[asterisk-users] Asterisk 1.6.2.6 and ENUM LOOKUP? E.164
Hello, All i have one issue regarding caller id, once i received a call from my SIP provider it always set caller id with append 1 into original callerID if a call from USA then there is no problem , but if i receive a call from other country like INDIA i have also found callerID part as 191 which is wrong as from provider says that you should support E.164 ?? is that true that we do enable E.164 as per reading from some forums and after goggling i think that this would be a part from provider , they should send me call with correct callerID and should not append 1 as prefix. the meaning for posting this question is what is E.164 ?? and if i want to do this then how should i start to enable this on asterisk 1.6.2.6 give me some tips,tricks regarding issue. hope for good help !!! regards dhaval -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Limit Call Duration with L-option of Dial : announcement
Hello, Limiting the call duration with the L-option of the Dial()-command is working fine, however the announcement is not played. Dialplan : exten = _367,n,Set(LIMIT_PLAYAUDIO_CALLER=yes) exten = _367,n,Dial(SIP/test6,,L(11000,5000,5000)) The call lasts for 11 seconds, but 5 minutes before time runs out an announcement should come. I hear no announcement, not on caller-side nor on callee side. Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Limit Call Duration with L-option of Dial : announcement
Take a look at /var/log/asterisk/main or full /if enabled. Perhaps there is a file not found. try: exten = _367,n,Set(LIMIT_PLAYAUDIO_CALLER=yes) exten = _367,n,Set(LIMIT_WARNING_FILE=/path_to_your_audiofiles/file) # do not add any extension! exten = _367,n,Dial(SIP/test6,,L(11000,5000,5000)) Am 11.11.2010 10:31, schrieb Jonas Kellens: Hello, Limiting the call duration with the L-option of the Dial()-command is working fine, however the announcement is not played. Dialplan : exten = _367,n,Set(LIMIT_PLAYAUDIO_CALLER=yes) exten = _367,n,Dial(SIP/test6,,L(11000,5000,5000)) The call lasts for 11 seconds, but 5 minutes before time runs out an announcement should come. I hear no announcement, not on caller-side nor on callee side. Kind regards, Jonas. -- Thorsten Gllner OVM Office Voice Media GmbH Herderstrasse 68 40237 Dsseldorf Tel.: +49(0)211 / 618 57 53 Fax: +49(0)211 / 618 57 54 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Limit Call Duration with L-option of Dial : announcement
On Thu, Nov 11, 2010 at 3:43 AM, Thorsten Göllner t...@ovm-group.com wrote: Take a look at /var/log/asterisk/main or full /if enabled. Perhaps there is a file not found. try: exten = _367,n,Set(LIMIT_PLAYAUDIO_CALLER=yes) exten = _367,n,Set(LIMIT_WARNING_FILE=/path_to_your_audiofiles/file) # do not add any extension! exten = _367,n,Dial(SIP/test6,,L(11000,5000,5000)) Am 11.11.2010 10:31, schrieb Jonas Kellens: Hello, Limiting the call duration with the L-option of the Dial()-command is working fine, however the announcement is not played. Dialplan : exten = _367,n,Set(LIMIT_PLAYAUDIO_CALLER=yes) exten = _367,n,Dial(SIP/test6,,L(11000,5000,5000)) The call lasts for 11 seconds, but 5 minutes before time runs out an announcement should come. I hear no announcement, not on caller-side nor on callee side. Kind regards, Jonas. -- Thorsten Göllner OVM Office Voice Media GmbH Herderstrasse 68 40237 Düsseldorf Tel.: +49(0)211 / 618 57 53 Fax: +49(0)211 / 618 57 54 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Gentlemen, the issue is the lack of proper separatorYou are supposed to split the x y and z with : not , So...it should look like this: exten = _367,n,Set(LIMIT_PLAYAUDIO_CALLER=yes) exten = _367,n,Dial(SIP/test6,,L(11000:5000:5000)) Slainte mates -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Limit Call Duration with L-option of Dial : announcement
Found the problem already : Dial(SIP/test6,,L(11000,5000,5000)) Correct syntax is : Dial(SIP/test6,,L(11000:5000:5000)) semicolon... Jonas. On 11/11/2010 10:43 AM, Thorsten Göllner wrote: Take a look at /var/log/asterisk/main or full /if enabled. Perhaps there is a file not found. try: exten = _367,n,Set(LIMIT_PLAYAUDIO_CALLER=yes) exten = _367,n,Set(LIMIT_WARNING_FILE=/path_to_your_audiofiles/file) # do not add any extension! exten = _367,n,Dial(SIP/test6,,L(11000,5000,5000)) Am 11.11.2010 10:31, schrieb Jonas Kellens: Hello, Limiting the call duration with the L-option of the Dial()-command is working fine, however the announcement is not played. Dialplan : exten = _367,n,Set(LIMIT_PLAYAUDIO_CALLER=yes) exten = _367,n,Dial(SIP/test6,,L(11000,5000,5000)) The call lasts for 11 seconds, but 5 minutes before time runs out an announcement should come. I hear no announcement, not on caller-side nor on callee side. Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] changing sip port
Hello How can I run the sip service on asterisk on another port beside 5080? I mean asterisk will still take sip requests on port:5080 and another custom port, lets say port:6080 Thanks for any help -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Playback sound dropping on linphone
Hi, I dial on A* from a linphonec to a Playback() extension, then suddenly the sound stops after a while, without any notice. I enabled debug both in linphone and A*, and the RTP packets are sent from A* and received from linphone. It doesn't matter whether I choose alaw, ulaw, gsm as codec (besides changing cpu load of course). How can I debug it? I'm using A* 1.6.2 and both linphone 2.x and 3.x. I just need a console scriptable softphone, so maybe there's an alternative to linphone (which seemed good enough anyway!)... Thank you, Matteo -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ISDN - Busy signal on 3rd call
