | quit | | | +---+ SQL which shows I can connect to the database on the command line using my DSN name. In the asterisk CLI however, the command: asterisk*CLI odbc show No such command 'odbc' (type 'help' for help) fails which is supposed to show connections to MySQL from the CLI. ANd lastly the command: asterisk*CLI realtime load voicemail mailbox 7521 No rows found matching search criteria. Nov 6 00:33:10 WARNING[2965]: config.c:920 find_engine: Realtime mapping for 'voicemail' found to engine 'odbc', but the engine is not available also fails. Where are I going wrong? Thanks. On Sun, 2006-11-05 at 13:39 +0200, Mosiuoa Tsietsi wrote: Hi all, I have been trying to get my asterisk (v1.2.10) to lookup voicemail config data from my mysql database as opposed to voicemail.conf + sip.conf for my users. Users register with SER and get passed through to asterisk when they dial out. I followed the instructions as per http://www.voip-info.org/wiki/view/Asterisk+voicemail+database so basically I have 1) Build asterisk-addons-1.2.5 and added the USE_MYSQL_VM_INTERFACE=1 to the asterisk/apps folder and built asterisk again 2) I configured my voicemail.conf appropriately: dbuser=username dbpass=password dbhost=localhost dbname=asterisk_vm and have a database called asterisk_vm with a table called users with the fields needed by asterisk 3) Populated my database with some values When I try to leave a voicemail, I get the following error in the CLI: -- Called [EMAIL PROTECTED] -- SIP/myserver-08c5ef80 is ringing -- Nobody picked up in 2 ms -- Executing VoiceMail(SIP/myserver-08c731a0, u7521) in new stack Nov 5 13:30:55 WARNING[18146]: app_voicemail.c:2412 leave_voicemail: No entry in voicemail config file for '7521' I was able (thanks to some guys on this list) to get my Prepaid application to read from the database but voicemail won't. Please help me with this error. 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 -- The Law of Unintended Consequences: for every action, there is an excellent chance of producing an opposite and totally disproportionate reaction. - Clyde Haberman ___ --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] names of SIP aware firewalls
There is firmware out there that is made for asterisk users that can be loaded on to some linksys routers. Dont remember the URL. Do a google search for linksys hacks. - Original Message - From: Erick Perez [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, November 05, 2006 11:54 PM Subject: [asterisk-users] names of SIP aware firewalls Besides ranch networks and borderware, what other SIP aware firewalls for the SOHO/medium market exists? -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] Definity Asterisk Caller ID Issue
You should run the 4ESS protocol ("a" on the Definity command "change DS1 board#) as the Definity may not send Display Name for the NI-2 protocol (settings "b" or "d")
Remember tomake the Asterisk zapata settings consistent with the Definity.
On the Definity trunk group form Page 1, change "Codeset to send Display" to "0 (zero)", also make sure Outgoing Display is "y" . On Page 2, make sure Send Name and Send Calling Number are "y".
Setting Send UCID = "n" as well as Send Codeset 6/7LAI = "n", will clean up those Unknown IEs.
The Zaptel software only supports CodeSet 0 Information Elements so the Display Name has to be in Codeset 0.
Mark !! Unknown IE 1544 (len = 6) !! Unknown IE 8 (cs6, Unknown Information Element) Progress Description: Calling equipment is non-ISDN TON: International Number xxx*CLI [ 02 01 d4 d2 08 02 0a e6 05 04 03 90 90 a2 18 03 a1 83 8b 1e 02 81 83 70 05 91 34 33 38 39 96 08 04 d0 35 30 80 ]
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Re: [asterisk-users] Call Quality Issues with IAX?
