[asterisk-users] Open source speech recognition engine?
Dear all, I am looking for an open source speech recognition engine for a hobby project. There used to be a Sphinx interface for the generic speech API (http://scribblej.com/svn/) but it does not compile on Asterisk versions later than 1.6.x Could anybody direct me on how to update this code, or should I simply change to the AGI script approach? Best regards, -- Carl-Fredrik Enell Tähteläntie 70B FIN-99600 Sodankylä, Finland - URL: http://www.is.kiruna.se/~fredrik Work URL: http://www.sgo.fi/~fredrik - -- _ -- 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] Fotos 18/08 .
11:09:12 AM Fotos 18/08..: Imagens Anexadas..: DSC_0401.jpg - DSC_0402.jpg - DSC_0403.jpg Videos Hotmail.com..: www.hotmail.com/videos.avi _ Brrr... its getting cold out there Find someone to snuggle up with http://a.ninemsn.com.au/b.aspx?URL=http%3A%2F%2Fdating%2Enz%2Emsn%2Ecom%2Fchannel%2Findex%2Easpx%3Ftrackingid%3D1048628_t=773568480_r=nzWINDOWSliveMAILemailTAGLINES_m=EXT___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help need to do Lookup from odbc database
Thanks. --- On Thu, 14/5/09, Tilghman Lesher tilgh...@mail.jeffandtilghman.com wrote: From: Tilghman Lesher tilgh...@mail.jeffandtilghman.com Subject: Re: [asterisk-users] Help need to do Lookup from odbc database To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Thursday, 14 May, 2009, 4:20 AM On Wednesday 13 May 2009 17:55:41 carl Lougher wrote: Howdy, How do i perform a lookup from a remote odbc database in the asterisk dialplan? I can do it with mysql but not sure of commands for odbc connection. See func_odbc.conf for examples. You'll also need to setup res_odbc.conf, as this is where func_odbc obtains its handles. -- Tilghman ___ -- 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
[asterisk-users] Help need to do Lookup from odbc database
Howdy, How do i perform a lookup from a remote odbc database in the asterisk dialplan? I can do it with mysql but not sure of commands for odbc connection. Cheers!!! ___ -- 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] Help with radius
Hi, I'm trying to get my Asterisk 1.4.24.1 server working with radius and aradial. I have radiusclient-ng installed and asterisk radius cdr. My cdr's fail to write to the database and i'm not sure how to authenticate each call. Anyone got this working or can offer any help. I've read all the radius docs and followed them to a tee.. Cheers!!! ___ -- 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] Parked calls for multiple customers
Ok cheers. Any idea when 1.6 goes stable for prod? - Original Message From: Mike l...@virtutel.ca To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, 24 April, 2009 0:54:59 Subject: Re: [asterisk-users] Parked calls for multiple customers No, but as I understand it 1.6 would have that possibility. Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of carl Lougher Sent: Thursday, April 23, 2009 4:54 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Parked calls for multiple customers Hi, Is there any method of getting call park working on different numbers for different customers on the same asterisk server? Currently running asterisk 1.4.23.1 Cheers!! ___ -- 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 ___ -- 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] Parked calls for multiple customers
Hi, Is there any method of getting call park working on different numbers for different customers on the same asterisk server? Currently running asterisk 1.4.23.1 Cheers!! ___ -- 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] Stun clients and canreinvite
Howdy, Scenario: Asterisk server Customer connected over internet using nat Customer phones are Linksys 942 with Stun enabled Issue: Inbound and Outbound calls work fine. But when phones call each other internally we have to carry the voice stream ie using t on dial commands. Question: Is there a better way of doing this or another way to get the media to stream internally on the customer network rather than us carrying it? We have to keep Stun on the phones to get the media to flick off on outbound calls. Cheers, Taff.. ___ -- 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] Canreinvite after media connection
Howdy, Is it possible to send a reinvite after the media has connected? Scenario: Inbound call hits asterisk ivr then is sent out to an extension using the dial command. We have to carry the rtp streams in this case as asterisk cant send the reinvite after the ivr has stopped playing the message as we already connected the call. Question: Any way around this or is there a better way we can do it? Cheers, Taff ___ -- 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] Call-limit=1 breaks attended transfer
Yeah but doesnt help for extensions that have or require call-limit=1. --- On Tue, 31/3/09, carl Lougher c_loug...@yahoo.co.uk wrote: From: carl Lougher c_loug...@yahoo.co.uk Subject: Re: [asterisk-users] Call-limit=1 breaks attended transfer To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Tuesday, 31 March, 2009, 2:20 AM We use call-limit set to 1 for hints. I guess i'll look into the dtmf method and debug the linksys phone to see what it uses for attended transfers. Cheers --- On Mon, 30/3/09, Mark Michelson mmichel...@digium.com wrote: From: Mark Michelson mmichel...@digium.com Subject: Re: [asterisk-users] Call-limit=1 breaks attended transfer To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Monday, 30 March, 2009, 10:50 PM carl Lougher wrote: Howdy, Was there ever a fix for this? I have Trix 2.6 running asterisk 1.4 and have to set an extension with call-limit=1. However that user can no longer do attended transfers from Linkys 962 ip phone. Is there anyway around this? Cheers, Taff.. Yes, set call-limit to something else :P Seriously though, there's no fix for that since it is behaving exactly as it should. When attempting to transfer the call, Asterisk has no way of knowing that the new SIP INVITE it receives (in order to call the transfer target) is an attempt to transfer the call. It appears that the same SIP peer is attempting to make a second call. Since the call-limit is set to 1, Asterisk rejects the second call attempt. I haven't tried this yet, but it may actually be possible to use DTMF transfers when the call limit is that low since Asterisk is the one that actually initiates the new call to the transfer target instead of the transferer's phone. To use DTMF transfers, you need to set a DTMF sequence in features.conf and use the 't' or 'T' flag (depending on which party should have the ability to transfer the call) in your calls to Dial() or Queue(). Why do you have the call-limit set to 1, anyway? Mark Michelson ___ -- 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 ___ -- 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] Call-limit=1 breaks attended transfer
Howdy, Was there ever a fix for this? I have Trix 2.6 running asterisk 1.4 and have to set an extension with call-limit=1. However that user can no longer do attended transfers from Linkys 962 ip phone. Is there anyway around this? Cheers, Taff.. ___ -- 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] Call-limit=1 breaks attended transfer
We use call-limit set to 1 for hints. I guess i'll look into the dtmf method and debug the linksys phone to see what it uses for attended transfers. Cheers --- On Mon, 30/3/09, Mark Michelson mmichel...@digium.com wrote: From: Mark Michelson mmichel...@digium.com Subject: Re: [asterisk-users] Call-limit=1 breaks attended transfer To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Monday, 30 March, 2009, 10:50 PM carl Lougher wrote: Howdy, Was there ever a fix for this? I have Trix 2.6 running asterisk 1.4 and have to set an extension with call-limit=1. However that user can no longer do attended transfers from Linkys 962 ip phone. Is there anyway around this? Cheers, Taff.. Yes, set call-limit to something else :P Seriously though, there's no fix for that since it is behaving exactly as it should. When attempting to transfer the call, Asterisk has no way of knowing that the new SIP INVITE it receives (in order to call the transfer target) is an attempt to transfer the call. It appears that the same SIP peer is attempting to make a second call. Since the call-limit is set to 1, Asterisk rejects the second call attempt. I haven't tried this yet, but it may actually be possible to use DTMF transfers when the call limit is that low since Asterisk is the one that actually initiates the new call to the transfer target instead of the transferer's phone. To use DTMF transfers, you need to set a DTMF sequence in features.conf and use the 't' or 'T' flag (depending on which party should have the ability to transfer the call) in your calls to Dial() or Queue(). Why do you have the call-limit set to 1, anyway? Mark Michelson ___ -- 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
[asterisk-users] Stun with hosted asterisk solution???