Paulo Santos wrote: Hello, Following my first mail about this issue [1], I think I know now what the problem is. When I have both lines being used and a third call comes in, the person calling doesn't get a busy tone, he gets something like line unavailable. I've been debugging mISDN and I think the reason is because asterisk is sending the release cause as 0. P[ 3] -- channel:0 mode:TE cause:0 ocause:0 rad: cad: The request from the telephone company's switch seems correct, a SETUP message (if 08 is Q.931, 05 is SETUP). 02 ff 03 08 01 04 05 a1 04 03 80 90 a3 18 01 80 6c 0b 01 83 39 31 36 33 39 31 37 34 32 70 03 c1 38 34 I've changed misdn.conf so it sends a release cause as 17 (user busy), but I get the same behaviour - cause:0 ocause:0. Anyone knows how can I force asterisk to send cause 16 or 17 in this situation? Thanks in advance. Best regards, Paulo Santos misdn.conf: http://pastebin.com/FmgECqkU misdn debug: http://pastebin.com/Tg6wPKBD [1] http://www.mail-archive.com/asterisk-users@lists.digium.com/msg244330.html Ok, I've encountered a similar issue on a different installation but instead of being PTP it's PTMP. Plus, it's a setup with 2 BRI lines with call forwarding between them - main number of BRI1 forwards to secondary number of BRI2 when busy/unavailable and vice-versa. I've called the phone company and confirmed that call waiting is disabled, yet I get a message in misdn debug saying: P[ 2] -- Call Waiting on PMP sending RELEASE_COMPLETE I don't know if this is the actual call waiting feature or if it is just an information of some kind. In the misdn debug I get this: http://pastebin.com/D7wv0qqm The P[ 2] is the port of the BRI line I called in the first place, then it is forwarded to P[ 1] where I get an error: P[ 1] Decoding FACILITY failed! (-1) And the same issue I said in the previews email: P[ 1] -- channel:0 mode:TE cause:0 ocause:0 rad: cad: I changed isdn_lib.c and now I'm sending ocause:17 (user busy). I've done this in the PTP line mentioned in the previews email as well. For the PTP line it appears to have worked, I have the regular busy signal. It worked only after the first time I tried to place a 3rd call. Now the 3rd call doesn't even reach Asterisk, which was what I wanted from the phone company in the first place. On the PTMP line it didn't work, I still don't get the busy signal. Maybe cause 17 isn't the right one? And what can be that FACILITY mentioned in the debug? Thanks in advance. Best regards, Paulo Santos -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP DNS SRV
On 11/09/2010 03:20 PM, Gareth Blades wrote: Jonas Kellens wrote: On 11/09/2010 02:12 PM, Gareth Blades wrote: Jonas Kellens wrote: On 11/08/2010 09:50 PM, Jonas Kellens wrote: Hello, SIP DNS SRV records are not working. My Grandstream uses the SRV records to find the first Asterisk server to register to. This works. But when I shut down the Asterisk proces on server 1 and I restart my GXP 2010, the phone does not register to server 2... No mather how long I wait, there is no registration coming in... When I start the Asterisk proces again on server 1, then here registration comes in. Kind regards, Jonas. More info : [jo...@jonas ~]$ host -t srv _SIP._udp.sip10.domain.tld _SIP._udp.sip10.domain.tld has SRV record 25 10 5060 sip2.domain.tld. _SIP._udp.sip10.domain.tld has SRV record 5 10 5060 sip1.domain.tld. It sounds like the grandstream phones are not fully compliant with the SRV standard. They are probably just looking for the lowest priority entry and hardcoding that to be used all the time internally. If you restart the phone does it work? It might try the 25 priority entry if it cannot initially contact the primary server. The way I test it : - Grandstream turned off. - Stop asterisk server1 (/sbin/service asterisk stop) - Turn on Grandstream (power up) Conclusion : Grandstream does not register. No register coming in on server2. Finally : - Start Asterisk again on server1 (/sbin/service asterisk start) Conclusion : Grandstream registers to server1. Jonas. Then it looks like the Grandstream phone dont fully support DNS SRV. maybe a firmware update will fix it. I use firmware version 1.2.4.3 http://grandstream.com/DOWNLOAD/FIRMWARE/BT200_GXP/Release_BT200_GXP_1.2.4.3.zip, which is the latest stable. Has anyone on this list implemented a successful Grandstream with DNS SRV support ?! Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] VoiceMail customizing
Hello We would like to customize the voicemail menues. So the intro should not be played if some user has recorded an own greeting message and we would also like to remove some options from the menue. Is this all hardcoded or is it somehow possible to redefine the voice menues and the order how messages are played via voicemail.conf? Mit freundlichen Grüssen Benoit Panizzon -- I m p r o W a r e A G- __ Zurlindenstrasse 29 Tel +41 61 826 93 07 CH-4133 PrattelnFax +41 61 826 93 02 Schweiz Web http://www.imp.ch __ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoiceMail customizing