Funny you mention this because I've run into some voice degradation problems with IAX2 myself recently... When I have an external call come in on a DiD I frequently have to send it back out to the PSTN (i.e. to a cell phone). When this happens I don't want my server in the media path, I want to hand it off to my ITSP and let them handle both ends of the call. I couldn't get it to work with SIP through the provider I'd been working with so I moved to a new ITSP and I switched from SIP to IAX2 at the same time. I had much better success transferring the call back to my ITSP using IAX2 - I could see the handshakes in the CLI and I could phyically disconnect my * server from the Ethernet once the call had been established. Unfortunately the call quality suffered terribly and was unacceptable. I had much better quality using SIP on my old ITSP, even with the media passing through my Asterisk box. So I'm curious whether this is an IAX2 problem or whether my new ITSP is simply not that good. Any thoughts? I don't think the problem can possibly be on my server given that the call is completly handed off but could I be missing something? Thanks, H On 11/5/06, Aaron J. Angel [EMAIL PROTECTED] wrote: Hey all, I recently got a message from my provider about IAX: We do not recommend the use of IAX. It is a lossy protocol that is known to cause crackling, loss of audio and other issues. You can use IAX if you want, but we will not assist with any issues you may encounter. Does anyone else know about these known problems? I'm not sure where this provided got this information, but it sounds like a crock. I've never experienced any of the above issues with IAX. I am concered about the reference to a lossy protocol. How is a protocol lossy? I've heard of lossy compression, which has nothing to do with the protocol used to trasmit compressed data...but I've never heard of a lossy protocol. Thoughts? Thanks, Aaron -- http://www.aaronjangel.us/ ___ --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] names of SIP aware firewalls
On Sunday 05 November 2006 13:54, Erick Perez wrote: Besides ranch networks and borderware, what other SIP aware firewalls for the SOHO/medium market exists? Netfilter's SIP connection-tracking module is ready for prime time, and will be included in 2.6.18 Linux kernels. Early birds can patch older kernels and not wait. http://www.enterprisenetworkingplanet.com/netos/article.php/3638441 -- ~ Carla Schroder Linux geek and random computer tamer check out my Linux Cookbook! best book for sysadmins and power 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] Very high translation costs for g729
On 06/11/2006, at 8:53 AM, Julian J. M. wrote: Try forcing asterisk recalculate those costs: Ok, that fixed it. Thanks! :) -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore StreetT: +61 (0) 3 9235 5400 Fitzroy, VIC F: +61 (0) 3 9235 5444 3065 W: http://www.squiz.net . Open Source - Own It - Squiz.net .. / ___ --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] SPA3k wired to PAP2 for echo testing
On 05/11/06, James Harper [EMAIL PROTECTED] wrote: Even in this configuration, with my impedance settings set to the Australian standard of 220+820||120nf, and the PSTN and PAP2 echo cancellers enabled (or not, and all combinations of) I get local echo as soon as I pick up the handset (I hear my voice bounced back to me). Surely this shouldn't be??? There is no hybrid involved at all! 'course there is. The telephone interface on the one end and the line interface on the other are both 2 wire. Ah. I was referring to the hybrid as the 2 wire to 4 wire interface at the telco. I have enlightened myself now :) Did you have a phone line connected to the other side. Running into an unconnected FXO port is likely to make echo because of the unbalanced impedance. The FXO (PSTN) interface on the SPA3000 is connected to the PAP2, which provides dialtone instead of the Telco providing it. I tried this to take the Telco out of the equation, but it didn't solve anything. I'm thinking there might be something wrong with my SPA3000, as with the settings I have used the impedance should be matched perfectly and there should be no echo. One strange thing happened once though (before this testing). I picked up the handset connected to the Line1 interface, which patched me straight through to the PSTN interface (dialplan = '(:@gw0S0)'), which was connected to the telco. I pressed '1' which stopped the Telco dialtone while it waited for more numbers, and there was no echo at all. It's never happened again, but the fact that it happened once gives me hope that the echo problem might be solvable. On the SPA3000 there is 'Echo Cancel', 'Adaptive Echo Cancel', and 'Echo Suppression'. Enabling 'Echo Cancel' doesn't seem to do anything. Enabling 'Adaptive Echo Cancel' causes a huge reduction in call quality and actually makes the echo worse. 'Echo Suppression' isn't really a solution. One question though... if everything is balanced properly, should I even need echo cancellation? I'll keep testing... Thanks James ___ --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] Hairpinning problems using IAX2 and SIP