Howdy, I have the following issue and would like to know if anyone has got around this before. IP Phones - Linksys 942 Sip server - Asterisk 1.4.13 Stun server - Vovida Ok heres the issue. We have multiple client phones on their own network behind a natted connection. We have setup the phones to be natted and also pointing to our stun server. Now when the phones make an outside call to the PSTN stun kicks in and their rtp streams are carried from the phones to the sip provider without any issues. Now when the phones dial each other internally the rtp stream is still carried via stun and therefore fails as its pointing to the same ip on the same router. Now by adding t to the asterisk dial commands for each internal phone the inbound calls work fine but the rtp streams are carried through asterisk rather than between themselves on their network. Also in this scenario when you try conference an outside phone with an inside phone it fails due to stun and outside address problems. So my question is can we set up or change something on the phones or asterisk to allow the phones rtp to go across the local network on internal calls and via stun for outbound pstn calls? 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] Music on hold for sub tenants
This seems to be related to inbound calls. So would this work for music on transfers within that context as well as hitting the hold key on calls? --- On Fri, 26/9/08, Darrick Hartman [EMAIL PROTECTED] wrote: From: Darrick Hartman [EMAIL PROTECTED] Subject: Re: [asterisk-users] Music on hold for sub tenants To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Friday, 26 September, 2008, 4:52 AM ...since everyone else top posted. Take a look at the application setmusiconhold. CLI core show application SetMusicOnHold You can use this in a dialplan as follows: [tenant1incoming] exten = s,1,Wait(1) exten = s,n,Answer() exten = s,n,Background(tenant1sounds/welcome) exten = s,n,SetMusicOnHold(tenant1) [tenant2incoming] exten = s,1,Wait(1) exten = s,n,Answer() exten = s,n,Background(tentant2sounds/welcome) exten = s,n,SetMusicOnHold(tenant2) Use that with the previously supplied info. Darrick carl Lougher wrote: Hi, I tried this but it still uses the default moh. Is there some way to define it based on a context in the sip.conf or extensions.conf??? Taff... --- On Fri, 26/9/08, Nhadie [EMAIL PROTECTED] wrote: From: Nhadie [EMAIL PROTECTED] Subject: Re: [asterisk-users] Music on hold for sub tenants To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Friday, 26 September, 2008, 4:10 AM Hi, i think you can define it like this: [moh-company-a] mode=files directory=/var/lib/asterisk/moh/companya [moh-company-b] mode=files directory=/var/lib/asterisk/moh/companyb regards, nhadie carl Lougher wrote: Howdy, Is there a way to apply a music on hold class to different context user groups? I have multiple clients on my asterisk server and they each want different music on hold. Company A Company B ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sip reload casuing issues
Howdy, Running asterisk 1.4.13 Sometime when running a sip reload the clients are unable to make and receive calls.. Any pointers? No errors in debug or asterisk console so far.. Cheers, Taff.. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Music on hold for sub tenants
Howdy, Is there a way to apply a music on hold class to different context user groups? I have multiple clients on my asterisk server and they each want different music on hold. Company A Company B Any help much appreciated.. Thanks, Taff... ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Monitoring simul calls
Howdy, Running asterisk 1.4 Is there a way to check the simultaneous sip calls in asterisk and display with mrtg or some graphing app??? Also is there a way to segregate these based on extension or context? Cheers, Taff.. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music on hold for sub tenants
Hi, I tried this but it still uses the default moh. Is there some way to define it based on a context in the sip.conf or extensions.conf??? Taff... --- On Fri, 26/9/08, Nhadie [EMAIL PROTECTED] wrote: From: Nhadie [EMAIL PROTECTED] Subject: Re: [asterisk-users] Music on hold for sub tenants To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Friday, 26 September, 2008, 4:10 AM Hi, i think you can define it like this: [moh-company-a] mode=files directory=/var/lib/asterisk/moh/companya [moh-company-b] mode=files directory=/var/lib/asterisk/moh/companyb regards, nhadie carl Lougher wrote: Howdy, Is there a way to apply a music on hold class to different context user groups? I have multiple clients on my asterisk server and they each want different music on hold. Company A Company B Any help much appreciated.. Thanks, Taff... ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to change Internal and external callerid
Howdy, Whats the best way to change the callerid for internal and external calls. At the moment using callerid- Fred 04412345 sends callerid as Fred 04412345 for internal calls when his internal extension is 200. How can i change the callerid for internal calls but also keep the specific external callerid for PSTN calls??? Much appreciated!!! Taff... __ Sent from Yahoo! Mail. More Ways to Keep in Touch. http://uk.docs.yahoo.com/nowyoucan.html ___ -- 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 call quality detection
Hi Chaps, Is there a way to detect/highlight poor quality voice calls going through an asterisk server? Was thinking of picking up a cdr record or some other variable showing poor quality on calls when the internet is having issues. Is there any qos or poor audio quality variables available? Cheers, Taff. ___ Yahoo! Answers - Got a question? Someone out there knows the answer. Try it now. http://uk.answers.yahoo.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] Asterisk call quality detection
Any explaination as to what it is, does it work and how to setup? Is the vnak found in the logs and is it only represented for iax calls? - Original Message - From: Henry Cobb [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, June 06, 2007 12:07 PM Subject: Re: [asterisk-users] Asterisk call quality detection On 6/6/07, carl Lougher [EMAIL PROTECTED] wrote: Hi Chaps, Is there a way to detect/highlight poor quality voice calls going through an asterisk server? Was thinking of picking up a cdr record or some other variable showing poor quality on calls when the internet is having issues. Is there any qos or poor audio quality variables available? I chart VNAKs per hour. -HJC ___ --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] Asterisk call quality detection
I've noticed that there is the odd vnak message displayed in my asterisk syslog traces. Would have to alert on those i'd assume.. - Original Message - From: Matt To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Wednesday, June 06, 2007 1:44 PM Subject: Re: [asterisk-users] Asterisk call quality detection I chart VNAKs per hour. Would you care to share how you accomplish this? What programs do you use? -- ___ --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] Re: Verizon Interconnection
I've connected to Verizon BRI circuits and had major echo issues. Moved to a Paetec PRI and bing all calls now work great. - Original Message - From: Klaverstyn, David C To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Wednesday, June 06, 2007 1:47 PM Subject: RE: [asterisk-users] Re: Verizon Interconnection I have connected with a PRI service with Verizon but not SIP. What is their SIP product as I am not familiar with it? -- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Sent: Wednesday, 6 June 2007 9:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; Commercial and Business-Oriented Asterisk Discussion Subject: [asterisk-users] Re: Verizon Interconnection So absolutely no one here was interconnected with Verizon? I am going to shoot this over to asterisk-biz, also, in hopes someone may have missed it that is on the biz list. The question again is: Has anyone on this list connected with Verizon's SIP product? We are currently undergoing interop testing with Verizon, and honestly, it seems like the most convoluted process. I'd be interested in talking with someone else who has gone through this and run a few things past you. -- ___ --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] Clicking Noise on Pure Voip Calls
Setup: Asterisk server in NY. Cisco 7960 IP Phones in NY and London. Dedicated T1 from NY to Ldn. T1: Latency - 100ms Qos applied No errors Default codec on Ldn IP Phones = g711alaw Default codec on NY IP Phones = g711ulaw Both codecs allowed on each phone. Issue: Calls on IP Phones from NY to London hear clicking noise on NY end. Anyone experienced something similar or can offer some assistance? Thanks, Taf.. Send instant messages to your online friends http://uk.messenger.yahoo.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
[asterisk-users] Choppy calls on IAX trunk but no problems on internal calls
We are running the default asterisk package on Ubuntu Dapper. Our connection to the PSTN is over an IAX trunk with our provider. We are getting really bad call quality on calls over the IAX trunk--voice seems to be garbled or out of order and often completely breaks up. But on internal calls between extensions, even with call recording turned on, which goes through our asterisk server, everything sounds fine. We also have some test SIP accounts with our provider. Phones connected directly to our provider on these accounts have no problem either, so we are confident that our network conditions are good and QoS is working properly. We are also confident that our provider is not the problem, since the phones that connect directly to our provider are working fine. We thought the problem might be hardware related, so we tried three different machines on it, each with adequate CPU, memory and disk performance. Every machine had the same problem. One of the machines we borrowed from our provider. They were using it with a hardware PRI and said their zttest results were consistenly 99.99 or greater and the server had performed great for them. But with our Ubuntu installation and no hardware, the same server gets results around 99.92. In fact, every one of the machines we tried got fairly bad zttest results, although we have discovered various info that indicate that zttest might not be a very accurate test (http://bugs.digium.com/view.php?id=4301 ___ --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] Choppy calls on IAX trunk but no problems on internal calls
We are running the default asterisk package on Ubuntu Dapper (which has the advanced timing options used by ztdummy). Our connection to the PSTN is over an IAX trunk with our provider. We are getting really bad call quality on calls over the IAX trunk--voice seems to be garbled or out of order and often completely breaks up. But on internal calls between extensions, even with call recording turned on, which goes through our asterisk server, everything sounds fine. We also have some test SIP accounts with our provider. Phones connected directly to our provider on these accounts have no problem either, so we are confident that our network conditions are good and QoS is working properly. We are also confident that our provider is not the problem, since the phones that connect directly to our provider without going through our asterisk server are working fine. We thought the problem might be hardware related, so we tried three different machines on it, each with adequate CPU, memory and disk performance. Every machine had the same problem. One of the machines we borrowed from our provider. They were using it with a hardware PRI and said their zttest results were consistenly 99.99 or greater and the server had performed great for them. But with our Ubuntu installation and no hardware, the same server gets results around 99.92. In fact, every one of the machines we tried got fairly bad zttest results, although we have discovered various info that indicate that zttest might not be a very accurate test (http://bugs.digium.com/view.php?id=4301), but it is the only benchmark we know of. We suspect there may be a problem with with the build options in the kernel or in the default asterisk package on dapper, so we are trying out trixbox at the moment. In the mean time, does anyone else have any suggestions? Are there some specific build options or kernel flags we should try? Are there any other approaches that someone might recommend? Thanks in advance for your time. Carl ___ --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] Do zttest results matter without telephony hardware?
I'm running an asterisk server that uses an IAX2 trunk with our voip provider for its PSTN gateway. We have no telephony hardware in our server. We are consistently getting 99.975% or better when we run zttest. I have heard that this is bad in some cases, but I'm wondering if it matters, since we are not using any telephony hardware. Can somebody please clear this up for us? Do zttest results matter when you don't have any telephony hardware in your system? We think there may be some call quality issues with out provider, but whenever we report them they turn around and say that our zttest results are bad and that our server is probably the source of the problem. However, our calls between extensions are superb, so I can't believe that our server is to blame, since only calls to and from the PSTN seem to have problems. Thanks in advance for your time. Carl ___ --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] Macro help needed!!!!
Upgrading to ver 1.2.10 fixed it.
--- carl Lougher [EMAIL PROTECTED] wrote:
Hi,
Need to get the following working:
1. User calls ext 750.
2. If no answer or busy go elsewhere.
3. If answered and press 1 accept call.
4. If answered and not pressed 1 or timed out then
send call to be redirected to the busy or no answer
option.
The issue is that the call gets accepted if any
number
is pressed or a timeout. How do i throw the call
back
out of the macro???
asterisk ver 1-0-9
[sip-clients]
exten = 750,1,Dial(SIP/225|60|gM(mobile))
exten = 750,2,GotoIf($[${DIALSTATUS} = BUSY]?7:3)
exten = 750,3,GotoIf($[${DIALSTATUS} = NOANSWER]?7)
exten = 750,4,Hangup()
exten = 750,7,Dial(SIP/226,15,t)
[macro-mobile]
exten = s,1,DigitTimeout(4)
exten = s,2,ResponseTimeout(5)
exten = s,3,Read(ACCEPT|press one now to accept|1)
;
exten = s,4,GotoIf($[${ACCEPT} = 1]?5:6)
exten = s,5,SetVar(MACRO_RESULT=CONTINUE)
exten = s,6,Hangup()
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[asterisk-users] Macro help needed!!!!
Hi,
Need to get the following working:
1. User calls ext 750.
2. If no answer or busy go elsewhere.
3. If answered and press 1 accept call.
4. If answered and not pressed 1 or timed out then
send call to be redirected to the busy or no answer
option.
The issue is that the call gets accepted if any number
is pressed or a timeout. How do i throw the call back
out of the macro???
asterisk ver 1-0-9
[sip-clients]
exten = 750,1,Dial(SIP/225|60|gM(mobile))
exten = 750,2,GotoIf($[${DIALSTATUS} = BUSY]?7:3)
exten = 750,3,GotoIf($[${DIALSTATUS} = NOANSWER]?7)
exten = 750,4,Hangup()
exten = 750,7,Dial(SIP/226,15,t)
[macro-mobile]
exten = s,1,DigitTimeout(4)
exten = s,2,ResponseTimeout(5)
exten = s,3,Read(ACCEPT|press one now to accept|1) ;
exten = s,4,GotoIf($[${ACCEPT} = 1]?5:6)
exten = s,5,SetVar(MACRO_RESULT=CONTINUE)
exten = s,6,Hangup()
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[asterisk-users] Finding far end echo in Verizon network
This is a weird one. Network: Asterisk ver 1-0-9 on DL360. 10 Cisco 7960g phones with 3.8.2 SIP Load. Gateway - Cisco 2811 router with 4 x verizon bri's. Network - Private vlan with 1ms response times to all devices. Issue: Intermittent echo on outbound/inbound calls. Users hearing their own voice about 0.5sec later. Tried so far: Upgraded firmware on some phones to 3.8.2 Upgraded software on Cisco router. Changed gain and attentuation settings on cisco router Got Verizon to test bris Moved rtp from asterisk direct to phone and router (canreinvite=yes) load tested asterisk None of the above made any difference. They are hearing their own voice so that means the issue is on the far end. But should it be up to me to control the possible delay or slippage in the verizon bri network? Any help much appreciated. Taf. ___ All new Yahoo! Mail The new Interface is stunning in its simplicity and ease of use. - PC Magazine http://uk.docs.yahoo.com/nowyoucan.html ___ --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] External call press 1
Hi,
Running asterisk ver 1-0-9
Trying to send a call to a mobile phone and playback a
message to the user to press one to accept the call.