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Benoit Panizzon Sent: Thursday, November 11, 2010 11:29 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] VoiceMail customizing Hello We would like to customize the voicemail menues. So the intro should not be played if some user has recorded an own greeting message and we would also like to remove some options from the menue. Is this all hardcoded or is it somehow possible to redefine the voice menues and the order how messages are played via voicemail.conf? Unfortunately, regular Voicemail/VoiceMailMain does not have customizable menus. However, if that is something you need/want, look into MiniVM. It's definitely a build-it-yourself approach, but will accomplish what you are looking to do. - Brad -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Playback sound dropping on linphone
I did some more tests, and it's not really a problem with linphone: the rtp capture shows empty packets sent by Asterisk. Since the channel which is doing Playback() is in a MeetMe conference, I tried also to speak on another phone on the same conference: well the rtp capture shows the stream from A* becoming silent, then the new sound from the phone comes up. Do I have to file a bug? Thank you, Matteo Il 11/11/2010 16:35, Matteo Fortini ha scritto: Hi, I dial on A* from a linphonec to a Playback() extension, then suddenly the sound stops after a while, without any notice. I enabled debug both in linphone and A*, and the RTP packets are sent from A* and received from linphone. It doesn't matter whether I choose alaw, ulaw, gsm as codec (besides changing cpu load of course). How can I debug it? I'm using A* 1.6.2 and both linphone 2.x and 3.x. I just need a console scriptable softphone, so maybe there's an alternative to linphone (which seemed good enough anyway!)... Thank you, Matteo -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4.37 Released
The Asterisk Development Team has announced the release of Asterisk 1.4.37. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.4.37 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this release: * Fix issue with decoding ^-escaped characters in realtime (res_pgsql) (Closes issue #17790. Reported denzs. Patched by Qwell) * Don't send a devstate change on poke_noanswer if the state did not change. (Closes issue #17741. Reported, patched by schmidts) * Transmit silence when reading DTMF in ast_readstring. Otherwise you could get issues with DTMF timeouts causing hangups. (Closes issue #17370. Reported, patched by makoto) * Fix to SIP extension state update (deadlock issues) (Closes issue #17888. Reported by zerohalo. Patched by dvossel) * Fix issue with MoH where it doesn't recover cleanly when it can't play a file and would just stop, instead of continuing to find the next playable file in the MoH class. (Closes issue #17807. Reported by kshumard. Patched by bbryant) For a full list of changes in the current release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.37 Thank you for your continued support of Asterisk! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.6.2.14 Released
The Asterisk Development Team has announced the release of Asterisk 1.6.2.14. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.2.14 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this release: * Fix issue where session timers would be advertised as supported even when session-timers=refuse was set in sip.conf. Also fix interoperability problems with session timer behavior in Asterisk. (Closes issue #17005. Reported by alexcarey. Patched by dvossel) * Parse all Accept headers for SIP SUBSCRIBE requests. (Closes issue #17758. Reported by ibc. Patched by dvossel) * Fix issue where queue stats would be reset on reload. (Closes issue #17535. Reported by raarts. Patched by tilghman) * Fix issue where MoH files were no longer rescanned on during a reload. (Closes issue #16744. Reported by pj. Patched by Qwell) * Fix issue with dialplan pattern matching where the specificity for pattern ranges and pattern characters was inconsistent. (Closes issue #16903. Reported, patched by Nick_Lewis) For a full list of changes in the current release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.14 Thank you for your continued support of Asterisk! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoiceMail customizing
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Watkins, Bradley Sent: Thursday, November 11, 2010 10:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] VoiceMail customizing -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Benoit Panizzon Sent: Thursday, November 11, 2010 11:29 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] VoiceMail customizing Hello We would like to customize the voicemail menues. So the intro should not be played if some user has recorded an own greeting message and we would also like to remove some options from the menue. Is this all hardcoded or is it somehow possible to redefine the voice menues and the order how messages are played via voicemail.conf? Unfortunately, regular Voicemail/VoiceMailMain does not have customizable menus. However, if that is something you need/want, look into MiniVM. It's definitely a build-it-yourself approach, but will accomplish what you are looking to do. - Brad A more (less?) drastic approach for those not inclined to BIY would be to silence the prompts for the unwanted VM options. I'm betting that 90% of end-users only know the functions that they are told by the system. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T38 re-invites issue
NAT? Firewall? On Thu, Nov 11, 2010 at 3:21 PM, Marek Soha ma...@snet.sk wrote: Hi all. I have an issue with T.38 and re-invites. Topology: provider - A (asterisk 1.6) - B (asterisk 1.6) - extension - - (software fax, gateway whatever). When between A and B trunk is canreinvite=no everything is working smooth. When I switch canreinvite to yes, it stop working. Do you have any idea where the issue can be? Any help will be much appreciated. Marek Soha -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T38 re-invites issue
