[asterisk-users] Fwd: Register SIP on Asterisk 1.6.2.14 on Centos 5.5 64bit
-- Forwarded message -- From: Phuong Hoang ducphuongbk200...@gmail.com Date: Thu, Nov 18, 2010 at 9:16 AM Subject: Register SIP on Asterisk 1.6.2.14 on Centos 5.5 64bit To: asterisk-users@lists.digium.com Hi all, I registered sip phone(X-Lite) on Asterisk 1.6.2.14 on CentOS 5.5 64 bit but not successful, Can anyone help me to do it? Thanks and best regards. Phuong Hoang. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to register SIP phone on Asterisk 1.6.2.14 on Centos 5.5 64bit
Hi all, I registered sip phone(X-Lite) on Asterisk 1.6.2.14 on CentOS 5.5 64 bit but not successful, Can anyone help me to do it? Thanks and best regards. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to register SIP phone on Asterisk 1.6.2.14 on Centos 5.5 64bit
On 18 Nov 2010, at 10:36, Phuong Hoang wrote: I registered sip phone(X-Lite) on Asterisk 1.6.2.14 on CentOS 5.5 64 bit but not successful, Can anyone help me to do it? How is this different to the other two posts? Please stop repeatedly sending messages! If nobody replies you're probably not giving enough information! S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fwd: Register SIP on Asterisk 1.6.2.14 on Centos 5.5 64bit
On 18 Nov 2010, at 10:33, Phuong Hoang wrote: I registered sip phone(X-Lite) on Asterisk 1.6.2.14 on CentOS 5.5 64 bit but not successful, Can anyone help me to do it? Given that you haven't given any error messages, any logs, or your sip.conf, or the manner in which it is not working No? Going to assume by the fact you said 'registered' rather than 'trying to register' - it's registered and you can't make a call? if so it could be something codec related I suppose. Please post your asterisk log, and SIP traces of when the problem occurs. S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Announcement Transfer with call-limit = 1
Hi people, Who knows as I can do Announcement Transfer with call-limit = 1 in Asterisk 1.6. Thanks so much. Renato -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to register SIP phone on Asterisk 1.6.2.14 on Centos 5.5 64bit
I think these will be helpful, if not for solving the problem, at least for trying to rephrase your questing in a meaningful manner: http://www.asteriskdocs.org http://www.asteriskguide.com and particularly http://www.asteriskguide.com/pdf/GettingStartedWithAsterisk.pdf Bert On 18/11/2010 11:36, Phuong Hoang wrote: Hi all, I registered sip phone(X-Lite) on Asterisk 1.6.2.14 on CentOS 5.5 64 bit but not successful, Can anyone help me to do it? Thanks and best regards. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Load Balance and Failover
Hi All Does anyone know about any tool that does to Asterisk what mod_jk does for JBoss/Tomcat: a load-balance/failover server that is constantly connected to Asterisk backend servers and is capable of identify loaded or down servers? Regards Antônio Theóphilo smime.p7s Description: S/MIME cryptographic signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GSM and SS7 Questions
On Wed, Nov 17, 2010 at 6:17 PM, Matthew Fredrickson cres...@digium.com wrote: On 11/17/10 2:44 PM, Cary Fitch wrote: In regard to #2, any T1 card should work. But the problem is you need SS7 software and SS7 connectivity in addition to the T1 card. Asterisk (as of version 1.6.0 or greater) has native support for SS7 with DAHDI interface cards in chan_dahdi. I obviously have used it with quite a few Digium cards that have worked well. Matthew, So 1.6.0 or newer, with a Digium card should talk T1/SS7 no problem? That's great news! I have a Nortel DMS100 that we have configured for DS1/SS7 and we were trying to figure out how to connect an Asterisk PBX to it. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Load Balance and Failover
You could use a sip proxy front end like Kamailio. Sent from my iPhone On Nov 18, 2010, at 7:39 AM, Antônio Theóphilo anto...@freeddom.com wrote: Hi All Does anyone know about any tool that does to Asterisk what mod_jk does for JBoss/Tomcat: a load-balance/failover server that is constantly connected to Asterisk backend servers and is capable of identify loaded or down servers? Regards Antônio Theóphilo -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recommended *WRT router to run Asterisk?
On Tue, 16 Nov 2010 09:42:33 -0500, Mark Deneen mden...@gmail.com wrote: Are you saying ADSL as in a generic term for broadband router or do you really mean that the router also acts as a DSL transceiver? Sorry about that. Ideally, the unit should be both an ADSL modem + router, but apparently, most of them are just routers so that the user would have to turn their ADSL router into a modem/bridge and connect the *WRT-moded router. If someone's been running Asterisk on that kind of hardware for SOHO use, what would you recommend? Apparently, those are hardware that come up often in forums: http://wiki.openwrt.org/toh/d-link/dir-825 http://wiki.openwrt.org/toh/buffalo/wzr-hp-g300h http://wiki.openwrt.org/toh/asus/wl500gp http://wiki.openwrt.org/toh/asus/wl600g -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recommended *WRT router to run Asterisk?
