Re: [asterisk-users] USA BRI -- any hope at all?
On Mon, Feb 09, 2009 at 11:42:43AM -0700, Wilton Helm wrote: I discussed my installation more with Tzafrir last week. He concluded that he thinks I don't have an HFC card. I think it is somewhat a matter of semantics. It's not a matter of what I think. It's a matter of what you actually have :-) As far as I have been able to determine, there are at least three general types of HFC cards. By far the most common are cards based on the Cologne chip--which are well supported. You refer here to The Cologn HFC-S chip, I guess. The second group are based on a multichannel chip, which seems to be fairly popular now, particularly for 2 and 4 port BRI cards where a single IC can form the basis of the card. E.g. the Cologn HFC-4S (and similar). Not useful if you have a single-port card (which are popular for the consumer market. Or at least were). But handy if you want to create a device for a PBX , that has more than one BRI port. The third group are based on the Winbond W6692 chip. I think the chip was released about 10 years ago. Which is what you have. It has not been well supported. That is the chip my card has. It appears that no form of Zaptel or Dahdi, including publicly available patches, supports it. Right. (Although if anybody wants to write a Zaptel/DAHDI driver for it, I'd welcome it) I'm not sure, but I think mISDN supports it, and I know my card is CAPI compliant, but that may assume a driver that may not exist for Linux. Whether either of these supports US NT1 format remains a question. Also, there is the issue of physical level signaling, which is different in the US than in Europe. IIRC (from personal correspondence with you) you have managed to get layer 1 working with some ISDN driver (hisax = isdn4linux? mISDN?) I don't know if my NT1 U to S/T adapter takes care of that or not. I do know that there have been products out there that could be used in either market with only a firmware change, so maybe this is a non-issue. One of my TAs (I have several with differing feature sets and in various states of repair and (non) support) will generate a D channel transaction log, which would let me know what exchange is required with the CO. That might be compared to what the channel software does. I might be able to modify the source if necessary, although there are two or three learning curves involved for me. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- 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] [NO ANSWER] Re: Asterisk and CIsco 1760 SIP ?
Anyone use CIsco 1760 with Asterisk Phibee Network Operation Center a écrit : Hi i am search a sample config (for asterisk and for cisco) for connect a cisco 1760 with a FXO card to my asterisk. Thanks for your help Jerome ___ -- 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] Disabling Echo Cancellation on a per Call basis
Hello all, i was just made aware on the Bristuff-Mailing list, that it is possible to disable echo cancellation per dialplan application. This comes in very handy, for terminating faxes. But the application seems only to be existing in the bristuff patches. Does there exist a solution for Asterisk 1.6.0.3 Digium Wildcard TE110P T1/E1 DAHDI Version: 2.1.0.3 Echo Canceller: MG2 without any Bristuff? At the Moment i have fax detection enabled. If a fax comes in a see that fax handling on the channel is set to true, but echo cancellation is also on. This makes problems with low quality fax calls, they break nearly 100%. I am guessing that disabling echo can might improve this. Kind Regards, -- Tobias Wolf ___ -- 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] put the hostname of asterisk in the callerid uri to avoid NAT problems
On Sat, Feb 7, 2009 at 8:31 AM, nik600 nik...@gmail.com wrote: hi is it possible to set up in the dialplan (on in sip.conf, or something else) the hostname of the outgoing uri call? This is my scenario: - CCM integrated with Asterisk via h323 - SIP user registerd to Asterisk - Asterisk is behind NAT - Asterisk ip is 10.10.10.2 - SIP user view Asterisk as 10.10.15.2 (Asterisk is behind NAT) When the CCM calls the SIP user the call works perfectly. The problem is that the SIP user receives the call with this uri: sip:x...@10.10.10.2 The call works properly and the audio goes in both directios, BUT if the SIP user does a redial (after the hangup) the call is forwarded to x...@10.10.10.2 that is the wrong address. I've tried to force SIP_HEADER(CONTACT) in the dialplan with a Set but it seems that i can't due to security reason. Is it possible to avoid this problem? Thanks -- /*/ nik600 http://www.kumbe.it Do you think that is a bug or a miss configuration, or simply is not possible to avoid that because it is hard-coded? Thanks -- /*/ nik600 http://www.kumbe.it ___ -- 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] USA BRI -- any hope at all?
Tzafrir Cohen schrieb: On Mon, Feb 09, 2009 at 11:42:43AM -0700, Wilton Helm wrote: I discussed my installation more with Tzafrir last week. He concluded that he thinks I don't have an HFC card. I think it is somewhat a matter of semantics. It's not a matter of what I think. It's a matter of what you actually have :-) LOL Philipp Kempgen -- AMOOCON 2009, May 4-5, Rostock / Germany - http://www.amoocon.de Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- 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 how many calls handle using H.323 to SIP conversion?
I have P4 2.50GHz RAM 4GB, Asterisk how many calls handle using H.323 to SIP conversion on this server? Regards, --- Muhammad Asif Raza ___ -- 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] Skip password option in voicemail.conf
Hi,
Would you find convenient to store in voicemail.conf, a per user option
saying if this user prefers to skip password when listening to its mailbox ?
Then a dialplan function would allow to use this value and tailor
VoiceMailMain options accordingly.
Example:
exten = 1000,1,Set(OPTION=$[ ${VM_SKIPPASSWORD(${CALLERID(num)})} =
yes]?,s:)
exten = 1000,n,VoiceMailmain(${CALLERID(num)}${OPTION})
Regards
___
-- 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] Disabling Echo Cancellation on a per Call basis
2009/2/10 Tobias Wolf tobias.w...@evision.de Hello all, i was just made aware on the Bristuff-Mailing list, that it is possible to disable echo cancellation per dialplan application. This comes in very handy, for terminating faxes. But the application seems only to be existing in the bristuff patches. Does there exist a solution for Asterisk 1.6.0.3 Digium Wildcard TE110P T1/E1 DAHDI Version: 2.1.0.3 Echo Canceller: MG2 without any Bristuff? At the Moment i have fax detection enabled. Do you mean a given DID receives voice or fax calls ? If positive, which app is detecting faxes ? If a fax comes in a see that fax handling on the channel is set to true, but echo cancellation is also on. This makes problems with low quality fax calls, they break nearly 100%. I am guessing that disabling echo can might improve this. Kind Regards, -- Tobias Wolf ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk how many calls handle using H.323 to SIP conversion?
hi the amount of calls is not calculated by the protocol. is by: transcodification monitor/mixmonitor other cpu eater preocess musiconhold and the best you can do is try... install other asterisk whit h323 and start making calls. David 2009/2/10 MianAsif asif4...@gmail.com I have P4 2.50GHz RAM 4GB, *Asterisk how many calls handle using H.323 to SIP conversion on this server*? Regards, --- Muhammad Asif Raza ___ -- 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 -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk - Trixbox
Mike, Please explain the problem more clearly and post a pastebin that shows the problem and only the problem, not a huge SIP dump. If you could point out the line numbers where you suspect an issue. Thanks, Steve On Mon, Feb 9, 2009 at 10:56 PM, Mike Hammett asterisk-us...@ics-il.net wrote: Can anyone help me determine where the problem lies and how to fix it? - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com From: Mike Hammett Sent: Thursday, January 15, 2009 1:00 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk - Trixbox My provider migrated from an old EOL softswitch to Trixbox. I have a number (8159093011) on a different server on a different network. It appears as though the incoming calls are trying to authenticate against that number, which isn't present on the box. Could someone help me decode this debugging output? I was calling 8159911010. My server is 208.100.1.33. Theirs is 208.1.87.235. I solved the s@ problem on the other server by adding insecure settings, but that didn't seem to solve it on this one. http://pastebin.com/f5151341f - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [NO ANSWER] Re: Asterisk and CIsco 1760 SIP ?
