Re: [asterisk-users] Working softphone for poket PC
Hi try put Speaq speaQ is a VoIP softphone which runs on either Windows Mobile 5.0 or Sharp Zaurus Linux. It can be used to make and record Internet phone calls using any SIP compliant Internet Phone Server. The free Beta Trial Version which can be downloaded from this page, lets you record phone calls, provides full call logging, DTMF and automatically integrates your speaQ phone contacts with the rest of your PDA's address book. Cheers , Giridhar On 5/23/07, Cosmin Prund [EMAIL PROTECTED] wrote: This is my SJphone story, this is why I removed it: I installed SJphone without too much trouble, I found a voip-info article on configuring it and tried configuring it. Apparently I failed to configure it properly since it did not attempt to register to my asterisk server (in fact, selecting the asterisk profile would do nothing, it would simply jump right back to the pc-to-pc sip profile). So I tried fixing the configuration - failed to that because the Options menu option failed to work! Every single other option would work, but NOT that one! So I uninstalled it :-) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Remco Post Sent: Tuesday, May 22, 2007 11:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Working softphone for poket PC Cosmin Prund wrote: Googling arround I found a number of pocket pc softphones. Of those I was only able to install SJ-something (removed it). SJphone, and why did you remove it? Is there one (pocket pc softphone) that works? SJphone ;-) At least I've made some successful calls using sjphone Thanks, Cosmin Prund___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Remco Post I didn't write all this code, and I can't even pretend that all of it makes sense. -- Glen Hattrup ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Working softphone for poket PC
Oops here is the link http://qtechinc.com/speaq_download.htm --Giridhar Bandi On 5/23/07, ram [EMAIL PROTECTED] wrote: On 5/23/07, Philipp von Klitzing [EMAIL PROTECTED] wrote: Hi! Googling arround I found a number of pocket pc softphones. Of those I was only able to install SJ-something (removed it). Is there one (pocket pc softphone) that works? Windows Mobile 6 comes with a SIP client, however on my HTC device I still need to use the speaker phone or a headset, the GSM phone speaker won't do: http://thinkabdul.com/2007/04/25/sip-config-loader-free-utility-to- automatically-configure-load-_setupxml-file-for-sip-voip-on-windows- mobile-60-device/ Other clients that I haven't tested yet (apart from SJphone - how do you register, I only manged to do URL dialing?): * Express Talk (free, http://www.nch.com.au/talk/ptalksetup.exe) * Kapanga (beta?) * voipsurfer (IAX, not free) * ppciax (IAX) * eScSoftphone (IAX, Demo available, http://www.electronicscience.com/ ) * agephone * gphone * x-pda * iFon (SIP, H.323, Video, Messaging, www.voip-info.org/wiki/view/iFon) HI any softphone for my sony erricson p990i SE says that its got SIP support but i dont see their releases or does ny one have source codes, for UIQ3 ram ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IVR sounds not on certain inbound route
I doubt this would be a codec issue . can you tell us what is the codec used on the sip and iax --Giridhar BandiOn 5/29/06, MC [EMAIL PROTECTED] wrote:Got 1 issue I can't seem to knock out of this particular box. The IVR works fine on the zap channels and the incoming SIP routes. Butcoming in via the IAX2 route leaves me with a silent phone.The prompts all work still letting me navigate the menu. But just can't hear anything.This is with[EMAIL PROTECTED] 2.8 (Asterisk 1.2.7.1, with FreePBX 2.1.0 also installed)Any thoughts on where to start looking to solve this? Console shows itall executing fine. ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IVR sounds not on certain inbound route