If 1 isn't pressed then the call needs to be re-routed
back into the asterisk dialplan.
Tried various macros etc but if one isn't pressed the
call still gets accepted?
Any clues???
exten = 333,1,Macro(test)
exten = 333,2,Hangup
exten = 334,1,Dial(SIP/XXX)
[macro-test]
exten = s,1,Wait(1)
exten = s,2,Read(ACCEPT|press-one |1)
exten = s,3,GotoIf($[${ACCEPT} = 1 ]?4:5)
exten = s,4,NoOp(Caller accepted)
exten = s,5,Goto(client,334,1)
exten = i,1,Set(MACRO_RESULT=CONTINUE)
exten = t,1,Set(MACRO_RESULT=CONTINUE)
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Re: [Asterisk-Users] Problem trying to SayDigits when an invalid
Thanks Doug, but this would not have helped me. Fortunately the
${INVALID_EXTEN} response was exactly what I needed, but your
suggestion would not have worked, because if an extension is found, it
no longer goes to exten = i. So the variable setting approach ends
up altering the flow of the dialplan.
On 6/16/06, Doug Lytle [EMAIL PROTECTED] wrote:
Carl Youngblood wrote:
No, ${EXTEN} contains i at that point in the dialplan.
exten = 123,1,Set(_TMPEXTEN=${EXTEN})
exten = i,1,SayDigit({$TEMPEXTEN})
You need to read the document in the Asterisk source directory on the
subject of variable inheritance.
asterisk/doc/README.variables.
Doug
--
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deserve neither Liberty nor Safety.
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Re: [Asterisk-Users] Problem trying to SayDigits when an invalid extension is dialed
No, ${EXTEN} contains i at that point in the dialplan.
On 16 Jun 2006 01:41:30 -, [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
Does SayDigits(${EXTEN}) not work in this case? I would imagine that it would
still maintain the dialled extension in that variable, would it not?
Undrhil
--- Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
wrote:
I am trying to modify a fairly complex digital receptionist dialplan
that has a number of included contexts. Right now the system is not
announcing the extension that the caller attempted to dial, so callers
get confused when they think they dialed a valid extension but
asterisk
didn't pick everything up. I would like to have the system
announce the
entension that they attempted to dial in addition to the
error message.
However, at the part where the error announcement is
made, the extension
is set to i, so I no longer know what digits the
caller dialed. I tried
inserting a wildcard extension before this
point that saves the dialed
digits in a variable, but since my
wildcard extension matches everything,
it no longer things that an
invalid extension was dialed, so it doesn't
go to the i extension.
Is there a way I can erase the fact that an extension
was matched? Or
is there some other way of accomplishing what I am trying
to do?
Thanks,
Carl
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Re: [Asterisk-Users] Need to track dropped calls
We use IAX for PSTN connectivity. Our phones are Linksys SPA-942s running SIP. We are connecting to our voip provider over the UTOPIA municipal network in SLC. Our network connectivity has been great until about two weeks ago, when we started to experience 5-10% packet loss due to a router malfunction or something. The packet loss seems to have gone away, but I just want to make sure that I am attacking these problems from all angles. I just wanted to know if there was some way I could examine the asterisk logs to see if any calls terminated abnormally. I wanted to put this into our monitoring system so that I could track these issues better. On 6/14/06, Steve Totaro [EMAIL PROTECTED] wrote: Little or no complaints means everything is working. Are your extensions IAX softphones or do you use IAX for PSTN connectivity? What network problems are we talking about? How about taking initiative and creating a user survey and sending it to everyone? You can see IAX stats on the CLI, have you looked at any of that? Have you run with IAX debugging turned on? Sounds like you are trying to build a solid house in the swamp. Get the foundation good and proper and take it from there. Thanks, Steve Totaro Carl Youngblood wrote: Of course I'm trying to deal with the network problems, but it's nice to have another method of verifying that everything is working. Frequently there are people who don't complain, so we don't realize that their call quality is sub-par. We are using iax. It seems like there should be a record somewhere if a call was terminated abnormally. Thanks, Carl On 6/14/06, Steve Totaro [EMAIL PROTECTED] wrote: Carl Youngblood wrote: I have been getting occasional reports of dropped calls from the users of our asterisk system. Is there anything I can monitor in my logs or in the console to see when a call is dropped? I'd like to see if these drops coincide with network traffic problems. Thanks, Carl If you have network traffic problems, address those first and then wait to hear of more dropped calls. Depending on your setup, you could use the monitor app to determine whether a call was actually dropped or a hangup by the context of the conversation. In a call center environment, dropped calls are quite frequently people hanging up for whatever reason and you can actually hear the handset go into the cradle (sometimes slam lol.) If it is happening alot and you have a pri, you can watch your pri debug span info and see if you are having errors or normal clearings If you are using IAX, maybe the jitterbuffer stats might indicate something. ___ --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] Problem trying to SayDigits when an invalid extension is dialed
Thank you! Thank you! I had been trying all sorts of convoluted ways
to get that information. That was very easy.
On 6/16/06, Lacy Moore - Aspendora [EMAIL PROTECTED] wrote:
http://www.voip-info.org/wiki/index.php?page=Asterisk+Variables
Use ${INVALID_EXTEN}
On 6/15/06, Carl Youngblood [EMAIL PROTECTED] wrote:
I am trying to modify a fairly complex digital receptionist dialplan
that has a number of included contexts. Right now the system is not
announcing the extension that the caller attempted to dial, so callers
get confused when they think they dialed a valid extension but
asterisk didn't pick everything up. I would like to have the system
announce the entension that they attempted to dial in addition to the
error message. However, at the part where the error announcement is
made, the extension is set to i, so I no longer know what digits the
caller dialed. I tried inserting a wildcard extension before this
point that saves the dialed digits in a variable, but since my
wildcard extension matches everything, it no longer things that an
invalid extension was dialed, so it doesn't go to the i extension.
Is there a way I can erase the fact that an extension was matched? Or
is there some other way of accomplishing what I am trying to do?
Thanks,
Carl
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[Asterisk-Users] Problem trying to SayDigits when an invalid extension is dialed
I am trying to modify a fairly complex digital receptionist dialplan that has a number of included contexts. Right now the system is not announcing the extension that the caller attempted to dial, so callers get confused when they think they dialed a valid extension but asterisk didn't pick everything up. I would like to have the system announce the entension that they attempted to dial in addition to the error message. However, at the part where the error announcement is made, the extension is set to i, so I no longer know what digits the caller dialed. I tried inserting a wildcard extension before this point that saves the dialed digits in a variable, but since my wildcard extension matches everything, it no longer things that an invalid extension was dialed, so it doesn't go to the i extension. Is there a way I can erase the fact that an extension was matched? Or is there some other way of accomplishing what I am trying to do? Thanks, Carl ___ --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] Need to track dropped calls
I have been getting occasional reports of dropped calls from the users of our asterisk system. Is there anything I can monitor in my logs or in the console to see when a call is dropped? I'd like to see if these drops coincide with network traffic problems. Thanks, Carl ___ --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] Need to track dropped calls
Of course I'm trying to deal with the network problems, but it's nice to have another method of verifying that everything is working. Frequently there are people who don't complain, so we don't realize that their call quality is sub-par. We are using iax. It seems like there should be a record somewhere if a call was terminated abnormally. Thanks, Carl On 6/14/06, Steve Totaro [EMAIL PROTECTED] wrote: Carl Youngblood wrote: I have been getting occasional reports of dropped calls from the users of our asterisk system. Is there anything I can monitor in my logs or in the console to see when a call is dropped? I'd like to see if these drops coincide with network traffic problems. Thanks, Carl If you have network traffic problems, address those first and then wait to hear of more dropped calls. Depending on your setup, you could use the monitor app to determine whether a call was actually dropped or a hangup by the context of the conversation. In a call center environment, dropped calls are quite frequently people hanging up for whatever reason and you can actually hear the handset go into the cradle (sometimes slam lol.) If it is happening alot and you have a pri, you can watch your pri debug span info and see if you are having errors or normal clearings If you are using IAX, maybe the jitterbuffer stats might indicate something. ___ --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] Are zttest results relevant on a system with no telephony hardware?
Our asterisk system gains access to the PSTN through a voip provider. We have no digium or other telephony hardware in our system. Do the zttest results still matter to us? Our results were as follows: --- Results after 1007 passes --- Best: 100.00 -- Worst: 99.780273 -- Average: 99.975763 ___ --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] Are zttest results relevant on a system with no telephony hardware?
Thanks. What is it in the 2.6.13-based kernel that improves timing? Should I expect to see a significant improvement if I upgrade to it? On 6/13/06, Mike Fedyk [EMAIL PROTECTED] wrote: IAX trunking and meetme conferences are some of the heaviest users of zaptel timing. I'd suggest if you don't have hardware timing (or at least a 2.6.13 based kernel), then use SIP all the way or at least turn off IAX trunking. ___ --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: Cell phone dialed digits too short to be recognized by asterisk
FYI, the digit timeout was simply too short in my IVR. After increasing that everything worked fine. This problem only showed up on cell phones, many of which don't allow you to type long digits so your keypresses have more silence in between them. On 5/12/06, Carl Youngblood [EMAIL PROTECTED] wrote: I'm having a big problem where digits dialed from certain cell phones are too short to be recognized by my asterisk server. I'm running AAH 2.8. Some cell phones don't allow the caller to hold down the digits and have the tones play as long as they hold them down for. They just play a short tone no matter how long you hold down the digits for. Has anyone run into this before, and if so what did you do about it? This is my larger problem but I have a smaller problem related to it. I'm trying to make the IVR play back the number it thinks the user dialed so that they can at least try again. But I'm having a hard time figuring out which asterisk variable contains the dialed digits. This seems like it should be pretty basic, but my research on voip-info hasn't turned up much. All I could find was some commentary on how DIALEDPEERNUMBER is supposed to hold the value but mysteriously doesn't. Thanks in advance for your help. Carl ___ --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] Cell phone dialed digits too short to be recognized by asterisk
I'm having a big problem where digits dialed from certain cell phones are too short to be recognized by my asterisk server. I'm running AAH 2.8. Some cell phones don't allow the caller to hold down the digits and have the tones play as long as they hold them down for. They just play a short tone no matter how long you hold down the digits for. Has anyone run into this before, and if so what did you do about it? This is my larger problem but I have a smaller problem related to it. I'm trying to make the IVR play back the number it thinks the user dialed so that they can at least try again. But I'm having a hard time figuring out which asterisk variable contains the dialed digits. This seems like it should be pretty basic, but my research on voip-info hasn't turned up much. All I could find was some commentary on how DIALEDPEERNUMBER is supposed to hold the value but mysteriously doesn't. Thanks in advance for your help. Carl ___ --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] How to determine if a device is in a call
Thanks to everyone who responded. I was able to modify the freepbx
paging code to use something like the suggested macro and it worked
well. For those who may be interested, the following Page macro works
for Linksys SPA942 phones:
[macro-page];
;
; Paging macro:
;
; Check to see if SIP device is in use and DO NOT PAGE if they are
;
; ${ARG1} - Device to page
exten = s,1,ChanIsAvail(${ARG1}|js) ; j is for dump and s is for ANY call
exten = s,2,Set(__SIPADDHEADER=Call-Info: \;answer-after=0)
exten = s,3,Set(__ALERT_INFO=Ring Answer)
exten = s,4,Set(__SIP_URI_OPTIONS=intercom=true)
exten = s,5,SIPAddHeader(Call-Info: \;answer-after=0) ; This is for
the Snoms and Others
exten = s,6,Dial(${ARG1}||)
exten = s,7,Hangup
exten = s,102,Hangup
On 5/6/06, Alexander Lopez [EMAIL PROTECTED] wrote:
See:
http://www.sineapps.com/news.php?rssid=1130
snip...