Thanks for responding. Yes I am doing pretty much exactly what you showed. When I try to dial without answering, I get a busy tone on the DiD (the local Telco offers to let them notify me when it becomes available). Sometimes I get half a ring on the destination cell phone b4 receiving the busy signal. Not sure whether this sheds any light. I'm going to have to answer the call anyway in order to implement an auto attendant, but it was worth trying. Cheers, H *CLI -- Accepting AUTHENTICATED call from ip: requested format = ulaw, requested prefs = (), actual format = g729, host prefs = (g729), priority = mine -- Executing Dial(IAX2/ITSP-1, IAX2/ITSP/6135551234) in new stack -- Called ITSP/6135551234 -- Call accepted by ip (format g729) -- Format for call is g729 -- IAX2/ITSP-2 is making progress passing it to IAX2/ITSP-1 -- Hungup 'IAX2/ITSP-2' == Spawn extension (incoming-iax, 6135551234, 1) exited non-zero on 'IAX2/ITSP1' -- Executing Hangup(IAX2/ITSP-1, ) in new stack == Spawn extension (incoming-iax, h, 1) exited non-zero on 'IAX2/ITSP-1' -- Hungup 'IAX2/ITSP-1' On 11/4/06, Andrew Joakimsen [EMAIL PROTECTED] wrote: When you say you answer the call, I assume you have something like this: exten = 5551212,1,Answer exten = 5551212,1,Dial(SIP/provider/10005551212) Try to not answer the call and see if the behviour changes, it could just be your ITSP configuration On 11/4/06, hugolivude [EMAIL PROTECTED] wrote: Asterisk 1.2.7 RedHat 9.0 I frequently have the need to redirect calls that come in on a DiD provisioned by my ITSP, back to the ITSP so that they can terminate the call on the PSTN. For example when an external call comes in, I often have to send it to a cell phone. I believe that this is referred to as hairpinning the call. I do this by answering the incoming call and then I use a simple dial command to send it back to my ISTP using a SIP or IAX channel and the ITSP terminates it on the cell phone.One of my main goals is to keep my Asterisk box out of the media path and let the ITSP handle all the provisioning for the call. I understand that the default behaviour of the dial command is supposed to do just that, but I've run into problems though on both SIP IAX channels. With IAX I use a simple dial command: Dial(IAX2/myIAX/7775551234) Things seem to work great, I can see the handshaking in the CLI as the call gets redirected and once both end points are connected, I can actually disconnect my box from the ethernet and the call is uninterruoted. Unfortuanately the call quality is terrible! Low volume, choppy and so on. It seemed to me that since I had stepped my * box out of the network, the problem must be with the ITSP. They suggested I try SIP. With SIP I use: Dial(SIP/[EMAIL PROTECTED]) Unfortuantely I don't get the handshakes and the whole call ends up passing through my box, which is something I'm desperate to avoid. I have canreinvite=yes as seen from my sip.conf: [mySIP] type=peer auth=md5 username=UID fromuser=UID fromdomain=domain secret=pw host=domain port=5060 nat=yes canreinvite=yes qualify=no disallow=all allow=g729 dtmfmode=rfc2833 insecure=very context=incoming-sip Now the questions: 1) Given that I can see the handshaking and I can disconnect my * box during the call, I think that the IAX call quality problems are on my ITSP's end, but I could be wrong. Is there anything I can do to improve call quality when using IAX this way? 2) What about SIP? Why doesn't that work? I always thought that dial would do exactly what I'm after (hairpin/redirect the call) if I avoided options like t or T. Any direction you can provide is highly appreciated. Thanks, H ___ --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] Very high translation costs for g729
newbie alert Glad to see this got fixed so quickly, but could someone give a brief explanation of what this was? What did Asterisk do? Where does the cost come into play or get calculated? Jason The place where you made your stand never mattered, only that you were there... and still on your feet Avi Miller wrote: Hey gang, I'm hoping someone can help me out here. I've just noticed that on two of my five Asterisk boxes (CentOS 4.4, Asterisk 1.2.12.1), I'm getting the following translation cost for g729: asterisk*CLI show translation Server 1: g729 -26252525252426 -5336 Server 2: g729 -66656565656469 -9075 On my other three boxes, I get much saner vaules (costs anywhere from 3 to 6). Any ideas why two boxes have such high costs? All the servers run the same OS, updated to the same versions of everything, including kernel. Four of the five boxes run x86_64 kernels, with the two that are playing up both running x86_64 kernels. I've switched the entire network to using Speex instead of g729 until I find out why I'm getting such high numbers here. I suspect (but can't prove) that this may have been the cause of some audio issues between these two servers as the phones on either end use alaw, so Asterisk is transcoding to g729 across the IAX2 link. Thanks, Avi -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore StreetT: +61 (0) 3 9235 5400 Fitzroy, VIC F: +61 (0) 3 9235 5444 3065 W: http://www.squiz.net . Open Source - Own It - Squiz.net .. / ___ --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] Anybody used Asterfax?
