[asterisk-users] Issues with dahdi show status output (and check IRQs)
Hi, On a (bogus) system equiped with a Digium HA8 and 8 BRI ports, I'm reading this: # asterisk -rx dahdi show status Description Alarms IRQbpviol CRC4 Fra Codi Options LBO HA8- RED 1090 0 CCS AMI YEL 0 db (CSU)/0-133 feet (DSX-1) HA8- UNCONFI 1090 0 CAS Unk YEL 0 db (CSU)/0-133 feet (DSX-1) HA8- RED 1090 0 CCS AMI YEL 0 db (CSU)/0-133 feet (DSX-1) HA8- RED 1090 0 CCS AMI YEL 0 db (CSU)/0-133 feet (DSX-1) HA8- RED 1090 0 CCS AMI YEL 0 db (CSU)/0-133 feet (DSX-1) HA8- RED 1090 0 CAS Unk YEL 0 db (CSU)/0-133 feet (DSX-1) HA8- RED 1090 0 CAS Unk YEL 0 db (CSU)/0-133 feet (DSX-1) HA8- RED 1090 0 CAS Unk YEL 0 db (CSU)/0-133 feet (DSX-1) (In this case, ports 2, 6, 7 and 8 are unconfigured). When I'm reading this output, I often can see a different IRQ value. On another (functional) system, with an HA8 and 4 BRI ports, I can read an IRQ value which is constant. But the displayed value is both different from the one I can read with lpsci -v and lspci -bv. 1. Where does this IRQ value from dahdi show status come from ? How can I check this value with linux tools such as lspci ? 2. Is it normal to see this IRQ changing from time to time ? 3. From dmesg | grep IRQ output: [ 13.565786] wctdm24xxp :04:06.0: PCI INT A - GSI 20 (level, low) - IRQ 20 [ 13.565823] IRQ 20/wctdm24xxp0: IRQF_DISABLED is not guaranteed on shared IRQs What does the last line mean ? I read it as implying my Digium HA8 board is sharing the IRQ 20 but cat /proc/interrupts and lspci -v do not confirm this. Regards -- _ -- 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] How to make app_meetme enable
Install DAHDI then !!? On Thu, Dec 8, 2011 at 12:46 PM, Durgesh Mishra durgesh.mis...@rancoretech.com wrote: In make menuselect =application=XXX app_meetme . I am doing confrence call using sip softphone. I checked It Depends on: dahdi(E) . How I can do app_meetme enable? Thanks -- _ -- 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] ISDN PRI configuration
Il 07/12/2011 23.45, Vieri ha scritto: As far as I know, E1 usually use 16 as D channel. Anyway, I tried as you suggested and set 1 as the D channel and 2-31 as B channels. In the asterisk log I got these messages: chan_dahdi.c: Channel 16 is reserved for D-channel. chan_dahdi.c: Unable to register channel '2-31' So doesn't this actually tell me that I should keep using 16 as the D channel? I your line is provisioned as NET5 (ETSI/EuroISDN) you should use channel 16 as D channel, no one else. As suggested from Kevin check your cabling using a loopback, if you don't have one make it using an rj45 socket simply wiring pin 1 to pin 4 ad pin 2 to pin 5. Also check you telco network termination, a standard one provide TX on pins 4/5 and RX on pins 1/2 but sometimes this isn't true (e.g. E1 gateways). -- _ -- 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 PRI configuration
Il 07/12/2011 23.45, Vieri ha scritto: As far as I know, E1 usually use 16 as D channel. Anyway, I tried as you suggested and set 1 as the D channel and 2-31 as B channels. In the asterisk log I got these messages: chan_dahdi.c: Channel 16 is reserved for D-channel. chan_dahdi.c: Unable to register channel '2-31' So doesn't this actually tell me that I should keep using 16 as the D channel? I your line is provisioned as NET5 (ETSI/EuroISDN) you should use channel 16 as D channel, no one else. As suggested from Kevin check your cabling using a loopback, if you don't have one make it using an rj45 socket simply wiring pin 1 to pin 4 ad pin 2 to pin 5. Also check you telco network termination, a standard one provide TX on pins 4/5 and RX on pins 1/2 but sometimes this isn't true (e.g. E1 gateways). -- _ -- 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 PRI configuration
On 12/07/2011 05:06 PM, Vieri wrote: --- On Wed, 12/7/11, Kevin P. Flemingkpflem...@digium.com wrote: Standard Ethernet cables do not always work for T-1/E-1 spans. They do work a rather large percentage of the time, but not always. Distance between the NIU and the T-1/E-1 card can be a factor, among other things. Many Digium products include span loopback devices, that you can plug a cable into and generate a hard loopback towards the card. If there is one of those on-site, have someone unplug the cable from the NIU and plug it into the loopback device instead; if the span goes green, then at least your cabling/wiring are OK. I bought several Digium products and for the site I'm managing now, there are at least these cards: Wildcard TE120P single-span T1/E1/J1 card (rev 11) A loopback connector should have been included with this card. It does not appear that our web store makes them (the T10i loopback connectors) available as individual items, although some distributors may sell them. ISDN controller: Digium, Inc. Wildcard B410 quad-BRI card and maybe more but right now I don't recall any loopback device although I won't be sure until I go to the site. Can a loopback device be bought seperately? What kind of cable should be used instead of an ethernet cable (I think they used a 5m long cat5 T-568B Straight-Through Ethernet Cable)? As I said before... an Ethernet cable will work nearly all the time, and at a 5m length it's probably fine. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA 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