On 11/11/2010 04:21 PM, Marek Soha wrote: Hi all. I have an issue with T.38 and re-invites. Topology: provider - A (asterisk 1.6) - B (asterisk 1.6) - extension - - (software fax, gateway whatever). When between A and B trunk is canreinvite=no everything is working smooth. When I switch canreinvite to yes, it stop working. Do you have any idea where the issue can be? Any help will be much appreciated. 'canreinvite' has *nothing* to do with T.38 re-INVITEs. 'canreinvite' controls whether Asterisk is allowed to setup a direct media path between endpoints, or must keep the media flowing through itself. If you don't need a direct media path, don't enable 'canreinvite' (which in Asterisk 1.8 has been renamed to avoid this confusion). If you do need a direct media path, you'll have a problem, because Asterisk cannot currently setup UDPTL (T.38) direct media paths, only RTP direct media paths. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T38 re-invites issue
Does it matter? Phones are working correctly...I tried also portforwarding. So corrected topology: provider - A (asterisk 1.6) - B (asterisk 1.6) - extension - - NAT/FIREWALL - (software fax, gateway whatever). Software fax ends with DIS sent, 9600Bbps Joel, dňa 11. novembra 2010 ste napísali: JM NAT? Firewall? JM On Thu, Nov 11, 2010 at 3:21 PM, Marek Soha ma...@snet.sk wrote: Hi all. I have an issue with T.38 and re-invites. Topology: provider - A (asterisk 1.6) - B (asterisk 1.6) - extension - - (software fax, gateway whatever). When between A and B trunk is canreinvite=no everything is working smooth. When I switch canreinvite to yes, it stop working. Do you have any idea where the issue can be? Any help will be much appreciated. Marek Soha -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T38 re-invites issue
On 11/11/2010 04:48 PM, Marek Soha wrote: Uf... you are perfectly clean about that confusion... Only thing I want to do, is to route stream out of local asterisk - to connect final extension directly to sender - provider. So what I can do if I need i.e: 1) canreinvite=yes AND send T38 faxes through the same trunk? 2) maybe canreinvite=no, but UDPTL routed outside local asterisk (called canreinvite in normal RTP traffic)? Yes and - many thanks Kevin for your perfect reply. There is no solution today; as I said, Asterisk only knows how to transfer UDPTL traffic through itself right now. The changes to support direct media paths for UDPTL wouldn't be terribly difficult, but nobody has done the work yet that I know of. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] scratchy sound on TE410P
On Tue, 9 Nov 2010, Daniel Tryba wrote: I am curious about the tool dahdi_maint... what do the various acronyms stand for? Yea there seemed to be a bit of confusion here as well so I patched trunk with some more descriptive error counter labels :O) FEC : 0: Framing Errors CEC : 0: CRC Errors CVC : 0: Code Violations EBC : 0: E-Bit Counter BEC : 0: PRBS: 0: Both of these were removed due to stale code GES : 76: General Errored Seconds Was this output a snapshot of your current system. Do you really have zeroed counters everywhere, but 76 errored seconds? If so, I'll probably need to investigate. -- Russ Meyerriecks Digium | Linux Kernel Developer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] scratchy sound on TE410P
On Thu, 11 Nov 2010, Russ Meyerriecks wrote: On Tue, 9 Nov 2010, Daniel Tryba wrote: I am curious about the tool dahdi_maint... what do the various acronyms stand for? Yea there seemed to be a bit of confusion here as well so I patched trunk with some more descriptive error counter labels :O) FEC : 0: Framing Errors CEC : 0: CRC Errors CVC : 0: Code Violations EBC : 0: E-Bit Counter BEC : 0: PRBS: 0: Both of these were removed due to stale code GES : 76: General Errored Seconds Nice! Perfect. Was this output a snapshot of your current system. Do you really have zeroed counters everywhere, but 76 errored seconds? If so, I'll probably need to investigate. Yes, this is a snapshot after about 24 hours since I cleared the counters. I see what you mean - how can I have 76 seconds of errors but no bumped error counters. I ran again just now: r...@vigw3:/etc/asterisk# dahdi_maint -s 1 Span 1: FEC : 0: CEC : 0: CVC : 0: EBC : 0: BEC : 0: PRBS: 0: GES : 248: Here's to hoping that the error in error is the GES, and that actually I have no errors ;) Cheers, j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] scratchy sound on TE410P
On 11/11/10 5:44 PM, Jeff LaCoursiere wrote: On Thu, 11 Nov 2010, Russ Meyerriecks wrote: On Tue, 9 Nov 2010, Daniel Tryba wrote: I am curious about the tool dahdi_maint... what do the various acronyms stand for? Yea there seemed to be a bit of confusion here as well so I patched trunk with some more descriptive error counter labels :O) FEC : 0: Framing Errors CEC : 0: CRC Errors CVC : 0: Code Violations EBC : 0: E-Bit Counter BEC : 0: PRBS: 0: Both of these were removed due to stale code GES : 76: General Errored Seconds Nice! Perfect. Was this output a snapshot of your current system. Do you really have zeroed counters everywhere, but 76 errored seconds? If so, I'll probably need to investigate. Yes, this is a snapshot after about 24 hours since I cleared the counters. I see what you mean - how can I have 76 seconds of errors but no bumped error counters. I ran again just now: r...@vigw3:/etc/asterisk# dahdi_maint -s 1 Span 1: FEC : 0: CEC : 0: CVC : 0: EBC : 0: BEC : 0: PRBS: 0: GES : 248: Here's to hoping that the error in error is the GES, and that actually I have no errors ;) Cheers, j Could you paste a cat of /proc/dahdi/1 for me here? -- Russ Meyerriecks Digium | Linux Kernel Developer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] scratchy sound on TE410P
