[asterisk-users] Question about DIALSTATUS NOANSWER
Hi, according to the wiki the value NOANSWER for the channel variable DIALSTATUS means: No answer. The dial command reached its number, the number rang for too long, then the dial timed out. In out dialplan we grap all these events with exten = s-NOANSWER,1,Playback(sometext) exten = s-NOANSWER,2,WAIT(1) exten = s-NOANSWER,3,Hangup() The dial commands for internal and external connections let the phone on the other ring for 60 seconds. But the cdr file Master.csv contains entries like the following: ,15,s-NOANSWER,fehler,User1 15,SIP/User1-b67e5c28,CAPI/ISDN1#02/062xx-6252,Playback,ungueltige_nummer,2008-02-11 10:32:31,2008-02-11 10:32:31,2008-02-11 10:32:32,1,1,ANSWERED,DOCUMENTATION,1202725951.43008, 1,1, means, that both duration and billsec are only 1 second. If NOANSWER stands for a timeout, how can the DIAL command timeout after only 1 second, when the phone should ring 60 seconds? Could it be, that NOANSWER catches more than one event? Thanks for your help, Stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Geschaeftsfuehrer Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice-over-IP-Loesungen ___ -- 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] Communication between two asterisk server
Hi, I want that an sjphone registered using serverA can call to an sjphone registered using serverB and vice vers. I want to know how two asterisk server communicate to each other. Please let me know, for that, what configuration file I have to change. Thanking you, Regards, Preeta Pandey The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.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] Problem with DTMF dialing
On Wed, Feb 13, 2008 at 10:48 AM, Andres Jimenez [EMAIL PROTECTED] wrote: On Tue, Feb 12, 2008 at 10:03 PM, Andrew Joakimsen [EMAIL PROTECTED] wrote: Maybe it is related but with PRI Asterisk does not generate any tone it sends a signal regarding your keypress. If you are using SIP phones make sure the dtmfmode in use is RFC2833. I have just double check and my phones use DTMF in RFC2833 mode. I wil try to downgrade my zaptel later today CONFIRMED. The problem disappears after downgrading zaptel from 1.4.8 to 1.4.7 -- Andres Jimenez GPG : http://www.andresin.com/gpg/[EMAIL PROTECTED] -- Andres Jimenez GPG : http://www.andresin.com/gpg/[EMAIL PROTECTED] ___ -- 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] gtalk and dtmf
Hi Adam, I've been googling for half an hour, i found some sort of jingle protocol which i'm not sure what to use for but it might be the solution? It seems to me that my problem is sending the dtmf tones, not receiving them, so this is really gtalk related. You've spotted the problem, you cannot send DTMF tones with your GoogleTalk client, even though Asterisk is capable of receiving them. I know the people of the Jabbin project were working on this topic, maybe you can try their gtalk compliant client : http://www.jabbin.com Philippe ___ -- 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] G729 transcoding and clicking
Hello, We have an Asterisk server receiving calls using G711 (ulaw). This server has rerouters de calls to other server using G729 (we bought the codecs, installed, sip show channels shows the codec properly, etc.) Using G729, there is a click while talking. Well, more than a click it seems that voice is missing during some ms (maybe 100 ms?) Using G711 we don't have any click. Where we could watch for it? Is it possible to add some Jitter buffer? Thank you, -- Carles Pina i EstanyGPG id: 0x8CBDAE64 http://pinux.info Manresa - Barcelona ___ -- 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] Pass arguments from extensions.conf
On Thu, Feb 14, 2008 at 9:52 PM, Naveen Palani [EMAIL PROTECTED] wrote: How can i pass the arguments from my dialplan to the ruby file. Is there a way i can do it with the agi script? Set them as variables in your extensions.conf and use them inside your agi scripts. raj ___ -- 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] UK -999 dialing issue
Jared Smith [EMAIL PROTECTED] writes: I've been suggesting that for about four years now (long before I ever started working for Digium), but the core Asterisk developers tell me it will have a very negative impact on Asterisk performance. The only reason why it has a negative impact is because everything is reparsed all the time. If the whole file was parsed ahead-of-time, having switchable regexp syntaxes would only cost a mispredicted jump each time. Ahead-of-time parsing is currently impossible because of realtime dial plans. That's not a showstopper for me personally, because I think the current realtime dial plan support is a bad idea anyway, but obviously there are others with different opinions. Another problem is the ability to add extensions at runtime, but it should not be impossible to call a parser function before inserting new extensions. /Benny ___ -- 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] 57iCT BLF problem
I guess we ought to add ...beyond downgrading the firmware to 2.0.2 to that. :) Paul Hales wrote: We upgrade 2 of our Aastra 57iCT to the latest firmware (2.1.2.30) and the BLF indicators no longer function. Has anyone had a similar issue? And a solution? PaulH ___ -- 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] Monitor Asterisk
14 feb 2008 kl. 22.35 skrev Benny Amorsen: Matthew J. Roth [EMAIL PROTECTED] writes: Yes, asterisk -rx will only allow you to execute CLI commands. It also tends to spew out a bunch of garbage that makes parsing difficult unless verbosity is always set to 0. It would be very handy if it was possible to turn off messages that aren't the direct result of a command in a particular instance of asterisk -r. Perhaps asterisk -r -q? In the long run we're trying to move to using the manager for all parsing by adding a lot of new manager events and actions. If there's something missing that you only can do or information you only can get in the CLI, please tell us. I would also like to see manager wrappers that produce data that is easy to handle for scripts, like a wrapper that produces show channels consise in various formats. Do we have a perl programmer on the list? Such a generic script could be added to the scripts library in the Asterisk distribution. /O ___ -- 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 DNS SIP issue
15 feb 2008 kl. 07.08 skrev Kevin Kiely: The other day my asterisk local SIP clients got hung when my provider had a DNS failure. All registrations went dead (even the ones that were IP addresses) and all sip peers went offline. I know this was know problem at one point is there any update on this when using a FQDN for one of the peer addresses in sip.conf? No, Asterisk (like many other pieces of software) is not very good in handling DNS failures. If you have a local caching DNS server on the same server, this wouldn't happen. At some point we need support for asynchronus DNS, meaning that Asterisk would time out and not hang when there's a lack of answer on DNS requests. /O ___ -- 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] Digium stopped TDM400P production: alternatives??
Hi, Digium stopped to produce TDM400P and the new TDM410 is too new to find it in our shops. The only alternative available is a fully-compatible Openvox product...but is it really fully-compatible? Any experience about Openvox products (card and zaptel versions, etc...)? Thank you! Giorgio. ___ -- 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] SNMP monitoring
Thanks guys, On two cloned machines, on one I tried: yum install lm_sensors-devel bzip2-devel (ignoring newt, and these were the only ones missing) ..and it compiled ok. Then on the other I just added lm_sensors-devel and the configure -with-net-snmp worked ok, but it didn't compile the snmp module (but didn't complain either). So then I added bzip2-devel and all was well on the second machine (so both needed). So now the res_snmp.so module is loaded. I'll continue to work out what else is needed (I've no res_snmp.conf file, or net-snmp config updates done yet). Adrian From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ricardo Carvalho Sent: 15 February 2008 00:29 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SNMP monitoring Maybe you'r right and newt isn't really necessary. I just read somewhere that those dependencies were needed, I've installed them and it worked... Try to only install the other ones and if res_snmp gets compiled without it, great! Regards, Ricardo Carvalho. On Fri, Feb 15, 2008 at 12:01 AM, Darrick Hartman (lists) [EMAIL PROTECTED] wrote: Ricardo Carvalho wrote: I had the same problem some time ago... You got to install also this packages: net-snmp-devel newt-devel lm_sensors-devel bzip2-devel That should do it! Why would this depend on newt? net-snmp and lm-sensor headers and libraries make sense. newt doesn't make any sense as a dependency. Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.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] Communication between two asterisk server
hi,preeta you have to change sip.conf in both server. suppose, server 1 and server 2 both are asterisk server. you want to call from server 1 to server 2. then, in ser-1, sip.conf [general] register= user:[EMAIL PROTECTED] [user] type=friend fromuser=user username=user secret=pass host=ipofserver2 context=any in server2, sip.conf [user] type=friend username=user secret=user host=dynamic context=anyyouwant Bhrugu Mehta (SAI INFO SYSTEM LTD.) On 2/15/08, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi, I want that an sjphone registered using serverA can call to an sjphone registered using serverB and vice vers. I want to know how two asterisk server communicate to each other. Please let me know, for that, what configuration file I have to change. Thanking you, Regards, Preeta Pandey The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.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] 1.4 and IAX Trunks ...
Something I've just noticed that might persuade me to move to 1.4 ... in iax.conf, there is a new option: transfer=mediaonly Does this mean that it keeps itself in the loop as far as signalling/CDR is concerned, but lets the media stream go between the 2 endpoints? ie. Asterisk A - Asterisk B - Asterisk C Where B keeps track of CDRs, so right now I have notransfer=yes in my 1.2 config files, but if I put transfer=mediaonly in my 1.4 config file, will it then let A and C talk directly without going via B, but keep B in the loop for billing information? And if so.. Does this require 1.4 at all 3 sites, or just Site B ? Thanks, Gordon ___ -- 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] Communication between two asterisk server
Hi Bhrugu , Thanks for the reply. I will check it off. Regards, Preeta -Original Message- From: [EMAIL PROTECTED] on behalf of Bhrugu Mehta Sent: Fri 2/15/2008 5:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Communication between two asterisk server hi,preeta you have to change sip.conf in both server. suppose, server 1 and server 2 both are asterisk server. you want to call from server 1 to server 2. then, in ser-1, sip.conf [general] register= user:[EMAIL PROTECTED] [user] type=friend fromuser=user username=user secret=pass host=ipofserver2 context=any in server2, sip.conf [user] type=friend username=user secret=user host=dynamic context=anyyouwant Bhrugu Mehta (SAI INFO SYSTEM LTD.) On 2/15/08, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi, I want that an sjphone registered using serverA can call to an sjphone registered using serverB and vice vers. I want to know how two asterisk server communicate to each other. Please let me know, for that, what configuration file I have to change. Thanking you, Regards, Preeta Pandey The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.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 The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com winmail.dat___ -- 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] HPEC
Al lists wrote: Just wondering how your experience is with HPEC, Is it just for analog interfaces or we can use it on TE122 as well? The HPEC can be used with any Zaptel-supported interface, but we don't provide free licenses for people to use them with T1/E1 cards, because the potential CPU load running HPEC on 24/30 channels in 128ms mode is quite high and could cause problems on the system. However, if you don't have that many active channels at once, or you have a very powerful system, or many other variables are in your favor, you can certainly give it a try. -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- 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 stopped TDM400P production: alternatives??