On 11/18/2010 07:52 AM, Gilles wrote: On Tue, 16 Nov 2010 09:42:33 -0500, Mark Deneenmden...@gmail.com wrote: Are you saying ADSL as in a generic term for broadband router or do you really mean that the router also acts as a DSL transceiver? Sorry about that. Ideally, the unit should be both an ADSL modem + router, but apparently, most of them are just routers so that the user would have to turn their ADSL router into a modem/bridge and connect the *WRT-moded router. If someone's been running Asterisk on that kind of hardware for SOHO use, what would you recommend? Apparently, those are hardware that come up often in forums: http://wiki.openwrt.org/toh/d-link/dir-825 http://wiki.openwrt.org/toh/buffalo/wzr-hp-g300h http://wiki.openwrt.org/toh/asus/wl500gp http://wiki.openwrt.org/toh/asus/wl600g I never saw the point of spending $100 for something that is so limited. You can spend a little more and get something like an ALIX board that is so much more capable and still fanless/low power. http://www.pcengines.ch/alix.htm The 2d3/2d13 are very nice for the price. If you really want to run on a small router like this, the Netgear WNR3500 is a decent device and can be found for around $90. Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recommended *WRT router to run Asterisk?
On Tue, Nov 16, 2010 at 8:28 AM, Gilles codecompl...@free.fr wrote: Hello For users who 1) don't have a QoS-capable ADSL router and 2) would like to run Asterisk with a couple of SIP trunks, I was wondering what hardware is recommend to run any of the main open-source *WRT projects to which Asterisk has been ported: I'm running Asterisk 1.4 on a WRT54GS that I picked up off ebay for $50. The WRT54GL doesn't have quite enough memory so I went with the GS model. I'm running OpenWRT on it. I was mostly experimenting with it but ended up installing it at my parents' house as a kind of batphone solution. I also hung a couple of SIP phones off of it giving them a couple of different extensions, one of which works across a WIFI connection. Their WRT54GS connects to my Asterisk 1.8.0 machine using IAX. Both endpoints are behind NAT. Works pretty well for me. -- Chris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recommended *WRT router to run Asterisk?
On 11/18/2010 10:02 AM, Chris Gentle wrote: On Tue, Nov 16, 2010 at 8:28 AM, Gilles codecompl...@free.fr mailto:codecompl...@free.fr wrote: Hello For users who 1) don't have a QoS-capable ADSL router and 2) would like to run Asterisk with a couple of SIP trunks, I was wondering what hardware is recommend to run any of the main open-source *WRT projects to which Asterisk has been ported: I'm running Asterisk 1.4 on a WRT54GS that I picked up off ebay for $50. The WRT54GL doesn't have quite enough memory so I went with the GS model. I'm running OpenWRT on it. I was mostly experimenting with it but ended up installing it at my parents' house as a kind of batphone solution. I also hung a couple of SIP phones off of it giving them a couple of different extensions, one of which works across a WIFI connection. Their WRT54GS connects to my Asterisk 1.8.0 machine using IAX. Both endpoints are behind NAT. Works pretty well for me. -- Chris I have a similar setup in an office but sip directly back to the main server - not sure what the value add to the local asterisk is, except intercom calls would not have to leave the lan, but isn't that the purpose of reinvite ? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Load Balance and Failover
Thank you for the answer Darren. In fact I have an application that requests a call to a real person through an AMI interface and get some client information. Using a SIP Proxy is an option but I prefer that the interface between the app and the Asterisk could be the AMI (or HTTP). Regards Antonio On 18/11/2010, at 11:51, Darren Sessions wrote: You could use a sip proxy front end like Kamailio. Sent from my iPhone On Nov 18, 2010, at 7:39 AM, Antônio Theóphilo anto...@freeddom.com wrote: Hi All Does anyone know about any tool that does to Asterisk what mod_jk does for JBoss/Tomcat: a load-balance/failover server that is constantly connected to Asterisk backend servers and is capable of identify loaded or down servers? Regards Antônio Theóphilo -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Antônio Theóphilo smime.p7s Description: S/MIME cryptographic signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recommended *WRT router to run Asterisk?