please check if you are able to hear the sounds using alaw on IAX i had some problem listening to the sounds using G729 on sip client --Giridhar BandiOn 5/29/06, MC [EMAIL PROTECTED] wrote:Coming in on the IAX route is G729. On the SIP lines it is alaw.Giridhar Reddy Bandi wrote: I doubt this would be a codec issue . can you tell us what is the codec used on the sip and iax --Giridhar Bandi On 5/29/06, *MC* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Got 1 issue I can't seem to knock out of this particular box. The IVR works fine on the zap channels and the incoming SIP routes. But coming in via the IAX2 route leaves me with a silent phone. The prompts all work still letting me navigate the menu. But just can't hear anything. This is with[EMAIL PROTECTED] 2.8 (Asterisk 1.2.7.1 http://1.2.7.1, with FreePBX 2.1.0 also installed) Any thoughts on where to start looking to solve this? Console shows it all executing fine. ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] features.conf *1 Call Recording
did you include automon = *1 in your features.conf ?? it should be somthing like this [featuremap]automon = *1 --Giridhar Bandi On 5/12/06, Dave Morrow [EMAIL PROTECTED] wrote: Thanks for the response.How would I change the DTMF transfer mode?David MorrowTechnical Systems LeadAutodata Solutions Company[EMAIL PROTECTED] http://www.autodatasolutions.comTel: (519) 963-3020Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of AutodataSolutions. This email and any files transmitted with it are confidentialand intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please deletethis message and notify the Autodata system administrator at[EMAIL PROTECTED] mailto: [EMAIL PROTECTED]-Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of FabioSent: Thursday, May 11, 2006 6:55 PMTo: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] features.conf *1 Call Recordingif you ar using SIP clients, try changing DTMF transfer mode.For test use sip debugon your * console, then place a call and watch the output. In INFO or rfc2833 mode you must see the codes like SIP messages. If you are usinginband transfer mode (DTMF codes aretransferred like sounds) you don'tsee the codes.Also, try adjusting featuredigittimeout in features.conf :[general]featuredigittimeout = 2000 ; 2 secondsbecause the default 500ms is a very short time.Fabio Olaechea3Tech SRLCalle 48 Nro 632, Of. 67.La Plata, CP B1900AMZBuenos Aires, Argentina. Tel. +54 221 445 0244 Ext. 301Fax. +54 221 445 0245www.trestech.com.ar-Mensaje original-De: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED]]En nombre de Dave MorrowEnviado el: Miercoles, 10 de Mayo de 2006 02:48 p.m.Para: Asterisk Users Mailing List - Non-Commercial DiscussionAsunto: RE: [Asterisk-Users] features.conf *1 Call RecordingOK. You lost me.David MorrowTechnical Systems LeadAutodata Solutions Company [EMAIL PROTECTED]http://www.autodatasolutions.comTel: (519) 963-3020Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attachedmaterial is the Confidential and Proprietary Information of AutodataSolutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom theyare addressed. If you have received this email in error please deletethis message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]-Original Message-From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED]] On Behalf Of AlejandroVargasSent: Wednesday, May 10, 2006 10:29 AM To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] features.conf *1 Call Recording2006/5/10, Dave Morrow [EMAIL PROTECTED] : I am attempting to setup Asterisk to allow me to press *1 while in a call to use automon to record the call but have had absolutely no success.Is there a trick to this?May be a problem with the way you are sending the dialtones. Try sending as data.--Alejandro Vargas___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] features.conf *1 Call Recording