Giorgio Incantalupo wrote: Digium stopped to produce TDM400P and the new TDM410 is too new to find it in our shops. The only alternative available is a fully-compatible Openvox product...but is it really fully-compatible? Any experience about Openvox products (card and zaptel versions, etc...)? Every distributor that carried the TDM400P should have TDM410s in stock already. Where are you located, and who do you buy Digium cards from? -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- 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 stopped TDM400P production: alternatives??
Sangoma makes a good card. On 2/15/08, Kevin P. Fleming [EMAIL PROTECTED] wrote: Giorgio Incantalupo wrote: Digium stopped to produce TDM400P and the new TDM410 is too new to find it in our shops. The only alternative available is a fully-compatible Openvox product...but is it really fully-compatible? Any experience about Openvox products (card and zaptel versions, etc...)? Every distributor that carried the TDM400P should have TDM410s in stock already. Where are you located, and who do you buy Digium cards from? -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- 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] Digium stopped TDM400P production: alternatives??
It is fairly easy on a fresh install since the Sangoma ./Setup install script can create all three configuration files for you. Thanks, Steve Totaro On Fri, Feb 15, 2008 at 8:11 AM, Rob Hillis [EMAIL PROTECTED] wrote: The cards themselves are okay, but the extra level of configuration is a pain in the proverbial. Zaptel is already double-configured in both zaptel.conf and zapata.conf (that's not a complaint - I understand the reason for the separation) but the Sangoma cards require a third level of configuration in Wanpipe. Steve Totaro wrote: Sangoma makes a good card. On 2/15/08, Kevin P. Fleming [EMAIL PROTECTED] wrote: Giorgio Incantalupo wrote: Digium stopped to produce TDM400P and the new TDM410 is too new to find it in our shops. The only alternative available is a fully-compatible Openvox product...but is it really fully-compatible? Any experience about Openvox products (card and zaptel versions, etc...)? Every distributor that carried the TDM400P should have TDM410s in stock already. Where are you located, and who do you buy Digium cards from? -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- 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 ___ -- 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 stopped TDM400P production: alternatives??
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I would say email Kevin what he asked. The problem with switching to a clone company is you get what you pay for. Sticking with Digium you at least have support. and 3 clone cards and hours of troubleshooting later you will wish you hadn't been all cheap. Rob Hillis wrote: The cards themselves are okay, but the extra level of configuration is a pain in the proverbial. Zaptel is already double-configured in both zaptel.conf and zapata.conf (that's not a complaint - I understand the reason for the separation) but the Sangoma cards require a /third/ level of configuration in Wanpipe. Steve Totaro wrote: Sangoma makes a good card. On 2/15/08, Kevin P. Fleming [EMAIL PROTECTED] wrote: Giorgio Incantalupo wrote: Digium stopped to produce TDM400P and the new TDM410 is too new to find it in our shops. The only alternative available is a fully-compatible Openvox product...but is it really fully-compatible? Any experience about Openvox products (card and zaptel versions, etc...)? Every distributor that carried the TDM400P should have TDM410s in stock already. Where are you located, and who do you buy Digium cards from? -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- 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 !DSPAM:47b591c7282271152562594! -- ___ -- 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 !DSPAM:47b591c7282271152562594! - -- James Finstrom Rhino Equipment Corp. Tel: 1-800-785-7073 ext. 6344 FAX: +1 (480) 961-1826 IP: asterisk.rhinoequipment.com ext 6344 FWD: 633686 ext 6344 THIS COMMUNICATION MAY CONTAIN CONFIDENTIAL AND/OR OTHERWISE PROPRIETARY MATERIAL and is thus for use only by the intended recipient. If you received this in error, please contact the sender and delete the email and its attachments from all computers. -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHtZVZdloC7YyaIOoRAuHXAJ0WD4UCOQzea43CCVXG32hDnxaADgCdHRUe 34tNh/zgUxxoOkAaQbB7z5Y= =TlLb -END PGP SIGNATURE- ___ -- 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 stopped TDM400P production: alternatives??
The cards themselves are okay, but the extra level of configuration is a pain in the proverbial. Zaptel is already double-configured in both zaptel.conf and zapata.conf (that's not a complaint - I understand the reason for the separation) but the Sangoma cards require a /third/ level of configuration in Wanpipe. Steve Totaro wrote: Sangoma makes a good card. On 2/15/08, Kevin P. Fleming [EMAIL PROTECTED] wrote: Giorgio Incantalupo wrote: Digium stopped to produce TDM400P and the new TDM410 is too new to find it in our shops. The only alternative available is a fully-compatible Openvox product...but is it really fully-compatible? Any experience about Openvox products (card and zaptel versions, etc...)? Every distributor that carried the TDM400P should have TDM410s in stock already. Where are you located, and who do you buy Digium cards from? -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- 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] Digium stopped TDM400P production: alternatives??
Hi Kevin, unfortunately I live in Italy and you is not so easy for us to get electronic stuff. Let's wait and see what happens.:) Giorgio Kevin P. Fleming wrote: Giorgio Incantalupo wrote: Digium stopped to produce TDM400P and the new TDM410 is too new to find it in our shops. The only alternative available is a fully-compatible Openvox product...but is it really fully-compatible? Any experience about Openvox products (card and zaptel versions, etc...)? Every distributor that carried the TDM400P should have TDM410s in stock already. Where are you located, and who do you buy Digium cards from? -- _ Giorgio Incantalupo, mailto:[EMAIL PROTECTED] FGA srl - http://www.fgasoftware.com - [EMAIL PROTECTED] - The Agile PBX http://www.voiceatwork.eu Tel: 02997663.14, Fax: 0291390172 ___ -- 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 stopped TDM400P production: alternatives??
Hi Steve, I've already tried a Sangoma card and it behaves the same as TDM400P. But the problem arises for example when I have to change a broken card on an old PBX keeping the modules, that's why I need a clone card like Openvox (Sangoma modules are different as you know) Moreover I'd like to avoid installing the Sangoma driver on the PBX. Steve Totaro wrote: It is fairly easy on a fresh install since the Sangoma ./Setup install script can create all three configuration files for you. Thanks, Steve Totaro On Fri, Feb 15, 2008 at 8:11 AM, Rob Hillis [EMAIL PROTECTED] wrote: The cards themselves are okay, but the extra level of configuration is a pain in the proverbial. Zaptel is already double-configured in both zaptel.conf and zapata.conf (that's not a complaint - I understand the reason for the separation) but the Sangoma cards require a third level of configuration in Wanpipe. Steve Totaro wrote: Sangoma makes a good card. On 2/15/08, Kevin P. Fleming [EMAIL PROTECTED] wrote: Giorgio Incantalupo wrote: Digium stopped to produce TDM400P and the new TDM410 is too new to find it in our shops. The only alternative available is a fully-compatible Openvox product...but is it really fully-compatible? Any experience about Openvox products (card and zaptel versions, etc...)? Every distributor that carried the TDM400P should have TDM410s in stock already. Where are you located, and who do you buy Digium cards from? -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- 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 ___ -- 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 -- _ Giorgio Incantalupo, mailto:[EMAIL PROTECTED] FGA srl - http://www.fgasoftware.com - [EMAIL PROTECTED] - The Agile PBX http://www.voiceatwork.eu Tel: 02997663.14, Fax: 0291390172 ___ -- 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 stopped TDM400P production: alternatives??
James, If you were replying to the original post about Openvox or specified that is what you were referring to, maybe I would not take issue but to reply to a suggesting to use Sangoma with what you did is absolutely misleading. There is nothing cheap or clone about Sangoma's cards. asterisk.rhinoequipment.com hm. Thanks, Steve Totaro James Finstrom wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I would say email Kevin what he asked. The problem with switching to a clone company is you get what you pay for. Sticking with Digium you at least have support. and 3 clone cards and hours of troubleshooting later you will wish you hadn't been all cheap. Rob Hillis wrote: The cards themselves are okay, but the extra level of configuration is a pain in the proverbial. Zaptel is already double-configured in both zaptel.conf and zapata.conf (that's not a complaint - I understand the reason for the separation) but the Sangoma cards require a /third/ level of configuration in Wanpipe. Steve Totaro wrote: Sangoma makes a good card. On 2/15/08, Kevin P. Fleming [EMAIL PROTECTED] wrote: Giorgio Incantalupo wrote: Digium stopped to produce TDM400P and the new TDM410 is too new to find it in our shops. The only alternative available is a fully-compatible Openvox product...but is it really fully-compatible? Any experience about Openvox products (card and zaptel versions, etc...)? Every distributor that carried the TDM400P should have TDM410s in stock already. Where are you located, and who do you buy Digium cards from? -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- 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] 57iCT BLF problem
Paul Hales wrote: We upgrade 2 of our Aastra 57iCT to the latest firmware (2.1.2.30) and the BLF indicators no longer function. Has anyone had a similar issue? And a solution? PaulH ___ -- 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 Paul, I have a couple of installations with Useragent: Aastra 57iCT/2.1.2.30 and Asterisk 1.4.15. If you do a show hints, does it show a watcher for each hint? ___ -- 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] Pass arguments from extensions.conf
On Thu, 2008-02-14 at 21:52 +0530, Naveen Palani wrote: How can i pass the arguments from my dialplan to the ruby file. Is there a way i can do it with the agi script? Sure... simply pass your arguments to the AGI() application, and they'll show up as if they were command-line arguments to your ruby program. For example, if you wanted to pass the arguments red and green, you would do something like: exten = 123,1,AGI(/path/to/your/ruby/program,red,green) -- Jared Smith Community Relations Manager Digium, Inc. ___ -- 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] 1.4 and IAX Trunks ...
On Fri, 2008-02-15 at 11:58 +, Gordon Henderson wrote: Something I've just noticed that might persuade me to move to 1.4 ... in iax.conf, there is a new option: transfer=mediaonly Does this mean that it keeps itself in the loop as far as signalling/CDR is concerned, but lets the media stream go between the 2 endpoints? Yes, that's exactly what it means. :-) -- Jared Smith Community Relations Manager Digium, Inc. ___ -- 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] 1.4 and IAX Trunks ...