On Thu, Nov 18, 2010 at 9:20 AM, jon pounder j...@inline.net wrote: I have a similar setup in an office but sip directly back to the main server - not sure what the value add to the local asterisk is, except intercom calls would not have to leave the lan, but isn't that the purpose of reinvite ? When I first set it up I was using only SIP connections without an Asterisk box on the remote end, just like you mentioned. I had numerous NAT problems, which I now believe were caused by a really lousy router on the far end. Since I'm behind NAT on both ends, I wanted to switch to IAX to see if that would help. It did and it was a fun learning experience to get Asterisk going on such a limited piece of hardware. Now it just works with almost no maintenance. -- Chris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GSM and SS7 Questions
On 11/18/10 7:40 AM, Matt wrote: On Wed, Nov 17, 2010 at 6:17 PM, Matthew Fredricksoncres...@digium.com wrote: On 11/17/10 2:44 PM, Cary Fitch wrote: In regard to #2, any T1 card should work. But the problem is you need SS7 software and SS7 connectivity in addition to the T1 card. Asterisk (as of version 1.6.0 or greater) has native support for SS7 with DAHDI interface cards in chan_dahdi. I obviously have used it with quite a few Digium cards that have worked well. Matthew, So 1.6.0 or newer, with a Digium card should talk T1/SS7 no problem? That's great news! I have a Nortel DMS100 that we have configured for DS1/SS7 and we were trying to figure out how to connect an Asterisk PBX to it. That is correct. Feel free to ask me any questions if you have any issues come up along the way. The sample chan_dahdi.conf has a section with an example of an SS7 setup in it, for reference on configuration. Matthew Fredrickson Hardware/Software Engineer Digium, Inc. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GSM and SS7 Questions
On 11/18/10 10:07 AM, Matthew Fredrickson wrote: On 11/18/10 7:40 AM, Matt wrote: On Wed, Nov 17, 2010 at 6:17 PM, Matthew Fredricksoncres...@digium.com wrote: On 11/17/10 2:44 PM, Cary Fitch wrote: In regard to #2, any T1 card should work. But the problem is you need SS7 software and SS7 connectivity in addition to the T1 card. Asterisk (as of version 1.6.0 or greater) has native support for SS7 with DAHDI interface cards in chan_dahdi. I obviously have used it with quite a few Digium cards that have worked well. Matthew, So 1.6.0 or newer, with a Digium card should talk T1/SS7 no problem? That's great news! I have a Nortel DMS100 that we have configured for DS1/SS7 and we were trying to figure out how to connect an Asterisk PBX to it. That is correct. Feel free to ask me any questions if you have any issues come up along the way. The sample chan_dahdi.conf has a section with an example of an SS7 setup in it, for reference on configuration. Oh yeah, and also, there's an asterisk-ss7 mailing list at lists.digium.com where SS7 related discussion and questions usually take place. Matthew Fredrickson Hardware/Software Engineer Digium, Inc. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.8 VM_DUR problems
Hi all, We have been using asterisk 1.8 for some while now, together with asterisk 1.6. We have the following problem. In asterisk 1.8 when you leave a voicemail, the person at that extension is notified by email that he has a voicemail. The voicemail is attached to that email along with other details like: duration of vm, caller name, caller number, date. The duration is always reported lover than 4 seconds despite the fact that the voicemail left has 20 seconds or more, I’ve looked on /var/spool/asterisk/voicemail/default/EXTEN/INBOX and there in the *.txt file of that voicemail the duration is still reported wrong. On asterisk 1.6 we didn’t had this problem and I even ported the configuration file (voicemail.conf) from there to asterisk 1.8 but it still doesn’t work. Do you have any ideas please ? Thanks for the help, Bogdan Sarandan-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk parking question
Hi, I`ve been using Asterisk parking lots (multiple parking lots) with relative success on 1.6.2.X. The problem that I just found was the following: When I do park someone, and the call is parked for the duration of the timeout, the person who parked the call gets back the parked calls (his phone rings). That much is as expected. Let`s say this person doesn`t answer. I can`t tell where do set the behavior. I get a no extension ‘t’ in context ‘’ Message in the CLI. So, an empty context. Where do I set where the call should do at this point, I don`t have a NULL context anywhere, and I don`t think I should….? Anyone knows what I can do to ensure the call isn’t dropped? Mike -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk parking question
Mike wrote: I get a no extension ‘t’ in context ‘’ Add this to your dial plan: [park-dial] exten = t,1,Where to send calls go here Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GSM and SS7 Questions
That is correct. Feel free to ask me any questions if you have any issues come up along the way. The sample chan_dahdi.conf has a section with an example of an SS7 setup in it, for reference on configuration. Oh yeah, and also, there's an asterisk-ss7 mailing list at lists.digium.com where SS7 related discussion and questions usually take place. Thanks - Last time I looked into this (back in the 1.2 days, things were still under extensive development) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk parking question