snip Anyone use CIsco 1760 with Asterisk /snip No, but I'm using 7941G-GE and 7961G-GE in a deployment of ~80 phones. Did you have a question about implementation or are you just curious? --Dave ___ -- 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] SMS /w Asterisk
On Tue, Feb 10, 2009 at 2:23 AM, randulo spamsucks2...@gmail.com wrote: On Tue, Feb 10, 2009 at 12:39 AM, Mike Diehl mdi...@diehlnet.com wrote: I'm looking into being able to send/receive SMS messages with my asterisk box in the US. I've seen the SMS command as well as the Kannel program. I'd prefer to do it from Asterisk. snip Do you have SMS service on the phone line? I've always thought SMS was uncommon if not inexistent in the USA. I used SMS with asterisk in both directions for years and it worked great, but that was in Europe where fixed line SMS is common. Suggest you look at http://voip-info.org as there is a lot of info there on SMS. hth /r SMS or Texting has definitely caught on with the younger generation. There are proposed laws all over, banning texting while driving. Scores of accidents are blamed on texting including a train crash. I only use it for reminders to my cell phone but not for communication. Kannel is probably the best way to go in the States, unless you want to sign up with an aggregator. I use Kannel and a bank of Sony Ericsson phones. To send SMS, you just have to hit a URL on the Kannel server with a properly formatted URL. I just use System() to call Lynx with the correct variables (the message you want to send). I have asked many carriers about SMS here in the States, and they all say, oh yes, we can do that. But their acronym for SMS means something else, not simple messaging. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [NO ANSWER] Re: Asterisk and CIsco 1760 SIP ?
Hi thanks for your answer, not curious, i have one 1760 with FXO card and ~100 Cisco 1751 with FXO Card to at connected to my asterisk in SIP but i don't have touch Asterisk since 18 mounth ... and never connected router at asterisk (only Linksys SPA941 voice unit) if you have a idea of the configuration of the Cisco 1760 .. it's help me bye jerome David Gibbons a écrit : snip Anyone use CIsco 1760 with Asterisk /snip No, but I'm using 7941G-GE and 7961G-GE in a deployment of ~80 phones. Did you have a question about implementation or are you just curious? --Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [ANSWER] Re: Asterisk and CIsco 1760 SIP ?
I'm using an FXS, but it is pretty much the same. What part do you need? It is virtually the same as using a Cisco PRI card. Here is the relevant part on the Cisco. On * you just set it up like a gateway. dial-peer voice 200 voip destination-pattern .T progress_ind setup enable 3 session protocol sipv2 session target ipv4:192.168.1.10 (IP of * box) session transport udp dtmf-relay rtp-nte codec g711ulaw fax rate disable fax nsf 00 no vad ! dial-peer voice 1 pots description DID 1234567890 destination-pattern 1234567890 port 2/0 voice-port 2/0 caller-id enable Phibee Network Operation Center wrote: Anyone use CIsco 1760 with Asterisk Phibee Network Operation Center a écrit : Hi i am search a sample config (for asterisk and for cisco) for connect a cisco 1760 with a FXO card to my asterisk. Thanks for your help Jerome ___ -- 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
Re: [asterisk-users] What do you use? .conf or AEL?
You should stick with your .conf, but work toward transitioning to AEL because all things in Asterisk will eventually deprecate (can I copyright that?) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lee, John (Sydney) Sent: Tuesday, February 10, 2009 1:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What do you use? .conf or AEL? Of course you should be using AEL. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Alan Lord (News) Sent: Tuesday, 10 February 2009 6:24 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] What do you use? .conf or AEL? Hi all, I built my first asterisk using the traditional (?) .conf files and constructs. I recall reading books at the time about AEL but it seemed new and untested so I left it alone. Now, I'm interested to poll the audience here to see if I should look into using AEL instead (or in addition to) for future work. TIA ___ -- 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
Re: [asterisk-users] SMS /w Asterisk
On Tue, 10 Feb 2009, Steve Totaro wrote: Kannel is probably the best way to go in the States, unless you want to sign up with an aggregator. I use Kannel and a bank of Sony Ericsson phones. To send SMS, you just have to hit a URL on the Kannel server with a properly formatted URL. I just use System() to call Lynx with the correct variables (the message you want to send). Just out of curiosity, how did you transfer the text you want to SMS to Asterisk? Can that be done through a call file? I want to SMS from a CRM app, the CRM app can create call files but for security reasons i do not want the sms server accessible from the network to each PC that can run the CRM app. On voip-info there are some affordable GSM adapters that also provide an SMS server accessible by URL, i would like to do something similar like you did, would you be willing to post your configs? Thanks a 1,000,000 :) ___ -- 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] SMS /w Asterisk
On Tue, Feb 10, 2009 at 9:23 AM, Remco Barendse aster...@barendse.to wrote: On Tue, 10 Feb 2009, Steve Totaro wrote: Kannel is probably the best way to go in the States, unless you want to sign up with an aggregator. I use Kannel and a bank of Sony Ericsson phones. To send SMS, you just have to hit a URL on the Kannel server with a properly formatted URL. I just use System() to call Lynx with the correct variables (the message you want to send). Just out of curiosity, how did you transfer the text you want to SMS to Asterisk? Can that be done through a call file? I want to SMS from a CRM app, the CRM app can create call files but for security reasons i do not want the sms server accessible from the network to each PC that can run the CRM app. On voip-info there are some affordable GSM adapters that also provide an SMS server accessible by URL, i would like to do something similar like you did, would you be willing to post your configs? Thanks a 1,000,000 :) In that case, you would not need Asterisk at all. If you can create call files can you hit a URL from your CRM as well? The config files for Kannel are very simple with slight alterations from the samples generated from what is created from building from source. Most of the alterations are to use the Sony phones as GSM modems. Kannel's learning curve is slight if you are just using it for simple SMS through a GSM modem, of course, doing WAP Push or proxying requires a bit more in the way of learning. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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] Disabling Echo Cancellation on a per Call basis
Olivier schrieb: 2009/2/10 Tobias Wolf tobias.w...@evision.de mailto:tobias.w...@evision.de Hello all, i was just made aware on the Bristuff-Mailing list, that it is possible to disable echo cancellation per dialplan application. This comes in very handy, for terminating faxes. But the application seems only to be existing in the bristuff patches. Does there exist a solution for Asterisk 1.6.0.3 Digium Wildcard TE110P T1/E1 DAHDI Version: 2.1.0.3 Echo Canceller: MG2 without any Bristuff? At the Moment i have fax detection enabled. Do you mean a given DID receives voice or fax calls ? If positive, which app is detecting faxes ? Since i have a dedicated DID for fax calls, i don't really need the fax detection. For this number i simply start the ReceiveFAX-Application and have some voodoo around it to name the file correctly. But if i do this, and look into the channel information from Dahdi i see that the fax handled flag is set to no. And this seems wrong to me. I have the feeling that the percentage of failed faxes is higher is this flag is set to no (or false, can't remember) ... Since i have a PRI connected to my Asterisk, i use the built-in fax detection of DAHDI. I have enabled it for incoming fax calls, in chan_dahdi.conf faxdetect=incoming The incoming call is answered and with an included Wait(4) the fax is detected and switched to the fax extension, where the ReceiveFAX-App is executed. Now the fax handled flag is set to yes and i am able to receive most of the fax calls. But i have massive problems receiving fax calls from certain people, especially from UK (i am in germany). I am not quite sure, if the echo canceller is automatically disabled if DAHDI knows that the call is a fax and the channel info doesn't indicate otherwise, since it says that echo canceller is active even if it says that it handles an fax. This is the reason why i was so happy to hear, that there seems to be the option to control the echo canceller with an dialplan app. But since this seems to be an Bristuff-only feature i am a little bit stuck. Kind regards, -- Tobias Wolf ___ -- 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] SMS /w Asterisk