On Fri, 15 Feb 2008, Jared Smith wrote: On Fri, 2008-02-15 at 11:58 +, Gordon Henderson wrote: Something I've just noticed that might persuade me to move to 1.4 ... in iax.conf, there is a new option: transfer=mediaonly Does this mean that it keeps itself in the loop as far as signalling/CDR is concerned, but lets the media stream go between the 2 endpoints? Yes, that's exactly what it means. :-) Excellent! What about the need for 1.4 at all sites? Is it sufficient to just have it in the man in the middle site? Thanks, Gordon ___ -- 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] ISDN PRIs and taking a server down formaintenance - blocking issue
Yes the 'stop gracefully' is what effectively blocks the calls as the telco seems to take it as we are answering the calls instead of seeing them as busy. I will look at implementing some sort of way of busying out all the zaptel channels, so that we eventually busy out all 120 channels (4x E1) and then can cleanly take the server offline while our telco presents the calls to the next Asterisk servers correctly. This would be a great way of busying out the server for maintenance while still allowing our inbound calls. Many thanks, Andrew _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brent Davidson Sent: 15 February 2008 00:30 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ISDN PRIs and taking a server down formaintenance - blocking issue Correct me if I'm wrong, but as I understand it your issue is that when you give Asterisk the stop gracefully command it waits until all active calls have finished before it takes the ISDN down but gives busy signals to new incoming calls on idle channels. If this is the case then it would seem that Asterisk is actually answering the call on the incoming channel and playing a busy signal. From reading a couple of threads on another list it appears this is the case (Google: Asterisk busy out PRI to find the discussion). There also appears to be some interest in making a function do what you need in the future. For the time being, however, a simple solution would be to create a temporary dial-plan that follows each outgoing hangup with a dial command to either a test number or some other service that will just keep playing audio down the line and not hangup. (You'd probably need to set some variable to know which channels had been busied) When you need to take down a server, load this dial plan and wait for all channels to call the busy number, then hang them all up and issue a stop now. It's a messy solution, but it's all I can think of without hacking code. The only other way I'd know would be to hack the code for the dial or answer command and build another command that simply takes the channel off-hook and leaves it there. Good luck, Brent Davidson Lyle Giese wrote: If you take Asterisk down, the PRI should go down as the D channel is down. Then the telco should KNOW that there is trouble with the PRI and those channels are in trouble busy and not availible. If the telco still tries to push a call to a channel on a PRI that is down, then the telco is at fault. Lyle Matt wrote: That does sound like what is happening.. Telco knows channel 1-23 are not busy (so far as they are concerned), however.. so far as you are concerned, they are busy.. so telco sends the call down... but the equipment doesn't take it. I would *think* the Telco could keep trying channels down the hunt group, but maybe not? We have, in the past, seen this issue with our dial-up modem banks.. especially if I would take one offline. However, it is not a big enough issue (i.e. we don't take things down that often) for me to look into it fully. On Thu, Feb 14, 2008 at 4:07 PM, Don Kelly [EMAIL PROTECTED] wrote: I think the problem is that the telco presents the call on a specific channel, then zaptel tells it that the channel is busy. We need to be able to tell the telco that each unused channel on a given span is unavailable, and it will determine that the others are in use and will present the call on a channel on another span. A rather ugly work-around (since Andrew seems to have lots of channels available, and one would assume that maintenance of this nature would occur during slow periods) would be to make calls to a DID in the same trunk group on all idle channels on the span shutting down then, when all channels on the span are in use and none of them are doing anything useful, take the span down hard so the telco will divert all calls to another span. --Don Don Kelly PCF Corp Real Support for your Virtual Office TM 651 842-1000 888 Don Kell(y) 651 842-1001 fax _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Sent: Thursday, February 14, 2008 2:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ISDN PRIs and taking a server down formaintenance - blocking issue Honestly.. this sounds like a telco issue.I understand what the other person is saying about the PRI still being technically up... BUT... if the channel is BUSY/BLOCKED/WHATEVER, the Telco should be forwarding the call to the next available channel, which they clearly are not doing. On Thu, Feb 14, 2008 at 8:29 AM, Andrew Smith [EMAIL PROTECTED] wrote: Hi Tim, Imagine the scenario where we had 10x Asterisk servers, with calls presenting sequentially starting from the first server, then server two, etc. If we took down the first server for maintenance with 'asterisk -rx stop gracefully' we then will block all incoming calls to all
Re: [asterisk-users] Digium stopped TDM400P production: alternatives??
Steve, Yes I work for Rhino that is no Secret. If you read the post I was responding to the thread not pimping my own products. I am not sure if your a Sangoma fanboy or employee since you are apparently offended by my response, however he wasn't asking to be sold to he was asking about specific products. So there it is yes I work for Rhino and I could have easily given one of our italian distributors but he didn't ask for that. It is not appropriate to troll the list and push your products unsolicited. If someone is looking for a recommendation for a card brand fine. If they need a solution like ADID or they need to accommodate funky CPC signals from their telco which Rhino does fine it is on subject. If someone asks should I use openvox to replace my digium you don't pimp your product because it wasn't asked for. If you want my honest opinion. I prefer people use Rhino products. I believe our products and support are superior but if you don't use our cards use Digium. If your reply is any indication on how Sandoma works I can honestly say go use a cheap clone before sangomaN they may not support you but at least they are open about being here just for the money. James Finstrom Rhino Equipment Corp. http://www.rhinoequipment.com -Original Message- From: Steve Totaro [EMAIL PROTECTED] Date: Fri, 15 Feb 2008 08:45:50 To:Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [asterisk-users] Digium stopped TDM400P production: alternatives?? James, If you were replying to the original post about Openvox or specified that is what you were referring to, maybe I would not take issue but to reply to a suggesting to use Sangoma with what you did is absolutely misleading. There is nothing cheap or clone about Sangoma's cards. asterisk.rhinoequipment.com hm. Thanks, Steve Totaro James Finstrom wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I would say email Kevin what he asked. The problem with switching to a clone company is you get what you pay for. Sticking with Digium you at least have support. and 3 clone cards and hours of troubleshooting later you will wish you hadn't been all cheap. Rob Hillis wrote: The cards themselves are okay, but the extra level of configuration is a pain in the proverbial. Zaptel is already double-configured in both zaptel.conf and zapata.conf (that's not a complaint - I understand the reason for the separation) but the Sangoma cards require a /third/ level of configuration in Wanpipe. Steve Totaro wrote: Sangoma makes a good card. On 2/15/08, Kevin P. Fleming [EMAIL PROTECTED] wrote: Giorgio Incantalupo wrote: Digium stopped to produce TDM400P and the new TDM410 is too new to find it in our shops. The only alternative available is a fully-compatible Openvox product...but is it really fully-compatible? Any experience about Openvox products (card and zaptel versions, etc...)? Every distributor that carried the TDM400P should have TDM410s in stock already. Where are you located, and who do you buy Digium cards from? -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- 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 !DSPAM:47b59f18311805637012918! ___ -- 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] Communication between two asterisk server
I am using IAX2, easier to get to work trow firewalls. //Mattias On Fri, Feb 15, 2008 at 1:14 PM, [EMAIL PROTECTED] wrote: Hi Bhrugu , Thanks for the reply. I will check it off. Regards, Preeta -Original Message- From: [EMAIL PROTECTED] on behalf of Bhrugu Mehta Sent: Fri 2/15/2008 5:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Communication between two asterisk server hi,preeta you have to change sip.conf in both server. suppose, server 1 and server 2 both are asterisk server. you want to call from server 1 to server 2. then, in ser-1, sip.conf [general] register= user:[EMAIL PROTECTED] [user] type=friend fromuser=user username=user secret=pass host=ipofserver2 context=any in server2, sip.conf [user] type=friend username=user secret=user host=dynamic context=anyyouwant Bhrugu Mehta (SAI INFO SYSTEM LTD.) On 2/15/08, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi, I want that an sjphone registered using serverA can call to an sjphone registered using serverB and vice vers. I want to know how two asterisk server communicate to each other. Please let me know, for that, what configuration file I have to change. Thanking you, Regards, Preeta Pandey The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.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 The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.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 -- Mattias Andersson Storskiftesvägen 6 145 60 Norsborg m. +46-70-799 44 41 h. +46-8-641 38 97 Email: [EMAIL PROTECTED] ___ -- 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 stopped TDM400P production: alternatives??
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Enough Said :) Buy Digium, or Rhino, or a Knock off but avoid the witch doctor Steve Totaro wrote: James, Huh? Trying to understand your rambling reply I just like Sangoma because they just work and have excellent support, I have no affiliation with them except being a very happy customer. You get what you pay for right? I also think Adtran or Adit are great products. Not sure about Rhino especially after your irrational response. Some spokesman, I will stick with Adtran and Adit, not some cheap knock-off.. Thanks, Steve Totaro On Fri, Feb 15, 2008 at 9:36 AM, James Finstrom [EMAIL PROTECTED] wrote: Steve, Yes I work for Rhino that is no Secret. If you read the post I was responding to the thread not pimping my own products. I am not sure if your a Sangoma fanboy or employee since you are apparently offended by my response, however he wasn't asking to be sold to he was asking about specific products. So there it is yes I work for Rhino and I could have easily given one of our italian distributors but he didn't ask for that. It is not appropriate to troll the list and push your products unsolicited. If someone is looking for a recommendation for a card brand fine. If they need a solution like ADID or they need to accommodate funky CPC signals from their telco which Rhino does fine it is on subject. If someone asks should I use openvox to replace my digium you don't pimp your product because it wasn't asked for. If you want my honest opinion. I prefer people use Rhino products. I believe our products and support are superior but if you don't use our cards use Digium. If your reply is any indication on how Sandoma works I can honestly say go use a cheap clone before sangomaN they may not support you but at least they are open about being here just for the money. James Finstrom Rhino Equipment Corp. http://www.rhinoequipment.com -Original Message- From: Steve Totaro [EMAIL PROTECTED] Date: Fri, 15 Feb 2008 08:45:50 To:Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [asterisk-users] Digium stopped TDM400P production: alternatives?? James, If you were replying to the original post about Openvox or specified that is what you were referring to, maybe I would not take issue but to reply to a suggesting to use Sangoma with what you did is absolutely misleading. There is nothing cheap or clone about Sangoma's cards. asterisk.rhinoequipment.com hm. Thanks, Steve Totaro James Finstrom wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I would say email Kevin what he asked. The problem with switching to a clone company is you get what you pay for. Sticking with Digium you at least have support. and 3 clone cards and hours of troubleshooting later you will wish you hadn't been all cheap. Rob Hillis wrote: The cards themselves are okay, but the extra level of configuration is a pain in the proverbial. Zaptel is already double-configured in both zaptel.conf and zapata.conf (that's not a complaint - I understand the reason for the separation) but the Sangoma cards require a /third/ level of configuration in Wanpipe. Steve Totaro wrote: Sangoma makes a good card. On 2/15/08, Kevin P. Fleming [EMAIL PROTECTED] wrote: Giorgio Incantalupo wrote: Digium stopped to produce TDM400P and the new TDM410 is too new to find it in our shops. The only alternative available is a fully-compatible Openvox product...but is it really fully-compatible? Any experience about Openvox products (card and zaptel versions, etc...)? Every distributor that carried the TDM400P should have TDM410s in stock already. Where are you located, and who do you buy Digium cards from? -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- 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 !DSPAM:47b5b32338121804284693! - -- James Finstrom Rhino Equipment Corp. Tel: 1-800-785-7073 ext. 6344 FAX: +1 (480) 961-1826 IP: asterisk.rhinoequipment.com ext 6344 FWD: 633686 ext 6344 THIS COMMUNICATION MAY CONTAIN CONFIDENTIAL AND/OR OTHERWISE PROPRIETARY MATERIAL and is thus for use only by the intended recipient. If you received this in error, please contact the sender and delete the email and its attachments from all computers.