I tried thator I think I did something similar, but that may or may not apply (depending on my understanding of parking lots) Here is my relevant contexts. The SIP phones are registered under this context: [some_context] include = parkinglotA include = outboundcalls exten = t,1,Verbose(1|parking timeout!!!) Here, in features.conf, here is parkinglotA's definition [parkinglotA] context = parkinglotA parkpos = 101-120 findslot = next parkingtime=60 The thing is, I never hit the Verbose command. So my questions: 1) Why won't this work? And more importantly: 2) what's this park-dial context you speak of ? Is this a hardcoded context calls go back to? Can I set one per parkilots (remember: I use multiple parking lots) Thanks for taking the time to answer my question. Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle Sent: Thursday, November 18, 2010 12:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk parking question Mike wrote: I get a no extension t in context Add this to your dial plan: [park-dial] exten = t,1,Where to send calls go here Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk parking question
On 18 November 2010 17:43, Mike l...@net-wall.com wrote: I tried thator I think I did something similar, but that may or may not apply (depending on my understanding of parking lots) Here is my relevant contexts. The SIP phones are registered under this context: [some_context] include = parkinglotA include = outboundcalls exten = t,1,Verbose(1|parking timeout!!!) Here, in features.conf, here is parkinglotA's definition [parkinglotA] context = parkinglotA parkpos = 101-120 findslot = next parkingtime=60 The thing is, I never hit the Verbose command. So my questions: 1) Why won't this work? And more importantly: 2) what's this park-dial context you speak of ? Is this a hardcoded context calls go back to? Can I set one per parkilots (remember: I use multiple parking lots) Your call is in the [parkinglotA] context, but you are adding the 't' to your [some_context] context, perhaps the following will work. I have not tried it: [some_context] include = parkinglotA include = outboundcalls [parkinglotA] exten = t,1,Verbose(1|parking timeout!!!) Regards, Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk parking question
Mike wrote: So my questions: 1) Why won't this work? I'm still running under 1.4.x And more importantly: 2) what's this park-dial context you speak of ? Is this a hardcoded context calls go back to? It is under 1.4 Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk parking question
Hi, I tried all combinaisons of parkinglot contexts. I always get the same CLI message: no extension t in context I can't find a way to define what context Asterisk should be going back to. The park-dial context that Doug suggested didn't help, neither did my variations on parkinglotA. The unnamed context seems like a bug, maybe I'll open something in the bug system if nobody can explain it... Or is there a secret place to define this in features.conf? Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Davies Sent: Thursday, November 18, 2010 12:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk parking question On 18 November 2010 17:43, Mike l...@net-wall.com wrote: I tried thator I think I did something similar, but that may or may not apply (depending on my understanding of parking lots) Here is my relevant contexts. The SIP phones are registered under this context: [some_context] include = parkinglotA include = outboundcalls exten = t,1,Verbose(1|parking timeout!!!) Here, in features.conf, here is parkinglotA's definition [parkinglotA] context = parkinglotA parkpos = 101-120 findslot = next parkingtime=60 The thing is, I never hit the Verbose command. So my questions: 1) Why won't this work? And more importantly: 2) what's this park-dial context you speak of ? Is this a hardcoded context calls go back to? Can I set one per parkilots (remember: I use multiple parking lots) Your call is in the [parkinglotA] context, but you are adding the 't' to your [some_context] context, perhaps the following will work. I have not tried it: [some_context] include = parkinglotA include = outboundcalls [parkinglotA] exten = t,1,Verbose(1|parking timeout!!!) Regards, Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Meetme and MOH
Hi, With a dynamic Meetme using: MeetMe(|DsMrc) How do I control which context MOH uses, other than default ? Asterisk: 1.4.15 Thanks, Adrian -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recommended *WRT router to run Asterisk?
On Tue, Nov 16, 2010 at 9:28 AM, Gilles codecompl...@free.fr wrote: Hello For users who 1) don't have a QoS-capable ADSL router and 2) would like to run Asterisk with a couple of SIP trunks, I was wondering what hardware is recommend to run any of the main open-source *WRT projects to which Asterisk has been ported: (http://en.wikipedia.org/wiki/List_of_wireless_router_firmware_projects Thank you. Are you saying ADSL as in a generic term for broadband router or do you really mean that the router also acts as a DSL transceiver? Most of the open source firmware projects do not support DSL and rely on separate hardware to do the DSL, just like they rely on separate hardware for cable modem internet. -M -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Incoming calls
Hi all, I'd like that each analog trunk of my TDM410p was received in different extension. So, in dahdi-channel.conf and chan-dahdi.conf I put each trunk in a different context and in my extensions.conf, under [default] I put such contexts and an especific estension to answer it. therefore, when I get call, it always is ringing on the first extensions, dont matter trunk . Anybody could teach me how can I organize that ? Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Avoiding deadlock