I have gotten intercom working on my office phones (Linksys SPA-942s),
but I have noticed that if someone is in a call, it places the call on
hold and sends the intercom audio to the person holding the phone that
is being paged. I'd like to add logic to my dialplan that doesn't
send a page to a phone that is currently in a call. But to do this I
need a function that will tell me if a device is in a call. Any
suggestions?
Thanks,
Carl
Snip.
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[Asterisk-Users] How to determine if a device is in a call
I have gotten intercom working on my office phones (Linksys SPA-942s), but I have noticed that if someone is in a call, it places the call on hold and sends the intercom audio to the person holding the phone that is being paged. I'd like to add logic to my dialplan that doesn't send a page to a phone that is currently in a call. But to do this I need a function that will tell me if a device is in a call. Any suggestions? Thanks, Carl ___ --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] Trying to set up automatic announcement upon transfer for IVR in AAH 2.8
I am running AAH 2.8. I have an IVR for our main phone number that allows users to dial an extension directly. I would like to have a this call may be recorded announcement played before the call gets transferred. There is not a built-in option for this in the IVR web interface, but one way I can do this is to create ring groups for each user with announcements and modify the dialplan to dial the ring groups instead of the extensions. The question is, where do I do this? What part of the dialplan should I modify to make it substitute a ring group for the dialed-in extension? Sorry to post on the asterisk users list, I know AAH is not exactly related, but there is something wrong on their forum right now. I can't post there, even though I'm logged into sourceforge. Thanks, Carl ___ --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] API or Call command
Is it possible to send an API command to dial an extension and playback a specific announcement using application and appdata commands. Scenario: User adds different announcements daily (can't used fixed name for Playback file). Call command dials user and plays back specific announcement message. I can do this manually by using the same Playback file name each time but is possible to specify the playback file to be played in the API command??? Any help much appreciated... ___ --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] Multiple ISDN BRI Units with Asterisk using Bristuff zaphfc in NT mode?
Maybe this is rather a hardware question, but I am posting it on this list because the probability of someone else of you having tried this is greater here than other places I can think of. I have an ISDN card that is setup in NT mode using the zaphfc driver in bristuff, and I got it working perfectly with one ISDN phone using a crossover cable and 100 ohm termination at the end of the cable. However, if I connect one more ISDN device to the ISDN bus both devices stop working, so the question is: Is it only possible to use one device with a HFC card in NT mode or is there something else I need to do first to make it work with two devices? -- Greetings, Carl Andersson a.k.a Zaphod Beeblebrox -=*=- FwD VOIP: 645040 - IaxTel VOIP: 17006906266 -=*=- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Multiple ISDN BRI Units with Asterisk usingBristuff zaphfc in NT mode?
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Zoltan Szecsei Sent: zaterdag 16 juli 2005 12:54 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Multiple ISDN BRI Units with Asterisk usingBristuff zaphfc in NT mode? Maybe this is rather a hardware question, but I am posting it on this list because the probability of someone else of you having tried this is greater here than other places I can think of. I have an ISDN card that is setup in NT mode using the zaphfc driver in bristuff, and I got it working perfectly with one ISDN phone using a crossover cable and 100 ohm termination at the end of the cable. However, if I connect one more ISDN device to the ISDN bus both devices stop working, so the question is: Is it only possible to use one device with a HFC card in NT mode or is there something else I need to do first to make it work with two devices? Hi Carl, I just started yesterday afternoon with exactly the same setup so you are a bit ahead of me. If anyone answers you directly then please be kind enough to forward their comments to me. I have not even tried to sort out trunks, bristuff or anything yet but it might be worth pointing out that my initial problems were that,using an old pIII motherboard with pci graphics, network plus 2 HFC bri isdn cards, I ran out of IRQs. I had to lock down exclude the irq for the network card before the 2 ISDN cards woke up. I now have the network 1 ISDN card on their own IRQ and the graphics and 2nd ISDN card sharing an IRQ. Maybe this could be a similar problem for you? I'm using this HW with AAH 1.1 Keep in touch, Cheers, Zoltan My problem turned out to be a termination problem. When using zaphfc together with other zap cards, it seems to be of importance in which order the drivers are loaded as well - At least in my case it would only work right if the X100P driver was loaded before the zaphfc driver. I have got verything working now, so if you have any questions you are more than welcome. You didn't write if you intended to use the ISDN cards to connect to ISDN lines, or if you wanted to create a setup like mine, with the card/cards in NT mode, acting as an ISDN switch of it's own. -- Greetings, Carl Andersson a.k.a Zaphod Beeblebrox Email: [EMAIL PROTECTED] - ICQ: 1705837. -=*=- FwD VOIP: 645040 - IaxTel VOIP: 17006906266 -=*=- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Multiple ISDN BRI Units with Asterisk usingBristuff zaphfc in NT mode?
Try terminating using 50 ohm resistors as suggested by this guide: http://home.foni.net/~jolly1/download/PBX4Linux-2.5.html in chapter 2.2 (Connect ISDN telephones to your ISDN card.) Best regards, Jan Snelders I did something along the lines of that, and it works great now. But instead of terminating with 50 Ohm at one end of the line, I put 100 Ohm termination in both ends of the line... Thanks for the help! -- Greetings, Carl Andersson a.k.a Zaphod Beeblebrox Email: [EMAIL PROTECTED] - ICQ: 1705837. -=*=- FwD VOIP: 645040 - IaxTel VOIP: 17006906266 -=*=- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Serial port DTMF Caller-ID reciever /w scripts for X100P/X101P/Clones....
Since the X100P/X101P/Clone cards does not work in all countries that
use DTMF based
Caller-ID systems, I've developed a hardware that you connect to a
serial port and the PSTN.
You then run a perl script cid_logger.pl as a daemon, and modify
extensions.conf to call
an agi script whenever a call comes in, and if it's on the X100 card it
will get the caller id
information collected by my cid daemon and return to asterisk in the
CALLERID variable.
get_cid.pl should be but in asterisk agi-bin directory, and the
following modifications should
be implemented in extensions.conf:
[from-pstn-reghours]
exten = s,1,GotoIf($[${FAX_RX} =
disabled]?from-pstn-reghours-nofax,s,1:2)
exten = s,2,Answer
exten = s,3,agi,getcid.pl
;exten = s,3,Wait(1)
[from-pstn-afthours]
exten = s,1,GotoIf($[${FAX_RX} =
disabled]?from-pstn-afthours-nofax,s,1:2)
exten = s,2,Answer
exten = s,3,agi,getcid.pl
;exten= s,3,Wait(1)
All plans and scripts are available under GPL license here:
http://www.area51.org.il/~zaphodb/asterisk/astcid/astcid.zip
A picture of the finished project:
http://www.area51.org.il/~zaphodb/asterisk/astcid/DSCN0906.JPG
The circuit board layout was made with an old version of Tango PCB, but I've
included Postscript versions of both the silkscreen and the back of the board.
--
Greetings, Carl Andersson a.k.a Zaphod Beeblebrox
-=*=- FwD VOIP: 645040 - IaxTel VOIP: 17006906266 -=*=-
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Re: [Asterisk-Users] X100P connected as extension to Panasonic 616 EASA-PHONE
So what you're saying is, you can make inbound/outbound calls on asterisk using an X100P connected to the PSTN. Then you unplug the X100P from the PSTN and plug it into a Panasonic 616 PABX and suddenly nothing works? Is that correct? And this is a problem with asterisk because.? I think you'll find the Panasonic box needs some attention... Jamie Guillermo Salas M wrote: Hi all. I`ve installed a X100P on my box and is working well with incoming and outgoing calls as a trunk with one PTSN line. I want to connect the X100P to my Panasonic 616 EASA-PHONE as an internal extension to permit to users to make calls to SIP devices from analog phones, the problem is when I dial the ext number where the X100P is connected I get busy tone. What config I need to change to my asterisk files to permit the commented in the last paragraph? Best Regards, Guillermo. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Jamie Carl Resident Geek Achieve, Corp. 82 Wentworth Ave Kingston ACT 2604 PO Box 4833 KINGSTON ACT 2604 T 1300 139 215 M 0413 955 956 F 1800 988 000 W www.achievecorp.com.au ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Nasty little incident ...
