Citando [EMAIL PROTECTED]: > Send Asterisk-Users mailing list submissions to > [email protected] > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users > or, via email, send a message with subject or body 'help' to > [EMAIL PROTECTED] > > You can reach the person managing the list at > [EMAIL PROTECTED] > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Asterisk-Users digest..." > > > Today's Topics: > > 1. Re: append # to dial string (Eric Wieling) > 2. Re: VAD/DTX implementation through zaptel cards (Eric Wieling) > 3. Re: CDR and TDS (Eric Wieling) > 4. RE: Zaptel Compile on a virtual dedicated host. > ([EMAIL PROTECTED]) > 5. Re: TE110P/Hipath3750 - Yellow Alarm (Henry Jensen) > 6. Re: Re: PTSN POTS Differences SOLVED (Robert Keller) > 7. Re: Can you comment on this Qos script? How does one shape > RTP? (Sean Kennedy) > 8. Interface bonding + asterisk (Jesus Mogollon) > 9. Re: Can you comment on this Qos script? How doesone shape > RTP? (Henry) > 10. Re: Can you comment on this Qos script? How does one shape > RTP? (trixter http://www.0xdecafbad.com) > 11. RE: Sangoma A101 + Rhino channelbank (mattf) > 12. Re: Can you comment on this Qos script? How does one shape > RTP? (Andrew Kohlsmith) > 13. Re: TDM400P power supply (Ricardo Peironcely) > 14. Problem with X101P (Yusuf Iqbal) > 15. Re: Can you comment on this Qos script? How does one shape > RTP? (trixter http://www.0xdecafbad.com) > 16. wcfxo problem (Dave Weis) > 17. (no subject) (Robert Webb) > 18. Re: Sipura SPA-841 Phone Review (Doug Millsaps) > 19. Re: From OH323 to SIP or OH323 without gatekeeper (Bruno Hertz) > 20. Re: wcfxo problem (Sahil Gupta) > 21. Re: TDM400P Revision question. (Robert Webb) > 22. Intercom with Aastra 480e? (Bobby Lacey) > 23. Manipulate Asterisk Database from manager? (Matt) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Mon, 11 Apr 2005 08:39:00 -0500 > From: Eric Wieling <[EMAIL PROTECTED]> > Subject: Re: [Asterisk-Users] append # to dial string > To: Asterisk Users Mailing List - Non-Commercial Discussion > <[email protected]> > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset=us-ascii; format=flowed > > John Breeden wrote: > > > Been there, done that - no joy :-) > > > > It appears the modifier only excepts a numeric, anyone know if/how you > > can feed it adecimal/hex for ascii #? > > > > Rich Adamson wrote: > > > >>> Is there anyway to append the '#' symbol to a dial string? - > >>> hex/octal whatever? I'm surprised that I can't find anything > >>> searching the wiki or google. > >>> > >> > >> > >> Try something like this: > >> > >> exten => _9XXXXXXX,1,Dial(Zap/4/${EXTEN}#) > > Then you are doing something wrong. The above syntax is correct. > > -- > Always do right. This will gratify some people and astonish the rest. > Mark Twain > > > ------------------------------ > > Message: 2 > Date: Mon, 11 Apr 2005 08:40:53 -0500 > From: Eric Wieling <[EMAIL PROTECTED]> > Subject: Re: [Asterisk-Users] VAD/DTX implementation through zaptel > cards > To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial > Discussion <[email protected]> > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset=us-ascii; format=flowed > > [EMAIL PROTECTED] wrote: > > > Hi, > > How can i implement VAD/DTX using zaptel with asterisk towards PSTN. > > TDM (PSTN/telcos) do not support VAD. The entire idea of VAD is not > even a valid idea. > > > ------------------------------ > > Message: 3 > Date: Mon, 11 Apr 2005 08:44:00 -0500 > From: Eric Wieling <[EMAIL PROTECTED]> > Subject: Re: [Asterisk-Users] CDR and TDS > To: Asterisk Users Mailing List - Non-Commercial Discussion > <[email protected]> > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset=us-ascii; format=flowed > > David Masure wrote: > > > > > Hi, > > > > I want to use the cdr to record the call log to my Microsoft SQL Server > > using unixodbc and freetds.... > > > > but when I compile, I've got this message.... > > > > Does anyone have the same problem and/or know how to solve it ? > > > Update of /usr/cvsroot/asterisk/doc > In directory mongoose.digium.com:/tmp/cvs-serv24936/doc > > Added Files: > README.tds > Log Message: > Add documentation for TDS noting compilation problem on 0.63+ > > > --- NEW FILE: README.tds --- > PLEASE NOTE > > The cdr_tds module is NOT compatible with version 0.63 of FreeTDS. > > The cdr_tds module is known to work with FreeTDS version 0.62.1; > it should also work with 0.62.2, 0.62.3 and 0.62.4, which are bug > fix releases. > > The cdr_tds module uses