Please post AsterFax enquiries on the AsterFax help forum at https://sourceforge.net/forum/forum.php?forum_id=510878. In the mean time please look for /var/log/asterfax_install.log and post any errors you see in there. Zeeshan Zakaria wrote: Trying to install asterfax-1.1-freeb2.i386.rpm, I get following error. How can I get rid of it. Installing jre Installing libtiff Installing ghostscript Installing Xvfb Installing openoffice.org http://openoffice.org Installing spandsp Installing spandsp0.0.3 Spandsp did not install correctly. error: %post(asterfax-1.1-freeb2.i386) scriptlet failed, exit status 1 ___ --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] Experiment: Dialplan size vs. Speed
On Sat November 4 2006 06:43, Steve Murphy [EMAIL PROTECTED] wrote: I was encouraged to post this notice on both asterisk-users and asterisk-dev; sorry if this is overkill, but it **is** applicable to both communities. Since the report is fairly large, has a pretty graph, and the whole bit, it was thought that posting on it a website, and letting you browse it would be better than sending hundreds a 50K message. http://www.asterisk.org/SpeedvsSizeExperiment Hope you enjoy it! murf Hi Murf, thanks for writing up what you've done so far. It was a very interesting read. Keep us posted with how it's coming along! Cheers, -- Nick E: [EMAIL PROTECTED] P: +61 7 5591 3588 F: +61 7 5591 6588 If you receive this email by mistake, please notify us and do not make any use of the email. We do not waive any privilege, confidentiality or copyright associated with it. ___ --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] TDM400 hungup problem
Ale, We had a simiarly problem here, not sure if its the same. The Telco here has 'ISDN suspension' (think thats the correct term) activated on landlines here by default. When you phone some one, the person who recieves the call can put down the reciever and goto another room and pick it up again and continue the call. If the person hangups and doesnt pick up another phone in the house for approx. 1 min the call is ended. Only the person being called can do this. I would connect a normal phone to the landline you're using for the TDM and then call it. Then hangup and pick it up again 30 seconds later and see if the call is still active. If it is then thats your problem. If this is your problem, you will have to take it up with the Telco suppling the landline. -Shaun On Saturday 04 November 2006 03:42, Ale wrote: Ciao, Giorgio Incantalupo wrote: Hi Ale, your problem is quite strange. Can you try with an analog phone or another SIP phone? We had try to call the local number attached to the tdm400 from a cell phone, another phone (real), and from a voip provider always with the same result... the phone of the caller doesn't hangup after the hangup command of asterisk. My zaptel and zapata are the same as yours but I have not the problem you describe. And Asterisk console?? Doesn't it say anything useful?? Nothing useful i think... voip*CLI -- Starting simple switch on 'Zap/2-1' Nov 3 18:39:10 NOTICE[2024]: chan_zap.c:6073 ss_thread: Got event 18 (Ring Begin)... Nov 3 18:39:11 NOTICE[2024]: chan_zap.c:6073 ss_thread: Got event 2 (Ring/Answered)... -- Executing Answer(Zap/2-1, ) in new stack -- Executing Dial(Zap/2-1, SIP/phone|20) in new stack -- Called phone -- SIP/phone-0815dbd8 is ringing -- SIP/phone-0815dbd8 answered Zap/2-1 -- Executing NoOp(Zap/2-1, asd 16) in new stack -- Hungup 'Zap/2-1' voip*CLI Notice that if the caller phone must be a normal phone...if it is a cell phone it does not automatically hang up. Notice that we have try also to call from a softphone sip and a realphone attached on the fxs of the tdm400 and viceversa and all works fine. Our operetor is Telecom Italia... Ciao Ciao Ale ___ --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] Use astbill to bill Trixbox
Hello everyone, I'm trying to set up a system wherein Trixbox handles the calls but it's astbill that's billing the calls. Has anyone set up something similar? How would you go about with this kind of set up? Best regards, Matt -- Stand before it and there is no beginning. Follow it and there is no end. Stay with the ancient Tao, Move with the present. ___ --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] Re: Newbie Questions - Grandstorm phones?