[asterisk-users] [OT]: Require suggestions - GSM Gateway - Asterisk
Hello, I am looking for ideas and suggestions. I want to use a 16 port GSM gateway as a trunk for outbound/inbound. I will also have two PSTN phone lines coming into the Asterisk server. All calls will go to an IVR on the Asterisk PBX. Outbound from extensions will route to GSM-GW when the dialed number matches a pattern set in the GSM-GW trunk. And accordingly for the PSTN trunks. But that's not the problem. The problem is that this GSM-GW (Eurotech VoIP2All 16 channel gateway) doesn't handle SMS properly. It supports Email-to-SMS (that too, only from Outlook) and when/if a recipient responds to the SMS sent by the GSM-GW.. we cannot just reply to it. We will need to compose a new email to the same number. So, no, no thread based reply system. I need to know if someone, who knows about this problematic setup, especially with SMS has an suggestions or alternatives.Thanks! -- _ -- 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 PRI configuration
In article 4ee0b0e2.3050...@digium.com, Kevin P. Fleming kpflem...@digium.com wrote: As I said before... an Ethernet cable will work nearly all the time, and at a 5m length it's probably fine. Kevin, under what circumstances would an Ethernet cable potentially not work with T1/E1? And in those circumstances, what should be used instead? I'm wondering because I had never realised it was an issue until you said. Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.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
[asterisk-users] libpri / ISDN feature ECT (explicit call transfer)
Hi, since version 1.4.12 the libpri package supports ETSI Explicit Call Transfer feature: http://downloads.asterisk.org/pub/telephony/libpri/ChangeLog-1.4.12 Does anyone know, how to use this feature in the dialplan? I can not find any hints in the asterisk doc. Best regards, -Thorsten- -- _ -- 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] random digits dialing during call
Hi List, When a user is on a call, sometimes they hear digits dialing as if the other end is randomly pressing the keypad with their face...but they aren't. It has happened while I've been on calls also, very odd and annoying. Has anyone come across this on Asterisk before? TIA, Skyler -- _ -- 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] random digits dialing during call
What are you using for hardware? I have experienced SPA2102s supplying a DTMF tone when someone was talking. This was caused by the talker reaching a certain frequency while talking in which the SPA popped out a DTMF tone. I haven't experienced this behavior on polycoms or anything else. --E -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Skyler Sent: Thursday, December 08, 2011 9:27 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] random digits dialing during call Hi List, When a user is on a call, sometimes they hear digits dialing as if the other end is randomly pressing the keypad with their face...but they aren't. It has happened while I've been on calls also, very odd and annoying. Has anyone come across this on Asterisk before? TIA, Skyler -- _ -- 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] libpri / ISDN feature ECT (explicit call transfer)
2011/12/8, Thorsten Göllner t...@ovm-group.com: Hi, since version 1.4.12 the libpri package supports ETSI Explicit Call Transfer feature: http://downloads.asterisk.org/pub/telephony/libpri/ChangeLog-1.4.12 Does anyone know, how to use this feature in the dialplan? I can not find any hints in the asterisk doc. Which Asterisk version are you using ? I think this feature need 1.8 and above (but I'm not very sure) Best regards, -Thorsten- -- _ -- 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] ISDN PRI configuration
I am not Kevin, but I'll tell you that I will not EVER use an Ethernet cable for T1 again. Kevin and I have discussed this at length, and the should work plays out poorly in the real world, or at least mine. I've had it be fine, and had major problems. I can't even find a pattern to it, like length of cable. In a colo cabinet that was direct-connected to a carrier, it worked great for years and then one day...no T1. Just gone. Go down there and put in a real T1 cable, came right up, still up years later. I usually make my own, since they are so expensive to buy. I just connect the four needed pins, pretty easy to do if you're not trying to stuff all eight wires into the connector. On Thu, Dec 8, 2011 at 5:57 AM, Tony Mountifield t...@softins.co.uk wrote: In article 4ee0b0e2.3050...@digium.com, Kevin P. Fleming kpflem...@digium.com wrote: As I said before... an Ethernet cable will work nearly all the time, and at a 5m length it's probably fine. Kevin, under what circumstances would an Ethernet cable potentially not work with T1/E1? And in those circumstances, what should be used instead? I'm wondering because I had never realised it was an issue until you said. Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.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 -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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] random digits dialing during call
