[asterisk-users] Compatible IP Phones for Asterisk
Hello, I have a cheap old BT 100 since first Asterisk came out and don't know that if with the lately IP phones that would work for Asterisk. I'm looking for the list of IP phones that work with Asterisk for all of the buttons on the phone like (MWI, DND, Message, Voicemail, Transfer, Conference, etc...) and advice of which one would be good for Small Home and Small Business (SOHO). Thanks. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Conference Calling
Here is where to get you start with this. http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO -Tri From: Faheem faheem_...@yahoo.com To: asterisk-users@lists.digium.com Sent: Sat, February 27, 2010 12:08:24 PM Subject: [asterisk-users] Conference Calling Hey All, I want to implement a conference calling scenario. Conference Call Procedure: User1 dial the User2. When call is connected put the current call on Hold and dial User3. When the call is connected between User1 and User3 join the User2 in a conference room! How I can implement this scenario. What are generic steps to do so! Thanks = Muhammad Faheem -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk n-way DTMF detection
I have figured this out and it's working fine now. Here is the key if anyone has the same issue. featuredigittimeout = 5000 Increase the digit timeout or you have to press the key codes quick enough in order for Asterisk to detect the keys. -Tri From: Tri Tu mtr...@yahoo.com To: asterisk-users@lists.digium.com Sent: Wed, February 24, 2010 10:38:25 PM Subject: [asterisk-users] Asterisk n-way DTMF detection Hello, I have setup the n-way conferencing with Asterisk and it's working when I use with my budgetone 100 phone but it doesn't work for any of the voip software or other ATA that I have. When I turned the debug on, I see that the correct keys (*0) were entered but asterisk doesn't detect the signal to trigger the features event. I have set a test extension to get the input dtmf key and say the digit out. They are getting correctly on the IVR but when using n-way conferencing, it's not taking it. Here is the output of testing DTMF with IVR. v103*CLI v103*CLI -- Executing [...@from-internal:1] Read(SIP/-b6807538, digito||10) in new stack -- Accepting a maximum of 10 digits. * DTMF-relay event received: 8 * DTMF-relay event received: 5 * DTMF-relay event received: 2 -- User entered '852' -- Executing [...@from-internal:2] SayDigits(SIP/-b6807538, 852) in new stack -- SIP/-b6807538 Playing 'digits/8' (language 'en') -- SIP/-b6807538 Playing 'digits/5' (language 'en') -- SIP/-b6807538 Playing 'digits/2' (language 'en') -- Executing [...@from-internal:3] Hangup(SIP/-b6807538, ) in new stack == Spawn extension (from-internal, 88, 3) exited non-zero on 'SIP/-b6807538' -- Executing [...@from-internal:1] Macro(SIP/-b6807538, hangupcall) in new stack -- Executing [...@macro-hangupcall:1] ResetCDR(SIP/-b6807538, vw) in new stack -- Executing [...@macro-hangupcall:2] NoCDR(SIP/-b6807538, ) in new stack -- Executing [...@macro-hangupcall:3] GotoIf(SIP/-b6807538, 1?skiprg) in new stack -- Goto (macro-hangupcall,s,6) -- Executing [...@macro-hangupcall:6] GotoIf(SIP/-b6807538, 1?skipblkvm) in new stack -- Goto (macro-hangupcall,s,9) -- Executing [...@macro-hangupcall:9] GotoIf(SIP/-b6807538, 1?theend) in new stack -- Goto (macro-hangupcall,s,11) -- Executing [...@macro-hangupcall:11] Hangup(SIP/-b6807538, ) in new stack == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/-b6807538' in macro 'hangupcall' == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/-b6807538' v103*CLI bash-3.1# anh here is the console log of the Asterisk when pressing the key during callerA is on the phone with CallerB. v103*CLI v103*CLI * DTMF-relay event received: * * DTMF-relay event received: 0 v103*CLI Wondering that if anyone know what could be wrong here. My asterisk version is Asterisk 1.4.20. -Tri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] X-Lite won't register
Turn debug on and watch on the console to see if the you see the x-lite extension talks to your asterisk box. CLI core set debug or CLI core set verbose 99 From: Girard, Jeffrey COL MIL USA jeffrey.gir...@us.army.mil To: asterisk-users@lists.digium.com Sent: Thu, February 25, 2010 6:35:52 AM Subject: [asterisk-users] X-Lite won't register Beginner to Asterisk, but not beginner to VoIP FreePBX front end running on a dell 1550 and XLite running on a different Woindows XP box Both boxes connected via switch on same subnet. No NAT involved On FreePBX I created a new extension 1001 with a SIP password of 1001 On Xlite, username is 1001, password is 1001, authorization user name is 1001, and domain is IP of Free PBX XLite tries to register then shows 408 error registration timeout Windows box pings Asterisk and firewall is disabled on XP machine What am I missing? Jeff -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No RTP from asterisk?