the raw "libtds" API of FreeTDS. It appears > that from 0.63 onwards, this is not considered a published API > of FreeTDS and is subject to change without notice. > > Between 0.62.x and 0.63 of FreeTDS, many incompatible changes > have been made to the libtds API. > > For newer versions of FreeTDS, it is recommended that you use the > ODBC driver. > > > > -- > Always do right. This will gratify some people and astonish the rest. > Mark Twain > > > ------------------------------ > > Message: 4 > Date: Mon, 11 Apr 2005 09:45:54 -0400 > From: [EMAIL PROTECTED] > Subject: RE: [Asterisk-Users] Zaptel Compile on a virtual dedicated > host. > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > <[email protected]> > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset="us-ascii" > > It appears to be Virtuozzo.. > > -----Original Message----- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Henry > Sent: Monday, April 11, 2005 9:34 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Zaptel Compile on a virtual dedicated > host. > > Hi, > > Do you happen to know what VPS system your host uses (e.g. UML, > Virtuozzo, VMWare, FreeVPS, etc.)? It could make a lot of difference, as > some platforms will allow changes that others will not. > > -- Henry Owens. > > > On 11/4/05 2:20 pm, "[EMAIL PROTECTED]" <[EMAIL PROTECTED]> > wrote: > Giles thank you for getting back so quickly, "dmesg" doesn't output > anything, but even if it did, I am not sure that I could recompile the > kernel. > > The server I am using is in a virtual dedicated hosting environment, I > do not have access to recompile the kernel, nor can I replace it. The > server prevents me from doing so. I do not have access to the "real" > /boot and don't have access as far as I can tell to the .config for the > kernel source. ("make oldconfig" seems to work) > > After a few more days of tech support, google searches and etc, I have > found that my provider is using kernel 2.24.21.4.0.1.elsmp. Of course, > cat /proc/version doesn't think so!! It thinks I am running Kernel > 2.4.20-021stab022.11.777-enterprise. I am able to use rpmfind to source > the corresponding rpm which installs without incident. The interesting > part is "rpm -qa kernel" doesn't see it :-(. I even tried to "rpm > -rebuilddb" > > Zaptel appears to compile fine, but when I run "modprobe zaptel" I get > the following: > > ----> > /lib/modules/2.4.20-021stab022.11.777-enterprise/misc/zaptel.o: > kernel-module version mismatch > /lib/modules/2.4.20-021stab022.11.777-enterprise/misc/zaptel.o > was compiled for kernel version 2.4.21-4.0.1.EL > while this kernel is version 2.4.20-021stab022.11.777-enterp. > /lib/modules/2.4.20-021stab022.11.777-enterprise/misc/zaptel.o: insmod > /lib/modules/2.4.20-021stab022.11.777-enterprise/misc/zaptel.o failed > /lib/modules/2.4.20-021stab022.11.777-enterprise/misc/zaptel.o: insmod > zaptel failed > <--- > > Is there a way to override zaptel's kernel check or have linux fool it > into thinking the kernel is 2.4.21-4.0.1.EL? > > thanks! > > > -----Original Message----- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] > <mailto:[EMAIL PROTECTED]> On Behalf Of Giles > Coochey > Sent: Wednesday, April 06, 2005 9:01 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: > > > >Anyone have any ideas on where I can find the right kernel source? I > have look at > > rpmfind.net and google'd with no avail! > > You could always download the Vanilla kernel source from > http://www.kernel.org and compile a kernel from source. I tend to always > use the Vanilla source, it's what everything has been tested against and > it tastes better. > > You should probably print out the "dmesg" output to help you configure > the kernel options prior to compilation so that your "hardware" is > correctly detected. > > I would also urge you to use a bootloader such as grub or lilo to ensure > that you can revert to the original kernel should it panic on boot, I > suspect Redhat already uses one of those anyway. > _______________________________________________ > Asterisk-Users mailing list > [email protected] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > _____ > > _______________________________________________ > Asterisk-Users mailing list > [email protected] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.digium.com/pipermail/asterisk- users/attachments/20050411/a0f7f809/attachment-0001.htm > > ------------------------------ > > Message: 5 > Date: Mon, 11 Apr 2005 15:54:44 +0200 > From: Henry