On Thu, 2010-11-11 at 18:28 -0600, Russ Meyerriecks wrote: On 11/11/10 5:44 PM, Jeff LaCoursiere wrote: On Thu, 11 Nov 2010, Russ Meyerriecks wrote: On Tue, 9 Nov 2010, Daniel Tryba wrote: I am curious about the tool dahdi_maint... what do the various acronyms stand for? Yea there seemed to be a bit of confusion here as well so I patched trunk with some more descriptive error counter labels :O) FEC : 0: Framing Errors CEC : 0: CRC Errors CVC : 0: Code Violations EBC : 0: E-Bit Counter BEC : 0: PRBS: 0: Both of these were removed due to stale code GES : 76: General Errored Seconds Nice! Perfect. Was this output a snapshot of your current system. Do you really have zeroed counters everywhere, but 76 errored seconds? If so, I'll probably need to investigate. Yes, this is a snapshot after about 24 hours since I cleared the counters. I see what you mean - how can I have 76 seconds of errors but no bumped error counters. I ran again just now: r...@vigw3:/etc/asterisk# dahdi_maint -s 1 Span 1: FEC : 0: CEC : 0: CVC : 0: EBC : 0: BEC : 0: PRBS: 0: GES : 248: Here's to hoping that the error in error is the GES, and that actually I have no errors ;) Cheers, j Could you paste a cat of /proc/dahdi/1 for me here? I seem to be having the same problem with a new server. I am using a TE220 with a VPM450 module. Using Dahdi 2.4.0 and Asterisk 1.6.2.13 on a Dell server. All calls to the outside have bad voice quality (echo and distortion). Internal calls between extensions sound fine. Dahdi_main for Span 2 (20 days uptime). Span 2: FEC : 103: CEC : 0: CVC : 0: EBC : 0: BEC : 0: PRBS: 72: GES : 4054: Span 2: TE2/0/2 T2XXP (PCI) Card 0 Span 2 (MASTER) HDB3/CCS Timing slips: 1566 32 TE2/0/2/1 Clear (In use) (EC: VPM450M) 33 TE2/0/2/2 Clear (In use) (EC: VPM450M) 34 TE2/0/2/3 Clear (In use) 35 TE2/0/2/4 Clear (In use) 36 TE2/0/2/5 Clear (In use) 37 TE2/0/2/6 Clear (In use) 38 TE2/0/2/7 Clear (In use) 39 TE2/0/2/8 Clear (In use) 40 TE2/0/2/9 Clear (In use) 41 TE2/0/2/10 Clear (In use) 42 TE2/0/2/11 Clear (In use) 43 TE2/0/2/12 Clear (In use) 44 TE2/0/2/13 Clear (In use) 45 TE2/0/2/14 Clear (In use) 46 TE2/0/2/15 Clear (In use) 47 TE2/0/2/16 HDLCFCS (In use) 48 TE2/0/2/17 Clear (In use) 49 TE2/0/2/18 Clear (In use) 50 TE2/0/2/19 Clear (In use) 51 TE2/0/2/20 Clear (In use) 52 TE2/0/2/21 Clear (In use) 53 TE2/0/2/22 Clear (In use) 54 TE2/0/2/23 Clear (In use) 55 TE2/0/2/24 Clear (In use) 56 TE2/0/2/25 Clear (In use) 57 TE2/0/2/26 Clear (In use) 58 TE2/0/2/27 Clear (In use) 59 TE2/0/2/28 Clear (In use) 60 TE2/0/2/29 Clear (In use) 61 TE2/0/2/30 Clear (In use) 62 TE2/0/2/31 Clear (In use) We are using span 2 at the moment because span 1 is reserved for another E1 that will be connected soon. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] scratchy sound on TE410P
On 11/11/10 7:23 PM, Carlos Chavez wrote: I seem to be having the same problem with a new server. I am using a TE220 with a VPM450 module. Using Dahdi 2.4.0 and Asterisk 1.6.2.13 on a Dell server. All calls to the outside have bad voice quality (echo and distortion). Internal calls between extensions sound fine. Dahdi_main for Span 2 (20 days uptime). Span 2: FEC : 103: CEC : 0: CVC : 0: EBC : 0: BEC : 0: PRBS: 72: GES : 4054: Span 2: TE2/0/2 T2XXP (PCI) Card 0 Span 2 (MASTER) HDB3/CCS Timing slips: 1566 With framing errors and slips it could be a timing issue. Let's see your /etc/dahdi/system.conf -- Russ Meyerriecks Digium | Linux Kernel Developer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TTS in Asterisk on Solaris
You try install debian in your sparc platform ? On Thu, Nov 11, 2010 at 8:52 PM, RR ranjt...@gmail.com wrote: Hello Group, I have been going through all the chit-chat about TTS and the various engines available to integrate with Asterisk incl. flite/festival, espeak, Nuance etc but I am wondering if anyone's tried any or all of these to compile on a Sparc based Solaris platform? If not, then what is the best way for me to accomplish a production environment TTS service when most of my servers or the core of the servers are Sparc based Solaris platforms. I found a group that seems to have done a fair bit of work on compiling Asterisk on Solaris, but I'm wondering if it'll be possible for me to have my core platform running Asterisk on Sparc Solaris and a set of Linux servers serving as a TTS cluster to which the calls can be thrown to for processing and then have them be played back to the user. Any ideas/advice? Thanks RR -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- - Luis Morales Consultor de Tecnologia Cel: +58(0412)2352745 OpenID: http://lmorales.myopenid.com/ Twitter: @magnadata Linux User ID : 470650 - Empieza por hacer lo necesario, luego lo que es posible... y de pronto estarás haciendo lo imposible Leonardo Da'Vinci - -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TTS in Asterisk on Solaris