Re: [asterisk-users] Automatically start after restart
I actually use daemon tools http://cr.yp.to/daemontools/daemontools-0.76.tar.gz I like it because its log handling features, it takes the stdout of asterisk and puts it in a log directory and automatically rotates the files. Doug Lytle wrote: bilal ghayyad wrote: Any script or something that can do that? The scripts are located in the Asterisk source directory under contrib/init.d Doug -- Thank you and have a wonderful day, Anthony Francis Rockynet VOIP (303) 444-7052 opt 2 [EMAIL PROTECTED] ___ -- 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 stopped TDM400P production: alternatives??
James, Huh? Trying to understand your rambling reply I just like Sangoma because they just work and have excellent support, I have no affiliation with them except being a very happy customer. You get what you pay for right? I also think Adtran or Adit are great products. Not sure about Rhino especially after your irrational response. Some spokesman, I will stick with Adtran and Adit, not some cheap knock-off. Thanks, Steve Totaro On Fri, Feb 15, 2008 at 9:36 AM, James Finstrom [EMAIL PROTECTED] wrote: Steve, Yes I work for Rhino that is no Secret. If you read the post I was responding to the thread not pimping my own products. I am not sure if your a Sangoma fanboy or employee since you are apparently offended by my response, however he wasn't asking to be sold to he was asking about specific products. So there it is yes I work for Rhino and I could have easily given one of our italian distributors but he didn't ask for that. It is not appropriate to troll the list and push your products unsolicited. If someone is looking for a recommendation for a card brand fine. If they need a solution like ADID or they need to accommodate funky CPC signals from their telco which Rhino does fine it is on subject. If someone asks should I use openvox to replace my digium you don't pimp your product because it wasn't asked for. If you want my honest opinion. I prefer people use Rhino products. I believe our products and support are superior but if you don't use our cards use Digium. If your reply is any indication on how Sandoma works I can honestly say go use a cheap clone before sangomaN they may not support you but at least they are open about being here just for the money. James Finstrom Rhino Equipment Corp. http://www.rhinoequipment.com -Original Message- From: Steve Totaro [EMAIL PROTECTED] Date: Fri, 15 Feb 2008 08:45:50 To:Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [asterisk-users] Digium stopped TDM400P production: alternatives?? James, If you were replying to the original post about Openvox or specified that is what you were referring to, maybe I would not take issue but to reply to a suggesting to use Sangoma with what you did is absolutely misleading. There is nothing cheap or clone about Sangoma's cards. asterisk.rhinoequipment.com hm. Thanks, Steve Totaro James Finstrom wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I would say email Kevin what he asked. The problem with switching to a clone company is you get what you pay for. Sticking with Digium you at least have support. and 3 clone cards and hours of troubleshooting later you will wish you hadn't been all cheap. Rob Hillis wrote: The cards themselves are okay, but the extra level of configuration is a pain in the proverbial. Zaptel is already double-configured in both zaptel.conf and zapata.conf (that's not a complaint - I understand the reason for the separation) but the Sangoma cards require a /third/ level of configuration in Wanpipe. Steve Totaro wrote: Sangoma makes a good card. On 2/15/08, Kevin P. Fleming [EMAIL PROTECTED] wrote: Giorgio Incantalupo wrote: Digium stopped to produce TDM400P and the new TDM410 is too new to find it in our shops. The only alternative available is a fully-compatible Openvox product...but is it really fully-compatible? Any experience about Openvox products (card and zaptel versions, etc...)? Every distributor that carried the TDM400P should have TDM410s in stock already. Where are you located, and who do you buy Digium cards from? -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- 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 !DSPAM:47b59f18311805637012918! ___ -- 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] multiple host in 1 context on sip.conf
Johansson Olle E wrote: Hi Mark! 13 feb 2008 kl. 23.42 skrev Mark Quitoriano: Is it possilble for a single context to have multiple host= something like this First context is something we use to describe a segment of the dialplan. I would call this section. [carrier] host=ip address1 host=ip address2 host=ip address3 type=peer disallow=all allow=g729 allow=ulaw canreinvite=no insecure=yes qualify=yes No. You can only add one. Yes You can, check this ticket http://bugs.digium.com/view.php?id=12005 --- * Olle E. Johansson - [EMAIL PROTECTED] * Asterisk Training http://edvina.net/training/ ___ -- 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] Variable setting in AMI Originate
Anthony Messina wrote:
Working with asterisk 1.4; using the AMI Originate command, it is possible to
do something like:
Variable: CDR(accountcode)123456
Or must the variable names be var[n] where n is a number?
I'd like to set the accountcode for a Local channel that originates a call.
Thanks. -A
Anthony,
I may not understand your question, but setting variables from the AMI is easy
enough:
Action: Originate
Channel: local/[EMAIL PROTECTED]
Context: to_meetme
Exten: s
Priority: 1
Variable: CALLERID(num)=${DEV_NAME}|CALLERID(name)=Conference Waiting
Async: true
--
Warm Regards,
Lee
Everything I needed to learn in life, I learned selling encyclopedias door to
door.
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Re: [asterisk-users] Digium stopped TDM400Pproduction: alternatives??
Steve Totaro wrote: If you were replying to the original post about Openvox or specified that is what you were referring to, maybe I would not take issue but to reply to a suggesting to use Sangoma with what you did is absolutely misleading. There is nothing cheap or clone about Sangoma's cards. Steve, The way I read it, James is suggesting that the original poster would be better served to use the Digium card. While James is obviously related to Rhino cards, since he is suggesting that it would be better to use the Digium card, I find no offense in his post. Had he suggested Rhino cards that would have been a different story. I also agree that there is nothing cheap about Sangoma cards. However even as you mention, the original poster specifically asked about Openvox, so your suggestion of Sangoma cards is as out of place as your claiming James to be. At least James kept his suggestions to the cards of question. Is your issue with his remarks somehow related to your consultation business? John Faubion ___ -- 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] Automatically start after restart
On Fri, Feb 15, 2008 at 09:05:43AM -0700, Anthony Francis wrote: I actually use daemon tools http://cr.yp.to/daemontools/daemontools-0.76.tar.gz I like it because its log handling features, it takes the stdout of asterisk and puts it in a log directory and automatically rotates the files. Asterisk is a daemon. Why do you need to look at its output? Check logger.conf, as well as your distro's logrotate. Or use rsyslog that already knows how to rotate logs (mind you: asterisk as well). Fedora have switched to it, and it seems that Debian will also do. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/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] is encrypted iax safe and secure?
Tim Panton wrote: The NEW frame doesn't _have_ to contain a dialed number, the digits can be sent later (I forget the frametype), but later means within the encrypted session :-) It's the DIAL command that you are thinking of. I'm considering implementing this, but it has one major caveat: to really do the job right, we wouldn't want any caller information (CLID or CNAM) to be in the NEW message either, it would have to be added as IEs to the DIAL command. Unfortunately no existing implementations are going to be prepared to receive that information as part of DIAL, so they would process this sort of call with an empty CLID and CNAM. We can of course enhance chan_iax2 to understand this method of doing things, but it won't be backward compatible with previous versions of Asterisk or any other IAX2 clients. -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- 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] DialPlan help with Analog Fax Machine
Jim Duda wrote: == Spawn extension (incoming-dial, fax, 0) exited non-zero on 'Zap/4-1' Yes, I DO think that's a little odd. It should be priority 1, shouldn't 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
[asterisk-users] Zaptel compilation problems.
When I try to make zaptel 1.4.8, I receive the following error: scripts/Makefile.build:46: *** CFLAGS was changed in /usr/src/zaptel-1.4.8/Makefile. Fix it to use EXTRA_CFLAGS. Stop. This is on a debian 4.0 machine running linux kernel 2.6.24.2. (gcc 4.1.2). TIA for any help in resolving this. ___ -- 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] multiple host in 1 context on sip.conf
Hi Olle, On Thu, Feb 14, 2008 at 5:35 PM, Johansson Olle E [EMAIL PROTECTED] wrote: Hi Mark! 13 feb 2008 kl. 23.42 skrev Mark Quitoriano: Is it possilble for a single context to have multiple host= something like this First context is something we use to describe a segment of the dialplan. I would call this section. [carrier] host=ip address1 host=ip address2 host=ip address3 type=peer disallow=all allow=g729 allow=ulaw canreinvite=no insecure=yes qualify=yes No. You can only add one. Normally I would add host=sip.mydomain.com and have multiple DNS entries or use SRV records to do failover and such, provided you use this for outbound calls. Since you call this peer carrier I assume you want to handle inbound calls. Today, you will have to define three different peers, but remember that you can use templates. [carrier](!) type=peer disallow=all allow=g729 allow=ulaw canreinvite=no insecure=yes qualify=yes [carrier-01](carrier) host=ip address1 [carrier-02](carrier) host=ip address2 [carrier-03](carrier) host=ip address3 You will now have three peers named carrier-01-03 but no peer named carrier in your sip driver when you run sip show peers. This looks interesting. Is there more documentation how to do this? And btw in dialplan can i call this just simply SIP/carrier and all the 3 ip will be used? 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] Variable setting in AMI Originate
Anthony Messina wrote:
On Friday 15 February 2008 10:21:33 am Lee Jenkins wrote:
Anthony Messina wrote:
Working with asterisk 1.4; using the AMI Originate command, it is
possible to do something like:
Variable: CDR(accountcode)123456
Or must the variable names be var[n] where n is a number?
I'd like to set the accountcode for a Local channel that originates a
call.
Thanks. -A
Anthony,
I may not understand your question, but setting variables from the AMI is
easy enough:
Action: Originate
Channel: local/[EMAIL PROTECTED]
Context: to_meetme
Exten: s
Priority: 1
Variable: CALLERID(num)=${DEV_NAME}|CALLERID(name)=Conference Waiting
Async: true
That was exactly my question (even though I forgot the =sign). However, I
am
not able to get that to work for reason. I'm trying to set the
CDR(accountcode) on the first leg of the call and am using Channel: Local/...
I am able to get it to work if I use Variable: var1=12345 then, use
CDR(accountcode)=${var1} in the dialplan, but I was hoping to avoid this
hack.
Not sure what could be the reason, maybe something in the cdr stuff and call
origination maybe?
--
Warm Regards,
Lee
Everything I needed to learn in life, I learned selling encyclopedias door to
door.
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Re: [asterisk-users] Digium stopped TDM400P production: alternatives??