On Tue, Feb 10, 2009 at 9:36 AM, Steve Totaro stot...@totarotechnologies.com wrote: On Tue, Feb 10, 2009 at 9:23 AM, Remco Barendse aster...@barendse.to wrote: On Tue, 10 Feb 2009, Steve Totaro wrote: Kannel is probably the best way to go in the States, unless you want to sign up with an aggregator. I use Kannel and a bank of Sony Ericsson phones. To send SMS, you just have to hit a URL on the Kannel server with a properly formatted URL. I just use System() to call Lynx with the correct variables (the message you want to send). Just out of curiosity, how did you transfer the text you want to SMS to Asterisk? Can that be done through a call file? I want to SMS from a CRM app, the CRM app can create call files but for security reasons i do not want the sms server accessible from the network to each PC that can run the CRM app. On voip-info there are some affordable GSM adapters that also provide an SMS server accessible by URL, i would like to do something similar like you did, would you be willing to post your configs? Thanks a 1,000,000 :) In that case, you would not need Asterisk at all. If you can create call files can you hit a URL from your CRM as well? The config files for Kannel are very simple with slight alterations from the samples generated from what is created from building from source. Most of the alterations are to use the Sony phones as GSM modems. Kannel's learning curve is slight if you are just using it for simple SMS through a GSM modem, of course, doing WAP Push or proxying requires a bit more in the way of learning. If you wanted to go through Asterisk, I would think a call file that drop into a context with a FastAGI to your CRM that returns the text would be my approach, or similarly, through the manager interface. Thanks, Steve ___ -- 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] What do you use? .conf or AEL?
Alan Lord (News) wrote: Hi all, I built my first asterisk using the traditional (?) .conf files and constructs. I recall reading books at the time about AEL but it seemed new and untested so I left it alone. Now, I'm interested to poll the audience here to see if I should look into using AEL instead (or in addition to) for future work. TIA I use AEL. I find it much cleaner to look at and not having to deal with priorities is a bonus. -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk - Trixbox
They must have changed something after I complained because it no longer references the incorrect phone number. I did disable However, it still wants to send everything to the s extension. Everything I have worked with before has sent calls the the DID's extension (a call to 888777 goes to exten = 888777,1,blah). Is this something they can change in Trixbox? http://pastebin.com/fa8b4f4e I highlighted the lines that contain the s extension. - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -- From: Steve Totaro stot...@first-notification.com Sent: Tuesday, February 10, 2009 7:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk - Trixbox Mike, Please explain the problem more clearly and post a pastebin that shows the problem and only the problem, not a huge SIP dump. If you could point out the line numbers where you suspect an issue. Thanks, Steve On Mon, Feb 9, 2009 at 10:56 PM, Mike Hammett asterisk-us...@ics-il.net wrote: Can anyone help me determine where the problem lies and how to fix it? - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com From: Mike Hammett Sent: Thursday, January 15, 2009 1:00 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk - Trixbox My provider migrated from an old EOL softswitch to Trixbox. I have a number (8159093011) on a different server on a different network. It appears as though the incoming calls are trying to authenticate against that number, which isn't present on the box. Could someone help me decode this debugging output? I was calling 8159911010. My server is 208.100.1.33. Theirs is 208.1.87.235. I solved the s@ problem on the other server by adding insecure settings, but that didn't seem to solve it on this one. http://pastebin.com/f5151341f - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk - Trixbox
Yes, they should fix this on their side, otherwise DID routing will not work. If you don't need it, you just need to create a DID entry for any/all or any/any, I cannot remember which it is right now, but it should be apparent when you look at it. The s extension is only used when no DID or extension is received. Thanks, Steve Totaro On Tue, Feb 10, 2009 at 9:49 AM, Mike Hammett asterisk-us...@ics-il.net wrote: They must have changed something after I complained because it no longer references the incorrect phone number. I did disable However, it still wants to send everything to the s extension. Everything I have worked with before has sent calls the the DID's extension (a call to 888777 goes to exten = 888777,1,blah). Is this something they can change in Trixbox? http://pastebin.com/fa8b4f4e I highlighted the lines that contain the s extension. - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -- From: Steve Totaro stot...@first-notification.com Sent: Tuesday, February 10, 2009 7:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk - Trixbox Mike, Please explain the problem more clearly and post a pastebin that shows the problem and only the problem, not a huge SIP dump. If you could point out the line numbers where you suspect an issue. Thanks, Steve On Mon, Feb 9, 2009 at 10:56 PM, Mike Hammett asterisk-us...@ics-il.net wrote: Can anyone help me determine where the problem lies and how to fix it? - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] unistim and transfer calls
Hi When i try to transfer calls from my ip2002 phone in asterisk 1.6, I can make the transfer and it rings on the extension I transfer to, but when I accept the call, asterisk dumps. How can I get it to work? And how do I save the dump error? Regards /ralf Ralf Träskman, IT AdLibris AB, Odengatan 106, 113 22 Stockholm, Sweden Dir: +46-(0)707548074, vxl: +46-(0)85460 60 00, fax: +46-(0)85460 60 99 r...@adlibris.commailto:r...@adlibris.com www.adlibris.comhttp://www.adlibris.com/ P Please consider the environment before printing this e-mail ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk - Trixbox
I disabled that last number's registration and moved to a new number (to test each number individually without the sip debugging from the others). I waited maybe 5 minutes and I restarted Asterisk to ensure the other side was done with whatever it was doing. I called the second number (8152641125) and the first number (8159911010) shows up as the peer. Not only that, but with this number, there's no compatible codecs. I ensured that both entries in sip.conf were the same other than things that needed to be different such as username. I even had that entry have allow=all. I still get the codec error. http://pastebin.com/f5b826d62 I highlighted the lines of interest. 34 is the peer issue whereas 42 is the codec issue. - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -- From: Steve Totaro stot...@first-notification.com Sent: Tuesday, February 10, 2009 7:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk - Trixbox Mike, Please explain the problem more clearly and post a pastebin that shows the problem and only the problem, not a huge SIP dump. If you could point out the line numbers where you suspect an issue. Thanks, Steve On Mon, Feb 9, 2009 at 10:56 PM, Mike Hammett asterisk-us...@ics-il.net wrote: Can anyone help me determine where the problem lies and how to fix it? - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com From: Mike Hammett Sent: Thursday, January 15, 2009 1:00 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk - Trixbox My provider migrated from an old EOL softswitch to Trixbox. I have a number (8159093011) on a different server on a different network. It appears as though the incoming calls are trying to authenticate against that number, which isn't present on the box. Could someone help me decode this debugging output? I was calling 8159911010. My server is 208.100.1.33. Theirs is 208.1.87.235. I solved the s@ problem on the other server by adding insecure settings, but that didn't seem to solve it on this one. http://pastebin.com/f5151341f - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- 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] What do you use? .conf or AEL?