hi Dave i get the following log on *CLI -- Attempting native bridge of SIP/200-39f4 and SIP/204-2ce4 -- Playing 'beep' (language 'en') -- User hit '*1' to record call. filename: wav|auto-1147452537-200-204|m -- Playing 'beep' (language 'en') -- User hit '*1' to stop recording call. -- Attempting native bridge of SIP/200-39f4 and SIP/204-2ce4what are you using as SIP client ( imean softphone/ analog phone + ATA / IPphone ) ? if you are using a softphone and that doesnot have a dtmf signaling then asterisk will not be able to recognize that you are pressing.--Giridhar Bandi On 5/12/06, Dave Morrow [EMAIL PROTECTED] wrote: It's quite strange. When I press *1 I do not hear a tone indicated that it's even trying to record. David Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodatasolutions.com Tel: (519) 963-3020 Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Dave MorrowSent: Friday, May 12, 2006 8:39 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] features.conf *1 Call Recording Yes. I did. David Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodatasolutions.com Tel: (519) 963-3020 Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Giridhar Reddy BandiSent: Friday, May 12, 2006 3:41 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] features.conf *1 Call Recording did you include automon = *1 in your features.conf ?? it should be somthing like this [featuremap]automon = *1 --Giridhar Bandi On 5/12/06, Dave Morrow [EMAIL PROTECTED] wrote: Thanks for the response.How would I change the DTMF transfer mode?David MorrowTechnical Systems LeadAutodata Solutions Company[EMAIL PROTECTED] http://www.autodatasolutions.comTel: (519) 963-3020Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of AutodataSolutions. This email and any files transmitted with it are confidentialand intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please deletethis message and notify the Autodata system administrator at[EMAIL PROTECTED] mailto: [EMAIL PROTECTED]-Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of FabioSent: Thursday, May 11, 2006 6:55 PMTo: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] features.conf *1 Call Recordingif you ar using SIP clients, try changing DTMF transfer mode.For test use sip debugon your * console, then place a call and watch the output. In INFO or rfc2833 mode you must see the codes like SIP messages. If you are usinginband transfer mode (DTMF codes aretransferred like sounds) you don'tsee the codes.Also, try adjusting featuredigittimeout in features.conf :[general]featuredigittimeout = 2000 ; 2 secondsbecause the default 500ms is a very short time.Fabio Olaechea3Tech SRLCalle 48 Nro 632, Of. 67.La Plata, CP B1900AMZBuenos Aires, Argentina. Tel. +54 221 445 0244 Ext. 301Fax. +54 221 445 0245www.trestech.com.ar-Mensaje original-De: [EMAIL PROTECTED][mailto: [EMAIL PROTECTED]]En nombre de Dave MorrowEnviado el: Miercoles, 10 de Mayo de 2006 02:48 p.m.Para: Asterisk Users Mailing List - Non-Commercial DiscussionAsunto: RE: [Asterisk-Users] features.conf *1 Call RecordingOK. You lost me.David MorrowTechnical Systems LeadAutodata Solutions Company [EMAIL PROTECTED] http://www.autodatasolutions.comTel: (519) 963-3020Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attachedmaterial is the Confidential and
Re: [Asterisk-Users] IVR: playing multiple streams simultaneously?
hi i don't know if we can do that . but i guess we can use audacity .. to mix both the files and get what you want.-Giridhar BandiOn 4/18/06, Herchi Silviu [EMAIL PROTECTED] wrote: Hi all, I'm setting up an IVR using Asterisk. Is there a way to have two streams played to the caller at the same time: for instance, one constant flow of background music, and the IVR contents at the same time? I've looked for solutions using (E)AGI and other things but nothing seems to work. Googling around and reading the list has not been helpful either... Thanks for your help, Silviu ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail problem