Hi, Where can I find that option? Thanks Jesus -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Gordon Henderson Enviado el: Jueves, 02 de Noviembre de 2006 11:44 a.m. Para: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [asterisk-users] Re: Newbie Questions - Grandstorm phones? On Wed, 1 Nov 2006, Henry.L.Coleman wrote: I came to the same conclusion. There is one thing however that the GXP2000 needs in my opinion. There is no dial plan avaiable in the configuration, this means that when dialing a number there is a slight delay before it actually dials. With a dial plan the dialed number is sent immeadiately the pattern is match ed so it saves a second or two. Maybe they will fix this? Set the Early Dial option - it's on a per-line basis, then as soon as Asterisk gets a number it can dial, it will. No need to wait the 4 seconds or press the send button... Gordon ___ --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 -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.409 / Virus Database: 268.13.23/513 - Release Date: 02/11/2006 ___ --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] xfsound=beep is not beeping
I have the value of xfersound = beep in my features.conf file but when a call is transferred there is no beep noise. Can someone please assist? features.conf xfersound = beep ; to indicate an attended transfer is complete ___ --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] Re: Newbie Questions - Grandstorm phones?
On the Budgetone 200 it is in the account tab settings of the web setup and it does work here with asterisk and my dialplans.. Doug On Sun, 5 Nov 2006, [iso-8859-1] Jes?s M?ndez Rom?n wrote: Hi, Where can I find that option? Thanks Jesus -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Gordon Henderson Enviado el: Jueves, 02 de Noviembre de 2006 11:44 a.m. Para: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [asterisk-users] Re: Newbie Questions - Grandstorm phones? On Wed, 1 Nov 2006, Henry.L.Coleman wrote: I came to the same conclusion. There is one thing however that the GXP2000 needs in my opinion. There is no dial plan avaiable in the configuration, this means that when dialing a number there is a slight delay before it actually dials. With a dial plan the dialed number is sent immeadiately the pattern is match ed so it saves a second or two. Maybe they will fix this? Set the Early Dial option - it's on a per-line basis, then as soon as Asterisk gets a number it can dial, it will. No need to wait the 4 seconds or press the send button... Gordon ___ --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] SPA3k wired to PAP2 for echo testing
Yes I agree, the SPA3000 can be a bear with echo on the PSTN. I did find that using older fimware helped some and that the levels - there are 4 settings - FXO/FXS in/out can be juggled to help. I also found out after adding a Budgetone 200 that I had much less echo problem going through it and the spa3000 FXO - vs. using the local analog phones on the spa3000 fxs port to FXO port. So some of the answer might be to get rid of as much (or all) local analog as you can. I plan to buy more hard sip phones and do that here eventually. This is ultimately more flexible as each extension has it's own number and they can dial each other as well as dial more then one place simutaneously. The big problem is that SIP phones are generally ugly and black and not styled for home use. Doug On Sun, 5 Nov 2006, James Harper wrote: In my seemingly endless search for the cause of echo on my SPA3000, I wired it up in the following configuration: Analogue Handset -- (FXS)SPA3000(FXO) -- PAP2 And set the Line1 dialplan on the SPA3k to '(:@gw0S0)' which means that as soon as I pick up the handset I get linked straight through to the PAP2, which gives me dialtone. Even in this configuration, with my impedance settings set to the Australian standard of 