On Tue, Feb 10, 2009 at 07:24:15AM +, Alan Lord (News) wrote: Hi all, I built my first asterisk using the traditional (?) .conf files and constructs. You still use them for most stuff, I guess. Anybody using Lua? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- 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] SMS /w Asterisk
On Tue, 10 Feb 2009, Steve Totaro wrote: In that case, you would not need Asterisk at all. If you can create call files can you hit a URL from your CRM as well? Not really, the app cannot open a browser, but it can create a file on a samba share quite easily. Therefore going through asterisk seemed to be the way. Alternatively i could try and recreate something similar like call files for asterisk, have the crm app create a text file with the complete url in it and feed the url to a browser like links. However with my skills at writing scripts being zero i thought that going through asterisk is the most obvious way for me. If you wanted to go through Asterisk, I would think a call file that drop into a context with a FastAGI to your CRM that returns the text would be my approach, or similarly, through the manager interface. OK, thanks, i will throw all that in Google-o-matic and see what comes up :) ___ -- 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] What t38pt_udptl is ? Explain T.38 in 1.4
On Tue, Feb 10, 2009 at 1:52 AM, Olivier oza-4...@myamail.com wrote: Have you tried to directly connect two T.38 enabled gateways without involving Asterisk at all (like this) ? ISDN --- Gateway --- SIP/T.38 --- ATA --- FXO/FXS --- fax machine No. As long as you can sufficiently debug that arrangement, I don't see any problems. As you have risk of corrupting the fax timing at each step, the fewer moving parts in that chain the better. However, I have done: channelized DS3 --- Adtran T1 gateway --- SIP/T.38 gateway --- Asterisk 1.6 as SIP fax machine and I was able to send / receive faxes wonderfully I verified this against T1 --- proprietary PBX --- FXO/FXS --- fax machine(s) both sending and receiving to / from the software fax in Asterisk. So I had 'real' traditional phones world in-between these two arrangements for testing purposes. ___ -- 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] SMS /w Asterisk
On Tue, Feb 10, 2009 at 10:15 AM, Remco Barendse aster...@barendse.to wrote: On Tue, 10 Feb 2009, Steve Totaro wrote: In that case, you would not need Asterisk at all. If you can create call files can you hit a URL from your CRM as well? Not really, the app cannot open a browser, but it can create a file on a samba share quite easily. Therefore going through asterisk seemed to be the way. Alternatively i could try and recreate something similar like call files for asterisk, have the crm app create a text file with the complete url in it and feed the url to a browser like links. However with my skills at writing scripts being zero i thought that going through asterisk is the most obvious way for me. If you wanted to go through Asterisk, I would think a call file that drop into a context with a FastAGI to your CRM that returns the text would be my approach, or similarly, through the manager interface. OK, thanks, i will throw all that in Google-o-matic and see what comes up :) How about a Samba share with a regular cron job or a process that watches that folder and fires off lynx then moves or deletes the file from the Samba share? The browser does not need to stay open, just a call to the URL and then close. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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] put the hostname of asterisk in the callerid uri to avoid NAT problems
On Tuesday 10 February 2009 04:36:09 nik600 wrote: On Sat, Feb 7, 2009 at 8:31 AM, nik600 nik...@gmail.com wrote: I've tried to force SIP_HEADER(CONTACT) in the dialplan with a Set but it seems that i can't due to security reason. Is it possible to avoid this problem? Do you think that is a bug or a miss configuration, or simply is not possible to avoid that because it is hard-coded? The issue isn't a security reason; it's that SIP_HEADER() is a read-only function. Try using the SIPAddHeader application, instead. -- 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
Re: [asterisk-users] Asterisk - Trixbox
How many accounts do you have? If just one, then a single peer should be fine but they should be sending the destination exten as a DID, obviously they are not. I think the burden of fixing it lies with them? What carrier is this? On Tue, Feb 10, 2009 at 10:09 AM, Mike Hammett asterisk-us...@ics-il.net wrote: I disabled that last number's registration and moved to a new number (to test each number individually without the sip debugging from the others). I waited maybe 5 minutes and I restarted Asterisk to ensure the other side was done with whatever it was doing. I called the second number (8152641125) and the first number (8159911010) shows up as the peer. Not only that, but with this number, there's no compatible codecs. I ensured that both entries in sip.conf were the same other than things that needed to be different such as username. I even had that entry have allow=all. I still get the codec error. http://pastebin.com/f5b826d62 I highlighted the lines of interest. 34 is the peer issue whereas 42 is the codec issue. - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -- From: Steve Totaro stot...@first-notification.com Sent: Tuesday, February 10, 2009 7:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk - Trixbox Mike, Please explain the problem more clearly and post a pastebin that shows the problem and only the problem, not a huge SIP dump. If you could point out the line numbers where you suspect an issue. Thanks, Steve On Mon, Feb 9, 2009 at 10:56 PM, Mike Hammett asterisk-us...@ics-il.net wrote: Can anyone help me determine where the problem lies and how to fix it? - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com From: Mike Hammett Sent: Thursday, January 15, 2009 1:00 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk - Trixbox My provider migrated from an old EOL softswitch to Trixbox. I have a number (8159093011) on a different server on a different network. It appears as though the incoming calls are trying to authenticate against that number, which isn't present on the box. Could someone help me decode this debugging output? I was calling 8159911010. My server is 208.100.1.33. Theirs is 208.1.87.235. I solved the s@ problem on the other server by adding insecure settings, but that didn't seem to solve it on this one. http://pastebin.com/f5151341f - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- 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 -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk - Trixbox
Do you know enough about Trixbox to tell me where they need to fix their misconfiguration, or is it a Trixbox design flaw? - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -- From: Steve Totaro stot...@first-notification.com Sent: Tuesday, February 10, 2009 8:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk - Trixbox Yes, they should fix this on their side, otherwise DID routing will not work. If you don't need it, you just need to create a DID entry for any/all or any/any, I cannot remember which it is right now, but it should be apparent when you look at it. The s extension is only used when no DID or extension is received. Thanks, Steve Totaro On Tue, Feb 10, 2009 at 9:49 AM, Mike Hammett asterisk-us...@ics-il.net wrote: They must have changed something after I complained because it no longer references the incorrect phone number. I did disable However, it still wants to send everything to the s extension. Everything I have worked with before has sent calls the the DID's extension (a call to 888777 goes to exten = 888777,1,blah). Is this something they can change in Trixbox? http://pastebin.com/fa8b4f4e I highlighted the lines that contain the s extension. - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -- From: Steve Totaro stot...@first-notification.com Sent: Tuesday, February 10, 2009 7:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk - Trixbox Mike, Please explain the problem more clearly and post a pastebin that shows the problem and only the problem, not a huge SIP dump. If you could point out the line numbers where you suspect an issue. Thanks, Steve On Mon, Feb 9, 2009 at 10:56 PM, Mike Hammett asterisk-us...@ics-il.net wrote: Can anyone help me determine where the problem lies and how to fix it? - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk - Trixbox