No, I want to use Solaris 10 on the Sparc platform. I've read a lot of reports and tests/benchmarks conducted that sow Solaris 10 actually performing better than all other Linux based Distros...not sure if that's been the experience of others in the group. I really want to know if someone has a high performance TTS based service running in a production environment. What product are they using as their core engine, does it handle and has available many different languages and can one build these independently of the telephony platform being used so I could use maybe Asterisk running on Solaris 10 and a cluster/farm of TTS servers for TTS processing. Thanks RR On Thu, Nov 11, 2010 at 10:21 PM, Luis Morales faston...@gmail.com wrote: You try install debian in your sparc platform ? On Thu, Nov 11, 2010 at 8:52 PM, RR ranjt...@gmail.com wrote: Hello Group, I have been going through all the chit-chat about TTS and the various engines available to integrate with Asterisk incl. flite/festival, espeak, Nuance etc but I am wondering if anyone's tried any or all of these to compile on a Sparc based Solaris platform? If not, then what is the best way for me to accomplish a production environment TTS service when most of my servers or the core of the servers are Sparc based Solaris platforms. I found a group that seems to have done a fair bit of work on compiling Asterisk on Solaris, but I'm wondering if it'll be possible for me to have my core platform running Asterisk on Sparc Solaris and a set of Linux servers serving as a TTS cluster to which the calls can be thrown to for processing and then have them be played back to the user. Any ideas/advice? Thanks RR -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- - Luis Morales Consultor de Tecnologia Cel: +58(0412)2352745 OpenID: http://lmorales.myopenid.com/ Twitter: @magnadata Linux User ID : 470650 - Empieza por hacer lo necesario, luego lo que es posible... y de pronto estarás haciendo lo imposible Leonardo Da'Vinci - -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TTS in Asterisk on Solaris
I use Nuance, festival, Ibm tts and Loquendo. Now in your case, i suggest use tts on the recommend tts environment. Solaris is not standart system for tts products. Then you can plug tts system into asterisk platform. I use Debian for sparc and work excelent!! don't discard this option may be an good choice. Regards, On Thu, Nov 11, 2010 at 11:36 PM, RR ranjt...@gmail.com wrote: No, I want to use Solaris 10 on the Sparc platform. I've read a lot of reports and tests/benchmarks conducted that sow Solaris 10 actually performing better than all other Linux based Distros...not sure if that's been the experience of others in the group. I really want to know if someone has a high performance TTS based service running in a production environment. What product are they using as their core engine, does it handle and has available many different languages and can one build these independently of the telephony platform being used so I could use maybe Asterisk running on Solaris 10 and a cluster/farm of TTS servers for TTS processing. Thanks RR On Thu, Nov 11, 2010 at 10:21 PM, Luis Morales faston...@gmail.com wrote: You try install debian in your sparc platform ? On Thu, Nov 11, 2010 at 8:52 PM, RR ranjt...@gmail.com wrote: Hello Group, I have been going through all the chit-chat about TTS and the various engines available to integrate with Asterisk incl. flite/festival, espeak, Nuance etc but I am wondering if anyone's tried any or all of these to compile on a Sparc based Solaris platform? If not, then what is the best way for me to accomplish a production environment TTS service when most of my servers or the core of the servers are Sparc based Solaris platforms. I found a group that seems to have done a fair bit of work on compiling Asterisk on Solaris, but I'm wondering if it'll be possible for me to have my core platform running Asterisk on Sparc Solaris and a set of Linux servers serving as a TTS cluster to which the calls can be thrown to for processing and then have them be played back to the user. Any ideas/advice? Thanks RR -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- - Luis Morales Consultor de Tecnologia Cel: +58(0412)2352745 OpenID: http://lmorales.myopenid.com/ Twitter: @magnadata Linux User ID : 470650 - Empieza por hacer lo necesario, luego lo que es posible... y de pronto estarás haciendo lo imposible Leonardo Da'Vinci - -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- - Luis Morales Consultor de Tecnologia Cel: +58(0412)2352745 OpenID: http://lmorales.myopenid.com/ Twitter: @magnadata Linux User ID : 470650 - Empieza por hacer lo necesario, luego lo que es posible... y de pronto estarás haciendo lo imposible Leonardo Da'Vinci - -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] scratchy sound on TE410P
On Thu, 11 Nov 2010 20:08:09 -0600, Russ Meyerriecks wrote On 11/11/10 7:23 PM, Carlos Chavez wrote: I seem to be having the same problem with a new server. I am using a TE220 with a VPM450 module. Using Dahdi 2.4.0 and Asterisk 1.6.2.13 on a Dell server. All calls to the outside have bad voice quality (echo and distortion). Internal calls between extensions sound fine. Dahdi_main for Span 2 (20 days uptime). Span 2: FEC : 103: CEC : 0: CVC : 0: EBC : 0: BEC : 0: PRBS: 72: GES : 4054: Span 2: TE2/0/2 T2XXP (PCI) Card 0 Span 2 (MASTER) HDB3/CCS Timing slips: 1566 With framing errors and slips it could be a timing issue. Let's see your /etc/dahdi/system.conf Here it is. # Span 1: TE2/0/1 T2XXP (PCI) Card 0 Span 1 span=1,2,0,cas,hdb3 cas=1-15:1101 cas=17-31:1101 # Span 2: TE2/0/2 T2XXP (PCI) Card 0 Span 2 (MASTER) span=2,1,0,ccs,hdb3 # termtype: te bchan=32-46 dchan=47 bchan=48-62 loadzone= mx defaultzone = mx -- Carlos Chavez Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TTS in Asterisk on Solaris