There are some tdm400 cards on ebay, http://search.ebay.com/tdm400 Moj Giorgio Incantalupo wrote: Hi, Digium stopped to produce TDM400P and the new TDM410 is too new to find it in our shops. The only alternative available is a fully-compatible Openvox product...but is it really fully-compatible? Any experience about Openvox products (card and zaptel versions, etc...)? Thank you! Giorgio. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Variable setting in AMI Originate
Anthony Messina wrote:
On Friday 15 February 2008 10:21:33 am Lee Jenkins wrote:
*snipped
Priority: 1
Variable: CALLERID(num)=${DEV_NAME}|CALLERID(name)=Conference Waiting
Async: true
That was exactly my question (even though I forgot the =sign). However, I
am
not able to get that to work for reason. I'm trying to set the
CDR(accountcode) on the first leg of the call and am using Channel: Local/...
I am able to get it to work if I use Variable: var1=12345 then, use
CDR(accountcode)=${var1} in the dialplan, but I was hoping to avoid this
hack.
why not just add
Account: 12345
to the originate?
(side note: you can also have multiple Variable: lines (some versions of
asterisk have issue with the | from what i hear)
so the above would look like
...
Variable: CALLERID(num)=${DEV_NAME}
Variable: CALLERID(name)=Conference Waiting
those are bad examples as you should just use CallerID:
Callerid: Conference Waiting DEVNUMBER
i hope this helps.
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Re: [asterisk-users] Variable setting in AMI Originate
On Friday 15 February 2008 10:21:33 am Lee Jenkins wrote:
Anthony Messina wrote:
Working with asterisk 1.4; using the AMI Originate command, it is
possible to do something like:
Variable: CDR(accountcode)123456
Or must the variable names be var[n] where n is a number?
I'd like to set the accountcode for a Local channel that originates a
call.
Thanks. Â -A
Anthony,
I may not understand your question, but setting variables from the AMI is
easy enough:
Action: Originate
Channel: local/[EMAIL PROTECTED]
Context: to_meetme
Exten: s
Priority: 1
Variable: CALLERID(num)=${DEV_NAME}|CALLERID(name)=Conference Waiting
Async: true
That was exactly my question (even though I forgot the =sign). However, I am
not able to get that to work for reason. I'm trying to set the
CDR(accountcode) on the first leg of the call and am using Channel: Local/...
I am able to get it to work if I use Variable: var1=12345 then, use
CDR(accountcode)=${var1} in the dialplan, but I was hoping to avoid this
hack.
--
Anthony - http://messinet.com - http://messinet.com/~amessina/gallery
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[asterisk-users] Connecting a Rolm CBX to Asterisk via T1?
Hi all,
So I'm trying to work on this complex fax server setup, and part of it involves
connecting my asterisk server to my Rolm CBX switch, via a T1 line. I plan on
using Asterisk simply as a T1-PRI Bridge to IAXmodem (which in turn, activates
HylaFax+ to handle the faxing). So far, though, I don't think I'm getting 100%
of the way there. When dialing the fax extension from my Rolm phone, I get
several seconds of silence followed by error tone. But on asterisk's CLI, I
see this:
-- Starting simple switch on 'Zap/2-1'
-- Starting simple switch on 'Zap/3-1'
-- Starting simple switch on 'Zap/4-1'
-- Starting simple switch on 'Zap/1-1'
-- Executing [EMAIL PROTECTED]:1] Dial(Zap/2-1, IAX2/iaxmodem0/s|10|r)
in new stack
-- Called iaxmodem0/s
-- Call accepted by 127.0.0.1 (format ulaw)
-- Format for call is ulaw
-- IAX2/iaxmodem0-5 is ringing
-- IAX2/iaxmodem0-5 answered Zap/2-1
-- Executing [EMAIL PROTECTED]:1] Dial(Zap/3-1, IAX2/iaxmodem0/s|10|r)
in new stack
-- Called iaxmodem0/s
[Feb 15 15:40:22] WARNING[24329]: chan_iax2.c:7542 socket_process: Call
rejected by 127.0.0.1: Busy
-- Hungup 'IAX2/iaxmodem0-1'
== Everyone is busy/congested at this time (1:0/0/1)
== Auto fallthrough, channel 'Zap/3-1' status is 'CHANUNAVAIL'
-- Executing [EMAIL PROTECTED]:1] Dial(Zap/4-1, IAX2/iaxmodem0/s|10|r)
in new stack
-- Called iaxmodem0/s
[Feb 15 15:40:30] WARNING[24327]: chan_iax2.c:7542 socket_process: Call
rejected by 127.0.0.1: Busy
-- Hungup 'IAX2/iaxmodem0-3'
== Everyone is busy/congested at this time (1:0/0/1)
== Auto fallthrough, channel 'Zap/4-1' status is 'CHANUNAVAIL'
-- Hungup 'Zap/3-1'
-- Executing [EMAIL PROTECTED]:1] Dial(Zap/1-1, IAX2/iaxmodem0/s|10|r)
in new stack
-- Called iaxmodem0/s
[Feb 15 15:40:35] WARNING[24327]: chan_iax2.c:7542 socket_process: Call
rejected by 127.0.0.1: Busy
-- Hungup 'IAX2/iaxmodem0-4'
== Everyone is busy/congested at this time (1:0/0/1)
== Auto fallthrough, channel 'Zap/1-1' status is 'CHANUNAVAIL'
-- Hungup 'Zap/4-1'
The Rolm gives me error tone just before the Starting simple switch messages
begin to appear, so it's almost like the Rolm is not waiting around long enough
for the asterisk server to answer, before it jumps to the next configured T1
channel, runs out of channels (I only configured four in the Rolm and on
asterisk).
Here's my configuration for asterisk. Is anything amiss by chance?
Standard T1
Signalling is EM Wink, 200ms wink time (as far as I can tell)
Mode is ESF and format is B8ZS
/etc/zaptel.conf is:
span=1,1,0,esf,b8zs
em=1-4
loadzone = us
defaultzone=us
/etc/asterisk/zapata.conf is:
[trunkgroups]
[channels]
language=en
context=default
switchtype=national
signalling=em_w
wink=200
channel = 1-4
usecallerid=yes
callerid=asreceived
cidsignalling=bell
hidecallerid=no
callwaiting=no
usecallingpres=yes
callwaitingcallerid=no
threewaycalling=no
transfer=no
canpark=no
cancallforward=no
callreturn=no
echocancel=yes
echocancelwhenbridged=no
echotraining=800
rxgain=0.0
txgain=0.0
group=0
busydetect=yes
busycount=6
faxdetect=incoming
/etc/asterisk/extensions.conf is:
[general]
static=yes
writeprotect=no
clearglobalvars=no
[globals]
CONSOLE=Console/dsp ; Console interface for demo
TRUNK=Zap/g0; Trunk interface
TRUNKMSD=1 ; MSD digits to strip (usually
1 or 0)
[fax-in]
exten = s,1,Dial(IAX2/iaxmodem0/${EXTEN},10,r)
Thoughts?
Thanks!,
--Josh
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Re: [asterisk-users] Connecting a Rolm CBX to Asterisk via T1?
Joshua Kinard wrote: So I'm trying to work on this complex fax server setup, and part of it involves connecting my asterisk server to my Rolm CBX switch, via a T1 line. I plan on using Asterisk simply as a T1-PRI Bridge to IAXmodem (which in turn, activates HylaFax+ to handle the faxing). So far, though, I don't think I'm getting 100% of the way there. When dialing the fax extension from my Rolm phone, I get several seconds of silence followed by error tone. But on asterisk's CLI, I see this: -- Starting simple switch on 'Zap/2-1' -- Starting simple switch on 'Zap/3-1' -- Starting simple switch on 'Zap/4-1' -- Starting simple switch on 'Zap/1-1' So, okay, there are four calls coming in on the Zap (strange, but...) -- Executing [EMAIL PROTECTED]:1] Dial(Zap/2-1, IAX2/iaxmodem0/s|10|r) in new stack -- Called iaxmodem0/s -- Call accepted by 127.0.0.1 (format ulaw) -- Format for call is ulaw -- IAX2/iaxmodem0-5 is ringing -- IAX2/iaxmodem0-5 answered Zap/2-1 iaxmodem0 correctly takes the first call... -- Executing [EMAIL PROTECTED]:1] Dial(Zap/3-1, IAX2/iaxmodem0/s|10|r) in new stack -- Called iaxmodem0/s [Feb 15 15:40:22] WARNING[24329]: chan_iax2.c:7542 socket_process: Call rejected by 127.0.0.1: Busy -- Hungup 'IAX2/iaxmodem0-1' == Everyone is busy/congested at this time (1:0/0/1) == Auto fallthrough, channel 'Zap/3-1' status is 'CHANUNAVAIL' -- Executing [EMAIL PROTECTED]:1] Dial(Zap/4-1, IAX2/iaxmodem0/s|10|r) in new stack -- Called iaxmodem0/s [Feb 15 15:40:30] WARNING[24327]: chan_iax2.c:7542 socket_process: Call rejected by 127.0.0.1: Busy -- Hungup 'IAX2/iaxmodem0-3' == Everyone is busy/congested at this time (1:0/0/1) == Auto fallthrough, channel 'Zap/4-1' status is 'CHANUNAVAIL' -- Hungup 'Zap/3-1' -- Executing [EMAIL PROTECTED]:1] Dial(Zap/1-1, IAX2/iaxmodem0/s|10|r) in new stack -- Called iaxmodem0/s [Feb 15 15:40:35] WARNING[24327]: chan_iax2.c:7542 socket_process: Call rejected by 127.0.0.1: Busy -- Hungup 'IAX2/iaxmodem0-4' == Everyone is busy/congested at this time (1:0/0/1) == Auto fallthrough, channel 'Zap/1-1' status is 'CHANUNAVAIL' -- Hungup 'Zap/4-1' And the other calls get busy and improperly run through the auto fallthrough process (you *need* a Hangup in your dialplan fax-in context). The Rolm gives me error tone just before the Starting simple switch messages begin to appear, so it's almost like the Rolm is not waiting around long enough for the asterisk server to answer, before it jumps to the next configured T1 channel, runs out of channels (I only configured four in the Rolm and on asterisk). I think that your zaptel/zapata configuration between the Rolm and Asterisk on that T1 is misconfigured. Set it up for PRI if you can... it'll be a lot easier, is my guess. Thanks, Lee. ___ -- 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] multiple host in 1 context on sip.conf
On Sat, Feb 16, 2008 at 12:31 AM, Faruk Kasumovic [EMAIL PROTECTED] wrote: Johansson Olle E wrote: Hi Mark! 13 feb 2008 kl. 23.42 skrev Mark Quitoriano: Is it possilble for a single context to have multiple host= something like this First context is something we use to describe a segment of the dialplan. I would call this section. [carrier] host=ip address1 host=ip address2 host=ip address3 type=peer disallow=all allow=g729 allow=ulaw canreinvite=no insecure=yes qualify=yes No. You can only add one. Yes You can, check this ticket http://bugs.digium.com/view.php?id=12005 this is good if the ip addresses are on the same network. But what if it has totally different networks? ___ -- 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] ISDN PRIs and taking a server down formaintenance - blocking issue