Possibly if I could take a look at their GUI and custom contexts. That could be quite a bit of work On Tue, Feb 10, 2009 at 10:26 AM, Mike Hammett asterisk-us...@ics-il.net wrote: Do you know enough about Trixbox to tell me where they need to fix their misconfiguration, or is it a Trixbox design flaw? - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -- From: Steve Totaro stot...@first-notification.com Sent: Tuesday, February 10, 2009 8:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk - Trixbox Yes, they should fix this on their side, otherwise DID routing will not work. If you don't need it, you just need to create a DID entry for any/all or any/any, I cannot remember which it is right now, but it should be apparent when you look at it. The s extension is only used when no DID or extension is received. Thanks, Steve Totaro On Tue, Feb 10, 2009 at 9:49 AM, Mike Hammett asterisk-us...@ics-il.net wrote: They must have changed something after I complained because it no longer references the incorrect phone number. I did disable However, it still wants to send everything to the s extension. Everything I have worked with before has sent calls the the DID's extension (a call to 888777 goes to exten = 888777,1,blah). Is this something they can change in Trixbox? http://pastebin.com/fa8b4f4e I highlighted the lines that contain the s extension. - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -- From: Steve Totaro stot...@first-notification.com Sent: Tuesday, February 10, 2009 7:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk - Trixbox Mike, Please explain the problem more clearly and post a pastebin that shows the problem and only the problem, not a huge SIP dump. If you could point out the line numbers where you suspect an issue. Thanks, Steve On Mon, Feb 9, 2009 at 10:56 PM, Mike Hammett asterisk-us...@ics-il.net wrote: Can anyone help me determine where the problem lies and how to fix it? - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- 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 -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk - Trixbox
It's a local CLEC, Essex Telcom. The burden does lie with them, but I doubt they'll fix it since if you provision a grandstream, it works just fine. I have a total of 5 numbers with them. Two are on the server that is experiencing issues. Another is on a different server with no issues. The remaining two aren't provisioned anywhere. I'm going to be adding another number shortly. - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -- From: Steve Totaro stot...@totarotechnologies.com Sent: Tuesday, February 10, 2009 9:50 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk - Trixbox How many accounts do you have? If just one, then a single peer should be fine but they should be sending the destination exten as a DID, obviously they are not. I think the burden of fixing it lies with them? What carrier is this? On Tue, Feb 10, 2009 at 10:09 AM, Mike Hammett asterisk-us...@ics-il.net wrote: I disabled that last number's registration and moved to a new number (to test each number individually without the sip debugging from the others). I waited maybe 5 minutes and I restarted Asterisk to ensure the other side was done with whatever it was doing. I called the second number (8152641125) and the first number (8159911010) shows up as the peer. Not only that, but with this number, there's no compatible codecs. I ensured that both entries in sip.conf were the same other than things that needed to be different such as username. I even had that entry have allow=all. I still get the codec error. http://pastebin.com/f5b826d62 I highlighted the lines of interest. 34 is the peer issue whereas 42 is the codec issue. - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -- From: Steve Totaro stot...@first-notification.com Sent: Tuesday, February 10, 2009 7:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk - Trixbox Mike, Please explain the problem more clearly and post a pastebin that shows the problem and only the problem, not a huge SIP dump. If you could point out the line numbers where you suspect an issue. Thanks, Steve On Mon, Feb 9, 2009 at 10:56 PM, Mike Hammett asterisk-us...@ics-il.net wrote: Can anyone help me determine where the problem lies and how to fix it? - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com From: Mike Hammett Sent: Thursday, January 15, 2009 1:00 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk - Trixbox My provider migrated from an old EOL softswitch to Trixbox. I have a number (8159093011) on a different server on a different network. It appears as though the incoming calls are trying to authenticate against that number, which isn't present on the box. Could someone help me decode this debugging output? I was calling 8159911010. My server is 208.100.1.33. Theirs is 208.1.87.235. I solved the s@ problem on the other server by adding insecure settings, but that didn't seem to solve it on this one. http://pastebin.com/f5151341f - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- 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 -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] USA BRI -- any hope at all?
It's not a matter of what I think. It's a matter of what you actually have :-) The third group are based on the Winbond W6692 chip. I think the chip was released about 10 years ago. Which is what you have. I've known that for several months, but I believe it is considered an HFC chip. I keep responding to posts here periodically in the hope that it will connect me with someone who can help me arrive at a solution. There definitely appears to be a need (albeit of limited quantity) for a) a working US BRI solution and possibly b) a W6692 driver. IIRC (from personal correspondence with you) you have managed to get layer 1 working with some ISDN driver (hisax = isdn4linux? mISDN?) Linux installs a driver (hisax, I believe) for the card. Whether that constitutes working, I don't know. I don't know what to configure or how to test at that level. F9 includes mISDN but I haven't figured enough of it out yet to know whether it can see the card. Key elements like config files and readme files aren't where the mISDN web page says they should be, so I don't have enough information to readily proceed. (Although if anybody wants to write a Zaptel/DAHDI driver for it, I'd welcome it) So would I. It isn't beyond the realm of possibility for me to write it, but it would take a very large amount of hand holding. I'm quite proficient in C. I understand the basic ideas involved in ISDN and have a working ISDN circuit, however I have very limited user knowledge of Asterisk and no past experience at the driver level or channel level and limited experience with linux. Probably my biggest weakness is that I don't have a clear picture of the levels involved--possibly because they may be historically fuzzy. If I understand correctly a DAHDI driver would cover layer 1 as well as higher layers up to what asterisk needs in a monolithic fashion. OTOH, other approaches, such as CAPI may split this up. I'm not familiar enough with what pieces cover what roles in the various possible scenarios. There are two problems that need to be solved: 1) Pieces in place to cover each layer that needs to be covered. 2) Making sure those pieces can work with US NT1 protocol. I am guessing that both have been solved by someone at some point, although probably not in the same file. I am also guessing that some modifications to existing code could accomplish this without writing a lot of code from scratch. I'm just not familiar enough with the options and available pieces to know where to look. Wilton ___ -- 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] What do you use? .conf or AEL?