Jensen <[EMAIL PROTECTED]> > Subject: Re: [Asterisk-Users] TE110P/Hipath3750 - Yellow Alarm > To: Asterisk Users Mailing List - Non-Commercial Discussion > <[email protected]> > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset=iso-8859-15 > > > On Tue, Apr 05, 2005 at 09:06:33PM +0200, Peter Svensson wrote: > > A yellow alarm means the remote end is sensing some error condition. Try > > looking for an error message at the remote end. It may be as easy as a > > broken cable (where the Hipath does not hear the Asterisk box). > > The problem is, that the TMS2-Card in the HiPath is not activated, > it says, that the line is dead. According to the Siemens-People the > Card should activate itself as soon as a signal reaches the card. > But it appears, that Asterisk sends no signal. > > > This is what the layout looks like: > > Asterisk|TE110P - TMS2|HiPath|TMS2 - PSTN > > > The cable is functional and the wiring is correct. But I'm not sure how I > must configure the TMS2 card. > > Regards, > Henry > > > > > > > ------------------------------ > > Message: 6 > Date: Mon, 11 Apr 2005 07:01:03 -0700 > From: Robert Keller <[EMAIL PROTECTED]> > Subject: Re: [Asterisk-Users] Re: PTSN POTS Differences SOLVED > To: Asterisk Users Mailing List - Non-Commercial Discussion > <[email protected]> > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset="US-ASCII" > > Tony, I don't see "${EXTEN}" anywhere in the [macro-dialout-trunk] context. > Am I missing something? > > Robert Andrew Keller > Ferndale School District #502 > [EMAIL PROTECTED] > 360-383-9228 PH. > 360-383-9218 FAX > "Paving the way for tomorrows genius." > > > From: [EMAIL PROTECTED] (Tony Mountifield) > > Organization: Software Insight Ltd., Winchester, UK > > Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion > > <[email protected]> > > Date: Mon, 11 Apr 2005 07:48:19 +0000 (UTC) > > To: [email protected] > > Subject: [Asterisk-Users] Re: PTSN POTS Differences SOLVED > > > > In article <[EMAIL PROTECTED]>, > > Robert Keller <[EMAIL PROTECTED]> wrote: > >> Thanks Rich, I wasn't sure where to find that context. I found the > outbound > >> context in the extensions_additional.conf and added w's in the following > >> manner: > >> > >> [outrt-001-Out1] > >> include => outrt-001-Out1-custom > >> exten => _1NXXNXXXXXX,1,Macro(dialout-trunk,1,w${EXTEN}) > >> exten => _1NXXNXXXXXX,2,Macro(outisbusy) ; No available circuits > >> exten => _9.,1,Macro(dialout-trunk,1,w${EXTEN:1}) > >> exten => _9.,2,Macro(outisbusy) ; No available circuits > >> exten => _NXXNXXXXXX,1,Macro(dialout-trunk,1,w${EXTEN}) > >> exten => _NXXNXXXXXX,2,Macro(outisbusy) ; No available circuits > >> exten => _NXXXXXX,1,Macro(dialout-trunk,1,w${EXTEN}) > >> exten => _NXXXXXX,2,Macro(outisbusy) ; No available circuits > > > > Couldn't you have just put the w in once, in the Dial command that > > is inside [macro-dialout-trunk] ? > > > > Cheers > > Tony > > -- > > Tony Mountifield > > Work: [EMAIL PROTECTED] - http://www.softins.co.uk > > Play: [EMAIL PROTECTED] - http://tony.mountifield.org > > _______________________________________________ > > Asterisk-Users mailing list > > [email protected] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ------------------------------ > > Message: 7 > Date: Mon, 11 Apr 2005 07:08:54 -0700 > From: Sean Kennedy <[EMAIL PROTECTED]> > Subject: Re: [Asterisk-Users] Can you comment on this Qos script? How > does one shape RTP? > To: [email protected] > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Honestly, the best script I've ever found is the wondershaper script ( > google it ). I tried the correct one posted in this thread, tried > modifying it, but in the end I just used wondershaper. > > Does a great job. My only fear is it doesn't specifically target IAX2 > traffic as high priority, but I can modify it later to do so if needed. > > On a 192 line I am able to get 4 ulaw ( IAX2 ) calls out with no > noticable problems. Along with someone streaming a shoutcast station ( > sigh ). The station broke up, but the calls didn't. > > cmisip wrote: > > >I got this from the voip wiki but the original script didn't seem to > >work right so I fiddled with it a little bit. I am no expert so maybe > >someone can look at it for errors. This is for my cable connection. So > >far asterisk seems to use 1:10 while all other traffic uses 1:102. How > >does one packet shape RTP? > > > >Thanks for any help. > > > > > ------------------------------ > > Message: 8 > Date: Mon, 11 Apr 2005 07:15:30 -0700 > From: Jesus Mogollon <[EMAIL PROTECTED]> > Subject: [Asterisk-Users] Interface bonding + asterisk > To: [email protected] > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset="iso-8859-1" > > Hi all > > I installed asterisk on a dual PIII 700 with two NICs. I then proceeded to > configure both NICs with bonding enable (bonding miimon=100 mode=1). I know > certain features (like load balancing) under a bonded configuration is not > understood by some switches, so I configured it using mode=1 (Failover > only). The problem I'm having is that, sometimes, calls start fine but then > one of the parties loses audio (it could be the caller of the callee who > loses audio, there is no pattern). I was wondering if someone has hit the > same wall as me. There are people using this server right now, so I haven't > tried the no-bonding option as it means downtime. Any help would be > appreciated. > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.digium.com/pipermail/asterisk- users/attachments/20050411/e427fb81/attachment-0001.htm > > ------------------------------ > > Message: 9 > Date: Mon, 11 Apr 2005 15:19:38 +0100 > From: Henry <[EMAIL PROTECTED]> > Subject: Re: [Asterisk-Users] Can you comment on this Qos script? How > doesone shape RTP? > To: Asterisk Users Mailing List - Non-Commercial Discussion > <[email protected]> > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset="US-ASCII" > > I agree that Wondershaper is a great script; prior to using it in an office > where I set up asterisk, there were some major problems with call quality, > but it seems to have helped hugely (the same DSL line is used for both VoIP > and everyday 'net usage for seven people - not ideal, but I didn't set the > budget :-) ). > > If you happen to modify it to to prioritize IAX2, drop me a copy! > > -- Henry Owens. > > > On 11/4/05 3:08 pm, "Sean Kennedy" <[EMAIL PROTECTED]> wrote: > > > Honestly, the best script I've ever found is the wondershaper script ( > > google it ). I tried the correct one posted in this thread, tried > > modifying it, but in the end I just used wondershaper. > > > > Does a great job. My only fear is it doesn't specifically target IAX2 > > traffic as high priority, but I can modify it later to do so if needed. > > > > On a 192 line I am able to get 4 ulaw ( IAX2 ) calls out with no > > noticable problems. Along with someone streaming a shoutcast station ( > > sigh ). The station broke up, but the calls didn't. > > > > cmisip wrote: > > > >> I got this from the voip wiki but the original script didn't seem to > >> work right so I fiddled with it a little bit. I am no expert so maybe > >> someone can look at it for errors. This is for my cable connection. So > >> far asterisk seems to use 1:10 while all other traffic uses 1:102. How > >> does one packet shape RTP? > >> > >> Thanks for any help. > >> > > _______________________________________________ > > Asterisk-Users mailing list > > [email protected] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ------------------------------ > > Message: 10 > Date: Mon, 11 Apr 2005 07:21:43 -0700 > From: "trixter http://www.0xdecafbad.com"; <[EMAIL PROTECTED]> > Subject: Re: [Asterisk-Users] Can you comment on this Qos script? How > does one shape RTP? > To: Asterisk Users Mailing List - Non-Commercial Discussion > <[email protected]> > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset="us-ascii" > > I used the one posted to this list and for a test did a > speedtest.dslreports.com bandwidth test duringa call, no loss in > quality. > > I set ports 10000-11024 to RTP in rtp.conf, I dont need 10k ports for > that as I have few calls being processed. I also added sip to the queue > although that prolly doesnt matter becuase its such a low bandwidth > protocol comparitevly speaking. > > > # udp/5060 is SIP > tc filter add dev $DSLDEV parent 1:0 protocol ip prio 1 u32 match ip > dport 506 > 0 0xffff match ip protocol 17 0xff flowid 1:0 > tc filter add dev $DSLDEV parent 1:0 protocol ip prio 2 u32 match ip > sport 506 > 0 0xffff match ip protocol 17 0xff flowid 1:0 > > # udp/10000-11024 is RTP > tc filter add dev $DSLDEV parent 1:0 protocol ip prio 1 u32 match ip > dport 100 > 00 0xf670 match ip protocol 17 0xff flowid 1:0 > tc filter add dev $DSLDEV parent 1:0 protocol ip prio 2 u32 match ip > sport 100 > 00 0xf670 match ip protocol 17 0xff flowid 1:0 > > > > On Mon, 2005-04-11 at 07:08 -0700, Sean Kennedy wrote: > > Honestly, the best script I've ever found is the wondershaper