220+820||120nf, and the PSTN and PAP2 echo cancellers enabled (or not, and all combinations of) I get local echo as soon as I pick up the handset (I hear my voice bounced back to me). Surely this shouldn't be??? There is no hybrid involved at all! If anyone on this list with a SPA3k (that doesn't have any local echo problems on the PSTN port) and an ATA with a FXS port, could they please try the above setup and post the results (including SPA3k hardware and firmware versions, and the ATA used)? I wonder if there is a problem with some versions of the SPA3k where there is some sort of inbalance on the PSTN port that causes echo right there rather than further down the line? Thanks James ___ --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 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 * ___ --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] Polycom SIP 2.0.2 firmware
I'm still waiting on the 2.0 firmware from Voipsupply. No luck. Don't hold your breath, I would have died a couple of weeks ago.On 11/4/06, Eric Bishop [EMAIL PROTECTED] wrote:I second that request. On 11/4/06, Kevin Bockman [EMAIL PROTECTED] wrote: Hi,Would anyone be kind enough to send me the 2.0.2 SIP firmware?I askedVoipSupply for it on Wednesday, nagged them again on Thursday and theydid not even send the request yet.I was supposed to have it 'Friday morning' at the latest.I'm doing equipment upgrades this weekend sothis is the time to do it.I've been having random phone crashing using2.0.1.I also asked VoipSupply for firmware a month or so ago and they never sent it.It is not listed yet on the freedom file site.Thanks,Kevin___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Lacy MooreI'm the guy that doesn't give a damn about anyone's problems but my own... ___ --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] names of SIP aware firewalls
Besides ranch networks and borderware, what other SIP aware firewalls for the SOHO/medium market exists? Anything Cisco ___ --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] light web user interface
FreePBX allows you to specify an extension range per login so that only extensions within the range are visible to that user. Cameron - Original Message - From: Curt Shaffer To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Wednesday, November 01, 2006 1:40 AM Subject: RE: [asterisk-users] light web user interface Basically I would like a page that would allow a user to log in and modify their extension only. So for example, I log in for extension 102 once in there I can turn on or off my call waiting. Add a number to call forward to. Change the email address my voice mail gets sent to. Add any numbers I may want to block via caller ID. Maybe view my voice mails that are saved and be able to download them in wav format from there. Add find me follow me extensions and numbers, etc I would also like it open enough that I can add features to it. Im not the best at PHP but I can work my way around in it. I thought maybe freePBX allowed this with its users but I cant see where you can lock them down to only see information on a particular extension. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dovid BSent: Tuesday, October 31, 2006 3:44 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [asterisk-users] light web user interface What attributes are you talking about ? Depending on what they are it may be real simple to set something up. - Original Message - From: Curt Shaffer To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Monday, October 30, 2006 9:51 PM Subject: [asterisk-users] light web user interface Does anyone know of a really lightweight web interface that allows users to log in and modify attributes of their extension only? Thanks Curt ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo 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] Anyone got a dialplan for SPA ATAs for ISN?