On Tue, 10 Feb 2009, Lee, John (Sydney) wrote: Of course you should be using AEL. Of course you should carry on using .conf. Gordon -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Alan Lord (News) Sent: Tuesday, 10 February 2009 6:24 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] What do you use? .conf or AEL? Hi all, I built my first asterisk using the traditional (?) .conf files and constructs. I recall reading books at the time about AEL but it seemed new and untested so I left it alone. Now, I'm interested to poll the audience here to see if I should look into using AEL instead (or in addition to) for future work. TIA ___ -- 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
Re: [asterisk-users] Transfer Asterisk 1.6 Telephone IP
Hi, My IP phone has an option to send the flash via DTMF. Enable the sending and the DEBUG I get the following: [ TYPE: Control (4) SUBCLASS: Flash (9) ] [SIP/34314730-b7714f38] [Feb 10 15:12:51] WARNING[29203]: chan_sip.c:5350 sip_indicate: Don't know how to indicate condition 9 [Feb 10 15:12:51] WARNING[29203]: channel.c:2858 ast_indicate_data: Unable to handle indication 9 for 'SIP/13649-2001-b7718f20' How do I set the flash to work as *2. [featuremap] blindxfer=## atxfer=*2 automon=*1 disconnect=** 2009/2/9 Daviramos Roussenq Fortunato daviramo...@gmail.com Hi List. I have a small problem in using the transfer key transfer of IP Phone in Asterisk 1.6, I think I spend some detail in the configuration but can not find. What happens is, when I do a transfer using the Transfer button, the phone, does not play the music on hold, which is waiting on the phone is silent, and I have the same settings on a 1.4 server, and the music plays correctly when using the same phone. When using the * 2 to transfer the connection or a softfone, the music plays correctly on this server with Asterisk 1.6. What the detail is missing in my configuration? My Configuration [featuremap] blindxfer=## atxfer=*2 automon=*1 disconnect=** I made a DEBUG to use the channel when the two key TRANSFER Server 1.4 and 1.6. Command: core set debug channel SIP/2720-b7d28d70 DEBUG no 1.4: [ TYPE: Control (4) SUBCLASS: Unknown control '16' (16) ] [SIP/2720-b7d28d70] -- Started music on hold, class 'default', on SIP/2001-08a56cf8 [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/2720-b7d28d70] DEBUG no 1.6: When tightening the TRANSFER button on the console does not show anything, but when any other key grip CLI appears in decimal value of the corresponding key. I'm using dtmfmode=rfc2833 How you help me? ___ -- 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] connecting 66 analog phones to asterisk - hardware suggestions
Hi, I am looking to connect 66 analog phones to an asterisk box. I was thinking of a Xorcom astribank 32port (2 of them and another 8 port). this is because the phones have no near connection to an ip network, so replacing the phones in favor of voip phones+network cabling is kinda out of the question. In your experience, will these units support all the phones talking at the same time with other units on the astribank, as well as to the pbx, pstn, etc? The asterisk pbx will be a server-class Hp Proliant unit (potentially a dl320). i must make sure the astribanks will not die when fully utilized. other hardware suggestions for this task will be nice. thanks, -- Erick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk AGX addons compile issues
2008/12/18 Michael mich...@networkstuff.co.nz Has anyone seen this before, and know what is happening? u...@host:~/asterisk/agx-ast-addons# ./build.sh -- Configuring done -- Generating done -- Build files have been written to: /root/asterisk/agx-ast-addons [ 11%] Building C object CMakeFiles/app_devstate.dir/app_devstate.o Linking C shared module dist/app_devstate.so [ 11%] Built target app_devstate [ 22%] Building C object CMakeFiles/app_nv_backgrounddetect.dir/app_nv_backgrounddetect.o Linking C shared module dist/app_nv_backgrounddetect.so [ 22%] Built target app_nv_backgrounddetect [ 33%] Building C object CMakeFiles/app_nv_faxdetect.dir/app_nv_faxdetect.o Linking C shared module dist/app_nv_faxdetect.so [ 33%] Built target app_nv_faxdetect [ 44%] Building C object CMakeFiles/app_pickup2.dir/app_pickup2.o Linking C shared module dist/app_pickup2.so [ 44%] Built target app_pickup2 [ 55%] Building C object CMakeFiles/app_rxfax.dir/app_rxfax.o cc1: warnings being treated as errors /root/asterisk/agx-ast-addons/app_rxfax.c: In function 'phase_e_handler': /root/asterisk/agx-ast-addons/app_rxfax.c:126: warning: implicit declaration of function 't30_get_local_ident' /root/asterisk/agx-ast-addons/app_rxfax.c:127: warning: implicit declaration of function 't30_get_far_ident' /root/asterisk/agx-ast-addons/app_rxfax.c: In function 'rxfax_exec': /root/asterisk/agx-ast-addons/app_rxfax.c:380: warning: implicit declaration of function 't30_set_local_ident' /root/asterisk/agx-ast-addons/app_rxfax.c:383: warning: implicit declaration of function 't30_set_header_info' /root/asterisk/agx-ast-addons/app_rxfax.c:385: warning: passing argument 2 of 't30_set_phase_b_handler' from incompatible pointer type /root/asterisk/agx-ast-addons/app_rxfax.c:386: warning: passing argument 2 of 't30_set_phase_d_handler' from incompatible pointer type make[2]: *** [CMakeFiles/app_rxfax.dir/app_rxfax.o] Error 1 make[1]: *** [CMakeFiles/app_rxfax.dir/all] Error 2 make: *** [all] Error 2 u...@host:~/asterisk/agx-ast-addons# ___ -- 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 Hi, I think the trick is to download trunk version from svn (see voip-info.orgfor instrcution). Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk registered as UA
Hi guys! Thank you for your help. I found that I didn't load the module dahdi-dummy just the dahdi module. After I loaded the module, my conference works fine. Best Regards Szasz Szabolcs ___ -- 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] What do you use? .conf or AEL?
.conf all the way, purely because i only noticed that extensions.ael even existed a couple of months back, i should pay more attention really :p but until it's broke, i can't be bothered to fix it. 2009/2/10 Alan Lord (News) alansli...@gmail.com Hi all, I built my first asterisk using the traditional (?) .conf files and constructs. I recall reading books at the time about AEL but it seemed new and untested so I left it alone. Now, I'm interested to poll the audience here to see if I should look into using AEL instead (or in addition to) for future work. TIA ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] chan_oss.c:585 setformat: Unable to re-open DSP device
Any Help ? On Mon, Feb 9, 2009 at 9:59 AM, David @ULC ucoms2...@gmail.com wrote: == Manager 'sendcron' logged off from 127.0.0.1 vicidialnow*CLI dial 919545090201 -- Executing AGI(OSS/dsp, agi://127.0.0.1:4577/call_log) in new stack -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 -- Executing Dial(OSS/dsp, SIP/19545090...@sip203||tTor) in new stack -- Called 19545090...@sip203 Feb 2 13:36:38 WARNING[2884]: chan_oss.c:585 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory -- SIP/sip203-086fb130 answered OSS/dsp Console call has been answered Feb 2 13:36:42 WARNING[8644]: chan_oss.c:585 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory Feb 2 13:36:43 WARNING[8644]: chan_oss.c:585 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory Feb 2 13:36:44 WARNING[8644]: chan_oss.c:585 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory Feb 2 13:36:45 WARNING[8644]: chan_oss.c:585 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory Feb 2 13:36:46 WARNING[8644]: chan_oss.c:585 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory Feb 2 13:36:47 WARNING[8644]: chan_oss.c:585 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory Feb 2 13:36:48 WARNING[8644]: chan_oss.c:585 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory Feb 2 13:36:49 WARNING[8644]: chan_oss.c:585 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory Feb 2 13:36:50 WARNING[8644]: chan_oss.c:585 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory Feb 2 13:36:50 NOTICE[8644]: rtp.c:331 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 216.168.169.103 == Spawn extension (local, 919545090201, 2) exited non-zero on 'OSS/dsp' -- Executing DeadAGI(OSS/dsp, agi://127.0.0.1:4577/call_log) in new stack -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 -- Executing DeadAGI(OSS/dsp, agi:// 127.0.0.1:4577/VD_hangup--HVcauses--PRI-NODEBUG-16-ANSWER-13-10)) in new stack -- AGI Script agi://127.0.0.1:4577/VD_hangup--HVcauses--PRI-NODEBUG-16-ANSWER-13-10) completed, returning 0 Hangup on console What is this error ? _ ___ -- 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] connecting 66 analog phones to asterisk - hardware suggestions
FYI: We use AudioCodes MP124D hardware for analog connections. They are priced just slightly over $1100 for 24 ports, and work with all PBX vendors (Cisco, Digium, Pingtel, openSER) that we are running. The users are mostly FAX lines and alarm circuits, but some are humans (analog phones). Slightly over 1100 of them so far... http://voip.psu.edu/ HTH, JDB ___ -- 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] What do you use? .conf or AEL?