script ( > > google it ). I tried the correct one posted in this thread, tried > > modifying it, but in the end I just used wondershaper. > > > > Does a great job. My only fear is it doesn't specifically target IAX2 > > traffic as high priority, but I can modify it later to do so if needed. > > > > On a 192 line I am able to get 4 ulaw ( IAX2 ) calls out with no > > noticable problems. Along with someone streaming a shoutcast station ( > > sigh ). The station broke up, but the calls didn't. > > > > cmisip wrote: > > > > >I got this from the voip wiki but the original script didn't seem to > > >work right so I fiddled with it a little bit. I am no expert so maybe > > >someone can look at it for errors. This is for my cable connection. So > > >far asterisk seems to use 1:10 while all other traffic uses 1:102. How > > >does one packet shape RTP? > > > > > >Thanks for any help. > > > > > _______________________________________________ > > Asterisk-Users mailing list > > [email protected] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > -- > Trixter http://www.0xdecafbad.com > -------------- next part -------------- > A non-text attachment was scrubbed... > Name: not available > Type: application/pgp-signature > Size: 189 bytes > Desc: This is a digitally signed message part > Url : > http://lists.digium.com/pipermail/asterisk- users/attachments/20050411/b94c9559/attachment-0001.pgp > > ------------------------------ > > Message: 11 > Date: Mon, 11 Apr 2005 10:34:51 -0400 > From: mattf <[EMAIL PROTECTED]> > Subject: RE: [Asterisk-Users] Sangoma A101 + Rhino channelbank > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > <[email protected]> > Message-ID: > <[EMAIL PROTECTED] m> > > Content-Type: text/plain; charset="iso-8859-1" > > Keep on bugging the Sangoma guys, I know they are working on several RBS T1 > issues right now(They called me Friday to go over a few things) They just > need help from users like you and I to find the bugs in their drivers. > > Have you tried any other signalling types other than LOOP? > > MATT--- > > > -----Original Message----- > From: Felician CHELU [mailto:[EMAIL PROTECTED] > Sent: Monday, April 11, 2005 9:52 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] Sangoma A101 + Rhino channelbank > > > Hello, > > I have Asterisk 1.0.6 - I try to setup Sangoma A101 T1 board together with > the Rhino fxs chanelbank. > Things done: > - T1 cross cable = I have carrier, signalling and framnig leds on > the channelbank green. > - channelbank configuration: > t1 - Proto: LOOP Frame: esf Clock: slave Coding: > b8zs > channels(analog) : Function:A-fxs Mode:loop > - zaptel.conf > span=2,1,0,esf,b8zs > fxols=32-55 > (i have a span 1 with a digium e1) > - zapata.conf > .... signalling=fxo_ls > - wanpipe1.conf > > [devices] > wanpipe1 = WAN_AFT, Comment > > [interfaces] > w1g1 = wanpipe1, , TDM_VOICE, Comment > > [wanpipe1] > CARD_TYPE = AFT > S514CPU = A > CommPort = PRI > AUTO_PCISLOT = NO > PCISLOT = 10 > PCIBUS = 2 > FE_MEDIA = T1 > FE_LCODE = B8ZS > FE_FRAME = ESF > FE_LINE = 1 > TE_CLOCK = MASTER > ACTIVE_CH = ALL > TE_HIGHIMPEDANCE = NO > LBO = 0DB > INTERFACE = V35 > CLOCKING = EXTERNAL > BaudRate = 0 > MTU = 1500 > UDPPORT = 9000 > TTL = 255 > IGNORE_FRONT_END = NO > > [w1g1] > PROTOCOL = HDLC > HDLC_STREAMING = YES > ACTIVE_CH = ALL > IDLE_FLAG = 0x7E > MTU = 1500 > MRU = 1500 > TDMV_SPAN = 2 > TDMV_ECHO_OFF = NO > MULTICAST = NO > TRUE_ENCODING_TYPE = NO > > > I already called Sangoma and Rhino support, but after hours of long distance > call conversation the problem is still not solved. Finnaly, a guy from Rhino > told me that their "asterisk expert" (which was not avaliable) knows about > this problem and that it is that the sangoma driver is not communicating > with asterisk. > > The wanrouter starts ok, after ztcfg I see the channels configured. > The problem: i don't have dialtone on phones. > > Question: When i enter zttoll, if i go to the sangoma span and I make "loop" > then it freezes. Is it normal? > > If someone has experienced this combination and made it work please give me > a sign. > > Thank you. > > PS: > > Felician > > _______________________________________________ > Asterisk-Users mailing list > [email protected] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > ------------------------------ > > Message: 12 > Date: Mon, 11 Apr 2005 10:32:26 -0400 > From: Andrew Kohlsmith <[EMAIL PROTECTED]> > Subject: Re: [Asterisk-Users] Can you comment on this Qos script? How > does one shape