For some reason we are seeing Avoiding deadlock for channel in our Asterisk logs, the logs are getting filled up with an amazing speed around 12000 lines a second, and all of them are Avoiding deadlock. What could be the potential reason for this to be happening? The Asterisk is used as auto dialler, therefore different channel types are involved SIP, DAHDI, Local's. [Nov 15 14:20:01] DEBUG[21740] channel.c: Avoiding deadlock for channel '0x9f17c88' [Nov 15 14:20:01] DEBUG[21740] channel.c: Avoiding deadlock for channel '0x9f17c88' [Nov 15 14:20:01] DEBUG[21740] channel.c: Avoiding deadlock for channel '0x9f17c88' [Nov 15 14:20:01] DEBUG[21740] channel.c: Avoiding deadlock for channel '0x9f17c88' [Nov 15 14:20:01] DEBUG[21740] channel.c: Avoiding deadlock for channel '0x9f17c88' [Nov 15 14:20:01] DEBUG[21740] channel.c: Avoiding deadlock for channel '0x9f17c88' [Nov 15 14:20:01] DEBUG[21740] channel.c: Avoiding deadlock for channel '0x9f17c88' [Nov 15 14:20:01] DEBUG[21740] channel.c: Avoiding deadlock for channel '0x9f17c88' [Nov 15 14:20:01] DEBUG[21740] channel.c: Avoiding deadlock for channel '0x9f17c88' [Nov 15 14:20:01] DEBUG[21740] channel.c: Avoiding deadlock for channel '0x9f17c88' [Nov 15 14:20:01] DEBUG[21740] channel.c: Avoiding deadlock for channel '0x9f17c88' [Nov 15 14:20:01] DEBUG[21740] channel.c: Avoiding deadlock for channel '0x9f17c88' [Nov 15 14:20:01] DEBUG[21740] channel.c: Avoiding deadlock for channel '0x9f17c88' Asterisk: 1.4.33.1 DAHDI: dahdi-2.3.0.1-3 Regards, Vilius. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming calls
On Thu, 18 Nov 2010, Flavio Miranda wrote: I'd like that each analog trunk of my TDM410p was received in different extension. So, in dahdi-channel.conf and chan-dahdi.conf I put each trunk in a different context and in my extensions.conf, under [default] I put such contexts and an especific estension to answer it. therefore, when I get call, it always is ringing on the first extensions, dont matter trunk. Anybody could teach me how can I organize that ? 0) Use a subject that gives a clue what you're looking for. Almost everybody has had a question about an incomig call at some point in time. Better bait = better fish. 1) It sounds like you have a clue about how to do it and are on the right track. 2) Including some details like the console output from: zap show channel 1 (I'm a 1.2 Luddite.) zap show channel 2 zap show channel 3 zap show channel 4 as well as the console log from a call coming in on each channel will help in assisting you in resolving this issue. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX2 and INVAL packets
Is anybody here familiar with the meaning of INVAL packets for IAX2? Every few days I get a dropped outgoing call in the middle of the conversation (the outgoing call has been connected for few minutes) when an incoming call comes in. The log reads the following when this happens: [Nov 17 15:25:04] DEBUG[5138] chan_iax2.c: Immediately destroying 2963, having received INVAL [Nov 17 15:25:04] DEBUG[5138] chan_iax2.c: Destroying call 2963 [Nov 17 15:25:04] DEBUG[11242] chan_iax2.c: We're hanging up IAX2/ihs_trunk_out-2963 now... [Nov 17 15:25:04] VERBOSE[11242] chan_iax2.c: -- Hungup 'IAX2/ihs_trunk_out-2963' And more setup details, for those who still have the will to live :-) Asterisk version: 1.6.2.13 Internal externsions: everything on SIP - 3 Grandstream GXP-2000, 2 analog phones on a pci OpenVox card and 2 Linphone softphones Trunks: IAX2 Trunks provider: Gradwell Asterisk machine: 800Mhz Intel Pentium, 512MB of RAM Internet connection: Tiscali business ADSL I am happy to post here any config files and logs you might think would be relevant. This is not consistent - and I've managed to have 4 concurrent calls which held 30 minutes (before I hung them up) when I tried. So not easy to replicate. Sebastian -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming calls
Hi Steve, thanks for the tips Better bait = better fish ! As you said, I am in the right track. Looking to dahdi show channles , I realized that all the trunks was in the same context. So, I have changed this and everything works! thanks you !! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda Date: Thu, 18 Nov 2010 11:53:26 -0800 From: asterisk@sedwards.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Incoming calls On Thu, 18 Nov 2010, Flavio Miranda wrote: I'd like that each analog trunk of my TDM410p was received in different extension. So, in dahdi-channel.conf and chan-dahdi.conf I put each trunk in a different context and in my extensions.conf, under [default] I put such contexts and an especific estension to answer it. therefore, when I get call, it always is ringing on the first extensions, dont matter trunk. Anybody could teach me how can I organize that ? 0) Use a subject that gives a clue what you're looking for. Almost everybody has had a question about an incomig call at some point in time. Better bait = better fish. 1) It sounds like you have a clue about how to do it and are on the right track. 2) Including some details like the console output from: zap show channel 1 (I'm a 1.2 Luddite.) zap show channel 2 zap show channel 3 zap show channel 4 as well as the console log from a call coming in on each channel will help in assisting you in resolving this issue. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming calls
On Thu, 18 Nov 2010, Flavio Miranda wrote: Looking to dahdi show channles , I realized that all the trunks was in the same context. So, I have changed this and everything works! That's why I prefer to work from what Asterisk parsed the file as, not what the poster thinks :) -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recommended *WRT router to run Asterisk?