I use them both; my legacy dialplan is all .conf and new stuff is .ael. I find AEL to be the better option when jumping around, but that's just my opinion. Mik Alan Lord (News) wrote: Hi all, I built my first asterisk using the traditional (?) .conf files and constructs. I recall reading books at the time about AEL but it seemed new and untested so I left it alone. Now, I'm interested to poll the audience here to see if I should look into using AEL instead (or in addition to) for future work. TIA ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Rewriting numbers while processing dial plan?
What I am trying to do, is get rid of the initial + in phone numbers
coming in from VoIP clients on mobile phones. I have outgoing extensions
that choose which of two providers to choose (based on cost for different
destinations), and I was hoping not having to have two sets of extension
rules - one for the 00 and one for the + variety.
You dont have to keep two sets. Just rewrite + by 00 and jump to appropriate
context/extension.
I use this for almost the same, just replacing 00xx, 011 and 1 by
+ or +1.
[long-distance]
exten = _00.,1,Goto(+${EXTEN:2},1)
exten = _1NXXNXX,1,Goto(+${EXTEN},1)
exten = _011.,1,Goto(+${EXTEN:3},1)
;USA
exten = _+1NXXNXX,1,Answer()
exten = _+1NXXNXX,n,Macro(enumdial,${EXTEN})
exten = _+1NXXNXX,n,Set(CALLERID(num)=+18579284409)
exten = _+1NXXNXX,n,Playback(pls-hold-while-try)
and so on...
Martin
An example of how I'm having to do this now:
[outgoing]
exten = _00.,1,Verbose(International call 00 - Vyke)
exten = _00.,n,Dial(SIP/vyke/$EXTEN,30,tr)
exten = _00.,n,Hangup
exten = _+.,1,Verbose(International call + - Vyke)
exten = _+.,n,Dial(SIP/vyke/00${EXTEN:1},30,tr)
exten = _+.,n,Hangup
I was however hoping that it'd be possible to have a general rule that
would match the initial +, rewrite it to 00 and continue with the first of
the two patterns shown above.
A banale example (which does not work):
[outgoing]
exten = _+.,1,Goto(outgoing,00${EXTEN:1},1)
exten = _00.,1,Verbose(International call 00 - Vyke)
exten = _00.,n,Dial(SIP/vyke/$EXTEN,30,tr)
exten = _00.,n,Hangup
What am I doing wrong here?
Thanks in advance for your kind assistance!
Best regards
Jan
_
Twice the fun—Share photos while you chat with Windows Live Messenger.
Learn more.
http://www.microsoft.com/uk/windows/windowslive/products/messenger.aspx
___
-- 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] Aastra phone crashes with Asterisk 1.6
I upgraded my office server from 1.4.22 to 1.6.0.5 on the weekend and after some testing there seems to be a compatibility problem when using Aastra phones. With 1.6.0.5 all incoming calls to all Aastra phones were dropped after a minute or so. I installed 1.6.1-rc1 and now the newer Aastra phones (5xi) work properly. The problem remains with the older phones (9112i, 9133i and 480i). If I dial any of those phones the call will drop after a minute or so and the phone will crash. After a reboot the phone is back up. If you dial from the phone you can talk as long as you want, only inbound calls have the problem. I have the latest firmware for all phones. What changes to SIP may cause this in Asterisk 1.6? Is there a way to be compatible with 1.4? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk + voxbone == Failed to authenticate user
Tobias Wolf a écrit : Johan Dindaine - Asterisk schrieb: Hi every all, since a few weeks I came back to asterisk and tried to install version 1.6. The installation went fine so I decided to buy new dids on Voxbone. I have added the sip peers of Voxbone Belgium1 like this in the sip.conf [81.201.82.39] host=dynamic type=friend insecure=very context=your_context canreinvite=no qualify=no deny=0.0.0.0/0.0.0.0 permit=81.201.82.39/255.255.255.255 but unfortunately when I receive a call I got this nice error: handle_request_invite: Failed to authenticate user 075 sip:075x...@voxbone.com;tag=76596. I am in doubt now because with the insecure=very, I must receive any incoming calls from from voxbone (81.201.82.39) without any problems. Do you know how to fix this please? Hi, we are also using Voxbone Dids and we have no problems: Here is a sample defintion from my sip.conf: [81.201.83.14] host = 81.201.83.14 type = friend insecure = port,invite context = voxbone canreinvite=no Hope this helps ... Regards Tobias Wolf ___ -- 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 I just modify my sip.conf file to match your configuration provided above and I also printed the debug that I received from voxbone from a SIP SET DEBUG that you can see below. What I don't get is with insecure=very or insecure=port,invite the IP Address of Voxbone should be able to send me an INVITE request without any problems. I simply don't get it. This is the log that I get for anyone who could help me. Thanks for the help --- SIP read from 81.201.82.39:5060 --- INVITE sip:442071xxx...@87.xx.xx.xx SIP/2.0 Call-ID: 7f9d62504fee06b1070e8a534cd92...@81.201.82.39 CSeq: 102 INVITE From: 075054X sip:075054xx...@voxbone.com;tag=7419 To: sip:4420710xx...@87.xx.xx.xx Via: SIP/2.0/UDP 81.201.82.39:5060;branch=z9hG4bKfe02774c6b7669b771be486f22b7bad9 Max-Forwards: 69 Content-Type: application/sdp Contact: sip:075054xx...@81.201.82.39:5060;transport=udp User-Agent: Vox Callcontrol Content-Length: 311 v=0 o=root 11023 11023 IN IP4 81.201.82.27 s=session c=IN IP4 81.201.82.27 t=0 0 m=audio 17574 RTP/AVP 8 0 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv - --- (11 headers 15 lines) --- Sending to 81.201.82.39 : 5060 (no NAT) Using INVITE request as basis request - 7f9d62504fee06b1070e8a534cd92...@81.201.82.39 Found no matching peer or user for '81.201.82.39:5060' [Feb 10 22:25:21] NOTICE[4313]: chan_sip.c:14422 handle_request_invite: Failed to authenticate user 075054X sip:075054xx...@voxbone.com;tag=7419 --- Reliably Transmitting (no NAT) to 81.201.82.39:5060 --- SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 81.201.82.39:5060;branch=z9hG4bKfe02774c6b7669b771be486f22b7bad9;received=81.201.82.39 From: 075054X sip:075054xx...@voxbone.com;tag=7419 To: sip:442071xxx...@87.xx.xx.xx;tag=as082cd51d Call-ID: 7f9d62504fee06b1070e8a534cd92...@81.201.82.39 CSeq: 102 INVITE User-Agent: XIVO PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 Scheduling destruction of SIP dialog '7f9d62504fee06b1070e8a534cd92...@81.201.82.39' in 32000 ms (Method: INVITE) barthez*CLI --- SIP read from 81.201.82.39:5060 --- ACK sip:442071xxx...@87.xx.xx.xx SIP/2.0 Call-ID: 7f9d62504fee06b1070e8a534cd92...@81.201.82.39 CSeq: 102 ACK From: 075054X sip:075054xx...@voxbone.com;tag=7419 To: sip:442071xxx...@87.xx.xx.xx;tag=as082cd51d Via: SIP/2.0/UDP 81.201.82.39:5060;branch=z9hG4bKfe02774c6b7669b771be486f22b7bad9 Max-Forwards: 69 User-Agent: Vox Callcontrol Content-Length: 0 - --- (9 headers 0 lines) --- Really destroying SIP dialog '7f9d62504fee06b1070e8a534cd92...@81.201.82.39' Method: ACK barthez*CLI --- SIP read from 81.106.106.35:8022 --- ___ -- 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] Digium TE220 card partially detected