___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Jamie Carl [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and grandstream weird call probs
Hey all. I've got a weird problem with the grandstream budgetone101 and asterisk that I'm having no luck finding any info on. I'm positive it's a grandstream problem but i'm hoping someone here can at least point me in the right direction. Basically, (and it's a simple problem) if a user taps the hook switch quickly they get dialtone again but it does not hangup the existing call. The user can then make another call, however, i have incominglimit=1 in sip.conf so they cannot. This means the original call get's lost. Does anyone know how to retrieve the call? Or at least where there is some documentation on this 'feature'? TIA -- Jamie Carl [EMAIL PROTECTED] Resident Geek Achieve Corp. +61262648200 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] how to make a dialplan on bases of Caller
Maybe have your uas registered in different contexts for outbound calling. Then have those contexts only available to dial the appropriate gw? Just a thought. I know you could probably do this with wildcard source routing but that seems like overkill. Jamie On Tue, 2005-06-14 at 18:09 -0700, Kamran Ahmad wrote: Hello i have two GWs and some uas. i want if ua (bw 3000 to 4010) is calling any number then this call will be routed to first GW and if ua (bw 4020 to 5000) want to call any number this call will be routed to second GW. Gateways=GW1,GW2 UAs=3000 to 5000 if 3000 wants to call any number ip or pstn then Dial(GW1), if 4500 want to call any number ip or pstn then Dial(GW2) thanks Kamran __ Discover Yahoo! Have fun online with music videos, cool games, IM and more. Check it out! http://discover.yahoo.com/online.html ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Jamie Carl [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Supervised/Attended transfers (working, but more Qs)
Ok, i just got it working with a CVS version. Cool. :) Answered my own question. One more question tho Will it only work with the keys defined in features.conf or is it possible to get it to work using the 'transfer' button on my grandstream budgetone101? I can already hear my users complaining that they'd like attended xfers using the transfer button. :( At the moment i have it mapped to *2 (the default). Is this possible with the budgetones? Jamie Jamie wrote: Hey all, I've been trying to get supervised transfers working without success. I'm currently running 1.0.7-stable and think it might be a version problem. Is the supervised transfer feature available in 1.0.7 or do i need to suck down a new version from CVS? Otherwise, apart from setting up features.conf, is there anything else i'm missing? TIA, Jamie. -- Jamie Carl Resident Geek Achieve, Corp. 82 Wentworth Ave Kingston ACT 2604 PO Box 4833 KINGSTON ACT 2604 T 1300 139 215 M 0413 955 956 F 1800 988 000 W www.achievecorp.com.au ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Nortel i2004 firmware upgrade.
On Wednesday, 25 May, 2005 19:13 : [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: I've been trying to look up information on upgrading firmware on a nortel i2004 ip phone. I have this phone leftover from a trial, and it's supposed to be upgradable to current firmwares. Since I also run a DMS I was able to login to nortel's site and get all the firmware files, but All the NTP's regarding firmware upgrading these are how to tell you BCM to send the file to it. I'm trying to use this with asterisk, and was wondering if any of you reading would have information like that. Well if someone have BCM + i2004 + firmwares, they can send me a dump of the network traffic. If it's ok, the firmware update will be available on the next version of chan_unistim. -- Carl ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] UNISTIM channel driver available
Hello, Cedric Hans has released an UNISTIM channel driver for asterisk (stable). You can download it at : http://mlkj.net/asterisk/chan_unistim-0.9.2.tar.bz2 Copy of README : This is a channel driver for Unistim protocol. You can use at least Nortel i2004 phones with it. Only few features are supported : Send/Receive CallerID, Redial, SoftKeys, SendText(), Music On Hold, Message Waiting Indication (MWI). Install : - This version works on asterisk stable (tested with 1.0.5 and 1.0.6) - tar xvjf chan_unistim-0.9.2.tar.bz2 cd chan_unistim-0.9.2 - make make install make config (should work with a default install of asterisk) - edit /etc/asterisk/unistim.conf - start asterisk How to configure i2004 phones : - Power on the phone - Wait for message Nortel Networks - Press quickly the four buttons just below the LCD screen, from left to right - If you see Locating server, power off and try again - DHCP : 0 - SET IP : a free ip of your network - NETMSK / DEF GW : netmask and default gateway - S1 IP : ip of the asterisk server - S1 PORT : 5000 - S1 ACTION : 1 - S1 RETRY COUNT : 10 - S2 : same as S1 Issues : - As always, NAT can be tricky. If a phone is behind a NAT, you should port forward UDP 5000 (or change [general] port= in unistim.conf) and UDP 1 (or change [yourphone] rtp_port=) - Only one phone per public IP (multiple phones behind the same NAT don't work). Setup a VPN if you want to do that. - If asterisk is behind a NAT, you must set [general] bindaddr= with your public IP. If you don't do that or the bindaddr is invalid (or no longer valid, eg dynamic IP), phones should be able to display messages but will be unable to send/receive RTP packets (no sound) - Don't forget : this work is based entirely on a reverse engineering, so you may encounter compatibility issues. At this time, I know three ways to establish a RTP session. You can modify [yourphone] rtp_method= with 0, 1 or 2. 0 is the default method, should work. 1 can be used on new firmware (black i2004) and 2 on old violet i2004. - If you have difficulties, try unistim debug and set verbose 3 on the asterisk CLI. For extra debug, uncomment #define DUMP_PACKET 1 and recompile chan_unistim. TODO : - bridging. Beware, it's a dangerous feature with the picky RTP stack of i2004 phones. RTP peer must send packets with a precise timing (every 20ms, 160 bytes payload for G711). - three way calling / multiple lines / transfer - call history - better codec negociation : if you enable g723/g729, although it works with asterisk in pass-thru, it's not great. - reload config ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] EICON DIVA prices
On Thursday, 24 February, 2005 12:54 : Peter Svensson [EMAIL PROTECTED] wrote: The Diva Server cards are expensive because they have on board dsp chips. Have you considered the cheaper alternative of using Junghanns quad/octobri cards for bri and Digium TE4xxP cards for PRI? Both of these use the cpu for the signal processing and are a lot cheaper. Yes, except if your main usage is faxing (in/out), in this case, use chan_capi + patch/hylafax. You'll get a reliable solution based on hardware DSP. If it's a voice only usage, buy a digium card. -- Carl ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Nortel i2004
On Monday, 14 February, 2005 02:30 : George Cohn [EMAIL PROTECTED] wrote: Stefan Gofferje wrote: Hi folks, has anybody knowledge about the Nortel i2004? Nortel calls it Internet Phone. I'm curious, which protocols it may understand... I just came back from a Nortel roadshow and was told it's H.323. Nop. This phone uses a proprietary protocol except for the audio part (RTP G711, G723 and G729). Some work are in progress right now for an UNISTIM support in asterisk. -- Carl ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Nortel i2004
On Tuesday, 15 February, 2005 02:46 : Stefan Gofferje [EMAIL PROTECTED] wrote: Carl Sempla schrieb: On Monday, 14 February, 2005 02:30 : George Cohn [EMAIL PROTECTED] wrote: Stefan Gofferje wrote: Hi folks, has anybody knowledge about the Nortel i2004? Nortel calls it Internet Phone. I'm curious, which protocols it may understand... I just came back from a Nortel roadshow and was told it's H.323. Nop. This phone uses a proprietary protocol except for the audio part (RTP G711, G723 and G729). Some work are in progress right now for an UNISTIM support in asterisk. Got even more curious. Nortel's website also says something about H.323. I sent an inquiry to them - will post the result here if anybody's interested... You're supposed to use these phones with a nortel PBX, who support H323 and SIP. If you buy an i2004 alone, it's quite useless for now. -- Carl ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Chan_Capi initial deadlock
On Friday, 04 February, 2005 14:16 : Sergio [EMAIL PROTECTED] wrote: I had applied the patch and it got much better. Now I only have problems every two days 563 usleep(1); 1 is too high you can safely lower it to sleep(1) there's a while over there otherwise it will lock the channel for 10 seconds. that's code from chan_capi fax patch right? Nop, it's in the original code. Don't forget that usleep use microseconds, not milli, so this line doesn't wait 10s. -- Carl ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Chan_Capi initial deadlock
On Thursday, 03 February, 2005 13:54 : Felix Deierlein [EMAIL PROTECTED] wrote: I had applied the patch and it got much better. Now I only have problems every two days eb 3 14:04:12 WARNING[23065]: channel.c:472 ast_channel_walk_locked: Avoided initial deadlock for 'CAPI[contr1/1429092]/279', 10 retries! Feb 3 14:04:12 WARNING[23065]: channel.c:472 ast_channel_walk_locked: Avoided initial deadlock for 'CAPI[contr1/1429092]/279', 10 retries! Feb 3 14:04:12 WARNING[23065]: channel.c:472 ast_channel_walk_locked: Avoided initial deadlock for 'CAPI[contr1/1429092]/279', 10 retries! Any idea? When you have this message, attach gdb to asterisk and type : thread apply all bt in the list, select the lastest asterisk function, just before libc, and type frame (this number). Example : (gdb) thread apply all bt Thread 13 (Thread 19469 (LWP 3466)): #0 0x4016bde1 in nanosleep () from /lib/libc.so.6 #1 0x40195e8e in usleep () from /lib/libc.so.6 #2 0x402af5f2 in capi_activehangup (c=0x40508ba8) at chan_capi.c:563 #3 0x402af7c7 in capi_hangup (c=0x40508ba8) at chan_capi.c:606 #4 0x0805945c in ast_hangup (chan=0x40508ba8) at channel.c:741 #5 0x08072b7f in ast_pbx_run (c=0x40508ba8) at pbx.c:1968 #6 0x08079036 in pbx_thread (data=0x40508ba8) at pbx.c:1980 #7 0x400200ba in pthread_start_thread () from /lib/libpthread.so.0 (gdb) frame 2 #2 0x402af5f2 in capi_activehangup (c=0x40508ba8) at chan_capi.c:563 563 usleep(1); And past the results of these commands. Good luck -- Carl ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Chan_Capi initial deadlock
On Thursday, 20 January, 2005 14:42 : Felix Deierlein
[EMAIL PROTECTED] wrote:
Jan 18 16:00:09 WARNING[2919]: Avoided initial deadlock for
'CAPI[contr1/1429092]/128', 10 retries!
2.) Patch to chan_capi
I did not tried it. The patch should solute that problems and enable
faxing? Has anybody experiences with it? If there is a problem why is
not kapejod solving that?
You should try :)
If you don't want the fax support, you can just change this line :
--- original/chan_capi.c Fri Aug 13 12:07:28 2004
+++ chan_capi/chan_capi.c Wed Oct 27 18:55:32 2004
@@ -556,7 +556,7 @@
}
}
// wait for the B3 layer to go down
- while (i-state != CAPI_STATE_CONNECTED) {
+ while ((i-state != CAPI_STATE_CONNECTED) (i-state !=
CAPI_STATE_DISCONNECTED)) {
usleep(1);
}
}
kapejod is (was ?) quite unresponsive.
--
Carl
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Re: [Asterisk-Users] Fax detection CAPI (doesn't work!)
On Tuesday, 14 December, 2004 22:17 : Humberto Aicardi [EMAIL PROTECTED] wrote: I'm currently using a ISDN-BRI with a Fritz ISDN card and the chan-capi. The problem is that the fax detection is not executed, Hi, The fax detection in chan_capi use the CAPI DTMF feature. So you need to set in /etc/asterisk/capi.conf the line softdtmf=0 Check if CFLAGS+=-DFORCE_SOFTWARE_DTMF in Makefile of chan_capi is commented (#). If you start asterisk with the -v option, you should see : CAPI[contrX] supports DTMF And obviously the card must report a DTMF when a fax tone is detected. It may be possible to alter the code and use the asterisk dsp code instead. Regards, -- Carl ___ 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] Setting up a Fritz AVM PCI card
Yes when I start Asterisk, I get: [chan_capi.so] = (Common ISDN API for Asterisk) == Parsing '/etc/asterisk/capi.conf': Found Nov 6 14:34:22 NOTICE[1076212352]: chan_capi.c:2636 load_module: CAPI not installed! I don't use AVM cards, but with Eicon you must configure it and load a firmware before. -- Carl ___ 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] chan_capi patch : fax support
Hello,
For those of you who have a CAPI card with an on-board DSP (like some Eicon
Diva Server), this patch allows you to receive faxes.