AstLinux on various embedded hardware really works well. I have several on older HP Thin Clients, 55xx series, some with only 128 Meg of ram. A replacement flash from Transcend and a 55xx of eBay can be easily under $50. the GUI in AstLinux makes life simple for users who want to make minor changes. If you want to go REALLY small, the Dockstar with a 1-2 Gig pen flash running Debian and 1.6.2.13 in a package not much larger than a pack of cigarettes Both run off 12VDC, so backup through power blips is easy. Either of these, IMO, is a better choice than the WRT on a router. But then, I don't much care for all the eggs in one basket. John Novack Chris Gentle wrote: On Thu, Nov 18, 2010 at 9:20 AM, jon pounder j...@inline.net mailto:j...@inline.net wrote: I have a similar setup in an office but sip directly back to the main server - not sure what the value add to the local asterisk is, except intercom calls would not have to leave the lan, but isn't that the purpose of reinvite ? When I first set it up I was using only SIP connections without an Asterisk box on the remote end, just like you mentioned. I had numerous NAT problems, which I now believe were caused by a really lousy router on the far end. Since I'm behind NAT on both ends, I wanted to switch to IAX to see if that would help. It did and it was a fun learning experience to get Asterisk going on such a limited piece of hardware. Now it just works with almost no maintenance. -- Chris -- Dog is my Co-pilot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ISDN-FAX with Asterisk
Hi everybody, since some time I am looking for a current and reliable solution to send and receive faxes (probably fax-2-mail and mail-2-fax) in conjunction with Asterisk. For testing I am using a HFC-ISDN passive PCI-card, in production a Digium Dual T1/E1 PCI-card will be used. I run CentOS 5.5 (Kernel 2.6.18) and Asterisk 1.4 (but I also can use 1.8) but did not find any solution where I think that's it. What are you using? mISDN, CAPI4linux, HylaFAX, IAXmodem, chan_misdn, ... ? Can you point me to the correct direction, may be there are some more or less current howto's (more current than the ones from 2007 and earlier you find everywhere in the net)? Thanks a lot, -- Chau y hasta luego, Thorolf -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recommended *WRT router to run Asterisk?
On Thu, 18 Nov 2010, John Novack wrote: AstLinux on various embedded hardware really works well. I have several on older HP Thin Clients, 55xx series, some with only 128 Meg of ram. A replacement flash from Transcend and a 55xx of eBay can be easily under $50. the GUI in AstLinux makes life simple for users who want to make minor changes. If you want to go REALLY small, the Dockstar with a 1-2 Gig pen flash running Debian and 1.6.2.13 in a package not much larger than a pack of cigarettes Holy crap! I just found these on CircuitCity's web site for $29... I bought three to play with :) Is anyone running on a Dockstar in production? For a small SIP only office this may just be ideal! j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ISDN-FAX with Asterisk
Im using
asterisk-1.6.2.13
asterisk-addons-1.6.2.2
dahdi-linux-complete-2.4.0+2.4.0
libpri-1.4.11.4
spandsp-0.0.6
Sangoma Hardware, using wanpipe-3.5.17
Extensions.conf:
[fax-in]
exten = s,1,Answer()
exten = s,n,Wait(1)
exten =
s,n,Set(BASEFILE=fax-${CALLERID(dnid)}-${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)})
;exten = s,n,Set(${LOCALSTATIONID})
exten = s,n,MixMonitor(/mnt/ramdisk/${BASEFILE}.wav)
exten = s,n,ReceiveFAX(/mnt/ramdisk/${BASEFILE}.tif)
exten = s,n,Hangup()
exten = h,1,System(/home/asterisk/dofax.sh ${EMAILADDRESS} ${FAXSTATUS}
${CALLERID(num)} ${REMOTESTATIONID} ${CALLERID(dn$
[inbound-pri]
;
exten = 00,1,Set(LOCALSTATIONID=${EXTEN})
exten = 00,2,Set(EMAILADDRESS=emailaddress)
exten = 00,3,Goto(fax-in,s,1)
the dofax.sh script checks if tif file exists, converts to pdf, emails, and
then archives on no errors,if missing tiff, or faxstatus success, it puts
the fax in a queue folder along with the mix monitor file for analysis. , of
the faxes the fail, you can usually here bad line quality from the sender.
/mnt/ramdisk is a 1gb ramdisk, the dofax script moves the tif/pdf/wavs to a
samba share, and deletes them out of the ramdisk folder.
William Stillwell
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Thorolf Godawa
Sent: Thursday, November 18, 2010 4:55 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] ISDN-FAX with Asterisk
Hi everybody,
since some time I am looking for a current and reliable solution to
send
and receive faxes (probably fax-2-mail and mail-2-fax) in conjunction
with Asterisk.
For testing I am using a HFC-ISDN passive PCI-card, in production a
Digium Dual T1/E1 PCI-card will be used.
I run CentOS 5.5 (Kernel 2.6.18) and Asterisk 1.4 (but I also can use
1.8) but did not find any solution where I think that's it.
What are you using? mISDN, CAPI4linux, HylaFAX, IAXmodem, chan_misdn,
... ?
Can you point me to the correct direction, may be there are some more
or
less current howto's (more current than the ones from 2007 and earlier
you find everywhere in the net)?
Thanks a lot,
--
Chau y hasta luego,
Thorolf
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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asterisk-users mailing list
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--
_
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http://www.asterisk.org/hello
asterisk-users mailing list
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Re: [asterisk-users] Recommended *WRT router to run Asterisk?