Googling for a while has turned up evidence that this can be corrected by a carefully-crafted dialplan for the Sipuras, at least, but the avaialable documentation is, let's say, a little convoluted. Try this on Sipura (*x.*x.) Seemed to work for me. Cameron ___ --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 DTMF detection
Hi, Hi All, I've just delved into the world of asterisk and I'm having a few dtmf issues. Internally, amongst sip phones, dtmf is fine. Externally, if you ring from a GSM mobile, DTMF is fine, however if you ring from a standard phone, DTMF fails to register. I am attempting to use a quad port HFC-4S Beronet Card. I've been searching the web most of the last week and haven't found anything that actually helps (although there seem to be a few people with DTMF boggles) Is there anything special we need to do for these cards to detect telecom dtmf tones? I find it a little confusing that incoming GSM, which rings in via the same lines works, and telecoms phones don't. I thought it would be the other way around if anything. I have attached logs of an incoming call via telecom, and an incoming call via vodafone. The only difference I can see is the codec (I think) type. Telecom seems to send the cal through as Audio 3.1K while vodafone uses Speech. Would this difference do anything? my mISDN config is also below should that be of any help. Telecom: (No DTMF) Connected to Asterisk 1.2.13 currently running on PROHOST100 (pid = 4481) Verbosity is at least 3 -- Remote UNIX connection P[ 4] set_channel: bc-channel:0 channel:1 P[ 4] I IND :SETUP oad: dad:9298 pid:3 state:none P[ 4] -- channel:1 mode:TE cause:16 ocause:16 rad: cad: P[ 4] -- info_dad: onumplan: dnumplan:0 rnumplan: cpnnumplan:0 P[ 4] -- caps:Audio 3.1k pi:3 keypad: sending_complete:1 P[ 4] -- Bearer: Audio 3.1k P[ 4] -- Codec: Alaw P[ 0] -- * NEW CHANNEL dad:9298 oad: P[ 4] -- CTON: Unknown P[ 4] EXPORT_PID: pid:3 P[ 4] -- PRES: Restricted (0) P[ 4] -- SCREEN: Unscreened (0) P[ 4] I SEND:PROCEEDING oad: dad:9298 pid:3 P[ 4] -- bc_state:BCHAN_CLEANED P[ 4] -- channel:1 mode:TE cause:16 ocause:16 rad: cad: P[ 4] -- info_dad: onumplan: dnumplan:0 rnumplan: cpnnumplan:0 P[ 4] -- caps:Audio 3.1k pi:3 keypad: sending_complete:1 -- Executing Set(mISDN/4-1, FROM_DID=9298) in new stack -- Executing Set(mISDN/4-1, FAX_RX=disabled) in new stack -- Executing Goto(mISDN/4-1, ivr-4|s|1) in new stack -- Goto (ivr-4,s,1) -- Executing Set(mISDN/4-1, LOOPCOUNT=0) in new stack -- Executing Set(mISDN/4-1, __DIR-CONTEXT=default) in new stack -- Executing Answer(mISDN/4-1, ) in new stack P[ 4] * ANSWER: P[ 4] -- Connection is without BF encryption P[ 4] -- ECHO OFF P[ 4] -- None P[ 4] -- empty cad using dad P[ 4] I SEND:CONNECT oad: dad:9298 pid:3 P[ 4] -- bc_state:BCHAN_CLEANED P[ 4] -- channel:1 mode:TE cause:16 ocause:16 rad: cad:9298 P[ 4] -- info_dad: onumplan: dnumplan:0 rnumplan: cpnnumplan:0 P[ 4] -- caps:Audio 3.1k pi:3 keypad: sending_complete:1 P[ 4] -- TRANSPARENT Mode P[ 4] ec_enable -- Executing Wait(mISDN/4-1, 1) in new stack P[ 4] BCHAN: bchan ACT Confirm pid:3 P[ 4] I IND :CONNECT_ACKNOWLEDGE oad: dad:9298 pid:3 state:CONNECTED P[ 4] -- channel:1 mode:TE cause:16 ocause:16 rad: cad:9298 P[ 4] -- info_dad: onumplan: dnumplan:0 rnumplan: cpnnumplan:0 P[ 4] -- caps:Audio 3.1k pi:3 keypad: sending_complete:1 -- Executing Set(mISDN/4-1, TIMEOUT(digit)=3) in new stack -- Digit timeout set to 3 -- Executing Set(mISDN/4-1, TIMEOUT(response)=10) in new stack -- Response timeout set to 10 -- Executing BackGround(mISDN/4-1, custom/pronet-incoming) in new stack -- Playing 