RTP? > To: [email protected] > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset="iso-8859-1" > > On April 11, 2005 10:08 am, Sean Kennedy wrote: > > Honestly, the best script I've ever found is the wondershaper script ( > > google it ). I tried the correct one posted in this thread, tried > > modifying it, but in the end I just used wondershaper. > > :-) I started out with wshaper and just didn't like it, which is where rc.tc > > came from. > > -A. > > > ------------------------------ > > Message: 13 > Date: Mon, 11 Apr 2005 16:48:27 +0200 > From: Ricardo Peironcely <[EMAIL PROTECTED]> > Subject: Re: [Asterisk-Users] TDM400P power supply > To: Asterisk Users Mailing List - Non-Commercial Discussion > <[email protected]> > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset="iso-8859-1" > > Thanks, > > I will try with external power supply. > > Rpr > > Rich Adamson escribi�: > > >>I've a problem with a TDM400P digium card. > >> > >>My box has no molex connectors for power supply. Simply has no any power > >>connector, because is not a normal PC) And I need to know if i can use a > >>external supply. But I've several questions: > >> > >>1.- Are both circuits (PCI-power and Phone-line-power) electrically > >>separated? > >>2.- A little voltage difference can create an undesired internal current? > >>3.- What are the current needs for this supply? > >> > >>I need the power supply because I want to use both FXS and FXO ports. > >>And I can't use a Y cable, because I've no molex connectors. > >> > >> > > > >Been discussed several times before and you should have found the > >answer using google. > > > >The TDM connector is only used for the fxs modules, and then only the > >+12 volt lead on that connector (and ground) is actually wired to > >anything on the TDM board. So, there is no conflict with internal > >system voltages. > > > >Yes you can use an external 12 volt power supply. > > > >The 12 volts is only used on the card to generate ringing voltage to > >the fxs modules. No ringing, no significant current draw. Just about > >any 12 volt supply should do, however I think I'd be looking for > >one that is at least somewhat regulated. No other idea on the power > >supply specs. > > > > > >_______________________________________________ > >Asterisk-Users mailing list > >[email protected] > >http://lists.digium.com/mailman/listinfo/asterisk-users > >To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.digium.com/pipermail/asterisk- users/attachments/20050411/aa53ab12/attachment-0001.htm > > ------------------------------ > > Message: 14 > Date: Mon, 11 Apr 2005 20:52:03 +0600 > From: "Yusuf Iqbal" <[EMAIL PROTECTED]> > Subject: [Asterisk-Users] Problem with X101P > To: [email protected] > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset="us-ascii" > > An HTML attachment was scrubbed... > URL: > http://lists.digium.com/pipermail/asterisk- users/attachments/20050411/4930a523/attachment-0001.htm > > ------------------------------ > > Message: 15 > Date: Mon, 11 Apr 2005 07:52:07 -0700 > From: "trixter http://www.0xdecafbad.com"; <[EMAIL PROTECTED]> > Subject: Re: [Asterisk-Users] Can you comment on this Qos script? How > does one shape RTP? > To: Asterisk Users Mailing List - Non-Commercial Discussion > <[email protected]> > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset="us-ascii" > > On Mon, 2005-04-11 at 10:32 -0400, Andrew Kohlsmith wrote: > > On April 11, 2005 10:08 am, Sean Kennedy wrote: > > > Honestly, the best script I've ever found is the wondershaper script ( > > > google it ). I tried the correct one posted in this thread, tried > > > modifying it, but in the end I just used wondershaper. > > > > :-) I started out with wshaper and just didn't like it, which is where > rc.tc > > came from. > you may want to pull at least the RTP lines I just posted and add them > to your rc.tc since that is what I got and tweaked since I use RTP :) > > -- > Trixter http://www.0xdecafbad.com > -------------- next part -------------- > A non-text attachment was scrubbed... > Name: not available > Type: application/pgp-signature > Size: 189 bytes > Desc: This is a digitally signed message part > Url : > http://lists.digium.com/pipermail/asterisk- users/attachments/20050411/c5e30ef3/attachment-0001.pgp > > ------------------------------ > > Message: 16 > Date: Mon, 11 Apr 2005 09:49:17 -0500 (CDT) > From: Dave Weis <[EMAIL PROTECTED]> > Subject: [Asterisk-Users] wcfxo problem > To: [email protected] > Message-ID: > <[EMAIL