I'm trying to configure Linksys 3102 for a short splash ring when someone leaves a message. in my sip.conf I have mailbox=number I have can see a visual indicator (light blinking on the phone) but there is no short splash ring) Linksys setting: Regional - tab Ring and Call Waiting Tone Spec Ring Waveform: Trapezoids Ring Frequency: 25 VMWI Refresh Intvl: 30 (was "0" I changed to 30 makes no difference) User 1 - tab Ring Settings: VMWI Ring Splash Len: 0.5 Did I miss any settings? Why isn't it working? -- #Joseph
Col Ferguson
2008-Aug-22 05:12 UTC
[asterisk-users] Problem with modem data calls and xorcom astribanks
Hello all, I have a system at a motel that is mostly analog phones with 2 32 port astribanks. I am having problems getting a modem data call to connect. There are many travelling salesmen that require this functionality to work to dial direct into their company systems. I am using Asterisk 1.4.18.1, and Zaptel 1.4.9.2 and freePBX 2.4.0.1 and Oslec echo can. I have now the simplest dialplan I can come up with and get a 4800 connection about 1 in 10 times. This should bypass any smarts that freePBX is adding in. The dialplan is [outbound-allroutes-custom] exten => 791,1,Dial(Zap/69/ww0198308888,300) exten => 791,n,Hangup In Hyperterminal I do atdt791 The number dialled is for a large dialup ISP. ww is needed to get a dialtone for the modem. Could this be causing the problem ? The log file shows [Aug 22 13:17:32] DEBUG[748] chan_zap.c: Deferring dialing... [Aug 22 13:17:32] VERBOSE[748] logger.c: -- Called 69/ww0198308888 [Aug 22 13:17:36] VERBOSE[748] logger.c: -- Zap/69-1 answered Zap/67-1 [Aug 22 13:17:36] DEBUG[748] chan_zap.c: Took Zap/67-1 off hook [Aug 22 13:17:36] DEBUG[748] chan_zap.c: master: 67, slave: 69, nothingok: 0 [Aug 22 13:17:36] DEBUG[748] chan_zap.c: Stopping tones on 67/0 talking to 69/0 [Aug 22 13:17:36] DEBUG[748] chan_zap.c: Stopping tones on 69/0 talking to 67/0 [Aug 22 13:17:36] DEBUG[748] chan_zap.c: Making 69 slave to master 67 at 0 [Aug 22 13:17:36] DEBUG[748] chan_zap.c: Added 18 to conference 9/67 [Aug 22 13:17:36] DEBUG[748] chan_zap.c: Added 76 to conference 9/69 [Aug 22 13:17:36] VERBOSE[748] logger.c: -- Native bridging Zap/67-1 and Zap/69-1 [Aug 22 13:20:00] DEBUG[748] chan_zap.c: Unlinking slave 69 from 67 [Aug 22 13:20:00] DEBUG[748] chan_zap.c: Removed 18 from conference 9/67 [Aug 22 13:20:00] DEBUG[748] chan_zap.c: Removed 76 from conference 9/69 [Aug 22 13:20:00] VERBOSE[748] logger.c: -- Hungup 'Zap/69-1' Does anyone know if there is some type of native echo canceller in the astribanks that could be affecting this ? Or anything else I could try ? Looking at /proc/oslec/info shows that oslec is not being used at the time. If I have the modem connected directly into the phone line, and completely bypass the astribank, I get a 50666 connection every time. Any suggestions gratefully accepted. Thanks, Col
On 08/21/08 21:10, Joseph wrote:>I'm trying to configure Linksys 3102 for a short splash ring when someone leaves a message. >in my sip.conf I have >mailbox=number > >I have can see a visual indicator (light blinking on the phone) but there is no short splash ring) > >Linksys setting: > >Regional - tab >Ring and Call Waiting Tone Spec >Ring Waveform: Trapezoids Ring Frequency: 25 >VMWI Refresh Intvl: 30 (was "0" I changed to 30 makes no difference) > >User 1 - tab > >Ring Settings: >VMWI Ring Splash Len: 0.5 > >Did I miss any settings? Why isn't it working?It seems to me this feature is sync with Line 1 "Register Expires:" under: Proxy and Registration The default setting is 3600 so it means the phone will get a short ring every hour. If I want to ring the phone every 30sec I need to set "Register Expires: 30" So I don't understand, what is the point of setting timer on: "VMWI Refresh Intvl:" since it doesn't get into effect until "Register Expires" I'm confused by this logic. - #Joseph