Hi, thanks for your reply. We're using PAP2T's. I've just recently found that this is most likely 'talk-off', a common issue with PAP2's. A new term for me, but once I found that it was much easier knowing what to search for. I found a few suggestions on changing dtmf to 'inband' so I've done that now and see how it goes. If it continues we'll just have to switch out the devices for another brand...maybe ht286's? couldn't find any talk-off related issues for those so that might be a good next option. S. On Thu, 2011-12-08 at 10:49 -0500, eherr wrote: What are you using for hardware? I have experienced SPA2102s supplying a DTMF tone when someone was talking. This was caused by the talker reaching a certain frequency while talking in which the SPA popped out a DTMF tone. I haven't experienced this behavior on polycoms or anything else. --E -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Skyler Sent: Thursday, December 08, 2011 9:27 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] random digits dialing during call Hi List, When a user is on a call, sometimes they hear digits dialing as if the other end is randomly pressing the keypad with their face...but they aren't. It has happened while I've been on calls also, very odd and annoying. Has anyone come across this on Asterisk before? TIA, Skyler -- _ -- 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] libpri / ISDN feature ECT (explicit call transfer)
since version 1.4.12 the libpri package supports ETSI Explicit Call Transfer feature: http://downloads.asterisk.org/pub/telephony/libpri/ChangeLog-1.4.12 Does anyone know, how to use this feature in the dialplan? I can not find any hints in the asterisk doc. Which Asterisk version are you using ? I think this feature need 1.8 and above (but I'm not very sure) Yes. You need Asterisk v1.8 and later to use the feature. ECT is initiated and accepted with the ETSI(EuroISDN) switch type. The dialplan is not involved with ECT when calls are transfered because the calls involved are already bridged. ECT is automatically initiated if the chan_dahdi.conf transfer=yes option is set and a call is natively bridged on the same span. ECT is also used to update connected line information. Richard -- _ -- 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 PRI configuration
2011/12/8, Carlos Alvarez car...@televolve.com: I am not Kevin, but I'll tell you that I will not EVER use an Ethernet cable for T1 again. Kevin and I have discussed this at length, and the should work plays out poorly in the real world, or at least mine. I've had it be fine, and had major problems. I can't even find a pattern to it, like length of cable. In a colo cabinet that was direct-connected to a carrier, it worked great for years and then one day...no T1. Just gone. Go down there and put in a real T1 cable, came right up, still up years later. I usually make my own, which type of cable are you then using ? since they are so expensive to buy. I just connect the four needed pins, pretty easy to do if you're not trying to stuff all eight wires into the connector. On Thu, Dec 8, 2011 at 5:57 AM, Tony Mountifield t...@softins.co.uk wrote: In article 4ee0b0e2.3050...@digium.com, Kevin P. Fleming kpflem...@digium.com wrote: As I said before... an Ethernet cable will work nearly all the time, and at a 5m length it's probably fine. Kevin, under what circumstances would an Ethernet cable potentially not work with T1/E1? And in those circumstances, what should be used instead? I'm wondering because I had never realised it was an issue until you said. Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.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 -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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] libpri / ISDN feature ECT (explicit call transfer)
snip if the chan_dahdi.conf transfer=yes option is set and a call is natively bridged on the same span. This is interesting as I didn't know that. What if a call comes in a BRI span in which one B-channel is already used ? Is dahdi still capable to ask Explicit Call Transfer using another span (within the same span set if I may call this like that) ? ECT is also used to update connected line information. Richard -- _ -- 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] Issues with dahdi show status output (and check IRQs)