If you want to answer a channel in fax mode, use capiAnswerFax() instead of
Answer()
If you use Answer(), you will be in voice mode. If the hardware DSP detects
a fax tone, you can switch from voice to fax mode by calling
capiAnswerFax().
Example of use :
line number 123, play something, if a fax tone is detected, handle it
line number 124, answer directly in fax mode
[incoming]
exten = 123,1,Answer()
exten = 123,2,BackGround(jpop)
exten = 124,1,Goto(handle_fax,s,1)
exten = fax,1,Goto(handle_fax,s,1)
[handle_fax]
exten = s,1,capiAnswerFax(/tmp/${UNIQUEID})
exten = s,2,Hangup()
exten = h,1,deadagi,fax.php // Run sfftobmp and mail it.
The output of capiAnswerFax is a SFF file. Use sfftobmp to convert it.
With a Diva Server, theses features are allowed : fax up to 33600, high
resolution. Color Fax /JPEG Compression is disabled (I can't test it).
You can download the patch at :
http://www.mlkj.net/asterisk/chan_capi-0.3.5-patch.tar.bz2
A fix for a dead lock issue is also included (Oct 22 18:06:00
WARNING[11275]: channel.c:472 ast_channel_walk_locked: Avoided initial
deadlock for 'CAPI[contr1/173720007]/7', 10 retries!)
--
Carl
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Re: [Asterisk-Users] Cannot start asterisk - CAPI issues
chan_capi.c:2603 load_module: Unable to load config capi.conf You need to create this file /etc/asterisk/capi.conf with the following content : [general] nationalprefix=0 internationalprefix=00 [interfaces] msn=50 incomingmsn=* controller=1 softdtmf=0 accountcode= context=incoming ;echosquelch=1 echocancel=no ;echotail=64 ;callgroup=1 ;deflect=12345678 devices=30 Adjust devices= with the number of B channels supported by your card. For ISDN BRI, it's 2, for PRI, it's 30. Now I *do* have some kind of ISDN card in the box which I have not worried about yet (Eicon Diva 2.01 S/T PCI according to lspci) and I understand that CAPI has something to do with ISDN, but I have no clue as why asterisk doesn't start... You need a kernel support for you card and you also need to load a firmware for some cards. If you have a message like CAPI not installed!, check your kernel. ___ 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] Cannot start asterisk - CAPI issues
You need a kernel support for you card and you also need to load a firmware for some cards. If you have a message like CAPI not installed!, check your kernel. I use the linux 2.6.7 kernel which came with the knoppix distro I've installed on the box. I have checked with make xconfig and the options for isdn support and capi support all seem to be there OK. It's not enough, you must compile the correct Eicon driver. Read /usr/src/linux/Documentation/isdn/README.eicon Usually, you also need to load a firmware (with eiconctrl). Check out behind your card, when succesfully started, LEDs are turned on. For old cards, you may try isdn4linux instead CAPI. ___ 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] Re: Using AVM C4 with fewer than four lines?
On Thursday, 28 October, 2004 14:08 : Louis van Dompselaar [EMAIL PROTECTED] wrote: I think changing devices=4 to devices=2 in capi.conf should do the trick. Of course you have to make sure that your ISDN lines are connected to the two ports that haven't been disabled. It's already at devices=2 but that doesn't make a difference. I think chan_capi sees the C4 as one single device with four controllers. Anyway, the current capi.conf has devices=2 (tried 4 and 1; doesn't matter) and controller=1 (tried 1, 2 and 1,2,3,4, doesn't matter). I always get: If your devices= is to low, for example =1, then when you receive a 2nd call, you'll get the following error : ERROR [3075]: chan_capi.c:1696 capi_handle_msg: did not find device for msn = 1234xxx and the caller get a busy signal. -- Carl ___ 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] GSM Audio Files on Windows w/o Quicktime
On Wednesday, 27 October, 2004 20:00 : Mark Halverson [EMAIL PROTECTED] wrote: I play em just fine using Media Player 10. It's odd, Media Player require a WAV header, and the GSM frame is different (MS GSM = 1625 bytes/sec, GSM used in asterisk = 1650 bytes/sec). Are you using files created by Record() ? You can also read raw GSM with winamp. http://winamp.com/plugins/details.php?id=142107 -- Carl ___ 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] Nortel Phones.
On Tuesday, 26 October, 2004 15:46 : Kostur, Andre [EMAIL PROTECTED] wrote: Say... no chance that you could post your server code somewhere? We've got some i2004's kicking around doing nothing The source code is available at : http://www.mlkj.net/UNISTIM/voi.tar.bz2 -- Carl ___ 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] Nortel Phones.
On Monday, 25 October, 2004 22:19 : Jim Van Meggelen [EMAIL PROTECTED] wrote: [EMAIL PROTECTED] wrote: Currently the Nortel IP phones only support Nortel's proprietary protocol, UNISTIM. There is currently no way for Asterisk to directly support these phones unless: a) Nortel releases a standards-complaint firmware image, or b) somebody is able to write UNISTIM compatibility into Asterisk. Actually, I've done some reverse engineering on the UNISTIM protocol. I have a fully functional server able to handle such phones. If someone want to port my standalone code into asterisk, it would be great. -- Carl ___ 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] Diva Server PRI/E1-30
Hello, Can I use the Diva Server PRI/E1-30 card with asterisk (with chan_capi or i4linux) ? If asterisk detect a fax, how divert the call to the onboard DSP ? How can I handle the fax with hylafax (capi or pseudo serial device ?) Thank you, -- Carl ___ 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] Optus Australia Multiline SHDSL service
What is the type/model of the Adtran box? I was under the impression the Optus Multinet network (of which MultiLine is a product of) used 2 boxes onsite. One an SHDSL NTU and the other a voice router. That is, unless things have changed since I left. The service has definately been completely installed? Usually they'd install the NTU and not put the voice router onsite until sometime later. Normally though you would be provided with E1s by Optus which the TE401p should work fine with. Regards, Jamie Carl Chief 'Stuff' Officer J-Code International Email: [EMAIL PROTECTED] PH: +61414365466 IAXTel: 17004250969 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of duncan hall Sent: Wednesday, 22 September 2004 5:14 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Optus Australia Multiline SHDSL service Hi, I am currently trying to find a replacement for a dinosaur PBX and want to replace it with a VoIP solution. We have just moved our lines over to an Optus Multiline from a Telstra ISDN Onramp 30 service with 100 lines. My question for you good people is what sort of hardware do I need to interface Asterix into the Optus Multiline? The Optus service is terminated in my office to a SHDSL NTU from Adtran and has two RJ45 conenctors on the back of it. Has anybody tried this yet? Thanks in advance. Duncan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/aster isk-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] TDM400 synch issue
Hello, I have the following configuration : E100P - * - TDM400 - Modem When I receive FAXes, about 20% of them are corrupted : pages are not always complete. If the fax is complex or with numerous pages, it's usually a mess. Before that, I was using spandsp with success. Unfortunately it's too picky with some broken fax (training failed). Since this failure only occurs with the same set of faxes and it's reproducible, I'm confident about my E100P configuration. That's why I suspect frame slips on the TDM400 side. How can I solve this issue ? It's a dual p3, highly idle without IRQ sharing : 20: 602869735 602487558 IO-APIC-level wctdm 21: 602485799 602863515 IO-APIC-level t1xxp zaptel.conf : span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 fxoks=32-35 zapata.conf : [channels] faxdetect=incoming context=default switchtype=euroisdn signalling=fxo_ls usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=no echocancelwhenbridged=no rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no signalling=pri_cpe context=incoming channel = 1-15,17-31 signalling=fxo_ks context=internal channel = 32-33 Thanks, -- Carl ___ 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] Softphone for PocketPC or iPaq
SJPhone from SJLabs. www.sjlabs.com Also, a lot of simple questions like this can be answered by looking at www.voip-info.org There is a large section there on different soft/hardware phones. Regards, Jamie Carl Chief 'Stuff' Officer J-Code International Email: [EMAIL PROTECTED] PH: +61414365466 IAXTel: 17004250969 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sudhir Kumar Sent: Thursday, 23 September 2004 1:03 PM To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Subject: [Asterisk-Users] Softphone for PocketPC or iPaq Is there a soft phone for PocketPC or iPaq? If not, is someone working on it? I will be more than willing to contribute my mite if needed. Thanks, -- sudhir ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/aster isk-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] Zaptel and Linux Distros
Jamie Carl [EMAIL PROTECTED] wrote: Just a quick question. Are there any known issues using the zaptel drivers on different linux distros? ie: is a 2.4 kernel the only requirement? That should read 2.4 or later. I'm using the Linux 2.6.8 kernel with the Gentoo distro. In theory, the Zaptel driver should work on any 2.4 through 2.6-based distro. I haven't tried 2.7. Excellent. If it runs on 2.6 I'll be able to give FC2 a shot and see if that makes any difference. I ask because I have an X100P i'm still trying to get work properly and I'm thinking there is nothing wrong with the card itself. It's setup correctly as far as I can tell and it used to work when I was using RedHat 8 and 9. But since I went to Fedora Core 1 I don't think it's ever worked properly. Anyone else using Fedora Core 1 and an X100P without issues? It behaves very strangely and has the same symptoms as setting it to ground start signalling instead of loop start on a loop start PSTN line. However it is definately set to fxsls and even fxsks has been tried. I have a X101P at home (using Koolstart) and that performs almost acceptably. I'll be dumping it in favour of a Sipura SPA-3000 soon. I've never used Red Hat Fedora, so I can't comment on that. Perhaps you should try FC2 instead of FC1 or, even better, try Gentoo. If FC2 doesn't make any difference I may try rolling back to maybe RH8 or 9 just to confirm if it's a hardware or OS issue. Thanx Regards, Jamie Carl Chief 'Stuff' Officer J-Code International Email: [EMAIL PROTECTED] PH: +61414365466 IAXTel: 17004250969 ___ 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] Zaptel and Linux Distros
Hey all, Just a quick question. Are there any known issues using the zaptel drivers on different linux distros? ie: is a 2.4 kernel the only requirement? I ask because I have an X100P i'm still trying to get work properly and I'm thinking there is nothing wrong with the card itself. It's setup correctly as far as I can tell and it used to work when I was using RedHat 8 and 9. But since I went to Fedora Core 1 I don't think it's ever worked properly. Anyone else using Fedora Core 1 and an X100P without issues? It behaves very strangely and has the same symptoms as setting it to ground start signalling instead of loop start on a loop start PSTN line. However it is definately set to fxsls and even fxsks has been tried. I've purchased another X100P just in case it IS a hardware issue. tia... -- Regards, Jamie Carl Chief 'Stuff' Officer J-Code International [EMAIL PROTECTED] PH: +61414365466 IAXTel: 17004250969 ___ 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] Zaptel 'Under the Hood' Project