Hi Daniel can you give us more information so that it would be easy to debug.like voice mail configuration etc Thanks,GIridhar Bandi.On 4/18/06, Daniel Korndorfer [EMAIL PROTECTED] wrote: Hi,when I call the voicemail app, it starts and die suddenly. Has anyonealready had this problem?Log:app.c:644 ast_play_and_record: No audio available on SIP/-6fca??-- User hung upTks, D.K.___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk service crashes
Hi Billy Try using safe_asterisk and see . safe_asterisk be useful if you fear asterisk may crash.--Giridhar BandiOn 4/19/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: List, The past few days the asterisk service on my server has crashed several times. I have had it running for months and have made no changes to it. When it crashes, I am unable to make calls or gain access to the CLI. The service has been stopped. If I try to start it again (service asterisk start), it will start and run for a few seconds then crash again. After a reboot, it will run successfully for several hours before doing it again. Here is a ps aux of the services while the server is crashed. Does anyone see any service that would have a conflict with the asterisk service? FYI, the only cron I have running is a reboot scheduled once a week. USER PID %CPU %MEM VSZ RSS TTY STAT START TIME COMMAND root 1 0.0 0.0 3096 656 ? S 17:39 0:00 init [3] root 2 0.0 0.0 0 0 ? S 17:39 0:00 [migration/0] root 3 0.0 0.0 0 0 ? SN 17:39 0:00 [ksoftirqd/0] root 4 0.0 0.0 0 0 ? S 17:39 0:00 [migration/1] root 5 0.0 0.0 0 0 ? SN 17:39 0:00 [ksoftirqd/1] root 6 0.0 0.0 0 0 ? S 17:39 0:00 [migration/2] root 7 0.0 0.0 0 0 ? SN 17:39 0:00 [ksoftirqd/2] root 8 0.0 0.0 0 0 ? S 17:39 0:00 [migration/3] root 9 0.0 0.0 0 0 ? SN 17:39 0:00 [ksoftirqd/3] root 10 0.0 0.0 0 0 ? S 17:39 0:00 [events/0] root 11 0.0 0.0 0 0 ? S 17:39 0:00 [events/1] root 12 0.0 0.0 0 0 ? S 17:39 0:00 [events/2] root 13 0.0 0.0 0 0 ? S 17:39 0:00 [events/3] root 14 0.0 0.0 0 0 ? S 17:39 0:00 [khelper] root 15 0.0 0.0 0 0 ? S 17:39 0:00 [kacpid] root 47 0.0 0.0 0 0 ? S 17:39 0:00 [kblockd/0] root 48 0.0 0.0 0 0 ? S 17:39 0:00 [kblockd/1] root 49 0.0 0.0 0 0 ? S 17:39 0:00 [kblockd/2] root 50 0.0 0.0 0 0 ? S 17:39 0:00 [kblockd/3] root 60 0.0 0.0 0 0 ? S 17:39 0:00 [pdflush] root 61 0.0 0.0 0 0 ? S 17:39 0:00 [pdflush] root 63 0.0 0.0 0 0 ? S 17:39 0:00 [aio/0] root 64 0.0 0.0 0 0 ? S 17:39 0:00 [aio/1] root 65 0.0 0.0 0 0 ? S 17:39 0:00 [aio/2] root 66 0.0 0.0 0 0 ? S 17:39 0:00 [aio/3] root 51 0.0 0.0 0 0 ? S 17:39 0:00 [khubd] root 62 0.0 0.0 0 0 ? S 17:39 0:00 [kswapd0] root 139 0.0 0.0 0 0 ? S 17:39 0:00 [kseriod] root 204 0.0 0.0 0 0 ? S 17:39 0:00 [scsi_eh_0] root 205 0.0 0.0 0 0 ? S 17:39 0:00 [aacraid] root 217 0.0 0.0 0 0 ? S 17:39 0:00 [kmirrord] root 218 0.0 0.0 0 0 ? S 17:39 0:00 [kmir_mon] root 226 0.0 0.0 0 0 ? S 17:39 0:00 [kjournald] root 1092 0.0 0.0 2312 556 ? Ss 17:39 0:00 udevd root 1125 0.0 0.0 0 0 ? S 17:39 0:00 [shpchpd_event] root 1377 0.0 0.0 0 0 ? S 17:39 0:00 [kauditd] root 1424 0.0 0.0 0 0 ? S 17:39 0:00 [kjournald] root 1962 0.0 0.0 2612 668 ? Ss 17:40 0:00 syslogd -m 0 root 1966 0.0 0.0 3532 532 ? Ss 17:40 0:00 klogd -x root 1976 0.0 0.0 2128 540 ? Ss 17:40 0:00 irqbalance rpc 1993 0.0 0.0 3432 644 ? Ss 17:40 0:00 portmap root 2012 0.0 0.0 1800 912 ? Ss 17:40 0:00 rpc.statd root 2038 0.0 0.0 5156 1140 ? Ss 17:40 0:00 rpc.idmapd root 2115 0.0 0.0 2092 648 ? Ss 17:40 0:00 /usr/sbin/acpid root 2124 0.0 0.1 8264 2188 ? Ss 17:40 0:00 cupsd root 2165 0.0 0.0 5120 1724 ? Ss 17:40 0:00 /usr/sbin/sshd root 2188 0.0 0.0 3152 896 ? Ss 17:40 0:00 xinetd -stayalive root 2295 0.0 0.1 9300 3084 ? Ss 17:40 0:00 sendmail: accepti smmsp 2303 0.0 0.1 7252 2668 ? Ss 17:40 0:00 sendmail: Queue r root 2316 0.0 0.0 2984 620 ? Ss 17:40 0:00 gpm -m /dev/input root 