On Thu, Dec 08, 2011 at 09:26:14AM +0100, Olivier wrote: Hi, On a (bogus) system equiped with a Digium HA8 and 8 BRI ports, I'm reading this: # asterisk -rx dahdi show status Description Alarms IRQbpviol CRC4 Fra Codi Options LBO HA8- RED 1090 0 CCS AMI YEL 0 db (CSU)/0-133 feet (DSX-1) HA8- UNCONFI 1090 0 CAS Unk YEL 0 db (CSU)/0-133 feet (DSX-1) HA8- RED 1090 0 CCS AMI YEL 0 db (CSU)/0-133 feet (DSX-1) HA8- RED 1090 0 CCS AMI YEL 0 db (CSU)/0-133 feet (DSX-1) HA8- RED 1090 0 CCS AMI YEL 0 db (CSU)/0-133 feet (DSX-1) HA8- RED 1090 0 CAS Unk YEL 0 db (CSU)/0-133 feet (DSX-1) HA8- RED 1090 0 CAS Unk YEL 0 db (CSU)/0-133 feet (DSX-1) HA8- RED 1090 0 CAS Unk YEL 0 db (CSU)/0-133 feet (DSX-1) (In this case, ports 2, 6, 7 and 8 are unconfigured). When I'm reading this output, I often can see a different IRQ value. On another (functional) system, with an HA8 and 4 BRI ports, I can read an IRQ value which is constant. But the displayed value is both different from the one I can read with lpsci -v and lspci -bv. 1. Where does this IRQ value from dahdi show status come from ? How can I check this value with linux tools such as lspci ? The number displayed in the IRQ field is actually the IRQ misses, or a count of frames where what was expected from the card didn't match up with what was actually received. This value should be the same as what you would see at the top of /proc/dahdi/span (when it's greater than 0) 2. Is it normal to see this IRQ changing from time to time ? Normally, after things have stabilized, it should remain constant on any of the newer cards that can adjust to system latency. On the Hx8 cards however, there are conditions where it will increase regularly if the card is looking for a new sync source. On BRI links this can happen in countries where the provider tries to take down layer 1 on idling spans. When the card internally switches it's timing source it will drop an in flight packet, and the IRQ miss will bump. 3. From dmesg | grep IRQ output: [ 13.565786] wctdm24xxp :04:06.0: PCI INT A - GSI 20 (level, low) - IRQ 20 [ 13.565823] IRQ 20/wctdm24xxp0: IRQF_DISABLED is not guaranteed on shared IRQs What does the last line mean ? I read it as implying my Digium HA8 board is sharing the IRQ 20 but cat /proc/interrupts and lspci -v do not confirm this. You would no longer see that message with the current release of DAHDI. It's an indication that the driver requested the kernel keep interrupts locked while running it's interrupt handler but the kernel doesn't guarantee that anymore. The current release of the wctdm24xxp driver in DAHDI does not request the kernel keep interrupts locked anymore. In this context shared IRQ means that it's technically possible for the card to share it's interrupt line, which is common for PCI devices. -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA 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] ISDN PRI configuration
A T1 cable according to this spec: http://www.cisco.com/en/US/products/hw/routers/ps214/products_tech_note09186a00801f5d89.shtml Crossing the 1/2 to 4/5 if needed. On Thu, Dec 8, 2011 at 9:37 AM, Olivier oza_4...@yahoo.fr wrote: 2011/12/8, Carlos Alvarez car...@televolve.com: I am not Kevin, but I'll tell you that I will not EVER use an Ethernet cable for T1 again. Kevin and I have discussed this at length, and the should work plays out poorly in the real world, or at least mine. I've had it be fine, and had major problems. I can't even find a pattern to it, like length of cable. In a colo cabinet that was direct-connected to a carrier, it worked great for years and then one day...no T1. Just gone. Go down there and put in a real T1 cable, came right up, still up years later. I usually make my own, which type of cable are you then using ? since they are so expensive to buy. I just connect the four needed pins, pretty easy to do if you're not trying to stuff all eight wires into the connector. On Thu, Dec 8, 2011 at 5:57 AM, Tony Mountifield t...@softins.co.uk wrote: In article 4ee0b0e2.3050...@digium.com, Kevin P. Fleming kpflem...@digium.com wrote: As I said before... an Ethernet cable will work nearly all the time, and at a 5m length it's probably fine. Kevin, under what circumstances would an Ethernet cable potentially not work with T1/E1? And in those circumstances, what should be used instead? I'm wondering because I had never realised it was an issue until you said. Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.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 -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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 -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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] FXO PCI Master Abort (was Re: Help! Logs filling up with errors!)