RTP is only firewall issue. Make sure that you can pass traffic from your client to the asterisk server. If it's on the same LAN, there shouldn't be any issue with RTP unless the Asterisk is setup with firewall to block RTP traffic (default is from 1 - 2 upd) Asterisk doesn't support G29 (pass-through is OK) but if you want to connect from your client to asterisk server with G729, you need to buy license. Using G711 is free and it taking about 68kbp. From: Peter Serwe peter.se...@gmail.com To: asterisk-users@lists.digium.com Sent: Sat, February 27, 2010 12:42:56 PM Subject: [asterisk-users] No RTP from asterisk? I've got an asterisk installation of 1.4.30-RC2 running, and while I can register lines and get call setup to pass, for some reason no RTP is being generated or received by asterisk. Debug doesn't seem to give me too much of relevance about it, especially rtp debug. I had a few other small issues, like trying to negotiate G729 when it's not capable, but since then, I've changed everything back to G711. I have connected to it, a SIP trunk, 3 registered users and I'm at a loss as to how to troubleshoot this further. Can anyone point me in the right direction? Peter -- Peter Serwe http://truthlightway.blogspot.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk n-way DTMF detection
Hello, I have setup the n-way conferencing with Asterisk and it's working when I use with my budgetone 100 phone but it doesn't work for any of the voip software or other ATA that I have. When I turned the debug on, I see that the correct keys (*0) were entered but asterisk doesn't detect the signal to trigger the features event. I have set a test extension to get the input dtmf key and say the digit out. They are getting correctly on the IVR but when using n-way conferencing, it's not taking it. Here is the output of testing DTMF with IVR. v103*CLI v103*CLI -- Executing [...@from-internal:1] Read(SIP/-b6807538, digito||10) in new stack -- Accepting a maximum of 10 digits. * DTMF-relay event received: 8 * DTMF-relay event received: 5 * DTMF-relay event received: 2 -- User entered '852' -- Executing [...@from-internal:2] SayDigits(SIP/-b6807538, 852) in new stack -- SIP/-b6807538 Playing 'digits/8' (language 'en') -- SIP/-b6807538 Playing 'digits/5' (language 'en') -- SIP/-b6807538 Playing 'digits/2' (language 'en') -- Executing [...@from-internal:3] Hangup(SIP/-b6807538, ) in new stack == Spawn extension (from-internal, 88, 3) exited non-zero on 'SIP/-b6807538' -- Executing [...@from-internal:1] Macro(SIP/-b6807538, hangupcall) in new stack -- Executing [...@macro-hangupcall:1] ResetCDR(SIP/-b6807538, vw) in new stack -- Executing [...@macro-hangupcall:2] NoCDR(SIP/-b6807538, ) in new stack -- Executing [...@macro-hangupcall:3] GotoIf(SIP/-b6807538, 1?skiprg) in new stack -- Goto (macro-hangupcall,s,6) -- Executing [...@macro-hangupcall:6] GotoIf(SIP/-b6807538, 1?skipblkvm) in new stack -- Goto (macro-hangupcall,s,9) -- Executing [...@macro-hangupcall:9] GotoIf(SIP/-b6807538, 1?theend) in new stack -- Goto (macro-hangupcall,s,11) -- Executing [...@macro-hangupcall:11] Hangup(SIP/-b6807538, ) in new stack == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/-b6807538' in macro 'hangupcall' == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/-b6807538' v103*CLI bash-3.1# anh here is the console log of the Asterisk when pressing the key during callerA is on the phone with CallerB. v103*CLI v103*CLI * DTMF-relay event received: * * DTMF-relay event received: 0 v103*CLI Wondering that if anyone know what could be wrong here. My asterisk version is Asterisk 1.4.20. -Tri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone working with NUFONE?
Hi Brian, By looking at the www.nufone.net, it doesn't have any much details of the services. What is the current rate that you have for domestic long distance rate? -Tri. - Original Message - From: Brian Capouch [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, February 24, 2004 10:42 AM Subject: Re: [Asterisk-Users] Anyone working with NUFONE? Sales wrote: Curious if anyone has any feedback on Nufone voip pbx. Perfectly happy customer. Most of my customers use NuFone as well--perhaps a dozen of us all told. Excellent uptime, reasonable rates. Email-based customer service for the most part. Others are bothered by that; I am not. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone working with NUFONE?
Do you know which one is the good VoIP termination? Is it Nufone or VoicePulse Connect? Any other suggestion for business plan. Thanks. -Tri. - Original Message - From: Brian Capouch [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, February 24, 2004 10:59 AM Subject: Re: [Asterisk-Users] Anyone working with NUFONE? Tri Tu wrote: Hi Brian, By looking at the www.nufone.net, it doesn't have any much details of the services. What is the current rate that you have for domestic long distance rate? 2.9c/min domestic. International varies by country, of course, but is very competitive according to those of my customers who are using it. Their customer service leaves a lot to be desired in terms of timeliness. It doesn't seem to be a problem once you're actually on board with them--I always get responses to my mails within 24 hours--but apparently if you're mailing them to sign up ([EMAIL PROTECTED]) it can take a few days. But they're very well connected on both the Internet and PSTN sides, and to my knowledge they never go down. That has been a problem with some of the other carriers. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] T1 inbound dialplan
Hello everyone, I'm trying to setup the T1 line with T100P card but could not get it accepts inbound calls. I only got busy signal. Anyone know what would be the dialplan (extension.conf) to accept T1 line calls. Thanks. -Tri.
[Asterisk-Users] T100P T1
Hello there, I have been using asterisk as VoIP for few month as gateway for both softphone hardphone. It's working great. I just bought two T100P cards to setup for the PBX system as gateway that would accept incoming calls and use VoIP for outbound calls. Here is my plan looks like: Current working PBX: PTSN T1 -- T1(CSU) -- Mitel SX200 PBX-- Phones New plan with Asterisk: PTSN T1 -- T1(CSU) -- T100P (first card) T100P (second card) -- Mitel SX200 PBX -- Phones The current PBX is currently running. It uses straight through cable from the T1 CSU (box) to Mitel PBX. When I unplug the cable to put in the T100P card, the light is still RED. I have test with both straight through crossover T1 cable. No luck. Can anyone help me how to make this work? Thanks. Tri Tu