On Thu, Nov 18, 2010 at 8:52 AM, Gilles codecompl...@free.fr wrote: On Tue, 16 Nov 2010 09:42:33 -0500, Mark Deneen mden...@gmail.com wrote: Are you saying ADSL as in a generic term for broadband router or do you really mean that the router also acts as a DSL transceiver? Sorry about that. Ideally, the unit should be both an ADSL modem + router, but apparently, most of them are just routers so that the user would have to turn their ADSL router into a modem/bridge and connect the *WRT-moded router. If someone's been running Asterisk on that kind of hardware for SOHO use, what would you recommend? Apparently, those are hardware that come up often in forums: http://wiki.openwrt.org/toh/d-link/dir-825 http://wiki.openwrt.org/toh/buffalo/wzr-hp-g300h http://wiki.openwrt.org/toh/asus/wl500gp http://wiki.openwrt.org/toh/asus/wl600g I use the Buffalo WZR-HP-G300NH with openwrt here. They are great for remote locations with a few sets -- you can have them hook up to the central server over an openvpn tunnel. I have ~20 sets hooked up to one with no issues at all. That being said, we probably only hit 7 or 8 concurrent calls at any point during the day. Even so, the resources on the router are not close to exhaustion. If you do try it, make sure that you do not have the iptables nat helper module installed. It's not helping you and causes problems when the router is the sip server and not hosting a sip client. -M -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Recommended *WRT router to run Asterisk?
Hello For users who 1) don't have a QoS-capable ADSL router and 2) would like to run Asterisk with a couple of SIP trunks, I was wondering what hardware is recommend to run any of the main open-source *WRT projects to which Asterisk has been ported: (http://en.wikipedia.org/wiki/List_of_wireless_router_firmware_projects Thank you. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recommended *WRT router to run Asterisk?
Jeff LaCoursiere wrote: On Thu, 18 Nov 2010, John Novack wrote: AstLinux on various embedded hardware really works well. I have several on older HP Thin Clients, 55xx series, some with only 128 Meg of ram. A replacement flash from Transcend and a 55xx of eBay can be easily under $50. the GUI in AstLinux makes life simple for users who want to make minor changes. If you want to go REALLY small, the Dockstar with a 1-2 Gig pen flash running Debian and 1.6.2.13 in a package not much larger than a pack of cigarettes Holy crap! I just found these on CircuitCity's web site for $29... I bought three to play with :) Is anyone running on a Dockstar in production? For a small SIP only office this may just be ideal! j Not really in production But for a SIP/IAX Asterisk box, it works! there is a Dockstar hacking site that de-nuts the boot code and allows booting from a 1-2 gig flash ( I have not had good luck with 4 and 8 gig flash, but it could be the flash sticks. Loading Debian squeeze onto the flash, configure Debian not to use the swap, then wget and compile Asterisk. the make file needs to be modified to specify arm5 rather than the longer name configure generates. adding some additional packages to the Debian load will be needed. Lenny also works. the Dockstar only has 128M of ram. the more expensive Sheeva I believe has more, but for a small office or home it just sits there and works! remember to noload all the unwanted modules as well. I am no Linux eggspurt, but I got this working, with the help of Google, in a few hours. Sometimes smaller is better, or at least it can be fun! John Novack -- Dog is my Co-pilot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recommended *WRT router to run Asterisk?
On Thu, Nov 18, 2010 at 9:26 AM, Darrick Hartman dhart...@djhsolutions.com wrote: On 11/18/2010 07:52 AM, Gilles wrote: On Tue, 16 Nov 2010 09:42:33 -0500, Mark Deneenmden...@gmail.com wrote: Are you saying ADSL as in a generic term for broadband router or do you really mean that the router also acts as a DSL transceiver? Sorry about that. Ideally, the unit should be both an ADSL modem + router, but apparently, most of them are just routers so that the user would have to turn their ADSL router into a modem/bridge and connect the *WRT-moded router. If someone's been running Asterisk on that kind of hardware for SOHO use, what would you recommend? Apparently, those are hardware that come up often in forums: http://wiki.openwrt.org/toh/d-link/dir-825 http://wiki.openwrt.org/toh/buffalo/wzr-hp-g300h http://wiki.openwrt.org/toh/asus/wl500gp http://wiki.openwrt.org/toh/asus/wl600g I never saw the point of spending $100 for something that is so limited. You can spend a little more and get something like an ALIX board that is so much more capable and still fanless/low power. http://www.pcengines.ch/alix.htm The 2d3/2d13 are very nice for the price. If you really want to run on a small router like this, the Netgear WNR3500 is a decent device and can be found for around $90. If you shop around / wait for deals, you can find the Buffalo for ~$70. I'm sure that the ALIX rigs are low power, just like most routers. However, most routers also come with a 4 port switch + plus one WAN interface. The ALIX boards get you at most 3 independent interfaces, but I don't believe that they can act as a switch. A 400 MHz MIPS is fairly close to a 500 MHz Geode. However, you can't get the asterisk g729 module for mips. I can't say I would want to transcode on the ALIX system, though. For a small setup or for a setup at home, it's really not a bad deal. Especially if you want something to do NAT for you. -M -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Meetme and MOH