I am still having issues with the error message Dec 7 14:25:06 servername kernel: FXO PCI Master abort filling up my log files. I've temporarily managed a work around by having the message log emptied every 10 minutes, but this is not a permanent solution. I expanded my google search to simple kernel pci master abort and came across a couple of sites recommending that the BIOS option PnP OS be set to No to solve these problems. Does anyone have any experience with this and think this might actually help? (The problem server is in a remote office and I don't want to make the 2 hour drive until I'm sure I have a solution.) Thanks, Brent -- _ -- 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 PRI configuration
Tony wrote: Kevin P. Fleming kpflem...@digium.com wrote: As I said before... an Ethernet cable will work nearly all the time, and at a 5m length it's probably fine. Kevin, under what circumstances would an Ethernet cable potentially not work with T1/E1? And in those circumstances, what should be used instead? I'm wondering because I had never realised it was an issue until you said. I've never had an issue with using Cat5 cable, but I have run into telco/techs that choose to use a pin out other than 1245, and of course defend it with 'That is our standard way to do it'. So a standard Ethernet cable would fail, but once one end was cut off an replaced with the required pin out it would work fine (but no longer be an Ethernet cable, semantics but important). Dan -- _ -- 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 PRI configuration
2011/12/8, Carlos Alvarez car...@televolve.com: A T1 cable according to this spec: http://www.cisco.com/en/US/products/hw/routers/ps214/products_tech_note09186a00801f5d89.shtml Crossing the 1/2 to 4/5 if needed. In fact I was rather referring to the previous example in which a cable did run OK for years and suddenly stopped to. Obviously, the connector pins were still correctly set. If it stopped to work, then it must come from the electric signals and should explained through cable impedance or things like that. My question was rather how could the replacement cable itself be precisely described (thickness, shield, category, ...) ? -- _ -- 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 PRI configuration
On Thu, Dec 8, 2011 at 9:37 AM, Olivier oza_4...@yahoo.fr wrote: I usually make my own, which type of cable are you then using ? I just realized that I may have not answered the right question. Did you mean what raw cable did I use to make T1 cables? Cat-3 or above is fine. I use whatever I have around, which is typically Cat-5e. Yes, I know that solid conductors aren't meant to be pushed into those connectors, yet my experience is 100% good doing that. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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 PRI configuration
On Thu, Dec 8, 2011 at 10:14 AM, Olivier oza_4...@yahoo.fr wrote: In fact I was rather referring to the previous example in which a cable did run OK for years and suddenly stopped to. My THEORY is that the driver chips on either end were wearing out and no longer able to send or receive as well as they once did. When you run the correct pairs, the wires are twisted together. This is important for a variety of electrical reasons, too lengthy to cover here, but a quick google search will give you a lot of info if you care. If you use an ethernet cable, you are using a pair of wires that is not twisted together, removing the electrical advantage of twisted-pair cable. Obviously, the connector pins were still correctly set. If it stopped to work, then it must come from the electric signals and should explained through cable impedance or things like that. Yes, exactly. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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] libpri / ISDN feature ECT (explicit call transfer)
snip if the chan_dahdi.conf transfer=yes option is set and a call is natively bridged on the same span. This is interesting as I didn't know that. What if a call comes in a BRI span in which one B-channel is already used ? Is dahdi still capable to ask Explicit Call Transfer using another span (within the same span set if I may call this like that) ? ECT is only initiated by Asterisk if the calls are on the same span. So in this case both calls must be on the same BRI. From chan_dahdi.conf: ; For FXS ports (either direct analog or over T1/E1): ; Support flash-hook call transfer (requires three way calling) ; Also enables call parking (overrides the 'canpark' parameter) ; ; For digital ports using ISDN PRI protocols: ; Support switch-side transfer (called 2BCT, RLT or other names) ; This setting must be enabled on both ports involved, and the ; 'facilityenable' setting must also be enabled to allow sending ; the transfer to the ISDN switch, since it sent in a FACILITY ; message. ; NOTE: This should be disabled for NT PTMP mode. Phones cannot ; have tromboned calls pushed down to them. ; transfer=yes Richard -- _ -- 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] Talk detection in meetme