You got me interested in this topic so I started doing some research. There is a discussion on the asterisk-dev list about adding true busy support to the Zaptel module. As it currently stands, when a call comes in on a PRI channel while asterisk is shutting down asterisk sends a signal back effectively rejecting the call, but the Telco sees it as Asterisk answering the call. What needs to happen is a mechanism needs to be implemented that will place the the channel in the off-hook state after the active call hangs up until the PRI can be truly taken down. I'm not a coder so I have no idea how to begin implementing that, but I suspect it would not bee too difficult for a coder to go in and take a couple of the pieces of code that answer or dial on channel and make an hook state function. -Brent Andrew Smith wrote: Yes the 'stop gracefully' is what effectively blocks the calls as the telco seems to take it as we are answering the calls instead of seeing them as busy. I will look at implementing some sort of way of busying out all the zaptel channels, so that we eventually busy out all 120 channels (4x E1) and then can cleanly take the server offline while our telco presents the calls to the next Asterisk servers correctly. This would be a great way of busying out the server for maintenance while still allowing our inbound calls. Many thanks, Andrew *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Brent Davidson *Sent:* 15 February 2008 00:30 *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] ISDN PRIs and taking a server down formaintenance - blocking issue Correct me if I'm wrong, but as I understand it your issue is that when you give Asterisk the stop gracefully command it waits until all active calls have finished before it takes the ISDN down but gives busy signals to new incoming calls on idle channels. If this is the case then it would seem that Asterisk is actually answering the call on the incoming channel and playing a busy signal. From reading a couple of threads on another list it appears this is the case (Google: Asterisk busy out PRI to find the discussion). There also appears to be some interest in making a function do what you need in the future. For the time being, however, a simple solution would be to create a temporary dial-plan that follows each outgoing hangup with a dial command to either a test number or some other service that will just keep playing audio down the line and not hangup. (You'd probably need to set some variable to know which channels had been busied) When you need to take down a server, load this dial plan and wait for all channels to call the busy number, then hang them all up and issue a stop now. It's a messy solution, but it's all I can think of without hacking code. The only other way I'd know would be to hack the code for the dial or answer command and build another command that simply takes the channel off-hook and leaves it there. Good luck, Brent Davidson Lyle Giese wrote: If you take Asterisk down, the PRI should go down as the D channel is down. Then the telco should KNOW that there is trouble with the PRI and those channels are in trouble busy and not availible. If the telco still tries to push a call to a channel on a PRI that is down, then the telco is at fault. Lyle Matt wrote: That does sound like what is happening.. Telco knows channel 1-23 are not busy (so far as they are concerned), however.. so far as you are concerned, they are busy.. so telco sends the call down... but the equipment doesn't take it. I would *think* the Telco could keep trying channels down the hunt group, but maybe not? We have, in the past, seen this issue with our dial-up modem banks.. especially if I would take one offline. However, it is not a big enough issue (i.e. we don't take things down that often) for me to look into it fully. On Thu, Feb 14, 2008 at 4:07 PM, Don Kelly [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I think the problem is that the telco presents the call on a specific channel, then zaptel tells it that the channel is busy. We need to be able to tell the telco that each unused channel on a given span is unavailable, and it will determine that the others are in use and will present the call on a channel on another span. A rather ugly work-around (since Andrew seems to have lots of channels available, and one would assume that maintenance of this nature would occur during slow periods) would be to make calls to a DID in the same trunk group on all idle channels on the span shutting down then, when all channels on the span are in use and none of them are doing anything useful, take the span down hard so the telco will divert all calls to another span.
Re: [asterisk-users] Variable setting in AMI Originate
On Friday 15 February 2008 01:49:46 pm Richard Lyman wrote:
Anthony Messina wrote:
On Friday 15 February 2008 10:21:33 am Lee Jenkins wrote:
*snipped
Priority: 1
Variable: CALLERID(num)=${DEV_NAME}|CALLERID(name)=Conference Waiting
Async: true
That was exactly my question (even though I forgot the =sign). However,
I am not able to get that to work for reason. I'm trying to set the
CDR(accountcode) on the first leg of the call and am using Channel:
Local/...
I am able to get it to work if I use Variable: var1=12345 then, use
CDR(accountcode)=${var1} in the dialplan, but I was hoping to avoid this
hack.
why not just add
Account: 12345
to the originate?
(side note: you can also have multiple Variable: lines (some versions of
asterisk have issue with the | from what i hear)
so the above would look like
...
Variable: CALLERID(num)=${DEV_NAME}
Variable: CALLERID(name)=Conference Waiting
those are bad examples as you should just use CallerID:
Callerid: Conference Waiting DEVNUMBER
i hope this helps.
that does work like a charm--it sets the accountcode, except that, for some
reason, i can't access the CDR(accountcode) value during call time.
i CAN see it in channel variables, etc. but ${CDR(accountcode)} evaluates to
nothing--it's blank. it even show up in the CDR after the call is over.
my dialplan basically says, set the callerid to the accountcode (which is my
real pstn number). since i have some users for which i need to block
outbound callerid on the pstn line, this was a convenient way to distinguish
between my devices with my accountcode and those without.
now that i'm trying to originate calls from a secured webpage using the
manager, it seems like my method isn't working well :(
i'd like to keep the callerid=Name internal exten settings in my devices
so internally, we see the extensions instead of a full pstn number.
how else would i be able to set the outbound/external callerid per device?
--
Anthony - http://messinet.com - http://messinet.com/~amessina/gallery
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Re: [asterisk-users] Digium stopped TDM400Pproduction: alternatives??
What about the word alternatives do you not understand. Read the title of the thread again. Thanks, Steve Totaro On Fri, Feb 15, 2008 at 11:13 AM, John Faubion [EMAIL PROTECTED] wrote: Steve Totaro wrote: If you were replying to the original post about Openvox or specified that is what you were referring to, maybe I would not take issue but to reply to a suggesting to use Sangoma with what you did is absolutely misleading. There is nothing cheap or clone about Sangoma's cards. Steve, The way I read it, James is suggesting that the original poster would be better served to use the Digium card. While James is obviously related to Rhino cards, since he is suggesting that it would be better to use the Digium card, I find no offense in his post. Had he suggested Rhino cards that would have been a different story. I also agree that there is nothing cheap about Sangoma cards. However even as you mention, the original poster specifically asked about Openvox, so your suggestion of Sangoma cards is as out of place as your claiming James to be. At least James kept his suggestions to the cards of question. Is your issue with his remarks somehow related to your consultation business? John Faubion ___ -- 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] Variable setting in AMI Originate
On Fri, 2008-02-15 at 14:45 -0600, Anthony Messina wrote:
that does work like a charm--it sets the accountcode, except that, for some
reason, i can't access the CDR(accountcode) value during call time.
i CAN see it in channel variables, etc. but ${CDR(accountcode)} evaluates to
nothing--it's blank. it even show up in the CDR after the call is over.
This definitely sounds like a bug to me -- would you mind creating a bug
report on the bug tracker (http://bugs.digium.com) so that the
developers can take a look at it? Thanks!
--
Jared Smith
Community Relations Manager
Digium, Inc.
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Re: [asterisk-users] Connecting a Rolm CBX to Asterisk via T1?
-Original Message-
From: Lee Howard
So, okay, there are four calls coming in on the Zap (strange, but...)
There's definitely some kind of a timing error here. I cut my channels back
down to 1, as the Rolm isn't waiting long enough for an answer back from the
asterisk server, and it gives up too early with a busy tone now. What I'm
seeing is the asterisk server taking too long to respond in kind, only to find
the Rolm's quit and gone home already.
Also, asterisk seems to have signalling=em and signalling=em_w mixed up, as I
have to use signalling=em to see a wink sent back down to my Rolm. em_w does
nothing.
An attached text file (rolm-asterisk-chatter.txt) is what my Rolm is seeing.
Notes on each line are on the right and are my additions.
Another attached text file shows what iaxmodem is doing during all of this.
Something about adjusting skew.
Here's what Asterisk itself sees (appears long after the Rolm went to busy
tone):
-- Starting simple switch on 'Zap/1-1'
-- Executing [EMAIL PROTECTED]:1] Dial(Zap/1-1, IAX2/iaxmodem0/s) in
new stack
-- Called iaxmodem0/s
-- Call accepted by 127.0.0.1 (format ulaw)
-- Format for call is ulaw
-- IAX2/iaxmodem0-3 is ringing
-- IAX2/iaxmodem0-3 answered Zap/1-1
-- Hungup 'IAX2/iaxmodem0-3'
== Spawn extension (fax-in, s, 1) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'
And the other calls get busy and improperly run through the auto
fallthrough process (you *need* a Hangup in your dialplan fax-in context).
Added, how does this look?
exten = s,1,Dial(IAX2/iaxmodem0/${EXTEN})
exten = s,2,Busy
exten = s,3,Hangup
I think that your zaptel/zapata configuration between the Rolm and
Asterisk on that T1 is misconfigured. Set it up for PRI if you can...
it'll be a lot easier, is my guess.
Unfortunately, the Rolm only speaks plain T1 talk. It's too old for PRI. We
have an Adtran Atlas unit infront of it that does the PRI-T1 translation that
we get from our carrier, but to get another card for the Adtran is more than
I'll be able to weasel out of my manager for now.
Cheers!,
--Josh
TRK#STATE INL/XDI CODE DIGITS PROCESS TEM SZ
--- -- --- --- --- --
1 IDLEOU
Idle
1 OUTPULS S01/011501 80 OU
fax ext. dialed (80=trk group)
1 OUTPULS S01/011501
1 IDLE
1 IDLE RESZ DELAY
Rolm quits here; busy tone
1 SI RSVD
1 RING-IN
Asterisk rings back
1 DIAL TO R01/011103
1 DIAL TO R01/011103W
Wink sent to Rolm
1 BUSY
Busy because no one answered
1 BUSY
Asterisk hangs up
1 IDLE
1 IDLE RESZ DELAY
1 IDLE
[2008-02-15 17:11:11] Incoming call connected s, , .
[2008-02-15 17:11:12] Answering
[2008-02-15 17:11:12] Adjusting skew to -50.
[2008-02-15 17:11:12] Adjusting skew to -100.
[2008-02-15 17:11:12] Adjusting skew to -150.
[2008-02-15 17:11:12] Adjusting skew to -200.
[2008-02-15 17:11:12] Adjusting skew to -250.
[2008-02-15 17:11:12] Adjusting skew to -300.
[2008-02-15 17:11:12] Adjusting skew to -350.
[2008-02-15 17:11:13] Adjusting skew to -400.