On Thu, Nov 18, 2010 at 12:37 PM, Adrian Marsh adrian.ma...@ubiquisys.com wrote: Hi, With a dynamic Meetme using: MeetMe(|DsMrc) How do I control which context MOH uses, other than “default” ? Asterisk: 1.4.15 Thanks, Adrian -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Use Set(CHANNEL(musicclass)=MUSICONHOLDCLASSYOUWANT). What I do is add a column to the conferences/meetme table in my database to hold the moh class I want and then retrieve that in the dialplan use the aforementioned command. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Meetme and MOH
On Thu, Nov 18, 2010 at 12:37 PM, Adrian Marsh adrian.ma...@ubiquisys.comwrote: Hi, With a dynamic Meetme using: MeetMe(|DsMrc) How do I control which context MOH uses, other than “default” ? Asterisk: 1.4.15 In 1.4.x you would use SetMusicOnHold(class) before you called your MeetMe() in the dialplan. In 1.6.x (at least 1.6.2.x), you would use Set(CHANNEL(musicclass)=...) instead. -- Thanks, --Warren Selby, dCAP http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2 and INVAL packets
On 11/18/2010 10:38 PM, Tilghman Lesher wrote: On Thursday 18 November 2010 14:01:49 Sebastian wrote: Is anybody here familiar with the meaning of INVAL packets for IAX2? Every few days I get a dropped outgoing call in the middle of the conversation (the outgoing call has been connected for few minutes) when an incoming call comes in. The log reads the following when this happens: [Nov 17 15:25:04] DEBUG[5138] chan_iax2.c: Immediately destroying 2963, having received INVAL [Nov 17 15:25:04] DEBUG[5138] chan_iax2.c: Destroying call 2963 [Nov 17 15:25:04] DEBUG[11242] chan_iax2.c: We're hanging up IAX2/ihs_trunk_out-2963 now... [Nov 17 15:25:04] VERBOSE[11242] chan_iax2.c: -- Hungup 'IAX2/ihs_trunk_out-2963' And more setup details, for those who still have the will to live:-) Asterisk version: 1.6.2.13 Internal externsions: everything on SIP - 3 Grandstream GXP-2000, 2 analog phones on a pci OpenVox card and 2 Linphone softphones Trunks: IAX2 Trunks provider: Gradwell Asterisk machine: 800Mhz Intel Pentium, 512MB of RAM Internet connection: Tiscali business ADSL I am happy to post here any config files and logs you might think would be relevant. This is not consistent - and I've managed to have 4 concurrent calls which held 30 minutes (before I hung them up) when I tried. So not easy to replicate. An INVAL response basically means that the remote Asterisk box received a packet for a call that it did not think existed. So likely, something else caused the call to hangup (such as an unrelated error crashing a process, and the replacement process had no record of such a call, so it sent an INVAL response to any subsequent packet). Technically, this could also be done as a MITM attack. If something were to see even a single packet related to the call, it is able to fake an INVAL packet. BTW, this is not unique to IAX2; a MITM attack can also fake a SIP CANCEL. Hi Tilghman and thank you for replying. I have been working on narrowing this down for a few months now - without much success. Do you have any suggestions on taking further steps to find the cause? In case it helps: 1. I use iptables to only allow in IAX2 connections from the IP addresses of the VoIP provider (Gradwell). 2. I also restrict incoming connections in iax.conf only to the same ip ranges. The drops occur randomly, once every few days normally (but there have been some cases of few drops in one day). Again, not sure if it is relevant - but here is the same log - only including few extra earlier lines. The strange thing is that, just before hanging up the call I'm interested in (2963), it seems to be hanging up another call - it reads destroying 706 - but I can't find any reference to a call 706 anywhere earlier in the log. So I can't understand what call is that, and why does it get hung-up: [Nov 17 15:25:01] DEBUG[5137] chan_iax2.c: Determining if address 212.11.91.202 with username x_in requires calltoken validation. Optional = 1 calltoken_required = 0 [Nov 17 15:25:01] DEBUG[5137] chan_iax2.c: ip callno count incremented to 1 for 212.11.91.202 [Nov 17 15:25:01] DEBUG[5132] chan_iax2.c: Immediately destroying 706, having received hangup [Nov 17 15:25:01] DEBUG[5132] chan_iax2.c: schedule decrement of callno used for 212.11.91.202 in 60 seconds [Nov 17 15:25:04] DEBUG[5138] chan_iax2.c: Immediately destroying 2963, having received INVAL [Nov 17 15:25:04] DEBUG[5138] chan_iax2.c: Destroying call 2963 [Nov 17 15:25:04] DEBUG[11242] chan_iax2.c: We're hanging up IAX2/ihs_trunk_out-2963 now... [Nov 17 15:25:04] VERBOSE[11242] chan_iax2.c: -- Hungup 'IAX2/ihs_trunk_out-2963' Any suggestion to help take this further is much appreciated. Sebastian -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] call forward problem
Hi,
I tried to perform call forward in asterisk by writing the following in the
dial plan.The data base is getting updated with the caller ID number how
ever the call is not getting forwarded.
[apps]
exten = _*21*XX,1,Set(DB(CFIM/${CALLERID(number)})=${EXTEN:4})
exten = _*21*XX,2,Hangup
exten = #21#,1,DBDel(CFIM/${CALLERID(num)})=${EXTEN:4}
exten = #21#,2,Hangup
Regards,
Aparna
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