PROTECTED]> > Content-Type: TEXT/PLAIN; charset=US-ASCII; format=flowed > > > I've got a X100P in a compaq proliant 3000. My system stops taking calls > and making calls. I had been getting the FXO PCI Master abort before > updating, I am now running a cvs head checkout from a week or so ago. Now > I still have the problem but get more error messages: > > Found a Wildcard FXO: Wildcard X101P > Registered tone zone 0 (United States / North America) > Registered tone zone 0 (United States / North America) > FXO PCI Master abort > wcfxo: Out of space to write register 05 with 02 > wcfxo: Out of space to write register 05 with 03 > wcfxo: Out of space to write register 05 with 0a > wcfxo: Out of space to write register 05 with 0a > wcfxo: Out of space to write register 05 with 0a > wcfxo: Out of space to write register 05 with 0a > > Any solution? > > -- > Dave Weis "I believe there are more instances of the abridgment > [EMAIL PROTECTED] of the freedom of the people by gradual and silent > encroachments of those in power than by violent > and sudden usurpations."- James Madison > > > ------------------------------ > > Message: 17 > Date: Mon, 11 Apr 2005 10:54:30 -0400 > From: "Robert Webb" <[EMAIL PROTECTED]> > Subject: [Asterisk-Users] (no subject) > To: [email protected] > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset="ISO-8859-1"; format="flowed" > > > Good morning all.. > > I was following a discussion on this list about the > TDM400P revisions. It is my understanding that the current > revision that one should have is the Rev. H and not the > E/F. I have not yet been able to verify the rev stamped on > the board, but zaptel is reporting that I have the Rev. > E/F. I just bought this card in January direct from Digium > and was wondering if I got the wrong Rev. somehow?? I have > been having some intermittent problems but only thought it > was my setup. > > > > ------------------------------ > > Message: 18 > Date: Mon, 11 Apr 2005 09:57:16 -0500 > From: Doug Millsaps <[EMAIL PROTECTED]> > Subject: Re: [Asterisk-Users] Sipura SPA-841 Phone Review > To: Asterisk Users Mailing List - Non-Commercial Discussion > <[email protected]> > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset="us-ascii"; format=flowed > > I use a headset w/out any problems, except for if my cell phone is close by > and rings. Otherwise, volume is ok and no humming. Could it be your > headset? > > At 01:56 PM 4/10/2005, you wrote: > > >Just make sure you don't have a cordless or cell phone near by or the > >headset jack will "receive" a considerable amount of interference into > >your conversation (when NOT using a headset). > > > >Also don't even try using a headset... volume is low and there is a loud > >humming noise. > > > > ------------------------------ > > Message: 19 > Date: Mon, 11 Apr 2005 17:03:32 +0200 > From: "Bruno Hertz" <[EMAIL PROTECTED]> > Subject: Re: [Asterisk-Users] From OH323 to SIP or OH323 without > gatekeeper > To: [email protected] > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset=us-ascii > > "Joe S" <[EMAIL PROTECTED]> writes: > > > Hi, > > > > I am new with asterisk. I was wondering if there is a way to call a > > OH323 user or SIP user using Netmeeting/SJPhone with H323 as the > > default protocol without having a gatekeeper. > > > > I can make a call from SIP to OH323 by specifying it in the > > extensions.conf file, like: > > > > exten=>1001, 1, Dial(OH323/10.10.10.1) > > > > so I was wondering if there was a way to call from OH323 to SIP or OH323. > > Sure. Just specify in oh323.conf the context where incoming calls > should go. That context then can include dial statements for any > protocol, SIP, H323, IAX, whatever. See the Wiki for details on how to > setup dial plans. > > Finally, instruct your H323 phone to use asterisk as a gateway > resp. proxy, not a gatekeeper. Any calls will then go through > asterisk, and to the context you specified. > > I'm doing that with Gnomemeeting all the time, and it works without > problems. > > Regards, Bruno. > > > > ------------------------------ > > Message: 20 > Date: Tue, 12 Apr 2005 01:03:51 +1000 (EST) > From: Sahil Gupta <[EMAIL PROTECTED]> > Subject: Re: [Asterisk-Users] wcfxo problem > To: Asterisk Users Mailing List - Non-Commercial Discussion > <[email protected]> > Message-ID: <[EMAIL PROTECTED]> > Content-Type: TEXT/PLAIN; charset=US-ASCII; format=flowed > > I'm having similar issues using an X100P Ambient Chipset Clone Card.... > any ideas? > > Regards, > > > Sahil