Eyal Mahalal wrote: I create Chat room with MEETME and now I have a problem. I want that the host of the room could identify the participants in the room by their speech, so that if a participant uses language the host could kick him from the room. Is there a way to do it? This is one of the features of the monitor page in Web-MeetMe. The key components are: 1. A web page that refreshes every x seconds. 2. Configuring Asterisk Manager interface to allow connections. 3. Code to connect to the manager interface an list the callers. a. I use PHPAGI in WMM, but there are other libraries to choose from. Dan -- _ -- 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] Issues with dahdi show status output (and check IRQs)
2011/12/8, Shaun Ruffell sruff...@digium.com: On Thu, Dec 08, 2011 at 09:26:14AM +0100, Olivier wrote: Hi, On a (bogus) system equiped with a Digium HA8 and 8 BRI ports, I'm reading this: # asterisk -rx dahdi show status Description Alarms IRQbpviol CRC4 Fra Codi Options LBO HA8- RED 1090 0 CCS AMI YEL 0 db (CSU)/0-133 feet (DSX-1) HA8- UNCONFI 1090 0 CAS Unk YEL 0 db (CSU)/0-133 feet (DSX-1) HA8- RED 1090 0 CCS AMI YEL 0 db (CSU)/0-133 feet (DSX-1) HA8- RED 1090 0 CCS AMI YEL 0 db (CSU)/0-133 feet (DSX-1) HA8- RED 1090 0 CCS AMI YEL 0 db (CSU)/0-133 feet (DSX-1) HA8- RED 1090 0 CAS Unk YEL 0 db (CSU)/0-133 feet (DSX-1) HA8- RED 1090 0 CAS Unk YEL 0 db (CSU)/0-133 feet (DSX-1) HA8- RED 1090 0 CAS Unk YEL 0 db (CSU)/0-133 feet (DSX-1) (In this case, ports 2, 6, 7 and 8 are unconfigured). When I'm reading this output, I often can see a different IRQ value. On another (functional) system, with an HA8 and 4 BRI ports, I can read an IRQ value which is constant. But the displayed value is both different from the one I can read with lpsci -v and lspci -bv. 1. Where does this IRQ value from dahdi show status come from ? How can I check this value with linux tools such as lspci ? The number displayed in the IRQ field is actually the IRQ misses, or a count of frames where what was expected from the card didn't match up with what was actually received. This value should be the same as what you would see at the top of /proc/dahdi/span (when it's greater than 0) OK 2. Is it normal to see this IRQ changing from time to time ? Normally, after things have stabilized, it should remain constant on any of the newer cards that can adjust to system latency. On the Hx8 cards however, there are conditions where it will increase regularly if the card is looking for a new sync source. OK this matches with the fact my /var/log/kern.log file is cluttered (say 25 lines per hour) with lines like this: wctdm24xxp :04:06.0: xhfc_set_sync_src - modpos 0: setting sync to be port -1 I specifically chose this line as this setting sync to be port -1 frightens me a bit as I would rather see a value ranging form 0 to 3 or 1 to 4. On BRI links this can happen in countries where the provider tries to take down layer 1 on idling spans. 1. How can I check this is happening ? Is pri debug, for instance, capable enough to confirm this ? 2. On a more general plan, is taking down layer 1 on idling spans something PBXs are negociating with each other (the public switch trying to take the layer one down, listening to an acknowledge from the private PBX) or is it more brutal than that ? 3. Is this feature worked on ? When the card internally switches it's timing source it will drop an in flight packet, and the IRQ miss will bump. 3. From dmesg | grep IRQ output: [ 13.565786] wctdm24xxp :04:06.0: PCI INT A - GSI 20 (level, low) - IRQ 20 [ 13.565823] IRQ 20/wctdm24xxp0: IRQF_DISABLED is not guaranteed on shared IRQs What does the last line mean ? I read it as implying my Digium HA8 board is sharing the IRQ 20 but cat /proc/interrupts and lspci -v do not confirm this. You would no longer see that message with the current release of DAHDI. Do you mean dahdi 2.6 ? It's an indication that the driver requested the kernel keep interrupts locked while running it's interrupt handler but the kernel doesn't guarantee that anymore. So, is it correct that : A it doesn't necessary imply the IRQ is shared, in this specific case, B that just means the driver is asking for something the kernel do not provide anymore. Then, why can't I read this warning line in every system ? Could this come from the way dahdi is configured (sometimes, dahdi is configured in a way that makes it ask for interrupts locks, and sometimes not) ? The current release of the wctdm24xxp driver in DAHDI does not request the kernel keep interrupts locked anymore. In this context shared IRQ means that it's technically possible but still not recommended ? for the card to share it's interrupt line, which is common for PCI devices. -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA 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 Thank you very much for your detailed answers. --