[2008-02-15 17:11:13] Adjusting skew to -450.
[2008-02-15 17:11:13] Adjusting skew to -500.
[2008-02-15 17:11:13] Adjusting skew to -550.
[2008-02-15 17:11:13] Adjusting skew to -600.
[2008-02-15 17:11:13] Adjusting skew to -650.
[2008-02-15 17:11:13] Adjusting skew to -700.
[2008-02-15 17:11:13] Adjusting skew to -750.
[2008-02-15 17:11:14] Adjusting skew to -800.
[2008-02-15 17:11:14] Adjusting skew to -850.
[2008-02-15 17:11:14] Adjusting skew to -900.
[2008-02-15 17:11:14] Adjusting skew to -950.
[2008-02-15 17:11:14] Adjusting skew to -1000.
[2008-02-15 17:11:14] Adjusting skew to -1050.
[2008-02-15 17:11:14] Adjusting skew to -1100.
[2008-02-15 17:11:14] Adjusting skew to -1150.
[2008-02-15 17:11:15] Adjusting skew to -1200.
[2008-02-15 17:11:15] Adjusting skew to -1250.
[2008-02-15 17:11:15] Adjusting skew to -1300.
[2008-02-15 17:11:15] Adjusting skew to -1350.
[2008-02-15 17:11:15] Adjusting skew to -1400.
[2008-02-15 17:11:15] Adjusting skew to -1450.
[2008-02-15 17:11:15] Adjusting skew to -1500.
[2008-02-15 17:11:15] Adjusting skew to -1550.
[2008-02-15 17:11:15] Adjusting skew to -1600.
[2008-02-15 17:11:16] Adjusting skew to -1650.
[2008-02-15 17:11:16] Adjusting skew to -1700.
[2008-02-15 17:11:16] Adjusting skew to -1750.
[2008-02-15 17:11:16] Adjusting skew to -1800.
[2008-02-15 17:11:16]
Re: [asterisk-users] Touch monitor file name format
Will Set(MONITOR_FILENAME=/blahblah/filename) work for you?
Moj
Jaap Winius wrote:
Hi list,
The default file name format for touch monitor (automon) recordings is:
auto-${EPOCH}-caller-calee
It's possible to use the ${TOUCH_MONITOR} variable to change the
'caller-calee' part, but what about the 'auto-${EPOCH}-' part?
I've been trying to use ${MONITOR_EXEC_ARGS} to add some more commands
after the somix sequence for mp3 conversion. This should work, but
I've so far failed to produce any mp3 files because I'm not able to
predict the above epoch number. If I could alter 'auto-${EPOCH}-', or
if it was stored in a variable I could use, then I'm sure my plan will
succeed.
Any ideas?
Thanks,
Jaap
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Re: [asterisk-users] is encrypted iax safe and secure?
Of course *it would be nice if* the IAX2 authentication parameters were also encrypted, so that there was no danger of a 3rd party hijacking your connection and generating a bunch of extra charges. S. On Fri, Feb 15, 2008 at 11:31 AM, Kevin P. Fleming [EMAIL PROTECTED] wrote: Tim Panton wrote: The NEW frame doesn't _have_ to contain a dialed number, the digits can be sent later (I forget the frametype), but later means within the encrypted session :-) It's the DIAL command that you are thinking of. I'm considering implementing this, but it has one major caveat: to really do the job right, we wouldn't want any caller information (CLID or CNAM) to be in the NEW message either, it would have to be added as IEs to the DIAL command. Unfortunately no existing implementations are going to be prepared to receive that information as part of DIAL, so they would process this sort of call with an empty CLID and CNAM. We can of course enhance chan_iax2 to understand this method of doing things, but it won't be backward compatible with previous versions of Asterisk or any other IAX2 clients. -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- 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] Monitor Asterisk
On Fri, Feb 15, 2008 at 08:55:11AM +0100, Johansson Olle E wrote: I would also like to see manager wrappers that produce data that is easy to handle for scripts, like a wrapper that produces show channels consise in various formats. Do we have a perl programmer on the list? Such a generic script could be added to the scripts library in the Asterisk distribution. The problem is that such a script needs to authenticate to the Asterisk manager. I can imagine a failed cron job yelling: But I'm root? why the f__ do I need to show an ID card to some stupid server?!?! Well, if you're capable of writing to the Asterisk socket you're also capable of reading /etc/asterisk/manager.conf , right? So here's an idea to work around that problem: At install time generate a random secret, e.g: # generates string with 128 random bits. Looks like an md5sum head -c 16 /dev/urandom | hexdump -e ' %4x ' This will server as a secret of an administrative account. It may be #include-d from another file. And anybody capable of reading that file is practically authorised to connect to the manager. This means that the manager connecting script has to assume something about the config files of Asterisk, which is probably a bad thing. Any better ideas? A similar example from a different program: mysql in Debian: pungenday:~# ls -l /etc/mysql/debian.cnf -rw--- 1 root root 312 Oct 26 13:22 /etc/mysql/debian.cnf pungenday:~# cat /etc/mysql/debian.cnf # Automatically generated for Debian scripts. DO NOT TOUCH! [client] host = localhost user = debian-sys-maint password = XX socket = /var/run/mysqld/mysqld.sock So now administrative scripts can use: MYSQLADMIN=mysqladmin --defaults-file=/etc/mysql/debian.cnf MYSQL=mysql --defaults-file=/etc/mysql/debian.cnf # and then something like: $MYSQL mysql -e 'select host,user from user' As a normal user: Could not open required defaults file: /etc/mysql/debian.cnf Fatal error in defaults handling. Program aborted As root: +---+--+ | host | user | +---+--+ | localhost | debian-sys-maint | | localhost | root | | pungenday | root | +---+--+ Note that in that example the password and the client configuration may not be in sync. And in fact if you read that file and the postinst script you'll see the exatra complexity that this has caused. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connecting a Rolm CBX to Asterisk via T1?
Joshua Kinard wrote:
Another attached text file shows what iaxmodem is doing during all of this.
Something about adjusting skew.
[2008-02-15 17:11:12] Adjusting skew to -50.
[2008-02-15 17:11:12] Adjusting skew to -100.
[2008-02-15 17:11:12] Adjusting skew to -150.
[2008-02-15 17:11:12] Adjusting skew to -200.
[2008-02-15 17:11:12] Adjusting skew to -250.
There is no mechanism for iaxmodem to pull clocking right from Asterisk
other than examining the IAX2 timestamps. So in the event that iaxmodem
isn't getting voice frames from Asterisk iaxmodem is left to use
clocking solely from the system clock... which may likely not be in-sync
with the T1 clocking... and so iaxmodem the skew messages you see is
an attempt by iaxmodem to compensate for a clock skew between the system
clock and the timestamps on the IAX2 frames... but because iaxmodem
isn't getting any voice frames you get a run of these skew messages
until the call disconnects.
Added, how does this look?
exten = s,1,Dial(IAX2/iaxmodem0/${EXTEN})
exten = s,2,Busy
exten = s,3,Hangup
Better. :-)
Thanks,
Lee.
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Re: [asterisk-users] Monitor Asterisk
Johansson Olle E wrote: In the long run we're trying to move to using the manager for all parsing by adding a lot of new manager events and actions. If there's something missing that you only can do or information you only can get in the CLI, please tell us. Olle, How does what you are describing compare to the action command http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+Command? Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer ___ -- 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] restart asterisk daily
I have multiple queues in my case. Do you mean multiple queues is one of the reason to consume memory? How to only reset the queue stats? You will see asterisk behave its worst with multiple queues and heavy dialplan logic. I restart my boxes with queues everynight at midnight just to reset the queue stats displayed with show queue. ___ -- 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] arris tm502g cablemodem FXS ports and zaptel 1.4.8
Hi there, I have a cablemodem, ARRIS brand, model tm502G. It has two FXS ports. I was wondering if anyone has details about the correct signalling of these FXS ports when connected to original X100p. Tests: fxsks on the zapata.conf and zaptel.conf files. From my cellphone I call the ARRIS, it starts ringing but the zap channel sees no call coming in. fxsls on the zapata.conf and zaptel.conf files. From my cellphone I call the ARRIS, it starts ringing, zap channel picks up the call. all good. fxsgs on the zapata.conf and zaptel.conf files. ztcfg reports error about invalid mode. Well, I used loopstart as the signal, however when using it I face one very nasty issue. My asterisk/zap channel does not detect hangups correctly. I have enabled busydetect but it's kind of unreliable. Specially when using DISA, if one of my external callers use DISA and the external caller hangsup, asteirsk wont see athing and will keep both zap channels open. I will like some suggestions with this as i am not sure if it's related to signalling in the ARRIS or maybe some tweaking i can do in the x100p (true x100p). Thanks, -- Erick Perez ___ -- 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 stopped TDM400P production: alternatives?? ?In-Reply-To: [EMAIL PROTECTED] ?References: [EMAIL PROTECTED] [EMAIL PROTECTED] ? [EMAIL PROTECTED] ? [EMAIL PROTECTE
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Re: [asterisk-users] Touch monitor file name format
Quoting Mojo with Horan Company, LLC [EMAIL PROTECTED]:
Will Set(MONITOR_FILENAME=/blahblah/filename) work for you?
No, that doesn't work. ${MONITOR_FILENAME} can influence the filenames
in the string that you can tack onto the somix sequence using
${MONITOR_EXEC_ARGS}, but not the file name that automon produces. I
suppose you could also regard the automon output file name format as:
auto-${EPOCH}-${TOUCH_MONITOR}
The default is:
auto-${EPOCH}-caller-calee
Once again, it's easy to change and/or predict what the
${TOUCH_MONITOR} part is going to be, but AFAIK not the
'auto-${EPOCH}-' part. Therefore, if I'm right, there's no way to
manipulate the automon output using ${MONITOR_EXEC_ARGS}.
Thanks anyway,
Jaap
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Re: [asterisk-users] Digium stopped TDM400P production: alternatives??