lspci can detect the card: 03:08.0 Communication controller: Digium, Inc. Device 0220 (rev 02) dahdi_hardware also: astpbx ~ # dahdi_hardware pci::03:08.0 wct4xxp+ d161:0220 Wildcard TE220 (4th Gen) Fine. The card is indeed handled by the module (+). astpbx ~ # But dahdi_tool does not show the card and dahdi_cfg gives DAHDI_SPANCONFIG failed on span 1: Invalid argument (22) Next: are you sure that it is configured for E1 and not for T1? What is the output of: cat /proc/dahdi/1 Please see what's there: NET: Registered protocol family 10 lo: Disabled Privacy Extensions eth0: no IPv6 routers present ACPI: PCI Interrupt Link [APC6] enabled at IRQ 16 ACPI: PCI Interrupt :03:08.0[A] - Link [APC6] - GSI 16 (level, low) - IRQ 16 Found TE2XXP at base address f510, remapped to f8d64000 TE2XXP version c01a016a, burst ON Octasic optimized! FALC version: 0005, Board ID: 09 Reg 0: 0x2b817400 Reg 1: 0x2b817000 Reg 2: 0x Reg 3: 0x Reg 4: 0x0001 Reg 5: 0x Reg 6: 0xc01a016a Reg 7: 0x90001300 Reg 8: 0x Reg 9: 0x00ff Reg 10: 0x004a Found a Wildcard: Wildcard TE220 (4th Gen) astpbx dahdi # cat /proc/dahdi/1 Span 1: DAHDI_DUMMY/1 DAHDI_DUMMY/1 (source: RTC) 1 (MASTER) astpbx dahdi # astpbx dahdi # ls /proc/dahdi/1 /proc/dahdi/1 astpbx dahdi # ___ -- 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] Digium TE220 card partially detected
Maxim Litnitskiy wrote: FALC version: 0005, Board ID: 09 The board ID switch (the rotary switch on the board) is set to 9 (nine). If you are going to have the switches set to anything other than zero, there must be at least one board with it set to zero, then one set to one, then one set to two, etc. Skipping numbers causes boards to be ignored by the driver at startup time, even though they were located. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- 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] Max person in meetme conference
Is it possible to restrict the maximum number of person in a particular conference room? for example, I have meetme.conf [general] [rooms] conf = c1,1234 conf = c2,1234 I want c1 to allow 10 persons max in the conference and c2 5 persons max How can I do that? 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] Max person in meetme conference
Hi, Try this: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+MeetMeCount Nhadie Emmanuel Bruno wrote: Is it possible to restrict the maximum number of person in a particular conference room? for example, I have meetme.conf [general] [rooms] conf = c1,1234 conf = c2,1234 I want c1 to allow 10 persons max in the conference and c2 5 persons max How can I do that? 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 ___ -- 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] Max person in meetme conference
Thanks! it works fine for me, I though there was another way maybe in the
config file for meetme.conf to set that option. The example given in that
link has a typo that needs to be fixed:
exten = s,2,Gotoif,$[${count} = ${CONFMAX}]?103
there's an extra , and the evaluation has to be between parenthesis.
exten = s,2,Gotoif($[${count} = ${CONFMAX}]?103)
On Tue, Feb 10, 2009 at 6:52 PM, Nhadie nha...@gmail.com wrote:
Hi,
Try this:
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+MeetMeCount
Nhadie
Emmanuel Bruno wrote:
Is it possible to restrict the maximum number of person in a particular
conference room?
for example, I have
meetme.conf
[general]
[rooms]
conf = c1,1234
conf = c2,1234
I want c1 to allow 10 persons max in the conference
and c2 5 persons max
How can I do that?
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
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] put the hostname of asterisk in the callerid uri to avoid NAT problems
Hi, Have you tried using externip in your sip.conf? By setting the correct localnet, any SIP packets that goes elsewhere will use the value in externip. This might solve your problem. Regards, Steve nik600 wrote: On Sat, Feb 7, 2009 at 8:31 AM, nik600 nik...@gmail.com wrote: hi is it possible to set up in the dialplan (on in sip.conf, or something else) the hostname of the outgoing uri call? This is my scenario: - CCM integrated with Asterisk via h323 - SIP user registerd to Asterisk - Asterisk is behind NAT - Asterisk ip is 10.10.10.2 - SIP user view Asterisk as 10.10.15.2 (Asterisk is behind NAT) When the CCM calls the SIP user the call works perfectly. The problem is that the SIP user receives the call with this uri: sip:x...@10.10.10.2 The call works properly and the audio goes in both directios, BUT if the SIP user does a redial (after the hangup) the call is forwarded to x...@10.10.10.2 that is the wrong address. I've tried to force SIP_HEADER(CONTACT) in the dialplan with a Set but it seems that i can't due to security reason. Is it possible to avoid this problem? Thanks -- /*/ nik600 http://www.kumbe.it Do you think that is a bug or a miss configuration, or simply is not possible to avoid that because it is hard-coded? 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] Aastra phone crashes with Asterisk 1.6
Carlos Chavez schrieb: I upgraded my office server from 1.4.22 to 1.6.0.5 on the weekend and after some testing there seems to be a compatibility problem when using Aastra phones. If I dial any of those phones the call will drop after a minute or so and the phone will crash. I'm not saying it's not an Asterisk problem. Maybe something in the SIP signaling/RTP is broken. However it's definitely an Aastra problem. No matter how broken the signaling -- that's no excuse for crashing. So make sure to report the issue to Aastra as well. Philipp Kempgen -- AMOOCON 2009, May 4-5, Rostock / Germany - http://www.amoocon.de Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- 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