Re: [asterisk-users] ISDN PRI configuration
Il 08/12/2011 18.17, Carlos Alvarez ha scritto: If you use an ethernet cable, you are using a pair of wires that is not twisted together, removing the electrical advantage of twisted-pair cable. This is wrong, in both T568A or T568B ethernet pins 1/2 and 4/5 runs on a properly twisted pair. Also 120 Ohm impedance is matching the ISDN pri specification. If a straight pri cable is needed then a straight ethernet cable fits the job (not the same for a pri cross cable vs an eth cross cable). -- _ -- 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 PRI configuration
On Thu, Dec 8, 2011 at 10:48 AM, giovanni.v i...@keybits.org wrote: This is wrong, in both T568A or T568B ethernet pins 1/2 and 4/5 runs on a properly twisted pair. Also 120 Ohm impedance is matching the ISDN pri specification. If a straight pri cable is needed then a straight ethernet cable fits the job (not the same for a pri cross cable vs an eth cross cable). It was probably the crossover I was thinking of, which is what I almost always end up needing. I stopped analyzing the situation when I found myself simply replacing them with the right cable and being successful. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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 PRI configuration
Try this instead: http://www.ahk.com/t1_cable.html That cisco link does not specify the cable itself, but only the pin outs. True T1 cable has a foil shield around each pair, also called ABAM cable in the telco world. Ethernet cable is twisted pair without any shielding between pairs. And one shield around all the pairs is not the same as ABAM. Lyle Giese LCR Computer Services, Inc. On 12/08/11 10:53, Carlos Alvarez wrote: A T1 cable according to this spec: http://www.cisco.com/en/US/products/hw/routers/ps214/products_tech_note09186a00801f5d89.shtml Crossing the 1/2 to 4/5 if needed. On Thu, Dec 8, 2011 at 9:37 AM, Olivier oza_4...@yahoo.fr mailto:oza_4...@yahoo.fr wrote: 2011/12/8, Carlos Alvarez car...@televolve.com mailto:car...@televolve.com: I am not Kevin, but I'll tell you that I will not EVER use an Ethernet cable for T1 again. Kevin and I have discussed this at length, and the should work plays out poorly in the real world, or at least mine. I've had it be fine, and had major problems. I can't even find a pattern to it, like length of cable. In a colo cabinet that was direct-connected to a carrier, it worked great for years and then one day...no T1. Just gone. Go down there and put in a real T1 cable, came right up, still up years later. I usually make my own, which type of cable are you then using ? since they are so expensive to buy. I just connect the four needed pins, pretty easy to do if you're not trying to stuff all eight wires into the connector. On Thu, Dec 8, 2011 at 5:57 AM, Tony Mountifield t...@softins.co.uk mailto:t...@softins.co.uk wrote: In article 4ee0b0e2.3050...@digium.com mailto:4ee0b0e2.3050...@digium.com, Kevin P. Fleming kpflem...@digium.com mailto:kpflem...@digium.com wrote: As I said before... an Ethernet cable will work nearly all the time, and at a 5m length it's probably fine. Kevin, under what circumstances would an Ethernet cable potentially not work with T1/E1? And in those circumstances, what should be used instead? I'm wondering because I had never realised it was an issue until you said. Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk mailto:t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org mailto:t...@mountifield.org - http://tony.mountifield.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 -- Carlos Alvarez TelEvolve 602-889-3003 tel:602-889-3003 -- _ -- 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 -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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] ISDN PRI configuration