Victor Rini wrote: Hello, After poking and prodding at Asterisk and Zaptel for over a couple years now, I've dedicated some time to actually reading the code and trying to figure it out. It's been fascinating. With the driver source on one part of the screen and a pdf of Linux Device Drivers on another part I've aquainted myself with device driver programming and the interesting hardware on the wildcards. I've always thought Asterisk and Zaptel were two of the coolest FOSS projects around and now that I've spelunked through the code a little bit I'm curious: Has anyone ever wrote a zaptel under the hood type of document, discussing how the pseudo tdm bus works, the zaptel hardware, etc? If so, please point me there. If not, I'd like to take a stab at compiling a paper or article about zaptel for a general audience, technically inclined but not hard core technical, i.e. people like me who have used asterisk but always wondered how it worked down to the hardware, spans, channels, chunks, samples level. Some help from the community of course would be great, perhaps through using a blog or wiki. Once the zaptel dragon is dispatched, I'd then focus on Asterisk. What do you all think? Regards, Victor ___ 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 I can't speak for anyone else, but I sure as hell would be interested in such a document. I don't think it would even just be for the technically inclined but not hard core technical guys either. I consider myself pretty hard core but I just don't have the time to sit down and learn about how it all works on the inside. There's just too many other projects that need to be done. So in my opinion, a document that just lays it out in plain english would save me a heck load of time and allow me to learn about something that I unfortunately just don't have the time (or motivation) to figure out for myself and therefore probably wouldn't end up learning about otherwise. :) My 2c. Jamie ___ 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] Mediatrix APA III-4FXO (or 1204) help. Anyone with user manual
Bob Knight wrote: I have MIBs for whatever version I am running that I am more than happy to share. Anyone know where I can place these for public access. Sort of like the freedomphones site for Polycom. We could then put pointers on the wiki. Thanks for the info tho. If mbrowse is console based it will be very useful. :) It has gui (X, gtk I think) if that is what you mean by console based. I can ssh into a remote * server and do get walks on my 1204's. Bob, I've managed to source the MIBs from another extremely helpful list member so hopefully I'm all sorted. :) As for posting them, as I'm sure there are others out there that are interested, there is a website called www.mibdepot.com which is trying to collect as many MIBs as possible and currently has a request for the APA III-4FXO MIB. If you email it to the webmaster of that site he'll post it as part of his collection. I found this site while I was looking for it myself so hopefully others will look there too as they already have quite a few MIBs available. Jamie ___ 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] Mediatrix APA III-4FXO (or 1204) help. Anyone with user manual
Thanks to everyone for their help and comments on this. You've all been very helpful. I've actually got outbound calls working on it fine right now without having to change the configuration on the Mediatrix box at all, as I don't have the Unit Manager Software at the moment. Outbount seems to work well but without inbound it means I can't put it in place for general use. I have my 'reseller' tracking down the software for me right now so hopefully he'll be able to find it for me. :) Asterisk doesn't seem to have any issues working with the APA III-4FXO at all as yet. Thanks again guys. J Gonzalo Gasca Meza wrote: Here is my configuration for MEdiatrix 1204, by default the 1204 strips one digit, so it is not necessary to use: To dial OUTSIDE EXTENSIONS.CONF [locales] ;ignorepat = 9 exten = _9,1,Dial(SIP/[EMAIL PROTECTED] mailto:SIP/[EMAIL PROTECTED]) exten = _9,2,Congestion exten = _9,102,Congestion To receive calls [from-pstn] ;Incoming calls from Mediatrix 1204, the 1204, sends an invite to [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] exten = ,1,Dial(SIP/100,20) exten = ,2,Voicemail(u100) exten = ,102,Voicemail(b100) exten = ,103,Hangup *** SIP.CONF ;Mediatrix Telecomm 1204 [Mediatrix] type=peer host=110.10.200.10 mask=255.255.255.255 context=from-sip qualify=yes canreinvite=yes disallow=g729 nat = yes In MEdiatrix 1204 use a program called Unit Manager Network a Configure the first port as extension for port 1, in option SIP. as user agent. also edit registar an dproxy SIP as the IP address of Asterisk. Works VERY GOOD with one line, although i have seen some scenarios with more than 1 line which experince problems. Do you Yahoo!? Win 1 of 4,000 free domain names from Yahoo! Enter now http://us.rd.yahoo.com/evt=26640/*http://promotions.yahoo.com/goldrush. ___ 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] Mediatrix APA III-4FXO (or 1204) help. Anyone with user manual
Bob Knight wrote: There is a linux package called mbrowse that you can use with your mediatrix mibs. I can get and walk everything in my 1204's. For some reason I have not had any success with writes, but I have not spent that much time on it. I don't even have the MIBs which is half the problem. I can do certain things using windoze SNMP software, but not exactly being a guru on SNMP i'm guessing that without the MIBs i'm pretty much stuffed. Anyone with MIBs they can send me? hehe Please? :) Thanks for the info tho. If mbrowse is console based it will be very useful. :) J ___ 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] Mediatrix APA III-4FXO (or 1204) help. Anyone with user manual?
Hi all, I just picked myself up a Mediatrix FXO SIP gateway to play around with and hook into Asterisk but have no documentation. Are there default passwords or IP's that I need to know if I do a factory reset? Or better still, would anyone have a User Manual they could send my way? Any help would be appreciated. TIA. Jamie ___ 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] FXO probs in Aus. Should I give up?
Hey all, I've been trying to get my X101P working again as of late (it used to work great) and before I decide to trash the card I thought I'd post up my symptoms to see if anyone has any ideas. My old working config was basically 1 channel running fxsks signalling. It was working great with no echo, busy detect worked well and I was very impressed considering this is all off and Australian PSTN line for which the X101P is not certified. (hhh). So one day I update the zaptel drivers (not sure if this caused it however), and now it cannot go off-hook on it's own. Outbound Symptoms are: Placing a call from a SIP softphone, * will cease the zap channel and look like it's working, but no audio can be heard (ring tone, etc) on the softphone. Now, if I go off-hook on a POTS phone running parallel to the X101p suddenly everything comes to life. If I go off-hook on the parallel phone before the X101p tries to dial, everything works fine. But on it's own, it's a no go. Inbound Symptoms: The zap channel detects ring, ceases the channel and begins normal call flow and in my test setup going straight to voicemail. The caller can hear the call is answered but again, no audio. Going off-hook again on the parallel phone kicks everything back into life. Now here's the kicker. I have an old frame-relay voice switch I 'borrowed' from an ex-employer and have configured some slots for FXS to run back-to-back with the X101p. It works first time, every time. Only difference I can think of between them is that the voice switch is from the US and therefore uses US tones, etc. ?? I have tried both Loopstart and Koolstart signalling. Groundstart will not load when I use 'ztcfg' for some reason. So is there something I'm missing. This used to work fine. Has something changed in the zaptel driver? Are there any undocumented settings I can tweak to possible get this working again? I'm about to chuck the card and go for a SIP or MGCP gateway but if I can not spend the cash, I will. Anyone with ideas? Thanks heaps. Regards, Jamie Carl Chief 'Stuff' Officer J-Code Web:http://www.j-code.net Email: [EMAIL PROTECTED] ___ 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] FXOs
The only FXO interface that I have at the moment is an X101p. It was working great up until about a year ago and then something weird happened and I haven't used it since (until recently). Now it would seem it just doesn't like my PSTN line however it works fine running back-to-back on another non-PSTN FXS interface. Still working on it tho so I may get it working again soon. Regards, Jamie Carl Chief 'Stuff' Officer J-Code Web:http://www.j-code.net Email: [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Saturday, 28 August 2004 1:41 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] FXOs Hi All, I'd really like to see a show of hands with regard to people's experience with FXO interfaces. I own a few X100p cards and have had nothing but problems with them. I also took part in Sipura's beta program, for the SPA-3000. While it can be an improvement over the X100p, it presently has echo problems that make it unusable. Sipura has not acknowledged the problem ( at least to me) although several in the user community make refernce to new firmware that might address the issue, real soon now. I see a lot of activity recently on-list about the TDM-400. Of course, mentions on-list are more than likely the result of people having problems. We don't hear about people who have no issues with a product. So, the nature of my inquiry is to explore how many people out here have good/great experiences with the various small FXO adapters? While the TDM-400 is my next possible purchase I'd also like to hear about devices from Welltech, Clipcomm, Micronet, Multitech, Immixtel, etc. With so many products being offered I would hope that we have some collective experience with each one. Thanks, Michael Michael Graves Sr Product Specialist Pixel Power Inc [EMAIL PROTECTED] o(713)861-4005 o(800)905-6412 f(713)864-8668 c(713)201-1262 ___ 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] DTMF Issues with SJPHONE
What ver of SJPHONE? Thanks for the voicemail stuff :-) - Original Message - From: Girish Gopinath [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, February 28, 2004 7:48 PM Subject: RE: [Asterisk-Users] DTMF Issues with SJPHONE Has anyone had a similar issue with Asterisk Voicemail being unable to detect the digits sent from an SJ Phone connection. I have included dtmfmode=inband and it works fine when calling other phones though not with Voicemail. Voicemail doesn't regonise the password. I am using SJPhone, and works fine for me. Is there a way to not send a password when logging into Voicemail as a temp measure. Try something like like this, it will not ask for your password: exten = your extension,1,Ringing exten = your extension,2,Wait(2) exten = your extension,3,VoicemailMain,s ; is the mail box number Also, check out this url: http://www.automated.it/guidetoasterisk.htm Regards, Girish _ Post Classifieds on MSN classifieds. http://www.sulekha.com/msnclassifieds Buy and Sell on MSN Classifieds. ___ 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] Iconnect behind NAT
Anyone got an example of sip and extensions confs for Iconnect outgoing calls behind NAT.
Re: [Asterisk-Users] DTMF Issues with SJPHONE
Same as mine. Strange! I'll keep trying. Cheers. - Original Message - From: Girish Gopinath [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, February 28, 2004 9:53 PM Subject: Re: [Asterisk-Users] DTMF Issues with SJPHONE What ver of SJPHONE? SJPhone Evaluation Version, release Jul 31, 2003, Build: 1.10.187c Girish _ All the news that matters. All the gossip from home. http://www.msn.co.in/NRI/ Specially for NRIs! ___ 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 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.576 / Virus Database: 365 - Release Date: 1/30/2004 ___ 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] Help needed setting up H323 gateway.