2329 0.0 0.0 5412 1176 ? Ss 17:40 0:00 crond xfs 2355 0.0 0.0 3392 1540 ? Ss 17:40 0:00 xfs -droppriv -da root 2373 0.0 0.0 3700 788 ? Ss 17:40 0:00 /usr/sbin/atd dbus 2382 0.0 0.0 3660 1300 ? Ss 17:40 0:00 dbus-daemon-1 --s root 2391 0.0 0.2 8456 5620 ? Ss 17:40 0:00 hald root 2471 0.0 0.2 16544 6000 ? Ss 17:40 0:00 /usr/sbin/httpd - root 2483 0.0 0.0 4700 1068 ? S 17:40 0:00 /usr/sbin/vsftpd root 2555 0.0 0.0 4544 1304 ? S 17:40 0:00 /bin/sh /usr/bin/ asterisk 2575 0.0 0.2 16544 6124 ? S 17:40 0:00 /usr/sbin/httpd - asterisk 2578 0.0 0.2 16544 6128 ? S 17:40 0:00 /usr/sbin/httpd - asterisk 2579 0.0 0.2 16544 6128 ? S 17:40 0:00 /usr/sbin/httpd - asterisk 2581 0.0 0.2 16544 6128 ? S 17:40 0:00 /usr/sbin/httpd - asterisk 2583 0.0 0.2 16544 6128 ? S 17:40 0:00 /usr/sbin/httpd - asterisk 2585 0.0 0.2 16544 6128 ? S 17:40 0:00 /usr/sbin/httpd - asterisk 2586 0.0 0.2 16544 6128 ? S 17:40 0:00 /usr/sbin/httpd - asterisk 2588 0.0 0.2 16544 6128 ? S 17:40 0:00 /usr/sbin/httpd - mysql 2600 0.0 0.9 125808 19424 ? Sl 17:40 0:00 /usr/libexec/mysq root 2697 0.0 0.0 1856 508 tty1 Ss+ 17:40 0:00 /sbin/mingetty tt root 2698 0.0 0.0 2600 508 tty2 Ss+ 17:40 0:00 /sbin/mingetty tt root 2699 0.0 0.0 1856 508 tty3 Ss+ 17:40 0:00 /sbin/mingetty tt root 2700 0.0 0.0 3184 508 tty4 Ss+ 17:40 0:00 /sbin/mingetty tt root 2701 0.0 0.0 2352 516 tty5 Ss+ 17:40 0:00 /sbin/mingetty tt root 2702 0.0 0.0 2488 508 tty6 Ss+ 17:40 0:00 /sbin/mingetty tt root 3754 0.0 0.1 6984 2244 ? Ss 17:56 0:00 sshd: bpiper [pri bpiper 3768 0.0 0.1 6984 2320 ? S 17:56 0:00
[Asterisk-Users] polycom unable to start recoding
Hi i have a Polycom Soundstation premier Basic Conference Telephone which is connected to Linksys pap2 boxi am unable to start recording using *1 . but with a normal analog phone connected to linksys pap2 i am able to start and stop recording .. i tried changing the DTMF setting but no use . can some one tell me where i have gone wrong ThanksGiridhar Bandi ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk IVR / Scalability
Hi i am looking for a good ivr system for my company. these are my question are there any good ivr's that can be easily integrated with asterisk ? and are there any large scale deployment of asterisk to date ? thanks Giridhar Bandi ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk behind NAT
IT works fine behind firewall .enable NAT in sip.conf and it works fine.Giridhar BandiOn 4/6/06, Joao Pereira [EMAIL PROTECTED] wrote:Hello to allCan we put Asterisk in a company that has an ADSL connection with just one public IP address? Because with just one public IP, Asterisk musthave a private (NATed) IP... but the idea is to make him dial other SIPdomains.Can Asterisk work behing NAT, and still route calls to the Internet? And he can still receive calls from the Internet?ThanksJoao Pereira___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk behind NAT
so that means that a sip client can access asterisk server which is behind NAT ( assuming that SIP and RTP ports are properly farwarded ) even is nat=no in sip.conf thanks,Giridhar Bandi. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DID registration status
HII have two sip accounts from two different ITSP's both configured on asterisk server. how can i know if these accounts have been successfully registered ?i generally look at the /var/log/asterisk/full suggest me if there are better way of doing this thanksGiridhar Bandi ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How is Teliax ?
Hi I am looking at purchasing some DID lines from Teliax to install it on my asterisk. i would like to know some feed back on Teliax before i purchase. suggest me if there are better sevice providers. thanks Giridhar Bandi ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How is Teliax ?
thank you all for the feed back. --Giridhar Bandi ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