Hi Luis, Thanks for your comments. How / Why are you using that many TTS products? Do you have a preference of one over the other? Also, do you have any documentation / install/configuration notes that you might be willing to share re: your experience with Debian on Sparc and the TTS configuration you have. I agree with you. I will use TTS in its own native environment and have Asterisk talk to it using UniMRCP or something but I need a lot of help. Any help will be appreciated. Thanks \RR On Thu, Nov 11, 2010 at 11:22 PM, Luis Morales faston...@gmail.com wrote: I use Nuance, festival, Ibm tts and Loquendo. Now in your case, i suggest use tts on the recommend tts environment. Solaris is not standart system for tts products. Then you can plug tts system into asterisk platform. I use Debian for sparc and work excelent!! don't discard this option may be an good choice. Regards, On Thu, Nov 11, 2010 at 11:36 PM, RR ranjt...@gmail.com wrote: No, I want to use Solaris 10 on the Sparc platform. I've read a lot of reports and tests/benchmarks conducted that sow Solaris 10 actually performing better than all other Linux based Distros...not sure if that's been the experience of others in the group. I really want to know if someone has a high performance TTS based service running in a production environment. What product are they using as their core engine, does it handle and has available many different languages and can one build these independently of the telephony platform being used so I could use maybe Asterisk running on Solaris 10 and a cluster/farm of TTS servers for TTS processing. Thanks RR On Thu, Nov 11, 2010 at 10:21 PM, Luis Morales faston...@gmail.com wrote: You try install debian in your sparc platform ? On Thu, Nov 11, 2010 at 8:52 PM, RR ranjt...@gmail.com wrote: Hello Group, I have been going through all the chit-chat about TTS and the various engines available to integrate with Asterisk incl. flite/festival, espeak, Nuance etc but I am wondering if anyone's tried any or all of these to compile on a Sparc based Solaris platform? If not, then what is the best way for me to accomplish a production environment TTS service when most of my servers or the core of the servers are Sparc based Solaris platforms. I found a group that seems to have done a fair bit of work on compiling Asterisk on Solaris, but I'm wondering if it'll be possible for me to have my core platform running Asterisk on Sparc Solaris and a set of Linux servers serving as a TTS cluster to which the calls can be thrown to for processing and then have them be played back to the user. Any ideas/advice? Thanks RR -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- - Luis Morales Consultor de Tecnologia Cel: +58(0412)2352745 OpenID: http://lmorales.myopenid.com/ Twitter: @magnadata Linux User ID : 470650 - Empieza por hacer lo necesario, luego lo que es posible... y de pronto estarás haciendo lo imposible Leonardo Da'Vinci - -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- - Luis Morales Consultor de Tecnologia Cel: +58(0412)2352745 OpenID: http://lmorales.myopenid.com/ Twitter: @magnadata Linux User ID : 470650 - Empieza por hacer lo necesario, luego lo que es posible... y de pronto estarás haciendo lo imposible Leonardo Da'Vinci
Re: [asterisk-users] TTS in Asterisk on Solaris
Well, I use many tts products because i work with diferents telphone systems. Now for asterisk the best way for free is Festival and noon free is Loquendo. I'm not have notes to install debian on Sparc, i just only use debian readme :-) It's too easy, debian work for you :D Just download sparc image, burn it and install. Regards, On Fri, Nov 12, 2010 at 1:23 AM, RR ranjt...@gmail.com wrote: Hi Luis, Thanks for your comments. How / Why are you using that many TTS products? Do you have a preference of one over the other? Also, do you have any documentation / install/configuration notes that you might be willing to share re: your experience with Debian on Sparc and the TTS configuration you have. I agree with you. I will use TTS in its own native environment and have Asterisk talk to it using UniMRCP or something but I need a lot of help. Any help will be appreciated. Thanks \RR On Thu, Nov 11, 2010 at 11:22 PM, Luis Morales faston...@gmail.com wrote: I use Nuance, festival, Ibm tts and Loquendo. Now in your case, i suggest use tts on the recommend tts environment. Solaris is not standart system for tts products. Then you can plug tts system into asterisk platform. I use Debian for sparc and work excelent!! don't discard this option may be an good choice. Regards, On Thu, Nov 11, 2010 at 11:36 PM, RR ranjt...@gmail.com wrote: No, I want to use Solaris 10 on the Sparc platform. I've read a lot of reports and tests/benchmarks conducted that sow Solaris 10 actually performing better than all other Linux based Distros...not sure if that's been the experience of others in the group. I really want to know if someone has a high performance TTS based service running in a production environment. What product are they using as their core engine, does it handle and has available many different languages and can one build these independently of the telephony platform being used so I could use maybe Asterisk running on Solaris 10 and a cluster/farm of TTS servers for TTS processing. Thanks RR On Thu, Nov 11, 2010 at 10:21 PM, Luis Morales faston...@gmail.com wrote: You try install debian in your sparc platform ? On Thu, Nov 11, 2010 at 8:52 PM, RR ranjt...