Col Ferguson
2008-Aug-22 09:22 UTC
[asterisk-users] Problem with modem data calls and xorcom astribanks
Hello Tzafrir, Yes the trunk is an FXO port in the astribank One astribank is 32 FXS ports, and one is 24 FXS and 8 FXO ports. Just in case it makes a difference, the testing I am doing is with the modem plugged in to the same astribank as the FXO ports. Zap/69 is an FXO port and Zap/67 is an FXS port Thanks, Col ----- Original Message ----- From: "Tzafrir Cohen" <tzafrir.cohen at xorcom.com> To: <asterisk-users at lists.digium.com> Sent: Friday, August 22, 2008 6:56 PM Subject: Re: [asterisk-users] Problem with modem data calls and xorcom astribanks> On Fri, Aug 22, 2008 at 03:12:41PM +1000, Col Ferguson wrote: > > Hello all, > > I have a system at a motel that is mostly analog phones with 2 32 port > > astribanks. > > What exactly is the trunk? FXO ports in the astribank? > > > > > I am having problems getting a modem data call to connect. > > There are many travelling salesmen that require this functionality towork> > to dial direct into their company systems. > > > > I am using Asterisk 1.4.18.1, and Zaptel 1.4.9.2 and freePBX 2.4.0.1 and > > Oslec echo can. > > > > I have now the simplest dialplan I can come up with and get a 4800 > > connection about 1 in 10 times. This should bypass any smarts thatfreePBX> > is adding in. > > > > The dialplan is > > [outbound-allroutes-custom] > > exten => 791,1,Dial(Zap/69/ww0198308888,300) > > exten => 791,n,Hangup > > > > In Hyperterminal I do > > atdt791 > > > > The number dialled is for a large dialup ISP. > > ww is needed to get a dialtone for the modem. Could this be causing the > > problem ? > > > > The log file shows > > [Aug 22 13:17:32] DEBUG[748] chan_zap.c: Deferring dialing... > > [Aug 22 13:17:32] VERBOSE[748] logger.c: -- Called 69/ww0198308888 > > [Aug 22 13:17:36] VERBOSE[748] logger.c: -- Zap/69-1 answeredZap/67-1> > [Aug 22 13:17:36] DEBUG[748] chan_zap.c: Took Zap/67-1 off hook > > [Aug 22 13:17:36] DEBUG[748] chan_zap.c: master: 67, slave: 69,nothingok: 0> > [Aug 22 13:17:36] DEBUG[748] chan_zap.c: Stopping tones on 67/0 talkingto> > 69/0 > > [Aug 22 13:17:36] DEBUG[748] chan_zap.c: Stopping tones on 69/0 talkingto> > 67/0 > > [Aug 22 13:17:36] DEBUG[748] chan_zap.c: Making 69 slave to master 67 at0> > [Aug 22 13:17:36] DEBUG[748] chan_zap.c: Added 18 to conference 9/67 > > [Aug 22 13:17:36] DEBUG[748] chan_zap.c: Added 76 to conference 9/69 > > [Aug 22 13:17:36] VERBOSE[748] logger.c: -- Native bridging Zap/67-1and> > Zap/69-1 > > [Aug 22 13:20:00] DEBUG[748] chan_zap.c: Unlinking slave 69 from 67 > > [Aug 22 13:20:00] DEBUG[748] chan_zap.c: Removed 18 from conference 9/67 > > [Aug 22 13:20:00] DEBUG[748] chan_zap.c: Removed 76 from conference 9/69 > > [Aug 22 13:20:00] VERBOSE[748] logger.c: -- Hungup 'Zap/69-1' > > > > Does anyone know if there is some type of native echo canceller in the > > astribanks that could be affecting this ? Or anything else I could try ? > > Looking at /proc/oslec/info shows that oslec is not being used at thetime.> > > > If I have the modem connected directly into the phone line, andcompletely> > bypass the astribank, I get a 50666 connection every time. > > > > Any suggestions gratefully accepted. > > > > Thanks, > > Col > > > > > > _______________________________________________ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > > Register Now: http://www.astricon.net > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > Tzafrir Cohen > icq#16849755 jabber:tzafrir.cohen at xorcom.com > +972-50-7952406 mailto:tzafrir.cohen at xorcom.com > http://www.xorcom.com iax:guest at local.xorcom.com/tzafrir > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users