'custom/pronet-incoming' (language 'en') P[ 4] I IND :DISCONNECT oad: dad:9298 pid:3 state:CONNECTED P[ 4] -- channel:1 mode:TE cause:16 ocause:16 rad: cad:9298 P[ 4] -- info_dad: onumplan: dnumplan:0 rnumplan: cpnnumplan:0 P[ 4] -- caps:Audio 3.1k pi:8 keypad: sending_complete:1 P[ 4] -- org:2 nt:0, inbandavail:1 state:10 P[ 4] hangup_chan P[ 4] - queue_hangup P[ 4] I SEND:RELEASE oad: dad:9298 pid:3 P[ 4] -- bc_state:BCHAN_ACTIVATED P[ 4] -- channel:1 mode:TE cause:16 ocause:-1 rad: cad:9298 P[ 4] -- info_dad: onumplan: dnumplan:0 rnumplan: cpnnumplan:0 P[ 4] -- caps:Audio 3.1k pi:8 keypad: sending_complete:1 == Spawn extension (ivr-4, s, 7) exited non-zero on 'mISDN/4-1' -- Executing Hangup(mISDN/4-1, ) in new stack == Spawn extension (ivr-4, h, 1) exited non-zero on 'mISDN/4-1' P[ 4] * IND : HANGUPpid:3 ctx:ivr-4 dad:h oad:(null) State:CONNECTED P[ 4] -- l3id:80003 P[ 4] -- cause:16 P[ 4] -- out_cause:16 P[ 4] -- state:CONNECTED P[ 4] Channel: mISDN/4-1 hanguped new state:CLEANING P[ 4] $$$ CLEANUP CALLED pid:3 P[ 4] $$$ Cleaning up bc with stid :10010400 pid:3 P[ 4] ec_disable P[ 4] Sending Control ECHOCAN_OFF P[ 4] I IND :RELEASE_COMPLETE oad: dad: pid:3 state:CLEANING P[ 4] -- channel:0 mode:TE cause:16 ocause:16 rad: cad: P[ 4] -- info_dad: onumplan:0 dnumplan:0 rnumplan:0 cpnnumplan:0 P[ 4] -- caps:Speech pi:0 keypad: sending_complete:0 P[ 4] ast_hangup already called, so we have no ast ptr anymore in event(RELEASE_COMPLETE) P[ 4] hangup_chan P[ 4] No need to queue hangup P[ 4] Cannot hangup chan, no ast P[ 4] release_chan: bc with l3id: 80003 P[ 4] BCHAN: DeACT Conf
Re: [asterisk-users] names of SIP aware firewalls
Intertex Not cheap, licensed per number of users But seem to work great and have some nifty tools very confusing picking models though On Nov 5, 2006, at 3:54 PM, Erick Perez wrote: Besides ranch networks and borderware, what other SIP aware firewalls for the SOHO/medium market exists? -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] names of SIP aware firewalls
Carla Schroder wrote: On Sunday 05 November 2006 13:54, Erick Perez wrote: Besides ranch networks and borderware, what other SIP aware firewalls for the SOHO/medium market exists? Netfilter's SIP connection-tracking module is ready for prime time, and will be included in 2.6.18 Linux kernels. Early birds can patch older kernels and not wait. http://www.enterprisenetworkingplanet.com/netos/article.php/3638441 AstLinux has a SIP-conntrack-enabled 2.6.18 in trunk right now. Looks pretty cool! -- Kristian Kielhofner ___ --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] Tampa Bay Asterisk Users Meetup on Monday
Matt Florell wrote: Hello, We will be having another Tampa Bay Area Asterisk Users Meetup on Monday, November 6th at 7:30 PM. Asterisk users from gurus to new users are welcome. Along with user discussions, we will be talking about Astricon and Asterisk 1.4 at this meeting. We will also have free items from Digium to be given away. go to the site for more info: http://asteriskpbx.meetup.com/1/calendar/5178348/ See you there, MATT--- Matt, Not only will I be there, I am bringing a Starbox-X for the same $50 deal that we had at Astricon! -- Kristian Kielhofner ___ --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] names of SIP aware firewalls
Erick Perez wrote: Besides ranch networks and borderware, what other SIP aware firewalls for the SOHO/medium market exists? Alcatel 610x (discontinued?)/620x. These routers can act as standalone SIP PBX or outbound proxy to allow phones to register to central registry. Leo ___ --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