Gupta > VoiceValley > > On Mon, 11 Apr 2005, Dave Weis wrote: > > > > > I've got a X100P in a compaq proliant 3000. My system stops taking calls > and > > making calls. I had been getting the FXO PCI Master abort before updating, > I > > am now running a cvs head checkout from a week or so ago. Now I still have > > > the problem but get more error messages: > > > > Found a Wildcard FXO: Wildcard X101P > > Registered tone zone 0 (United States / North America) > > Registered tone zone 0 (United States / North America) > > FXO PCI Master abort > > wcfxo: Out of space to write register 05 with 02 > > wcfxo: Out of space to write register 05 with 03 > > wcfxo: Out of space to write register 05 with 0a > > wcfxo: Out of space to write register 05 with 0a > > wcfxo: Out of space to write register 05 with 0a > > wcfxo: Out of space to write register 05 with 0a > > > > Any solution? > > > > -- > > Dave Weis "I believe there are more instances of the > abridgment > > [EMAIL PROTECTED] of the freedom of the people by gradual and silent > > encroachments of those in power than by violent > > and sudden usurpations."- James Madison > > _______________________________________________ > > Asterisk-Users mailing list > > [email protected] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > ------------------------------ > > Message: 21 > Date: Mon, 11 Apr 2005 11:11:45 -0400 > From: "Robert Webb" <[EMAIL PROTECTED]> > Subject: Re: [Asterisk-Users] TDM400P Revision question. > To: Asterisk Users Mailing List - Non-Commercial Discussion > <[email protected]> > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset="ISO-8859-1"; format="flowed" > > Sorry for the initial no subject line. Was in a hurry when > I typed this and somehow missed putting it in. > > Please accept my apologies.... > > On Mon, 11 Apr 2005 10:54:30 -0400 > "Robert Webb" <[EMAIL PROTECTED]> wrote: > > > > Good morning all.. > > > > I was following a discussion on this list about the > >TDM400P revisions. It is my understanding that the > >current revision that one should have is the Rev. H and > >not the E/F. I have not yet been able to verify the rev > >stamped on the board, but zaptel is reporting that I have > >the Rev. E/F. I just bought this card in January direct > >from Digium and was wondering if I got the wrong Rev. > >somehow?? I have been having some intermittent problems > >but only thought it was my setup. > > > > _______________________________________________ > > Asterisk-Users mailing list > > [email protected] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ------------------------------ > > Message: 22 > Date: Mon, 11 Apr 2005 11:13:45 -0400 > From: "Bobby Lacey" <[EMAIL PROTECTED]> > Subject: [Asterisk-Users] Intercom with Aastra 480e? > To: <[email protected]> > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset="us-ascii" > > Hello list, > > I have been successful in setting up my first * box with a pair of > x100p's, Cisco 7960, and a Digium iAXy. > > I would like to incorporate an Aastra 480e using my iAXy and ADSI. I > want to be able to answer phone calls with my 7960 in the back of the > house and park the call, then in turn call the intercom on the 480e in > the front (using two way audio) to announce that there is a call that > needs to be picked up on 701. > > Also, by using the Aastra 480e, can I see my Zap line status to see what > lines are available and also if extensions are in use? > > Thanks in advance. > > B. Lacey > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.digium.com/pipermail/asterisk- users/attachments/20050411/2f2fcc73/attachment-0001.htm > > ------------------------------ > > Message: 23 > Date: Mon, 11 Apr 2005 11:16:56 -0400 > From: Matt <[EMAIL PROTECTED]> > Subject: [Asterisk-Users] Manipulate Asterisk Database from manager? > To: Asterisk Users Mailing List - Non-Commercial Discussion > <[email protected]> > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset=ISO-8859-1 > > Hi, > Is there anyway to manipulate the asterisk internal database from the > manager (the one you can telnet to)? And if so.. how does one do it? > (ie for enabling call forwarding, etc) > > > ------------------------------ > > _______________________________________________ > Asterisk-Users mailing list > [email protected] > http://lists.digium.com/mailman/listinfo/asterisk-users > > > End of Asterisk-Users Digest, Vol 9, Issue 93 > ********************************************* >
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