How about a technical comparision. What makes the Rhino better than the Sangoma? On a scale of 1 to 10 I would give Sangoma a 9 for support based on personal experience so I strongly disagree with that part of your argument. -Original Message- From: James Finstrom [mailto:[EMAIL PROTECTED] Sent: Friday, February 15, 2008 6:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Digium stopped TDM400P production: alternatives?? Steve, Yes I work for Rhino that is no Secret. If you read the post I was responding to the thread not pimping my own products. I am not sure if your a Sangoma fanboy or employee since you are apparently offended by my response, however he wasn't asking to be sold to he was asking about specific products. So there it is yes I work for Rhino and I could have easily given one of our italian distributors but he didn't ask for that. It is not appropriate to troll the list and push your products unsolicited. If someone is looking for a recommendation for a card brand fine. If they need a solution like ADID or they need to accommodate funky CPC signals from their telco which Rhino does fine it is on subject. If someone asks should I use openvox to replace my digium you don't pimp your product because it wasn't asked for. If you want my honest opinion. I prefer people use Rhino products. I believe our products and support are superior but if you don't use our cards use Digium. If your reply is any indication on how Sandoma works I can honestly say go use a cheap clone before sangomaN they may not support you but at least they are open about being here just for the money. James Finstrom Rhino Equipment Corp. http://www.rhinoequipment.com -Original Message- From: Steve Totaro [EMAIL PROTECTED] Date: Fri, 15 Feb 2008 08:45:50 To:Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [asterisk-users] Digium stopped TDM400P production: alternatives?? James, If you were replying to the original post about Openvox or specified that is what you were referring to, maybe I would not take issue but to reply to a suggesting to use Sangoma with what you did is absolutely misleading. There is nothing cheap or clone about Sangoma's cards. asterisk.rhinoequipment.com hm. Thanks, Steve Totaro James Finstrom wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I would say email Kevin what he asked. The problem with switching to a clone company is you get what you pay for. Sticking with Digium you at least have support. and 3 clone cards and hours of troubleshooting later you will wish you hadn't been all cheap. Rob Hillis wrote: The cards themselves are okay, but the extra level of configuration is a pain in the proverbial. Zaptel is already double-configured in both zaptel.conf and zapata.conf (that's not a complaint - I understand the reason for the separation) but the Sangoma cards require a /third/ level of configuration in Wanpipe. Steve Totaro wrote: Sangoma makes a good card. On 2/15/08, Kevin P. Fleming [EMAIL PROTECTED] wrote: Giorgio Incantalupo wrote: Digium stopped to produce TDM400P and the new TDM410 is too new to find it in our shops. The only alternative available is a fully-compatible Openvox product...but is it really fully-compatible? Any experience about Openvox products (card and zaptel versions, etc...)? Every distributor that carried the TDM400P should have TDM410s in stock already. Where are you located, and who do you buy Digium cards from? -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- 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 !DSPAM:47b59f18311805637012918! ___ -- 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 stopped TDM400P production: alternatives??
How about a technical comparision. What makes the Rhino better than the Sangoma? On a scale of 1 to 10 I would give Sangoma a 9 for support based on personal experience so I strongly disagree with that part of your argument. -Original Message- From: James Finstrom [mailto:[EMAIL PROTECTED] Sent: Friday, February 15, 2008 6:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Digium stopped TDM400P production: alternatives?? Steve, Yes I work for Rhino that is no Secret. If you read the post I was responding to the thread not pimping my own products. I am not sure if your a Sangoma fanboy or employee since you are apparently offended by my response, however he wasn't asking to be sold to he was asking about specific products. So there it is yes I work for Rhino and I could have easily given one of our italian distributors but he didn't ask for that. It is not appropriate to troll the list and push your products unsolicited. If someone is looking for a recommendation for a card brand fine. If they need a solution like ADID or they need to accommodate funky CPC signals from their telco which Rhino does fine it is on subject. If someone asks should I use openvox to replace my digium you don't pimp your product because it wasn't asked for. If you want my honest opinion. I prefer people use Rhino products. I believe our products and support are superior but if you don't use our cards use Digium. If your reply is any indication on how Sandoma works I can honestly say go use a cheap clone before sangomaN they may not support you but at least they are open about being here just for the money. James Finstrom Rhino Equipment Corp. http://www.rhinoequipment.com -Original Message- From: Steve Totaro [EMAIL PROTECTED] Date: Fri, 15 Feb 2008 08:45:50 To:Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [asterisk-users] Digium stopped TDM400P production: alternatives?? James, If you were replying to the original post about Openvox or specified that is what you were referring to, maybe I would not take issue but to reply to a suggesting to use Sangoma with what you did is absolutely misleading. There is nothing cheap or clone about Sangoma's cards. asterisk.rhinoequipment.com hm. Thanks, Steve Totaro James Finstrom wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I would say email Kevin what he asked. The problem with switching to a clone company is you get what you pay for. Sticking with Digium you at least have support. and 3 clone cards and hours of troubleshooting later you will wish you hadn't been all cheap. Rob Hillis wrote: The cards themselves are okay, but the extra level of configuration is a pain in the proverbial. Zaptel is already double-configured in both zaptel.conf and zapata.conf (that's not a complaint - I understand the reason for the separation) but the Sangoma cards require a /third/ level of configuration in Wanpipe. Steve Totaro wrote: Sangoma makes a good card. On 2/15/08, Kevin P. Fleming [EMAIL PROTECTED] wrote: Giorgio Incantalupo wrote: Digium stopped to produce TDM400P and the new TDM410 is too new to find it in our shops. The only alternative available is a fully-compatible Openvox product...but is it really fully-compatible? Any experience about Openvox products (card and zaptel versions, etc...)? Every distributor that carried the TDM400P should have TDM410s in stock already. Where are you located, and who do you buy Digium cards from? -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- 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 !DSPAM:47b59f18311805637012918! ___ -- 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] 57iCT BLF problem
No. That's how we determined it was the phone and (therefore) most likely the firmware at fault. After we downgraded the firmware, the phone did correctly pick up it's hints. Sigma Networks wrote: Paul Hales wrote: We upgrade 2 of our Aastra 57iCT to the latest firmware (2.1.2.30) and the BLF indicators no longer function. Has anyone had a similar issue? And a solution? PaulH ___ -- 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 Paul, I have a couple of installations with Useragent: Aastra 57iCT/2.1.2.30 and Asterisk 1.4.15. If you do a show hints, does it show a watcher for each hint? ___ -- 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] Digium stopped TDM400P production: alternatives??
I told myself that I was going to stay out of this one, but since you find this important enough to reply twice to the mailing list with the same content, it must be worth my time to reply. If you carefully read the thread, the person who replied from Rhino went out of his way to NOT try to sell his hardware, until someone claimed that Sangoma is the best. I do not have first hand experience with Sangoma hardware. I am however one of the Astlinux developers. In that capacity, I can easily say that compiling the necessary modules for the Rhino cards is much easier than what is required to get the Sangoma stuff working. I have no doubt that in both cases the hardware is good. (I personally have not experienced significant trouble with Digium analog cards either, but they do take more time to adjust properly). Both hardware companies did work with us to ensure their cards will work properly with Astlinux. With the Rhino hardware, there is no need to compile extra utilities, only a zaptel module. The Sangoma setup is more complex. One plus in my mind is the Rhino card is made in the USA. I've also found the Rhino tech support to be excellent. Bottom line, use what works for you. I've used several Digium TDM400P cards and several Rhino analog cards. Both work well, but like I said earlier, the Rhino cards (with built in echo cancellation) were much easier to configure and get working out of the box. Darrick shadowym wrote: How about a technical comparision. What makes the Rhino better than the Sangoma? On a scale of 1 to 10 I would give Sangoma a 9 for support based on personal experience so I strongly disagree with that part of your argument. -Original Message- From: James Finstrom [mailto:[EMAIL PROTECTED] Sent: Friday, February 15, 2008 6:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Digium stopped TDM400P production: alternatives?? Steve, Yes I work for Rhino that is no Secret. If you read the post I was responding to the thread not pimping my own products. I am not sure if your a Sangoma fanboy or employee since you are apparently offended by my response, however he wasn't asking to be sold to he was asking about specific products. So there it is yes I work for Rhino and I could have easily given one of our italian distributors but he didn't ask for that. It is not appropriate to troll the list and push your products unsolicited. If someone is looking for a recommendation for a card brand fine. If they need a solution like ADID or they need to accommodate funky CPC signals from their telco which Rhino does fine it is on subject. If someone asks should I use openvox to replace my digium you don't pimp your product because it wasn't asked for. If you want my honest opinion. I prefer people use Rhino products. I believe our products and support are superior but if you don't use our cards use Digium. If your reply is any indication on how Sando ma works I can honestly say go use a cheap clone before sangomaN they may not support you but at least they are open about being here just for the money. James Finstrom Rhino Equipment Corp. http://www.rhinoequipment.com -Original Message- From: Steve Totaro [EMAIL PROTECTED] Date: Fri, 15 Feb 2008 08:45:50 To:Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [asterisk-users] Digium stopped TDM400P production: alternatives?? James, If you were replying to the original post about Openvox or specified that is what you were referring to, maybe I would not take issue but to reply to a suggesting to use Sangoma with what you did is absolutely misleading. There is nothing cheap or clone about Sangoma's cards. asterisk.rhinoequipment.com hm. Thanks, Steve Totaro James Finstrom wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I would say email Kevin what he asked. The problem with switching to a clone company is you get what you pay for. Sticking with Digium you at least have support. and 3 clone cards and hours of troubleshooting later you will wish you hadn't been all cheap. Rob Hillis wrote: The cards themselves are okay, but the extra level of configuration is a pain in the proverbial. Zaptel is already double-configured in both zaptel.conf and zapata.conf (that's not a complaint - I understand the reason for the separation) but the Sangoma cards require a /third/ level of configuration in Wanpipe. -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.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] Digium stopped TDM400P production: alternatives??
You are kidding, right ??? A small user that just buys one card won't get a good support from Digium. It'll be just a waste of time on the phone. Practically any manufacturer gives similar support including ssh'ing in the users box. Right now they push the user to buy a 4 channel echo canceller which you can get from Octasic for $40. The card with 4 ports is retail around $640. You can get OpenVox or another brand TDM400P compatible for 1/3 of that + $40 for echo canceller. Now that's a Digium high marigin right there .. someone has to pay the CEO salary and the mortgage for a new building :) cheers On 2/15/08, James Finstrom [EMAIL PROTECTED] wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I would say email Kevin what he asked. The problem with switching to a clone company is you get what you pay for. Sticking with Digium you at least have support. and 3 clone cards and hours of troubleshooting later you will wish you hadn't been all cheap. ___ -- 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 stopped TDM400P production: alternatives??
On Friday 15 February 2008 23:53:19 [EMAIL PROTECTED] wrote: You are kidding, right ??? A small user that just buys one card won't get a good support from Digium. It'll be just a waste of time on the phone. Do you have experience with this or are you just talking out of your ass? Digium support prides itself on giving customers who buy even just a single card the best possible support. Practically any manufacturer gives similar support including ssh'ing in the users box. Really? Which manufacturers, specifically, will allow you to call up, get remote assistance, and help you get the card working like this? I'd also like to know why you're posting anonymously, instead of standing behind your words. -- 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] Digium stopped TDM400P production: alternatives??
At 11:07 PM 2/15/2008, you wrote: Really? Which manufacturers, specifically, will allow you to call up, get remote assistance, and help you get the card working like this? Well, Digium did this for me when I had trouble getting something to work right with my TDM04. Took about 5 minutes with support logged in and all was well in my world. Ira, a happy Digium customer! ___ -- 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