Hi, Can someone offer some assistance in setting up Asterisk as a gateway to connect to a third party gatekeeper. I have looked at the h323.conf.sample file but not sure of the following: Do I need to create a new h323.conf file? Where should this file reside i.e., h323 directory? Do you need to add info to extensions file to point to context in the h323.conf file? How do u send an account number with the call so that the third party gatekeeper can verify? Your help will be much appreciated! Carl. ---Outgoing mail is certified Virus Free.Checked by AVG anti-virus system (http://www.grisoft.com).Version: 6.0.576 / Virus Database: 365 - Release Date: 1/30/2004
Re: [Asterisk-Users] Iconnect behind NAT
I'll give them a whirl. Cheers C. - Original Message - From: Darren Wiebe [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, February 29, 2004 11:00 AM Subject: Re: [Asterisk-Users] Iconnect behind NAT I signed up with nufone. Their customer service is a little bit slow but they seem to be pretty decent. I'd recommend checking them out. www.nufone.net Darren Wiebe [EMAIL PROTECTED] Carl wrote: Ha ha I get the picture :-) I've tried Voicepulse but can't manage to get through with them either. Emailed their customer support a week ago and heard nothing since. They get the destination numbers as I can see it on their cdr records. Any other providers offering IAX interconnects? - Original Message - From: John Fraizer [EMAIL PROTECTED] To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Sent: Sunday, February 29, 2004 2:43 AM Subject: Re: [Asterisk-Users] Iconnect behind NAT carl wrote: Anyone got an example of sip and extensions confs for Iconnect outgoing calls behind NAT. Here you go: [Scene starts out with you on the phone with IConnect technical support.] You: I know that Asterisk isn't one of your supported platforms. I'm not asking you to support my 'device'... I'm asking you to support your 'service'... Why is it that I can't have multiple outbound calls at a time? Why doesn't inbound caller-ID work right when someone is calling from a Nextel phone? Why do calls I make show up with no caller-ID? I need them to show caller-ID or the people I'm calling won't answer the phone. Why do I have to wait several (10-15) seconds between calls to prevent getting congestion tone from IConnect? Iconnect: We do not support Asterisk. You: Cancel my account. I'm going to find a REAL provider. [curtain closes - both on the scene and on IConnect.] Seriously, you're much better off finding a provider that will support IAX interconnect as well as address the problems in our scene. I'll be happy to get you set up with IAX peering. Drop me an email if you're interested. John ___ 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 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.576 / Virus Database: 365 - Release Date: 1/30/2004 ___ 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 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.576 / Virus Database: 365 - Release Date: 1/30/2004 ___ 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] H323 SETUP ON ASTERISK??
Hi, Whats involved in getting H323 working on Asterisk with Redhat 9??? Cheers, Carl.
[Asterisk-Users] DTMF Issues with SJPHONE
Has anyone had a similar issue with Asterisk Voicemail being unable to detect the digits sent from an SJ Phone connection. I have included dtmfmode=inband and it works fine when calling other phones though not with Voicemail. Voicemail doesn't regonise the password. Is there a way to not send a password when logging into Voicemail as a temp measure.
[Asterisk-Users] RE:Poor Voicemail / Ivr announcement quality
Howdy, The first 5 secs of each Voicemail or IVR announcement is stuttered and u can hardly hear the sound. After that its ok. Running TOP showed a high CPU usage on start up of the announcement as running command X?? Is this a PC CPU/RAM issue or something else related to Asterisk OS : Redhat v9 PC : AMD K2 512 Cheers, Carl. _ Store more e-mails with MSN Hotmail Extra Storage 4 plans to choose from! http://click.atdmt.com/AVE/go/onm00200362ave/direct/01/ ___ 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] Newbie Qu.
When I call Voicemail I get a very slow underwater sounding voice for the first few seconds then it corrects itself. Any idea? Output from Console: -- Executing VoiceMailMain(SIP/2101-20db, ) in new stack -- Playing 'vm-login' (language 'en') Feb 26 14:45:58 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! Feb 26 14:45:58 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! Feb 26 14:45:59 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! Feb 26 14:45:59 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! Feb 26 14:45:59 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! Feb 26 14:45:59 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! Feb 26 14:45:59 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! Feb 26 14:46:00 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! Feb 26 14:46:00 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! Feb 26 14:46:00 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! Feb 26 14:46:00 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! Feb 26 14:46:00 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! Feb 26 14:46:01 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! Feb 26 14:46:01 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! Feb 26 14:46:01 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! Feb 26 14:46:01 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! Feb 26 14:46:02 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! Feb 26 14:46:02 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! Feb 26 14:46:02 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! Feb 26 14:46:02 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! Feb 26 14:46:02 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! Feb 26 14:46:02 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! Feb 26 14:46:02 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! Feb 26 14:46:03 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! Feb 26 14:46:03 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! Feb 26 14:46:03 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! Feb 26 14:46:03 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! Feb 26 14:46:03 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! Feb 26 14:46:03 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! Feb 26 14:46:03 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! Feb 26 14:46:03 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! Feb 26 14:46:03 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! Feb 26 14:46:04 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! Feb 26 14:46:04 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! Feb 26 14:46:04 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! Feb 26 14:46:04 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! Feb 26 14:46:04 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! Feb 26 14:46:04 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! Feb 26 14:46:04 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! Feb 26 14:46:04 NOTICE[393234]: sched.c:218 sched_settime: Request to schedule in the past?!?! _ Watch high-quality video with fast playback at MSN Video. Free! http://click.atdmt.com/AVE/go/onm00200365ave/direct/01/ ___ 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] New to asterisk? RUN... don't walk.
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Wednesday 31 December 2003 03:24 pm, asterisk wrote: Here's the deal: It does almost anything. I can make it open my garage door. My installation records all conversations and then archives them as timestamped stereo MP3s. Our VB windows application can dial out with a click. All for free. No argument here. I think 80% of us n00bs can get by with the docs as-is (all I ask is to not be attacked), although if listserv gets repeated questions, maybe it's a symptom. Thing is, a novice or journeyman can't really fix the docs to the best technical info; takes a master, who is understandably doing more important things. Looks to me at this point, that asterisk has the potential of being (is?) one of the great open-source projects. Kudos. BTW, is anyone participating in the ENum trial? With Asterisk? -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.3 (GNU/Linux) iEYEARECAAYFAj/zR9MACgkQnQ18+PFcZJvGuQCfSjwr0WQhy3l9tUH9tgjL8L0K laEAnRsFlpC+kcU81c+imhB7WOpZJw3u =X/ME -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FAX, IAX and *....Maybe I'm dreaming...:-)
Hi Jeff, I live in Provo and I think I understand the application you're referring to. Some folks in my neighborhood have been getting to be the beta testers for these cool new fiber links that the city is supposed to be laying out. If I only lived a few blocks over, I would be able to get one too. Darn. Anyway, I've been following this thread, and I'm wondering if an alternative might be to provide some sort of fax jack on the hardware you provide the customer that your network could notice and then treat differently from regular voice data? An even better alternative would be if asterisk could recognize a fax machine on the end of the line and use a different protocol or codec that would work with faxes. It sounds like some of the contributors to this thread were saying this is possible. But I'm not sure--I'm pretty new to asterisk and VoIP in general, so I could be wrong. Carl ProvoCityPower wrote: Did DVD players have to accommodate VHS tapes? Did VHS players have to accept beta? Why does VoIP have to deal with an accent protocol that can't handle lossy audio, nor irregular delays? Also why should we be soo wasteful when fax machines need a 80K codec to get the data across IP, and the faster machines I see say 15 secs per page. So why should we send 1.2meg when 150k is fine? Also who says Fax should ever be required on IP? My office has been using VoIP for all voice traffic for over a year now, but always left the fax machine on a analog line. The analog line was cheap enough to not be a concern. -- Steven Critchfield [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] I'm fairly new here and don't mean to be contentious. We all have different perspectives as to what VOIP should be. My goal is to replace analog lines, not supplement them. I'm talking residential installations. I don't think I can ask these folks to leave their fax on an anolog line? I think that if we start deciding things for the Customer, then VOIP will be seen as an elitist toy for digitally inclined, instead of an acceptable alternative for the masses. No offense to the anti-fax coalition. Jeff ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoicePulse for outbound dialing
Next get a VOIP service provider to provide you with a PSTN DID (A phone number) VoicePulse will do this for about $8.00/month pluss outgoing per minute cost. So you get as many incomming lines as you need and you have zero hardware interface at your site. (other then your DSL line.) Can I use this type of service for outbound dialing without any SIP phones? I just want to have a server that sends voice messages to our customers. And if so, what part of asterisk do I want to examine to develop this? It would be great to write an outbound dialer that didn't require any specialized hardware. Does a service like this let you make more than one phone call simultaneously, or must you pay an additional $8.00 for each line that gets used at the same time? Sorry if these questions are stupid but I'm new to asterisk. Thanks, Carl Youngblood ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoicePulse for outbound dialing
What about the G.729 codec? From what I've heard it allows you to stuff an analog call into 8 Kbps. This would give you a theoretical maximum of 80 simultaneous connections on a 640 Kbps DSL line. I would expect this to be much lower in practice, say 20 simultaneous streams, but still, that's not bad. Adding to my own question, VoicePulse doesn't appear to support G.729. Here is their list of supported codecs: GSM, G.711ulaw, G.711alaw, ADPCM, ILBC, SPEEX (from http://connect.voicepulse.com/specifications.aspx). Does anyone know if one of these is as good as G.729? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoicePulse for outbound dialing
From what I read here: http://www.globalipsound.com/pdf/gips_iLBC.pdf iLBC is free and better quality than G.729A, same quality as G.729E and offers substantially better quality over congested networks. Its bandwidth requirements are a little higher (13-15 kbps) but they aren't bad. Adam Hart wrote: Adding to my own question, VoicePulse doesn't appear to support G.729. Here is their list of supported codecs: GSM, G.711ulaw, G.711alaw, ADPCM, ILBC, SPEEX (from http://connect.voicepulse.com/specifications.aspx). Does anyone know if one of these is as good as G.729? All of thoses besides GSM are as good as G.729 IMO (not sure what ADPCM is, just raw PCM?). My recommendation is iLBC. -Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] unixODBCget/put/del/deltree
Sorry, but would someone mind giving a brief explanation to newbies as to why this is cool? I am interested in creating call trees from a postgres database, so this looks like it might be useful, but I still don't understand much of what's going on here. Thanks, Carl Youngblood On Dec 6, 2003, at 2:41 PM, Brian West wrote: -- Executing unixODBCput(SIP/10-cc1b, BLAH/blah=bkw) in new stack -- unixodbcput: family=BLAH, key=blah, value=bkw -- Executing unixODBCput(SIP/10-cc1b, BLAH/blah=bk2) in new stack -- unixodbcput: family=BLAH, key=blah, value=bk2 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