Interesting: If you cannot obtain T1 specific cable, then use two runs of CAT 5. Use one CAT5 cable for the Transmit (Tx) signal and one CAT5 cable for the Receive (Rx) signal. It is necessary for the Tx and Rx signals to be in separate sheaths to prevent cross talk interference So pins 1 and 2 on one cable and pins 4 and 5 on another. --- On Thu, 12/8/11, Lyle Giese l...@lcrcomputer.net wrote: Try this instead: http://www.ahk.com/t1_cable.html That cisco link does not specify the cable itself, but only the pin outs. True T1 cable has a foil shield around each pair, also called ABAM cable in the telco world. Ethernet cable is twisted pair without any shielding between pairs. And one shield around all the pairs is not the same as ABAM. Lyle Giese LCR Computer Services, Inc. On 12/08/11 10:53, Carlos Alvarez wrote: A T1 cable according to this spec: http://www.cisco.com/en/US/products/hw/routers/ps214/products_tech_note09186a00801f5d89.shtml Crossing the 1/2 to 4/5 if needed. On Thu, Dec 8, 2011 at 9:37 AM, Olivier oza_4...@yahoo.fr mailto:oza_4...@yahoo.fr wrote: 2011/12/8, Carlos Alvarez car...@televolve.com mailto:car...@televolve.com: I am not Kevin, but I'll tell you that I will not EVER use an Ethernet cable for T1 again. Kevin and I have discussed this at length, and the should work plays out poorly in the real world, or at least mine. I've had it be fine, and had major problems. I can't even find a pattern to it, like length of cable. In a colo cabinet that was direct-connected to a carrier, it worked great for years and then one day...no T1. Just gone. Go down there and put in a real T1 cable, came right up, still up years later. I usually make my own, which type of cable are you then using ? since they are so expensive to buy. I just connect the four needed pins, pretty easy to do if you're not trying to stuff all eight wires into the connector. On Thu, Dec 8, 2011 at 5:57 AM, Tony Mountifield t...@softins.co.uk mailto:t...@softins.co.uk wrote: In article 4ee0b0e2.3050...@digium.com mailto:4ee0b0e2.3050...@digium.com, Kevin P. Fleming kpflem...@digium.com mailto:kpflem...@digium.com wrote: As I said before... an Ethernet cable will work nearly all the time, and at a 5m length it's probably fine. Kevin, under what circumstances would an Ethernet cable potentially not work with T1/E1? And in those circumstances, what should be used instead? I'm wondering because I had never realised it was an issue until you said. Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk mailto:t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org mailto:t...@mountifield.org - http://tony.mountifield.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 -- Carlos Alvarez TelEvolve 602-889-3003 tel:602-889-3003 -- _ -- 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 -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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:
[asterisk-users] Sip.conf and extensions.conf configuration for Exchange 2010 U.M.
Hi All, I'm using Exchange as our voicemail system. Everything works fine until the 1 week mark when Exchange changes the port number used, then Asterisk 1.8 seg faults and I have no phones (unless I restart the U.M. service before the 1 week period is up). Since that is a hack, I'm hoping someone can post their working configs that accomodates the port change. The documentation I've seen is still a little unclear to me. I'm not using secured mode, so just using ports 5065/5067. Thanks for your help. Jim -- _ -- 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] Preparing to store vm in database
Hi all,
I'm getting ready to start storing all of my voicemail in a mysql database.
I've already got RT sip and RT voicemailboxes working.
I understand that vm storage only works via odbc. I've read all of the
documentation I could find, including:
http://www.voip-info.org/wiki/view/Asterisk+Voicemail+ODBC+storage
===
But there was this comment, at the bottom of the page:
I tried adding ODBC message storage to a 1.2.5 system already using MySQL for
RealTime... Not a good idea, but using ODBC for both Realtime and msg storage
seems good so far.
Is this still true with Asterisk 1.6.2.9? Or do I need to migrate all of my
RT configuration to ODBC?
===
I also read this comment:
Make sure you load the .WAV file and not the .wav or .gsm or it won't work!
-- is that true? I couldnt get .WAV to work. But when I used a 16bit 8000Hz
file with a .wav extension, it worked fine.
Will I be ok if I just load the .wav file?
===
Also, I see that I'll have to pre-load all of the voicemail messages and
greeting files into the database, like so:
INSERT INTO voicemail (msgnum,dir,mailboxuser,mailboxcontext,recording) VALUES
(-1,'/var/spool/asterisk/voicemail/CONTEXT/USER/busy','CONTEXT','USER',LOAD_FILE('/var/spool/asterisk/voicemail/CONTEXT/USER/busy.WAV'));
I assume this has to be done for every CONTEXT and every USER, and those
values need to be substituted into both the fields, and the directory path
value.
===
Finally, is there an AGI command that will play a .wav file from the database?
TIA,
--
Take care and have fun,
Mike Diehl.
--
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Re: [asterisk-users] Trying to send customer mwi updates
On 12/9/2011 12:55 AM, Mike Diehl wrote: What am I doing wrong? perhaps try: http://jeremy.kister.net/code/asterisk/mwi.pl -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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