@gmail.com wrote: Hello Group, I have been going through all the chit-chat about TTS and the various engines available to integrate with Asterisk incl. flite/festival, espeak, Nuance etc but I am wondering if anyone's tried any or all of these to compile on a Sparc based Solaris platform? If not, then what is the best way for me to accomplish a production environment TTS service when most of my servers or the core of the servers are Sparc based Solaris platforms. I found a group that seems to have done a fair bit of work on compiling Asterisk on Solaris, but I'm wondering if it'll be possible for me to have my core platform running Asterisk on Sparc Solaris and a set of Linux servers serving as a TTS cluster to which the calls can be thrown to for processing and then have them be played back to the user. Any ideas/advice? Thanks RR -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- - Luis Morales Consultor de Tecnologia Cel: +58(0412)2352745 OpenID: http://lmorales.myopenid.com/ Twitter: @magnadata Linux User ID : 470650 - Empieza por hacer lo necesario, luego lo que es posible... y de pronto estarás haciendo lo imposible Leonardo Da'Vinci - -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --
Re: [asterisk-users] changing sip port
asterisk by default listen on port 5060.You simply need open the file /etc/asterisk/sip.conf and change these. udpbindaddr=0.0.0.0:6080tcpbindaddr=0.0.0.0:6080save the file and open asterisk console and execute sip reload. Muhammad Faheem --- On Fri, 11/12/10, Baha @ SH i...@saudihome.com wrote: From: Baha @ SH i...@saudihome.com Subject: [asterisk-users] changing sip port To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Date: Friday, November 12, 2010, 3:25 AM Hello How can I run the sip service on asterisk on another port beside 5080? I mean asterisk will still take sip requests on port:5080 and another custom port, lets say port:6080 Thanks for any help -Inline Attachment Follows- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TTS in Asterisk on Solaris
Sure, no worries. Will try that. What about advice on TTS setup. Would you have any notes on how best to setup high-volume TTS environment, like maybe a cluster of TTS servers and how Asterisk talks to those? Recommendations on how to set that up? I'm thinking about trying Festival/FLite and maybe Cepstral? How expensive is Loquendo? Thanks RR On Fri, Nov 12, 2010 at 1:35 AM, Luis Morales faston...@gmail.com wrote: Well, I use many tts products because i work with diferents telphone systems. Now for asterisk the best way for free is Festival and noon free is Loquendo. I'm not have notes to install debian on Sparc, i just only use debian readme :-) It's too easy, debian work for you :D Just download sparc image, burn it and install. Regards, On Fri, Nov 12, 2010 at 1:23 AM, RR ranjt...@gmail.com wrote: Hi Luis, Thanks for your comments. How / Why are you using that many TTS products? Do you have a preference of one over the other? Also, do you have any documentation / install/configuration notes that you might be willing to share re: your experience with Debian on Sparc and the TTS configuration you have. I agree with you. I will use TTS in its own native environment and have Asterisk talk to it using UniMRCP or something but I need a lot of help. Any help will be appreciated. Thanks \RR On Thu, Nov 11, 2010 at 11:22 PM, Luis Morales faston...@gmail.com wrote: I use Nuance, festival, Ibm tts and Loquendo. Now in your case, i suggest use tts on the recommend tts environment. Solaris is not standart system for tts products. Then you can plug tts system into asterisk platform. I use Debian for sparc and work excelent!! don't discard this option may be an good choice. Regards, On Thu, Nov 11, 2010 at 11:36 PM, RR ranjt...@gmail.com wrote: No, I want to use Solaris 10 on the Sparc platform. I've read a lot of reports and tests/benchmarks conducted that sow Solaris 10 actually performing better than all other Linux based Distros...not sure if that's been the experience of others in the group. I really want to know if someone has a high performance TTS based service running in a production environment. What product are they using as their core engine, does it handle and has available many different languages and can one build these independently of the telephony platform being used so I could use maybe Asterisk running on Solaris 10 and a cluster/farm of TTS servers for TTS processing. Thanks RR On Thu, Nov 11, 2010 at 10:21 PM, Luis Morales faston...@gmail.com wrote: You try install debian in your sparc platform ? On Thu, Nov 11, 2010 at 8:52 PM, RR ranjt...@gmail.com wrote: Hello Group, I have been going through all the chit-chat about TTS and the various engines available to integrate with Asterisk incl. flite/festival, espeak, Nuance etc but I am wondering if anyone's tried any or all of these to compile on a Sparc based Solaris platform? If not, then what is the best way for me to accomplish a production environment TTS service when most of my servers or the core of the servers are Sparc based Solaris platforms. I found a group that seems to have done a fair bit of work on compiling Asterisk on Solaris, but I'm wondering if it'll be possible for me to have my core platform running Asterisk on Sparc Solaris and a set of Linux servers serving as a TTS cluster to which the calls can be thrown to for processing and then have them be played back to the user. Any ideas/advice? Thanks RR -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com-- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- - Luis Morales Consultor de Tecnologia Cel: +58(0412)2352745 OpenID: http://lmorales.myopenid.com/ Twitter: @magnadata Linux User ID : 470650 - Empieza por hacer lo necesario, luego lo que es posible... y de pronto estarás haciendo lo imposible Leonardo Da'Vinci - -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com-- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello
