RE: [Asterisk-Users] Future WinCE IP Phone
[Kevin Walsh Wrote] Marvellous. Microsoft will bring their legendary stability, security and reliability to the VoIP world. Oops - there goes my lunch. Maybe but looking past that what the unit will bring is a programmable touch screen GUI on a hard VOIP phone. And being a Microsoft product it's going to have the familar look and feel, outlook synchronisation, office integration etc. etc. that 90% of the computer users on the planet know how to work. Aaron __ Do you Yahoo!? Yahoo! Mail - 50x more storage than other providers! http://promotions.yahoo.com/new_mail ___ 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
RE: [Asterisk-Users] Really basic stuff :(
65.39.205.111 is not local, substituting externip Check for res for is not a local user build_route: Contact hop: sip:65.39.205.111:5060 -- Executing Dial(SIP/fwd.pulver.com-0811c948, IAX2/janie|20|tr) in new stack SIMPLE DIAL (NO URL) -- Called janie -- Call accepted by 10.0.0.74 (format ULAW) -- Format for call is ULAW -- IAX2[janie]/4 is ringing And so it is. I answer the softphone and: Dropping incompatible voice frame on IAX2[janie]/4 of format GSM since our native format has changed to ULAW screams up the screen for each frame... Does this make sense to anyone? FWD only supports ULAW comment out the line allow=GSM in the general section of the iax.conf Jason ___ 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
RE: [Asterisk-Users] Compiling zaptel under 9.1 Suse
Hi, at SuSE 9.0 helped: I am not able to compile zaptel... Could you give me a hint? Have you tried the following, which is suggested in the output? 'make cloneconfig make dep' in /usr/src/linux/ Felix -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael George Sent: Thursday, June 24, 2004 8:53 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Compiling zaptel under 9.1 Suse Try building the kernel and the build the zaptel drivers. That worked for me. On Jun 24, 2004, at 1:20 PM, Tony Nichols wrote: Still no go I have asked Digium tech support to look into it. I need the later cvs to get around a bug with the latest tdm400 card (load driver - unload driver - load driver again to make it work. t o n y On Thu, 2004-06-24 at 08:15, Tony Nichols wrote: On Wed, 2004-06-23 at 14:32, asterisk wrote: Have some errors with the above. I have tried make and make linux26 Anyone got any clues ? I've googled but only got the make linux26 help Asterisk compiles and runs great, libpri compiles with no problems. TIA Julian. pbx:~ # cd /usr/src/zaptel pbx:/usr/src/zaptel # make linux26 make -C /usr/src/linux-2.6 SUBDIRS=/usr/src/zaptel modules make[1]: Entering directory `/usr/src/linux-2.6.4-52' CHK include/linux/version.h *** Warning: Overriding SUBDIRS on the command line can cause *** inconsistencies make[2]: `arch/i386/kernel/asm-offsets.s' is up to date. CC [M] /usr/src/zaptel/zaptel.o /usr/src/zaptel/zaptel.c: In function `zt_net_open': /usr/src/zaptel/zaptel.c:1166: warning: passing arg 1 of `hdlc_open' from incompatible pointer type /usr/src/zaptel/zaptel.c: In function `zt_net_stop': /usr/src/zaptel/zaptel.c:1238: warning: passing arg 1 of `hdlc_close' from incompatible pointer type /usr/src/zaptel/zaptel.c: In function `zt_xmit': /usr/src/zaptel/zaptel.c:1294: error: structure has no member named `netdev' /usr/src/zaptel/zaptel.c:1294: warning: type defaults to `int' in snip This happened to me too (same dist/kernel) with cvs head 6/21/2004 - older version 4/24/2004 worked ok. I'm going to try latest cvs today and see if it works. t o n y ___ 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 -Michael ___ 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
RE: [Asterisk-Users] Which Linux ?
Hi, From recent experience: If you want to use digium hardware dont use suse 9.0. It seems to think the E1 card is a tigerjet bri card and the kernel hangs on ztcfg. I have a WT405P running under SuSE 9.0 and it works great. But I had only choosen SuSE because I also need capi... Bye FElix ___ 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
RE: [Asterisk-Users] Which Linux ?
Mike, I've been trying to install under SuSE 9.1, but cannot compile zaptel What's the secret incantation ?? TIA I was helped with: I am not able to compile zaptel... Could you give me a hint? Have you tried the following, which is suggested in the output? 'make cloneconfig make dep' in /usr/src/linux/ Felix ___ 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
Re: [Asterisk-Users] Compiling zaptel under 9.1 Suse
Tried that. Tried rebuilding kernel and rebooting. Same errors encountered. Ah well. I've reloaded the machine with FC1. Thanks for all the help and support anyway - it's been a great lesson. I built my first kernel :) Julian - Original Message - From: ePyron Felix Deierlein [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, June 25, 2004 9:17 AM Subject: RE: [Asterisk-Users] Compiling zaptel under 9.1 Suse Hi, at SuSE 9.0 helped: I am not able to compile zaptel... Could you give me a hint? Have you tried the following, which is suggested in the output? 'make cloneconfig make dep' in /usr/src/linux/ Felix -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael George Sent: Thursday, June 24, 2004 8:53 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Compiling zaptel under 9.1 Suse Try building the kernel and the build the zaptel drivers. That worked for me. On Jun 24, 2004, at 1:20 PM, Tony Nichols wrote: Still no go I have asked Digium tech support to look into it. I need the later cvs to get around a bug with the latest tdm400 card (load driver - unload driver - load driver again to make it work. t o n y On Thu, 2004-06-24 at 08:15, Tony Nichols wrote: On Wed, 2004-06-23 at 14:32, asterisk wrote: Have some errors with the above. I have tried make and make linux26 Anyone got any clues ? I've googled but only got the make linux26 help Asterisk compiles and runs great, libpri compiles with no problems. TIA Julian. pbx:~ # cd /usr/src/zaptel pbx:/usr/src/zaptel # make linux26 make -C /usr/src/linux-2.6 SUBDIRS=/usr/src/zaptel modules make[1]: Entering directory `/usr/src/linux-2.6.4-52' CHK include/linux/version.h *** Warning: Overriding SUBDIRS on the command line can cause *** inconsistencies make[2]: `arch/i386/kernel/asm-offsets.s' is up to date. CC [M] /usr/src/zaptel/zaptel.o /usr/src/zaptel/zaptel.c: In function `zt_net_open': /usr/src/zaptel/zaptel.c:1166: warning: passing arg 1 of `hdlc_open' from incompatible pointer type /usr/src/zaptel/zaptel.c: In function `zt_net_stop': /usr/src/zaptel/zaptel.c:1238: warning: passing arg 1 of `hdlc_close' from incompatible pointer type /usr/src/zaptel/zaptel.c: In function `zt_xmit': /usr/src/zaptel/zaptel.c:1294: error: structure has no member named `netdev' /usr/src/zaptel/zaptel.c:1294: warning: type defaults to `int' in snip This happened to me too (same dist/kernel) with cvs head 6/21/2004 - older version 4/24/2004 worked ok. I'm going to try latest cvs today and see if it works. t o n y ___ 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 -Michael ___ 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 ___ 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
Re: [Asterisk-Users] X101P on a UK BT line ---- txgain issue
I do get echo, lots of it, I am waiting until the new patch they are all on about on the list gets into a stable release, then I will upgrade and see if that does the trick. I am told that some of the echo may be to do with a mismatch in the impedance with the BT line. I had an adsl problem a while back and the engineer fixing it said I had a constant, 50 something ohm, loop condition if the X100p was plugged in. If we could fiddle the impedance matching maybe it would fix things a bit. But I am no phone engineer so am not sure on any of this. Chris. Chris Stenton wrote: Chris, Do you get echo issues? If not could you let us have your config and which echo canceller you use. Thanks Chris On Thu, 2004-06-24 at 20:40, Chris Lee wrote: Chris Stenton wrote: I am finding that I have to increase the txgain in zapata.conf to 8 when my X101P is connected to my BT phone line, otherwise people can hardly hear me. This then gives echo issues. Do other people have the same problem on BT lines. I was wondering if I should give BT a call and get them to increase the gain on the line. Strange though as the rxgain is OK and I don't have this problem with an ordinary phone. Yes I have this too (BT LINE), I upped mine to 10 in order that other people could hear me without a problem. I can hear them fine. Chris. ___ ___ 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
Re: [Asterisk-Users] host=dynamic vs host=xxx.xxx.xxx.xxx
Jeremy It seems you misunderstood my question. I was talking about SIP not IAX. It wasn't about access control - it was about having a problem with phones on a poor connection that is prone to occasional packet loss or disconnection. How much clearer do you need to be? Asterisk is telling you exactly what the problem is. Interpreting an error message is easy finding, as you so rightly point out asterisk is telling you exactly what the problem is. The solution however wasn't clear hence my reason for posting. Finding a solution on software you're not particularly familiar with and in a configuration you don't have 100% confidence in isn't as simple as reading an error message; hence my posting. Matt Matt wrote: NOTICE[-1147675728]: Peer '004' is trying to register, but not configured as host=dynamic How much clearer do you need to be? Asterisk is telling you exactly what the problem is. Have you tried simply doing host=dynamic into your iax.conf? If you still want IP based access control you can use permit and deny directives. Jeremy McNamara ___ 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
RE: [Asterisk-Users] X101P on a UK BT line ---- txgain issue
I do get echo, lots of it, I am waiting until the new patch they are all on about on the list gets into a stable release, then I will upgrade and see if that does the trick. The patch didn't seem to work for me. I am told that some of the echo may be to do with a mismatch in the impedance with the BT line. Problem is do we really want BT messing with gain there end and impedance cos it might mess our ADSL lines up =) I know im on the limits. Kind Regards, Chris Bond ___ 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] Bridging two calls together with Eicon card - help please :)
Hello all, I'm not familiar with Asterisk at all, so any help would be appreciated. I have an ISDN card lspci: 07:06.0 Network controller: Eicon Technology Corporation DIVA Server PRI-30M 2.0 which has 8 channels active. I am wondering if a:, this card is supported/can be made to work with Asterisk, and b:, if it is possible to make Asterisk initiate 2 outgoing voice calls, which it conferences together. Unfortunately I only have binary drivers for the Eicon card ( gah ). I am not a PBX expert, but am pretty handy with Linux, and can get my hands dirty with C if necessary. Calum -- Random russian saying: One does not look for good when he is well. jabber: [EMAIL PROTECTED] pgp: http://gk.umtstrial.co.uk/~calum/keys.php Linux 2.6.5-gentoo 09:32:07 up 16 days, 22:00, 1 user, load average: 0.27, 0.18, 0.11 ___ 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] wcfxs CPU usage
Hi, I'm using 12 fxo modules on tdm cards. When I do 'modprobe wcfxs', the cpu usage in kernel mode varies from 2% to 100%. While monitoring using top, there is no process using much cpu resource. Is this ok? Thanks in advance.
[Asterisk-Users] Latest CVS fax detection grandstream bug
As a follow up to my previous post, I have now identified what is causing the bug with the grandstream phones. When the line faxdetection=incoming is in the zapata.conf file, the grandstream phones will not ring, nor connect a call to the zaptel interface. Can anyone else confirm this bug? I'm going to play with the different options (incoming/outgoing/both) to see if it makes a difference. Thanks, Andrew _ Andrew Yager Real World Technology Solutions Real People, Real SolUtions (tm) ph: (02) 9945 2567 fax: (02) 9945 2566 mob: 0405 15 2568 http://www.rwts.com.au/ _ ___ 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
Re: [Asterisk-Users] X101P on a UK BT line ---- txgain issue
I can't see that any echo cancelling is going to work with a 10db difference between rx and txgain. If the difference is due to impedance mismatch reflections then the reflected tx signal is going to be of greater amplitude than the callers signal. Chris - Original Message - From: Chris Bond [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, June 25, 2004 9:35 AM Subject: RE: [Asterisk-Users] X101P on a UK BT line txgain issue I do get echo, lots of it, I am waiting until the new patch they are all on about on the list gets into a stable release, then I will upgrade and see if that does the trick. The patch didn't seem to work for me. I am told that some of the echo may be to do with a mismatch in the impedance with the BT line. Problem is do we really want BT messing with gain there end and impedance cos it might mess our ADSL lines up =) I know im on the limits. Kind Regards, Chris Bond ___ 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
RE: [Asterisk-Users] chan_capi problem - hangup???
Hi Tobi, I installed Asterisk with CAPI support. Everything works fine while starting Asterisk, but when a call comes in Asterisk hangsup the call after two times of ringing. The output is like: Jun 24 22:19:49 NOTICE[1082178480]: chan_capi.c:1931 capi_handle_msg: CONNECT_IND ID=002 #0x011d LEN=0048 Controller/PLCI/NCCI= 0x101 CIPValue= 0x10 CalledPartyNumber = c1**some_number** CallingPartyNumber = 21 83**some_number** CalledPartySubaddress = default CallingPartySubaddress = default BC = 80 90 a3 LLC = default HLC = 91 81 AdditionalInfo = default == CONNECT_IND (PLCI=0x101,DID=**some_number**,CID=**some_number**,CIP=0x10,C ONTROLLER=0x1) Jun 24 22:19:49 WARNING[1086696368]: pbx.c:1819 ast_pbx_run: Channel 'CAPI[contr1/**some_number**]/0' sent into invalid extension 's' in context 'default', but no invalid handler -- CAPI Hangingup activehangingup -- started pbx on channel (callgroup=0)! -- INFO_IND ID=002 #0x011e LEN=0023 Controller/PLCI/NCCI= 0x101 InfoNumber = 0x70 InfoElement = c1**some_number** I read in the mailing list archives of commenting out line 2615 in chan_capi.c, but that did not change anything. Has anybody got an idea what the error: Channel 'CAPI[contr1/**some_number**]/0' sent into invalid extension 's' in context 'default', but no invalid handler Do you have DIDs (PTP-ISDN)? Bye Felix ___ 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
Re: [Asterisk-Users] chan_capi problem - hangup???
ePyron Felix Deierlein wrote: Do you have DIDs (PTP-ISDN)? Bye Felix yes ___ 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] Howto: Installing Asterisk and ISDN on Fedora Core 1
I've managed to install and run Asterisk on a Fedora Core 1 server using the fritz avm ISDN card, and thought I'd share how it was done. This worked for me: Server: Dell 6450 Quad Xeon 700 2Mb Cache 4GB Ram 2x18GB SCSI (Mirrored) ISDN card (fritz!pci). This is packaged as a BT ISDN card. * Install FC1 from CD. Select only server components, development environment and kernel development * Yum update Important - you must do this - it installs a new kernel, not the buggy release kernel. * Reboot * Install atrpms-55-1.rhfc1.at.i386.rpm (from http://atrpms.net/dist/fc1/atrpms) * Install kernel-module-fcpci-2.4.22-1.2194.nptlsmp-03.11.02-3.rhfc1.at.i686.rpm (from http://atrpms.net/dist/fc1/fcpci) * vi /etc/capi.conf should contain a single line fcpci - - - - - - * modprobe fcpci * lsmod Should now show fcpci and kernelcapi loaded * capiinit * Get Asterisk from a shell, cd /usr/src export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot cvs login (anoncvs is the password) cvs checkout zaptel libpri asterisk * Build Asterisk cd /usr/src/zaptel make clean;make install;make config cd /usr/src/libpri make clean;make install;make config cd /usr/src/asterisk make clean;make install;make samples * Get chan_capi (from http://www.junghanns.net/asterisk/downloads/chan_capi.0.3.4a.tar.gz) * Extract chan_capi cd /usr/src tar -zxvf /tmp/chan_capi* (assuming the file was downloaded into /tmp) * Build chan_capi cd chan_capi make clean; make install; make config * Add chan_capi into Asterisk by changing modules.conf [modules] autoload=yes noload = pbx_gtkconsole.so noload = pbx_kdeconsole.so noload = app_intercom.so load = chan_modem.so load = res_musiconhold.so ; following two lines added load = res_parking.so load = chan_capi.so noload = chan_alsa.so [global] chan_modem.so=yes ; following line added chan_capi.so=yes * Start Asterisk asterisk -c (should be no errors) * Place a call to the ISDN line. As I said, this worked for me. The real problem I had before was getting the drivers for the isdn card to work. However, downloading them from the atrpms site worked first time no errors! Please feel free to tear this apart if you want. Julian. ___ 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] Failure in RTP streaming
hi, I use the oh323 driver to answer H323 calls. The connection is set up normally. In my extensions.conf file I use: exten = s,1,Answer exten = s,2,Playback(demo-instruct) exten = s,3,Hangup So that when a call is answered i get: *CLI -- Executing Answer(H323/ip$10.0.3.23:32782/6502, ) in new stack -- Executing Playback(H323/ip$10.0.3.23:32782/6502, demo-instruct) in new stack -- Playing 'demo-instruct' (language 'en') which is the normal procedure. The connexion is well built between the client and asterisk (H225 H245) and well negociated with the codec (gsm). But no RTP stream comes out of the asterisk (I tcpdumped to be sure). My question is: 1/Is there a way to explain this ? (lack of configuration, compilation options) if not, 2/ Is there a way to investigate deeper in order to understand where does the RTP stream faint inside Asterisk ? regards, -- Kiel ___ 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] Howto: Use setgroup, checkgroup to check incoming and outgoing client limits
Hi there, I was wondering how I can use setgroup and checkgroup for perfom incoming and outgoing limitation checks. I've have some users that doesn't what to be able to recieve more than 1 call at a time, and I also want to limit a users outgoing call abilities. Any help would be greatly appreciated. Kind regards Cf --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.708 / Virus Database: 464 - Release Date: 18-06-2004 ___ 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
Re: [Asterisk-Users] Latest CVS, Grandstream and Zaptel bug?
Have you upgraded the firmware on the Grandstream phones?
I had the same problem on newly purchased Grandstreams until I
upgraded the firmware (currently using 1.0.5.0).
Another thing I had to do to get ringing indications, etc. working on
my Grandstream was:
- Create a separate context for the Grandstream phones. For example,
if I had [cos-idd], I would create [cos-idd-grandstream] and in this,
put:
[cos-grandstream-idd]
exten = _.,1,Answer
exten = _.,2,ResetCDR
exten = _.,3,Goto(cos-idd,${EXTEN},1)
The ResetCDR part is required so that the CDR doesn't log ANSWERED
for every call.
That fixed my problems regarding ringing.
Hope that helps.
Regards,
Shaun
On Fri, 25 Jun 2004 15:40:39 +1000, Andrew Yager [EMAIL PROTECTED] wrote:
Already set.
Andrew
_
Andrew Yager
Real World Technology Solutions
Real People, Real SolUtions (tm)
ph: (02) 9945 2567 fax: (02) 9945 2566
mob: 0405 15 2568
http://www.rwts.com.au/
_
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Re: [Asterisk-Users] chan_capi problem - hangup???
Hi Felix, then I guess that I have the same problem. If I get a overlaped dial from PSTN, i get only the first did-digit as extension , p.e: my number 8993-12 then it goes to 89931 and that extension does not exist If I get a call from ISDN (or maybe mobile) with block transfer, I get 899312 and it works. For me it seems that chan_capi does not supply inbound overlap-dial. Could anybody clearify that, please? well, that seems not to be the same problem, as I get all digits from mobile and PSTN. Could you please share your capi.conf and extensions.conf? Would be great...maybe I've configured something wrong in there. Tobi ___ 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
RE: [Asterisk-Users] Record call from switch using serviceobserve? (execute command after dial?)
Hmmm. Now I have another problem... After the call goes to the extension 100 in this example, I get a jump to the t extension for the context. I can't find a way to make the call not time out, and asterisk acts like it needs to do something after the record start (i.e. the 100,2 in your example). When it jumps to the non-existing t (for timeout) it hangs up. This is kind of kludgey, but I sent it to a quiet (no announcements) meetme room. Any better way to handle this? Otherwise it works fine! Thanks! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adam Goryachev Sent: Thursday, June 24, 2004 7:51 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Record call from switch using serviceobserve? (execute command after dial?) Use the call file, and set the channel to something like: Zap/1/160wextension then set the extension/context/etc to point to something like this: exten = 100,1,playback(Now call will be recorded) exten = 100,2,Record(some file) exten = 100,3,playback(beep) Now, to stop recording you have two choices, either record to the end of the call, or else use the manager interface to signal a soft hangup. Actually, better is to use the manager interface to transfer the x100P to another extension like this: exten = 101,1,playback(Call no longer being recorded) exten = 101,2,StopRecord exten = 101,3,playback(beep) exten = 101,4,softhangup Check the proper applications for stoprecording and softhangup Regards, Adam On Fri, 2004-06-25 at 04:54, Garry Adkins wrote: Hi, I am working on a project to record agent calls when completing specific transactions with customers. Since these calls do not go through the asterisk box (They go through a lucent G3), we're thinking that service observe would be the easiest way to accomplish our goal. Here's what I need: On demand, I need to be able to attach to the switch, dial the service observe code, make an announcement, record. On the second event, I need to make an announcement, stop the recording, and hang-up the channel to the switch. Here's my plan: 1) Agent software calls a CGI on the asterisk box. This passes extension the agent is talking on. 2) CGI program somehow makes asterisk call to the switch, dials 160wextension which does a service observe (i.e. attaches the extension audio to our channel) 3) Asterisk play recording about transaction being recorded 4) Start recording 5) Software calls CGI again to notify asterisk to stop the recording. 6) Asterisk plays recording that the transaction is recorded 7) Asterisk disconnects channel. Eventually I will have a T1 interface into the switch, but for testing I'm just using the X100P and an analog port on the switch. The two communicate properly, I can call the asterisk box and have it answer, and I can generate a call to the switch from a different extension on the Asterisk box. Here's my attempted solutions: 1) When I try to generate the call from a SIP phone, it works fine. The extensions.conf contains a dial(zap/1/160wextension) 2) When I try to generate the call from the manager interface, I cannot do it without having a different input. action: originate channel: zap/1 exten: 555 context: default priority: 1 Extension 555 does a dial(zap/1/160wextension) Three problems: a) The problem is I have no other channels but the ZAP channel for the X100p. I can't connect both ends to the same channel. b) Also, I cannot send audio to this channel from the manager channel (for the announcement of the recording) c) Dial doesn't exit until hang-up, so I cannot background() the audio to the channel. 3) When I try to dial by generating a call file in the proper outbound call directory, I still get stuck on the dial command. Any ideas? Am I just not understanding something critical? Thanks for any help! I've search the archives and the WIKI for about 3 days. I'm stumped! -G ___ 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 -- -- Adam Goryachev Website Managers Ph: +61 2 9345 4395 [EMAIL PROTECTED] Fax: +61 2 9345 4396www.websitemanagers.com.au ___ 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]
Re: [Asterisk-Users] Dell 400SC and X100P
Actually ACPI is enabled no matter what on the machine. I wasn't talking about ACPI in the CMOS, but ACPI support in the Linux Kernel. I'm using 2.4.26 (kernel.org+latest libata patches) on a Debian Sarge box. Kind regards, Martin List-Petersen On Thu, 2004-06-24 at 14:25, Isamar Maia wrote: Thanks a lot for replying. I turned on the ACPI in the CMOS and it got better. At least I call receive several calls in sequence and call out but it hangs up right after the person gets the phone in the other side. So, something is still missing. What is your ACPI mode in the CMOS ? S1 or S3? Which kernel version are you using? Can you send me your .config ? Thanks again, Isamar On Thu, 24 Jun 2004, Martin List-Petersen wrote: Is your kernel ACPI enabled ? The motherboard in the PE400SC is basically the Dimension 8300, which i use for my development box with 1 X101P, 1 TDM400P and two ISDN cards here at home and that works without problems. One thing to make sure with these boards is that ACPI is enabled, since they are ACPI only. Kind regards, Martin List-Petersen On Thu, 2004-06-24 at 02:58, Isamar Maia wrote: I have a Dell PowerEdge 400SC with a X100P and a TDM01b. The board works wonderfully in another machine but in this brand new one, it just get in nuts. The problem is: 1) Zaptel recognizes it perfectly 2) No IRQ conflicts, two-wire new cable. 3) Asterisk starts up and listen the ring and answer the cal 4) RIght after answering the call, it's dropped. 5) The following calls, even with asterisk off, the driver(???) answers the call and hang it up. With the * running, it doesn't even get any ring, and the call is answered and dropped right away. Isamar ___ 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 ___ 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
[Asterisk-Users] Playtones problem
Hi:
I have an Asterisk server (version 0.9.0) working with a Digium
E100P card that makes an EuroISDN connection with a Siemens Hicom 300 PBX.
Hicom 300 is pri_net and Asterisk is pri_cpe.
Furthermore, makes an OH323 connection with a GNU gatekeeper
When you make a call between a Hicom 300 extension an IP phone its
generates succefully, but a Hicom 300 extension still hear a ringback when
an IP phone
pick up.
My extensions.conf includes:
exten = _30XX,1,Playtones(ring)
exten = _30XX,2,Dial,OH323/${EXTEN}
exten = _30XX,3,StopPlaytones
but StopPlaytones never executes.
If use Ringing instead Playtones(ring), a Hicom 300 extension can`t hear a
ringback sound
when a call progress.
Could you give me some hints?
Thanks.
Rafael Olivieri.
Mayor Rafael Mario Olivieri
Comando de Comunicaciones e Informática
Dpto Comunicaciones - Jefe Div C4
4346-6137
4346-6100 int 6137
Este mensaje y sus adjuntos son de caracter confidencial para uso de
los destinatarios a los que está dirigido. Las opiniones vertidas en este
correo son exclusivas de su autor y no representa la opinión del Ejército
Argentino.
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Re: [Asterisk-Users] Howto: Use setgroup, checkgroup to check incoming and outgoing client limits
At 13:00 25/06/2004 +0200, you wrote: Hi there, I was wondering how I can use setgroup and checkgroup for perfom incoming and outgoing limitation checks. I've have some users that doesn't what to be able to recieve more than 1 call at a time, and I also want to limit a users outgoing call abilities. Any help would be greatly appreciated. exten = 999,1,SetGroup(moh);Set Current Group to moh exten = 999,2,CheckGroup(1);Check moh does not have more than 1 exten = 999,3,Answer;Answer the call exten = 999,4,MusicOnHold(default);Play default Music on hold exten = 999,103,Busy;Play busy if 1 person is already listening This will allow only one call to use the resource music on hold. Jason
[Asterisk-Users] HT286, fax and FXS impedance for Europe?
Hi there, the latest firmware (at least 1.5.0.0) for the Grandstream HT286 HandyTone now has an option to set the FXS impedance. It appears that for EU countries the CTR21 is the correct setting. My question: Does changing this setting improve fax operation for you? Cheers, Philipp Source: http://www.cmlmicro.com/Products/Reports/DE8681%20CTR21%20Compliance.pdf In 1998, many European telecom standards were replaced with a single harmonized standard CTR21, which applied to all EU member countries. Compliance with CTR 21 was mandatory for CE marking. On 8th April 2000 however, a new directive, 1999/5/EC, commonly known by its short name Radio and Telecommunication Terminal Equipment (RTTE), established a new regulatory framework for telecom approvals, the RTTE Directive repealed CTR21 and declared that it no longer be mandatory for devices that connect to public network phone lines. Under the RTTE directive, the only mandatory requirements for CE marking of a typical modem are those for safety and EMC. Otherwise, products are presumed to comply with the Directive when they meet the requirements within the usage conditions for which they are intended. While many EU countries now maintain certain recommended requirements for interfacing to their phone lines, compliance with these requirements is not mandatory for CE marking. To deal with the uncertainty regarding telecom compliance, some manufacturers have continued to design for compliance with CTR 21. This is perfectly acceptable, but it is no longer mandatory. ___ 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
Re: [Asterisk-Users] help needed with read()
On Wed, 2004-06-23 at 17:12, Sathya wrote:
Hi,
Greatly appreciate if some one help me with the application read().
I have added a feature to reload asterisk from a phone...
it uses 'read' to get a 3 digit password
I was using '#' to end the sequence until I realised I could specify the
number should be only three digits long...
My voice prompts (posix-...) are described in the text comments...
; 307 = Restart Asterisk
exten = 307,1,DigitTimeout(4) ; Set Digit Timeout 4 seconds
exten = 307,2,ResponseTimeout(5); Set Response Timeout 5 sec
exten = 307,3,Read(Secret,posix-pass-restart-ast,3)
; to restart type the passwd
exten = 307,4,NoOp(${Secret})
exten = 307,5,Gotoif($[${Secret} = 123]?6:9)
exten = 307,6,Playback(posix-restarting) ; Restarting asterisk
exten = 307,7,Wait(1)
exten = 307,8,System(/usr/sbin/asterisk -rx reload)
exten = 307,9,Hangup
--
. . ___. .__ Posix Systems - Sth Africa
/| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE
/ |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496
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Re: [Asterisk-Users] Problem with music on hold...
Hi! Using a sip phone, x-lite, after I dialed 6601 I get the following: Have you set Transmit silence to YES in X-Lite? -- Reloading module 'cdr_csv.so' (Comma Separated Values CDR Backend) -- Executing WaitMusicOnHold(SIP/666-f408, 30) in new stack Jun 24 22:35:07 WARNING[458766]: res_musiconhold.c:331 moh1_exec: Unable to start music on hold (class '30') on channel SIP/666-f408 == Spawn extension (default, 6601, 1) exited non-zero on 'SIP/666-f408' I assume you did load musiconhold in /etc/asterisk/modules.conf, and that you configured /etc/asterisk/musiconhold.conf correctly !? I am using gentoo linux, I have mpg123 installed. Which version of mpg123 are you using exactly? The recommended version is ends with an 'r'. Finally: What type of mp3 file are you trying to play, is it 128 kbit, mono and without VBR (variable bit rate) and without any ID3 tags? Cheers, Philipp ___ 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
RE: [Asterisk-Users] Transfer - to your own number
Hi! Um - If my secretary transfer's a call from her BT101 to her own number - she looses the call. What can I do to stop this from happening - apart from dyeing her hair from blond to brunette ??? - method a) SetGroup() and GetGroupcount() in extensions.conf - method b) incominglimit= and outgoinglimit= in sip.conf Method a) is to be preferred. Use it together with GotoIf() and a group name that includes the secretary extension. Cheers, Philipp ___ 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
RE: [Asterisk-Users] Really basic stuff :(
Hi! FWD only supports ULAW comment out the line allow=GSM in the general section of the iax.conf Nonsense - FWD *does* permit the use of GSM. Cheers, Philipp ___ 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
RE: [Asterisk-Users] Really basic stuff :(
At 14:48 25/06/2004 +0200, you wrote: Hi! FWD only supports ULAW comment out the line allow=GSM in the general section of the iax.conf Nonsense - FWD *does* permit the use of GSM. Cheers, Philipp Not in iax only with sip Jason ___ 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
Re: [Asterisk-Users] Howto: Installing Asterisk and ISDN on Fedora Core 1
ISDN Carlos Arnt [EMAIL PROTECTED] Diretor de Informática. Divisão de Tecnologia e Desenvolvimento TI. Intellissence do Brasil. http://www.intellissence.com/brasil Tel:(+55)-(21)-(3908-4667) Tel(Direto):(+55)-(21)-(3905-1561) Cel:(+55)-(21)-(9169-8537) --VoIP Contact Method-- World VoIP Pin/Code: 31872 Uk/Japan VoIP Pin/Code: 17009881356 - "Thinking is the hardest work there is, which is the probable reason so few engage in it". - Henry Ford - On Fri, 25 Jun 2004 11:46:28 +0100, Asterisk wrote: I've managed to install and run Asterisk on a Fedora Core 1 server using the fritz avm ISDN card, and thought I'd share how it was done. This worked for me: Server: Dell 6450 Quad Xeon 700 2Mb Cache 4GB Ram 2x18GB SCSI (Mirrored) ISDN card (fritz!pci). This is packaged as a BT ISDN card. * Install FC1 from CD. Select only server components, development environment and kernel development * Yum update Important - you must do this - it installs a new kernel, not the buggy release kernel. * Reboot * Install atrpms-55-1.rhfc1.at.i386.rpm (from http://atrpms.net/dist/fc1/atrpms) * Install kernel-module-fcpci-2.4.22-1.2194.nptlsmp-03.11.02- 3.rhfc1.at.i686.rpm (from http://atrpms.net/dist/fc1/fcpci) * vi /etc/capi.conf should contain a single line fcpci - - - - - - * modprobe fcpci * lsmod Should now show fcpci and kernelcapi loaded * capiinit * Get Asterisk from a shell, cd /usr/src export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot cvs login (anoncvs is the password) cvs checkout zaptel libpri asterisk * Build Asterisk cd /usr/src/zaptel make clean;make install;make config cd /usr/src/libpri make clean;make install;make config cd /usr/src/asterisk make clean;make install;make samples * Get chan_capi (from http://www.junghanns.net/asterisk/downloads/chan_capi.0.3.4a.tar.gz) * Extract chan_capi cd /usr/src tar -zxvf /tmp/chan_capi* (assuming the file was downloaded into /tmp) * Build chan_capi cd chan_capi make clean; make install; make config * Add chan_capi into Asterisk by changing modules.conf [modules] autoload=yes noload = pbx_gtkconsole.so noload = pbx_kdeconsole.so noload = app_intercom.so load = chan_modem.so load = res_musiconhold.so following two lines added load = res_parking.so load = chan_capi.so noload = chan_alsa.so [global] chan_modem.so=yes following line added chan_capi.so=yes * Start Asterisk asterisk -c (should be no errors) * Place a call to the ISDN line. As I said, this worked for me. The real problem I had before was getting the drivers for the isdn card to work. However, downloading them from the atrpms site worked first time no errors! Please feel free to tear this apart if you want. Julian. ___ 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
Re: [Asterisk-Users] Failure in RTP streaming
On Fri, Jun 25, 2004, kiel hedjam wrote: hi, I use the oh323 driver to answer H323 calls. The connection is set up normally. In my extensions.conf file I use: exten = s,1,Answer exten = s,2,Playback(demo-instruct) exten = s,3,Hangup So that when a call is answered i get: *CLI -- Executing Answer(H323/ip$10.0.3.23:32782/6502, ) in new stack -- Executing Playback(H323/ip$10.0.3.23:32782/6502, demo-instruct) in new stack -- Playing 'demo-instruct' (language 'en') which is the normal procedure. The connexion is well built between the client and asterisk (H225 H245) and well negociated with the codec (gsm). But no RTP stream comes out of the asterisk (I tcpdumped to be sure). My question is: 1/Is there a way to explain this ? (lack of configuration, compilation options) if not, 2/ Is there a way to investigate deeper in order to understand where does the RTP stream faint inside Asterisk ? The version I used was the last cvs snapshot, I've just been trying with the 0.9.0 (the tar.gz version) and evrything is all right. I don't why I didn't get RTP streams with the cvs version, if I got time I would investigate a little bit. If anybody here have an idea ... -- Kiel ___ 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
RE: [Asterisk-Users] Leave one call to pick up another
Andrew Thompson wrote: Eric Wieling wrote: How is this different from the way standard call waiting works when provided from your telco? Um, he actually has two phone lines, not just one that he's flash-ing back and forth between. If he hangs up the line, does the second call not continue ringing? I take it once he hangs up the line both calls are gone? I can't help with any solution, but I can add my voice describing another symptom of this exact problem, in direct response to your last question. I have two lines, each handled by a Digium X100P card. If I am on the phone (whether I initiated a call, or received one, whether it uses one of the POTS lines or whether it's a VoIP call), if another call comes in, I hear the Call Waiting signal. If I simply hang up the current call, I lose _both_ calls. Meaning, the phone does not start ringing with the pending call any longer. This is _not_ the same behavior that I had with the same exact phone, when it was connected to the POTS line directly. Hanging up on the current call yields a ringing for the second call, after a second or two delay... P.S. If I flash the call, I can indeed speak to the second caller, and bounce back and forth between the calls, so I get the same behavior that the original poster (Brian Capouch) described. ___ 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
Re: [Asterisk-Users] Failure in RTP streaming
What version of asterisk-oh323 do you use? Michael. kiel hedjam wrote: hi, I use the oh323 driver to answer H323 calls. The connection is set up normally. In my extensions.conf file I use: exten = s,1,Answer exten = s,2,Playback(demo-instruct) exten = s,3,Hangup So that when a call is answered i get: *CLI -- Executing Answer(H323/ip$10.0.3.23:32782/6502, ) in new stack -- Executing Playback(H323/ip$10.0.3.23:32782/6502, demo-instruct) in new stack -- Playing 'demo-instruct' (language 'en') which is the normal procedure. The connexion is well built between the client and asterisk (H225 H245) and well negociated with the codec (gsm). But no RTP stream comes out of the asterisk (I tcpdumped to be sure). My question is: 1/Is there a way to explain this ? (lack of configuration, compilation options) if not, 2/ Is there a way to investigate deeper in order to understand where does the RTP stream faint inside Asterisk ? regards, ___ 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
Re: [Asterisk-Users] chan_capi problem - hangup???
Hi, Philipp von Klitzing wrote: Hi! Channel 'CAPI[contr1/**some_number**]/0' sent into invalid extension 's' in context 'default', but no invalid handler Look at Asterisk's standard extensions like s, i, o and so forth. Insert this in your context [default] in extensions.conf: exten = s,1,Answer exten = s,2,Dial(Local/1234/n); replace 1234 with any valid extension Cheers, Philipp This doesn't change a thing... I've this config: capi.conf: ; ; CAPI config ; ; [general] nationalprefix=0 internationalprefix=00 rxgain=0.8 txgain=0.8 [interfaces] msn=mymsn incomingmsn=* controller=1 softdtmf=1 accountcode= context=default devices=2 extensions.conf: [general] static=yes writeprotect=yes TRUNK=CAPI [default] exten = s,1,Answer exten = s,2,Background(welcome) I'm expecting the Asterisk welcome-message??? But instead comes the hangup? Any solutions would be great Thanks Tobi ___ 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
RE: [Asterisk-Users] FXO impedance matching
Rich Adamson wrote: From: Nik Martin [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] FXO impedance matching Date: Wed, 23 Jun 2004 11:02:00 -0500 To: [EMAIL PROTECTED] Michael Welter wrote: Jason A. Pattie wrote: Robert Hajime Lanning wrote: Echo echo ech ech ec ec e e . . :) quote who=[EMAIL PROTECTED] What's the importance of the impedance matching in a FXO interface ? My experience is with excessive buzz and hum on the line. When I plug a vintage Western Electric phone into the line, there is no buzz or hum because the phone has its own impedance matching circuitry. When I plug my ATT 954 set into the line, I hear a lot of hum. I'm told the X100P does not have impedance matching. Rich Adamson is the fellow to talk with about impedance. Apparently the hum on my lines is caused by a partial ground on either the tip or ring (or both) wire. If both leads have the same resistance to ground (matched) then there is no hum. I don't experience echo with the buzz and hum. I've been told that echo is caused when the circuit goes from four wire to two wire. I'm trying to locate a schematic of an impedance matching circuit so I can breadboard a device but haven't found one so far. I anyone has experience with this I invite him to reply. Mike If you KNOW the impedances of the two lines, a simple impedance matching transformer available at any electronics distributer (Mouser, Digi-Key, etc) carries many differdnt types, that are just for this purpose. Nik, Could you pass along something more specific then 'any electronics dist'? The old Western Electric repeat coils (transformer) are no longer available, and research with several manufacturers did not turn up anything usable. An ordinary two-winding transformer won't work as it does not pass the DC (supervision) component. If you know of a four-winding 1:1 transformer, please enlighten us. Several are looking for such a component. The original posting was oriented towards the x100p card (which apparently does not have any form of impedence matching support), while Mike's posting relates to a channel bank where the manufacturer states the fxo interface is high impedence (1,000 ohms) unbalanced with no impedence matching on board. In Mike's case building an impedence matching interface for the channel bank is highly likely to improve the noise/hum/buzz that he's getting on pstn lines. Not sure the same is going to be true for the x100p though. Rich http://www.digikey.com/scripts/DkSearch/dksus.dll?Criteria?Ref=25939Site=US Cat=33096753 I may have misspoken, but these are T1 specific matching/isolation transformers. Nik ___ 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
Re: [Asterisk-Users] Latest CVS fax detection grandstream bug
Hi, I have finally tracked down this problem a bit more. The problem is not actually related to the Grandstream phones - just they were the only obvious exhibitors of the problem. The problem seems to be that when faxdetect is set in zapata.conf, asterisk does not inform the sip phones that a voice call has started actually been answered. The console shows this: -- Executing Dial(SIP/1000-b180, Zap/1/0405152568||m) in new stack -- Called 1/0405152568 -- Started music on hold, class 'default', on SIP/1000-b180 (mobile ringing answered, but SIP phone still playing hold music) -- Stopped music on hold on SIP/1000-b180 (hungup sip phone) -- Hungup 'Zap/1-1' == Spawn extension (local, 0405152568, 1) exited non-zero on 'SIP/1000-b180' In this case, not even my fantastic Cisco phone can hold or transfer the call. I still thinks it is ringing. If I remove the faxdetect line, I get the following behaviour: -- Remote UNIX connection -- Executing Dial(SIP/1000-ac04, Zap/1/0405152568||m) in new stack -- Called 1/0405152568 -- Started music on hold, class 'default', on SIP/1000-ac04 -- Zap/1-1 answered SIP/1000-ac04 (phone mobile ringing, answered mobile call) -- Stopped music on hold on SIP/1000-ac04 -- Hungup 'Zap/1-1' == Spawn extension (local, 0405152568, 1) exited non-zero on 'SIP/1000-ac04' This time, it is all good. The Cisco phone realises the call is in progress and works perfectly. Setting faxdetect=no causes everything to behave absolutely properly, with no problems whatsoever. For reference sake - upgrading the grandstream phone to the 1.5.0 firmware has meant that it now behaves similarly to the Cisco phone and carries audio to the handset even during the non-ringing stage. Any help on what to do from here would be useful. Thanks, Andrew _ Andrew Yager Real World Technology Solutions Real People, Real SolUtions (tm) ph: (02) 9945 2567 fax: (02) 9945 2566 mob: 0405 15 2568 http://www.rwts.com.au/ _ On 25/06/2004, at 7:16 PM, Andrew Yager wrote: As a follow up to my previous post, I have now identified what is causing the bug with the grandstream phones. When the line faxdetection=incoming is in the zapata.conf file, the grandstream phones will not ring, nor connect a call to the zaptel interface. Can anyone else confirm this bug? I'm going to play with the different options (incoming/outgoing/both) to see if it makes a difference. Thanks, Andrew _ Andrew Yager Real World Technology Solutions Real People, Real SolUtions (tm) ph: (02) 9945 2567 fax: (02) 9945 2566 mob: 0405 15 2568 http://www.rwts.com.au/ _ ___ 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
Re[2]: [Asterisk-Users] Transfer - to your own number
Dear Philipp, I see that you are from Germany. I would like to ask you about the configuration for Caller ID and Zone Data (zaptel.conf/loadzone/defaultzone). I am asking you that because our standards in Bulgaria are similar to German because our equipment is Siemens. Best Regards, Miroslav Nachev ___ 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
RE: [Asterisk-Users] X101P on a UK BT line ---- txgain issue
I get a problem with what appears to be a slow oscillation on the line if the rxgain + txgain adds up to more than -1db. If I use rxgain=-1.0 and txgain=0.0, it doesn't oscillate but the levels are far too low. The card is an X100P. The oscillation (even on the standard built-in Asterisk echo test) comes over as a loud hiss and crackle at about 1-2 per second making the line unusable. I have tried the settings below and it is dreadful. Any ideas? I am using yesterday's CVS Head. Peter -Original Message- From: Chris Bond [mailto:[EMAIL PROTECTED] Sent: 24 June 2004 18:03 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] X101P on a UK BT line txgain issue I am finding that I have to increase the txgain in zapata.conf to 8 when my X101P is connected to my BT phone line, otherwise people can hardly hear me. This then gives echo issues. Im having the same issue so far im on rxgain=2.0 and txgain=6.0. Seems to work perfectly apart from the echo issue. Im just about to checkout the latest cvs and apply the echotraining=800 Kind Regards, Chris Bond ___ 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 This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you. ___ 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] SIP/IAX to PSTN setup time
Hi, I have started some users terminating calls from my asterisk server to the PSTN through a couple of termination providers. The biggest problem I am having is the time it takes to initially set the call up. It regularly exceeds twenty seconds. I can work around this with failing over to another provider or increasing the timeout but people are used to call setup times of 5 to 10 seconds. I imagine this is a fairly common situation. Does anyone know the reason for the large setup time and/or how to reduce it? Aaron __ Do you Yahoo!? Yahoo! Mail Address AutoComplete - You start. We finish. http://promotions.yahoo.com/new_mail ___ 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] problems compiling shadydial-asterisk on gentoo
hello there: did some one compiled shadydial with asterisk on gentoo successfully, if some one plz help me I am getting compilation errors during asterisk compilation after replacing the files provided with shadydial thank you here is my log, please help gcc -pipe -I=/usr/local/pgsql/include -pipe -Wall -Wstrict-prototypes -Wmissing -prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GN U_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\CVS-HEAD- 05/19/04-04:15:26\ -DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\ -DASTLIB DIR=\/usr/lib/asterisk\ -DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\ /var/run\ -DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk \ -DASTCONFPATH=\/etc/asterisk/asterisk.conf\ -DASTMODDIR=\/usr/lib/asterisk /modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN -DNEW_PRI_HANGUP -Wno-missing-prototypes -Wno-missing-declarations -DZAPATA_ PRI -DIAX_TRUNKING -DCRYPTO -fPIC -c -o chan_agent.o chan_agent.c chan_agent.c: In function `agent_hangup': chan_agent.c:566: warning: implicit declaration of function `ast_say_digit_str' chan_agent.c: In function `agent_new': chan_agent.c:736: warning: assignment from incompatible pointer type chan_agent.c:754: error: too many arguments to function `ast_queue_frame' chan_agent.c: At top level: chan_agent.c:787: warning: function declaration isn't a prototype make[1]: *** [chan_agent.o] Error 1 make[1]: Leaving directory `/usr/src/cvs-src/asterisk/channels' make: *** [subdirs] Error 1 Sent via the WebMail system at convergence.com.pk ___ 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] Manager originate command from SIP to Zap not working
I'm running Asterisk CVS-HEAD-06/07/04. When I try to originate a call from a SIP channel to a ZAP channel using manager everything works up to the point when I pickup the ringing ZAP phone. Originate ZAP to SIP works fine. This is the error from my asterisk debug. Jun 25 09:41:26 WARNING[770069]: chan_sip.c:1718 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 4/4) I don't have any problem calling directly(using only the phones) from SIP to ZAP. My SIP phone is a Grandstream and I'm using ULAW and ALAW only. This is from my sip.conf file [roger] type=friend disallow=all allow=ulaw allow=alaw host=dynamic username=roger [EMAIL PROTECTED] context=home callerid=roger 805 This is the manager command sent: --- Action: Originate Channel: SIP/ROGER Callerid: [EMAIL PROTECTED] Exten: 820 Context: home Priority: 1 These are the manager events received: -- Event: Newchannel Channel: SIP/ROGER-9f51 State: Down Callerid: Uniqueid: 1088170907.40 Event: Newcallerid Channel: SIP/ROGER-9f51 Callerid: [EMAIL PROTECTED] Uniqueid: 1088170907.40 Event: Newchannel Channel: SIP/ROGER-9f51 State: Ringing Callerid: [EMAIL PROTECTED] Uniqueid: 1088170907.40 Event: Newstate Channel: SIP/ROGER-9f51 State: Up Callerid: [EMAIL PROTECTED] Uniqueid: 1088170907.40 Response: Success Message: Originate successfully queued Event: Newexten Channel: SIP/ROGER-9f51 Context: home Extension: 824 Priority: 1 Uniqueid: 1088170907.40 Event: Newchannel Channel: Zap/9-1 State: Rsrvd Callerid: marvin 824 Uniqueid: 1088170908.41 Event: Newstate Channel: Zap/9-1 State: Ringing Callerid: [EMAIL PROTECTED] Uniqueid: 1088170908.41 Event: Newstate Channel: Zap/9-1 State: Up Callerid: [EMAIL PROTECTED] Uniqueid: 1088170908.41 Event: Link Channel1: SIP/ROGER-9f51 Channel2: Zap/9-1 Uniqueid1: 1088170907.40 Uniqueid2: 1088170908.41 Event: Unlink Channel1: SIP/ROGER-9f51 Channel2: Zap/9-1 Uniqueid1: 1088170907.40 Uniqueid2: 1088170908.41 Event: Hangup Channel: Zap/9-1 Uniqueid: 1088170908.41 Cause: 0 Event: Hangup Channel: SIP/ROGER-9f51 Uniqueid: 1088170907.40 Cause: 1 ___ 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] SS7 to Pri
Does anyone know of a device that will take an SS7 link and convert it to a PRI? -- respectfully, Joseph - (606) 477-2355 x140 --= ___ 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
Re: [Asterisk-Users] Speex
thanx that works here is wat i want to do sip (xlite)-*(iax2)(codec speex)-internet*(iax2)--pstn when i do that all i get is noise in the other end..am sure am doing something stupid... any help would be appreciated ( i want to meassure the bandwidth will be using the program called rate) sriram - Original Message - From: Jon Radon [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, June 24, 2004 12:55 PM Subject: RE: [Asterisk-Users] Speex Inband dtmf does not work with speex(only ulaw). Switch your dtmf mode to rfc2833. :) From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sola 2000 Sent: Thursday, June 24, 2004 11:16 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Speex I am a new to asterisk, i wanted to test the opensource codec speex i have installed speex, and recompiled asterisk i can see the speex_codec.so getting loaded i have xten lite, i used the registry patch ( http://bugs.digium.com/bug_view_page.php?bug_id=918 ) but still when i use xten lite i get the following errors Jun 24 10:46:15 WARNING[-1305486416]: codec_speex.c:167 speextolin_framein: Out of buffer space Jun 24 10:46:15 WARNING[-1305486416]: dsp.c:1478 ast_dsp_process: Unable to process inband DTMF on 512 frames am i doing something wrong? any pointers is helpful thanx sriram ___ 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
Re: [Asterisk-Users] Latest CVS fax detection grandstream bug
OMG - I couldn't be happier to see this!
On Friday 25 June 2004 07:39, Andrew Yager wrote:
The problem seems to be that when faxdetect is set in zapata.conf,
asterisk does not inform the sip phones that a voice call has started
actually been answered.
I've been plaqued by a different problem in CVS HEADs since June 2 - turns
out, it's the same problem!
Andrew, try this:
With faxdetect enabled ('both' or 'incoming'), make an SIP-ZAP-PSTN call.
While that call is active, type 'show channels' at the * CLI.
You will see the STATE of your ZAP and SIP channels are reported incorrectly
as DIALING and RING (the state whould be UP).
I was having a problem where Uniden phones would drop any call with the PSTN
after 3 minutes - cuz they didn't realize that they were on a call.
So FAXDETECT _is_ the culprit. Turning it off, makes this problem go away.
BUG# 0001909 has been updated with this info.
Turns out I'm not crazy, you made my day.
--
..
Ryan Courtnage
Coalescent Systems
403.830.9410
[EMAIL PROTECTED]
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RE: [Asterisk-Users] X101P on a UK BT line ---- txgain issue
I do get echo, lots of it, I am waiting until the new patch they are all on about on the list gets into a stable release, then I will upgrade and see if that does the trick. Not likely the patch will get applied to the Stable release since its been stated several times that's all but dead. The patch didn't seem to work for me. I am told that some of the echo may be to do with a mismatch in the impedance with the BT line. There are at least several sources of echo, which have been noted several times in the past six months or so: a. echo can in * not functioning correctly in some circumstances (patch) b. mismatched x100p - pstn line c. pstn line problems (eg, imbalance between tip/ring and ground) d. 2-wire to 4-wire conversion along the end-to-end voice path Any single pstn line could have one or more of those happening. Problem is do we really want BT messing with gain there end and impedance cos it might mess our ADSL lines up =) I know im on the limits. The telco isn't going to mess with changing impedance on their stuff as that would require re-engineering their entire outside plant (cables), line interfaces within the CO, etc. You couldn't pay them enough money to do it. Its also highly unlikely they have any technician adjustable transmission level adjustments on ordinary pstn CO line interfaces, as those are engineered to 'standards', and manufacturing engineers typically don't want support technicians to muck with those for lots of very valid reasons. I'm in the US and don't have any real clue what the UK standards are for impedance, transmission levels, etc. (It would be somewhat interesting to here from someone who knows for sure what those are.) The x100p card (from digium) uses the Silicon Labs ( www.silabs.com ) 3012 chip to interface with the pstn line, the 3021 chip to interface the 3012 (analog-to-digital converter) to the Tigerjet PCI controller. The 3012 is responsible for matching pstn line impedance. The spec sheets at their site tend to suggest the 3012 was built to interface to 600 ohm pstn lines and is not adjustable/setable to other values. If the UK pstn lines are not 600 ohm impedance, then its unlikely the x100p is going to properly match up with UK lines from an impedance matching perspective. However, imedance mismatches have to be rather dramatic to cause a lot of echo. The tdm fxo module uses the 3019 and 3050 chipset, where the 3019 pstn line interface chip has many different pstn line impedance settings including 600, 900, 270, 220, 370, 320, 275, 120, 350, etc, ohms. Have no clue which countries use which settings, but obviously Silicon Labs intended this chip set to operate in different countries, whereas the 3012 spec sheet doesn't seem to support those objectives. So, backing into exactly what is causing the echo in the three UK cases noted yesterday on this list... - not likely to be d (2-wire to 4-wire conversion along the end-to-end voice path) as that would impact all telco users, not just * users. - item c (pstn line problems) can contribute to echo depending upon how bad the pstn line actually happens to be. Most telco's have the equipment to measure line quality, however most will stop at the cable entrance to your home/business, leaving you to guess at what's happening inside. - item b (mismatched x100p - pstn line) can contribute, but without knowing the exact specs (and probably more info from Silicon Labs), its impossible to guess at this one. - item a (echo can in *) is still a very real possibility, and Mark is about the only person I know of that has the knowledge of the spec's and * to weight in on this one. One of the methods that I used to help determine whether my tdm echo was a pstn line or * issue was to eval echo on three different pstn lines using the exact same physical * port (x100p). If the lines are clean from an analog phone perspective (eg, no hum, no noise) and the lines cause equal echo when used with *, then its highly likely the issue is either a or b. If the answers to b rule out impedance mismatches, then a is likely. Why? Your not likely to have multiple pstn lines with exactly the same fault. (Could happen but not likely.) Second, if your pstn line is also a DSL line with appropriate filters, the DSL line is far more critical of pstn imperfections then is the * interface to the analog pstn line. In the majority of cases, a poor pstn line will cause significantly more DSL problems and probably a total DSL failure way before the pstn analog path is impacted. Third, I obtained a Mediatrix 1204 sip gateway to displace the x100p's to eliminate the echo. It worked fine (zero echo) on all three lines, essentially proving the pstn lines were not the issue causing echo. After that test, there was nothing left other then * echo can functions (which turned out to be the case with my tdm fxo _and_ CO operation, and resulted in this recent patch). Given that Mark
RE: [Asterisk-Users] Really basic stuff :(
FWD only supports ULAW comment out the line allow=GSM in the general section of the iax.conf Nonsense - FWD *does* permit the use of GSM. At http://www.freeworlddialup.com/advanced/iax it currently says: Q: what codecs does FWD support? A: All codecs will pass through FWD, but FWD servers support G711 (PCMU) only. When questioned recently, Ed Guy confirmed the above and indicated they were not going to support gsm (didn't want to deal with transcoding on their servers). ___ 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
RE: [Asterisk-Users] SS7 to Pri
A switch ? ;-) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Joseph Sent: Friday, June 25, 2004 4:25 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] SS7 to Pri Does anyone know of a device that will take an SS7 link and convert it to a PRI? -- respectfully, Joseph - (606) 477-2355 x140 --= ___ 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
Re: [Asterisk-Users] Which Linux ?
Kevin Walsh wrote: Freddy Setiawan [EMAIL PROTECTED] wrote: Hi there, linux got so many distro, but which one that have more compability with the Asterisk? My Asterisk server is running on Gentoo, with the 2.6.7-gentoo-r5 kernel. The Zaptel drivers work nicely too. I should think that any of the GNU/Linux distros would work. Selecting a distro is usually just a matter of personal preference; They all run the same kernel and usually have the same tools, libraries and compiler etc. Kevin, I am about to build my first asterisk box, I want to make it Gentoo based with a 2.4 kernel. When performing the initial system emerge on the Gentoo bos, are there any special USE flags you would recommend setting to make the asterisk build go smoothly? Ed ___ 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] Re: SS7 to Pri
Try looking into FastComm. They do C7 to E-ISDN. Kurt __ Do you Yahoo!? Yahoo! Mail Address AutoComplete - You start. We finish. http://promotions.yahoo.com/new_mail ___ 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
RE: [Asterisk-Users] Leave one call to pick up another
Hadar Pedhazur wrote: Andrew Thompson wrote: Eric Wieling wrote: How is this different from the way standard call waiting works when provided from your telco? Um, he actually has two phone lines, not just one that he's flash-ing back and forth between. If he hangs up the line, does the second call not continue ringing? I take it once he hangs up the line both calls are gone? I can't help with any solution, but I can add my voice describing another symptom of this exact problem, in direct response to your last question. I have two lines, each handled by a Digium X100P card. If I am on the phone (whether I initiated a call, or received one, whether it uses one of the POTS lines or whether it's a VoIP call), if another call comes in, I hear the Call Waiting signal. If I simply hang up the current call, I lose _both_ calls. Meaning, the phone does not start ringing with the pending call any longer. This is _not_ the same behavior that I had with the same exact phone, when it was connected to the POTS line directly. Hanging up on the current call yields a ringing for the second call, after a second or two delay... P.S. If I flash the call, I can indeed speak to the second caller, and bounce back and forth between the calls, so I get the same behavior that the original poster (Brian Capouch) described. Unless someone has a configuration that doesn't exhibit this behavior, I'd say it's time for some Mark Spencer branded Raid* I can attempt to recreate this behavior tonight, for commenting on a bug report. Can the original poster open a bug at bugs.digium.com (if they haven't already)? *(bug spray) ___ 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] forced ring on dial?
I am routing outgoing calls through a sip gateway. The calls go through no problem, however the ringing in the callers ear begins as soon as the last digit is dialed. This has two nasty side effects. First, the caller hears 1-2 more rings than the callee. Second, and more importantly, if the callee's line is busy, the caller continues to get hear ringing, even though the gateway has returned a busy indication. The whole problem seems to be * is not waiting for the proper call progress signal from the sip gateway before giving the caller a ring indication. Is there any way to control this so that * waits for call progress from the gateway before giving the caller the appropriate indication, i.e., ring or busy tone? I have been told this is a result of setting * to forced ring and this should be turned off, but of course, on * it is probably called something else. Thanks Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 ___ 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] NO AUDIO IN BOTH DIRECTIONS
hello all, I am having a trouble with Audio using h.323 channel... I am doing this Call comes into cisco 5300 and is sent to Asterisk, asterisk catch call with h.323 driver and send call to a SoftSwitch that routes the call, I can see log debug telling me, CALLED XXX, and then RINGING, and I can hear ring tones... but when call is answered, I DONT HEAR ANYTHING... I am using lastest ASTERISK download somebody can help me to solve this problem thanks..!!
RE: [Asterisk-Users] X101P on a UK BT line ---- txgain issue
BT do occasionally tweak up line gain a bit if you keep complaining that you have a modem and are getting a very slow speed. I have had a 40k V90 come up to 48k after this was done on my line at home (System X switch). You have to get a sympathetic engineer though - frequently they will tell you that it can't be done. Peter -Original Message- From: Rich Adamson [mailto:[EMAIL PROTECTED] Sent: 25 June 2004 14:33 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] X101P on a UK BT line txgain issue I do get echo, lots of it, I am waiting until the new patch they are all on about on the list gets into a stable release, then I will upgrade and see if that does the trick. Not likely the patch will get applied to the Stable release since its been stated several times that's all but dead. The patch didn't seem to work for me. I am told that some of the echo may be to do with a mismatch in the impedance with the BT line. There are at least several sources of echo, which have been noted several times in the past six months or so: a. echo can in * not functioning correctly in some circumstances (patch) b. mismatched x100p - pstn line c. pstn line problems (eg, imbalance between tip/ring and ground) d. 2-wire to 4-wire conversion along the end-to-end voice path Any single pstn line could have one or more of those happening. Problem is do we really want BT messing with gain there end and impedance cos it might mess our ADSL lines up =) I know im on the limits. The telco isn't going to mess with changing impedance on their stuff as that would require re-engineering their entire outside plant (cables), line interfaces within the CO, etc. You couldn't pay them enough money to do it. Its also highly unlikely they have any technician adjustable transmission level adjustments on ordinary pstn CO line interfaces, as those are engineered to 'standards', and manufacturing engineers typically don't want support technicians to muck with those for lots of very valid reasons. I'm in the US and don't have any real clue what the UK standards are for impedance, transmission levels, etc. (It would be somewhat interesting to here from someone who knows for sure what those are.) The x100p card (from digium) uses the Silicon Labs ( www.silabs.com ) 3012 chip to interface with the pstn line, the 3021 chip to interface the 3012 (analog-to-digital converter) to the Tigerjet PCI controller. The 3012 is responsible for matching pstn line impedance. The spec sheets at their site tend to suggest the 3012 was built to interface to 600 ohm pstn lines and is not adjustable/setable to other values. If the UK pstn lines are not 600 ohm impedance, then its unlikely the x100p is going to properly match up with UK lines from an impedance matching perspective. However, imedance mismatches have to be rather dramatic to cause a lot of echo. The tdm fxo module uses the 3019 and 3050 chipset, where the 3019 pstn line interface chip has many different pstn line impedance settings including 600, 900, 270, 220, 370, 320, 275, 120, 350, etc, ohms. Have no clue which countries use which settings, but obviously Silicon Labs intended this chip set to operate in different countries, whereas the 3012 spec sheet doesn't seem to support those objectives. So, backing into exactly what is causing the echo in the three UK cases noted yesterday on this list... - not likely to be d (2-wire to 4-wire conversion along the end-to-end voice path) as that would impact all telco users, not just * users. - item c (pstn line problems) can contribute to echo depending upon how bad the pstn line actually happens to be. Most telco's have the equipment to measure line quality, however most will stop at the cable entrance to your home/business, leaving you to guess at what's happening inside. - item b (mismatched x100p - pstn line) can contribute, but without knowing the exact specs (and probably more info from Silicon Labs), its impossible to guess at this one. - item a (echo can in *) is still a very real possibility, and Mark is about the only person I know of that has the knowledge of the spec's and * to weight in on this one. One of the methods that I used to help determine whether my tdm echo was a pstn line or * issue was to eval echo on three different pstn lines using the exact same physical * port (x100p). If the lines are clean from an analog phone perspective (eg, no hum, no noise) and the lines cause equal echo when used with *, then its highly likely the issue is either a or b. If the answers to b rule out impedance mismatches, then a is likely. Why? Your not likely to have multiple pstn lines with exactly the same fault. (Could happen but not likely.) Second, if your pstn line is also a DSL line with appropriate filters, the DSL line is far more critical of pstn imperfections then is the * interface to the analog pstn line. In the majority of cases, a poor pstn line will
[Asterisk-Users] IAX2 authentication confusion
We spent some time yesterday trying to understand how IAX2 authentication works, and now I'm confused... Let's say that the receiving end has this entry in their iax.conf file: [remote-site] type=user secret=foo auth=md5 context=incoming host=dynamic The way I see it, there are two ways to initiate an outbound IAX2 connection to this system: 1) Use Dial, as in: Dial(IAX2/remote-site:[EMAIL PROTECTED]/extension) In this mode, the IAX2 setup message contains a USERNAME (remote-site), and the receiving system compares it to the entity name in iax.conf, before comparing the secret. This is fine. 2) Use Dial and iax.conf, as in: Dial(IAX2/local-site/extension) and in remote-site's iax.conf: [local-site] type=peer secret=foo auth=md5 host=local-site.domain.com In this mode, the IAX2 setup message _does not_ contain a USERNAME, and the receiving system somehow manages to find the proper entry and authenticate the connection. However, the only way that I could see that this would be possible is that the receiving system is comparing the supplied secret against all secrets in it's iax.conf file to try to find a match. I don't know how that is possible using md5 authentication, but even if it is, I don't particularly like it. That means someone can connect to my Asterisk server over IAX if they can guess _any_ secret that happens to be in my iax.conf file. I really would prefer to not embed the username/password information in my Dial commands (that way it doesn't have to be duplicated in multiple contexts, and it's more logically stored in iax.conf anyway), but unless I do that Asterisk does not send a USERNAME to the receiving server and thus the authentication is not very secure. Is there a reason why Asterisk allows incoming IAX2 calls without USERNAME specified at all? ___ 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
Re: [Asterisk-Users] Asterisk with PostgreSQL
Caleb Kow wrote: Here we go: [EMAIL PROTECTED] root]# netstat -ap Active Internet connections (servers and established) Proto Recv-Q Send-Q Local Address Foreign Address State PID/Program name tcp0 0 *:32768 *:* LISTEN 3221/ tcp0 0 *:imaps *:* I didn't see Postgres running but did notice mysql. They run on different ports so that not a problem unless you are mistaking one for the other. Another poster stated that Postgres runs local socekts by default and that a change in the config is needed to get it working with TCP/IP. I'd investigate that as that's what it looks like. I hope this helps. -- Linux Home Automation Neil Cherry[EMAIL PROTECTED] http://home.comcast.net/~ncherry/ (Text only) http://linuxha.sourceforge.net/ (SourceForge) http://hcs.sourceforge.net/ (HCS II) ___ 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
Re: [Asterisk-Users] Works for a while and then rings off hook
On Wednesday 23 June 2004 14:28, Joseph Finley wrote: I have a X100P that works great for a couple days maybe even a week and then outside callers say my phone just rings and rings. When I try to dial out during this period, it waits dead air You aren't alone (i've spoken to several people on the IRC that have had this). Most people will hear static on the line and not dead-air. I've been fortunate enough to experience both silence and static (with tdm400p + fxo). It's tough to duplicate this one, and even tougher to experiment and run tests when it happens, especially when it only happens every other day at a client site. If you have the luxury of being able to reproduce this in a non-production environment, _many_ people would benefit if you submitted a bug. (bugs.digium.com) FYI - it seems most people eventually get around the problem by moving the card to different pci slots, or even changing computers all-together. Of course, this doesn't help in determining the root cause of the problem and fixing it. Cheers and then Allison says Goodbye and hangs up. I have to stop *, modprobe -r wcfxo, modprobe wcfxo, and ztcfg then run safe_asterisk. After that it works for a day or a little longer. This never happened on the previous server, but I was running an older version of * at the time. I'm running .9.0. Upgrading it from previous versions to .9.0 didn't fix it either. Any suggestions would be fine. I looked for similar posts, but the other posts we're slightly different from my circumstances. Joe ___ 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 -- .. Ryan Courtnage Coalescent Systems 403.830.9410 www.voxbox.ca ___ 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
RE: [Asterisk-Users] forced ring on dial?
I'd be willing to bet you have r in your dialout string (i.e.
something like: Dial(${TRUNK}/${EXTEN},120,r)...
Get rid of that in the outbound dialing, and you otta be ok.
Jeremy Jones
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Bruce Komito
Sent: Friday, June 25, 2004 8:52 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] forced ring on dial?
I am routing outgoing calls through a sip gateway. The calls
go through
no problem, however the ringing in the callers ear begins as
soon as the
last digit is dialed. This has two nasty side effects.
First, the caller
hears 1-2 more rings than the callee. Second, and more
importantly, if
the callee's line is busy, the caller continues to get hear
ringing, even
though the gateway has returned a busy indication.
The whole problem seems to be * is not waiting for the proper call
progress signal from the sip gateway before giving the caller a ring
indication. Is there any way to control this so that * waits for call
progress from the gateway before giving the caller the appropriate
indication, i.e., ring or busy tone? I have been told this
is a result of
setting * to forced ring and this should be turned off, but
of course,
on * it is probably called something else.
Thanks
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
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Re: [Asterisk-Users] SS7 to Pri
On Fri, 2004-06-25 at 09:24, Joseph wrote: Does anyone know of a device that will take an SS7 link and convert it to a PRI? I think it's called an ILEC or CLEC. 8-) -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ 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
Re: [Asterisk-Users] forced ring on dial?
On Fri, 2004-06-25 at 09:52, Bruce Komito wrote: I am routing outgoing calls through a sip gateway. The calls go through no problem, however the ringing in the callers ear begins as soon as the last digit is dialed. This has two nasty side effects. First, the caller hears 1-2 more rings than the callee. Second, and more importantly, if the callee's line is busy, the caller continues to get hear ringing, even though the gateway has returned a busy indication. The whole problem seems to be * is not waiting for the proper call progress signal from the sip gateway before giving the caller a ring indication. Is there any way to control this so that * waits for call progress from the gateway before giving the caller the appropriate indication, i.e., ring or busy tone? I have been told this is a result of setting * to forced ring and this should be turned off, but of course, on * it is probably called something else. Remove the r option from your Dial line. -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ 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] Asterisk SIP
Good morning all, I'm setting up Asterisk for the first time with no prior PBX experience. I'm following Andy Powell's 'Getting Started with Asterisk' (http://www.automated.it/guidetoasterisk.htm). This is my second time through that document - as I did something weird the first time and really upset it somehow - and I wanted to ask a few general questions of the list. First, a little on what I'm trying to do: I need to setup the PBX to answer on multiple 'lines' (I use that word with trepidation as I'm not sure if it's the right term in the absence of modems actual lines) and play a brief message identifying itself as the 'line' connected to. The originator of that call will be a softphone. Before rolling this out to my lab, I'm trying to work out the proper config on my laptop. Therein I have Windows XP w/ VMware - Red Hat 7.3 is running in a VMware session. My connection to the net is NAT'd. My internal IPs for both XP Linux are of the 192.168.x.x private network variety. I have the Xten X-Lite softphone on XP to test with. (I also have another called SJPhone, but haven't done much with that past installing it.) I've configured a number for that through freeworlddialup.com. X-Lite appears to be working fine. At least I can dial their echo test numbers without a problem and get the expected responses. So the questions: 1. A general will this work? (vmware linux, same pc as phone, NAT'd addresses,etc) 2. Has anyone done it this way before and/or followed Andy Powell's doc, and have any suggestions or things to watch out for? 3. Reading the various published SIP documentation (Ubiquity's 'Understanding SIP' for instance), it seems like freeworlddialup is acting as Registrar, Proxy Redirect server. Is that accurate? 4. How do I tell the freeworlddialup registrar 'where' to find my PBX? Should I setup an account from it - like I did with the softphone on XP - so it will have a 'phone number' of its own? Or is the proxy/redirect server expecting to talk to the Asterisk PBX in some other way? I appreciate any and all responses. Please cc my email address directly on replies as I have the list configured in digest mode to stem the flow a bit and don't want to miss any of them in the mix. Thanks to one and all Fletcher Bonds Operations Software Tester TeleCommunication Systems, Inc. (TCS) Enabling Convergent Technologies www.telecomsys.com [EMAIL PROTECTED] ___ 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] SIP extension outside of IP tables firewall
I have an Asterisk PBX on the private lan, which is protected from the public Internet with a Linux iptables machine. The firwall is it's own seperate box running NAT with SPI. I want to drop a SIP phone at my brothers house, and have it be an extension off my Asterisk box. I've been looking around at some FAQ info on forwarding ports, and also looked at siproxd. Anyway, I'm posting this, because this seems like a really standard thing people would want to do with Asterisk/Iptables, so is there a standard solution that I can apply? ___ 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
Re: [Asterisk-Users] Re: SS7 to Pri
Thanks, that looks interesting. On Fri, 2004-06-25 at 10:47, Kurt wrote: Try looking into FastComm. They do C7 to E-ISDN. Kurt __ Do you Yahoo!? Yahoo! Mail Address AutoComplete - You start. We finish. http://promotions.yahoo.com/new_mail ___ 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 -- respectfully, Joseph - (606) 477-2355 x140 --= ___ 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
Re: [Asterisk-Users] Latest CVS fax detection grandstream bug
On 2004.06.25 06:39 Andrew Yager wrote: The problem seems to be that when faxdetect is set in zapata.conf, asterisk does not inform the sip phones that a voice call has started actually been answered. Setting faxdetect=no causes everything to behave absolutely properly, with no problems whatsoever. For reference sake - upgrading the grandstream phone to the 1.5.0 firmware has meant that it now behaves similarly to the Cisco phone and carries audio to the handset even during the non-ringing stage. I use current CVS, a Grandstream BT100 (1.5.0), and have faxdetect=incoming, and I have been experiencing no problems of the type you mention. Lee. ___ 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
RE: [Asterisk-Users] H.323 - NO AUDIO IN BOTH DIRECTIONS
Just checking that you have installed the proper versions of both OpenH323 and PWLib, as mentioned in ~/asterisk/channels/h232/README, and have rebuilt asterisk after those installations as specified? If so, then you are having the same problem I'm experiencing: no audio on H.323. I'm also connecting through a Cisco 5300. I'm just generating audio in one direction: outbound from asterisk - I hear nothing. This used to work I'm pretty sure... There is an outstanding bug report covering H.323 problems (#1334), not sure what the current status is. Cheers Scott Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com http://www.evtmedia.com/ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sebastian Nocetti Sent: Friday, June 25, 2004 7:55 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] NO AUDIO IN BOTH DIRECTIONS hello all, I am having a trouble with Audio using h.323 channel... I am doing this Call comes into cisco 5300 and is sent to Asterisk, asterisk catch call with h.323 driver and send call to a SoftSwitch that routes the call, I can see log debug telling me, CALLED XXX, and then RINGING, and I can hear ring tones... but when call is answered, I DONT HEAR ANYTHING... I am using lastest ASTERISK download somebody can help me to solve this problem thanks..!! ___ 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
Re: [Asterisk-Users] SS7 to Pri
Joseph wrote: Does anyone know of a device that will take an SS7 link and convert it to a PRI? It could be * - depending which version of * you have. :-) Regards, Steve ___ 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
RE: [Asterisk-Users] SS7 to Pri
Didn't I hear a week or two ago (on this list) that someone had taken it upon themselves to write an asterisk module for the openss7-modified digium t1/e1 cards? Maybe soon asterisk'll do it. Jeremy Jones -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Friday, June 25, 2004 9:28 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] SS7 to Pri On Fri, 2004-06-25 at 09:24, Joseph wrote: Does anyone know of a device that will take an SS7 link and convert it to a PRI? I think it's called an ILEC or CLEC. 8-) -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ 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
Re: [Asterisk-Users] Speex
thanx that works here is wat i want to do sip (xlite)-*(iax2)(codec speex)-internet*(iax2)--pstn when i do that all i get is noise in the other end..am sure am doing something stupid... any help would be appreciated ( i want to meassure the bandwidth will be using the program called rate) sriram - Original Message - From: Jon Radon [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, June 24, 2004 12:55 PM Subject: RE: [Asterisk-Users] Speex Inband dtmf does not work with speex(only ulaw). Switch your dtmf mode to rfc2833. :) From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sola 2000 Sent: Thursday, June 24, 2004 11:16 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Speex I am a new to asterisk, i wanted to test the opensource codec speex i have installed speex, and recompiled asterisk i can see the speex_codec.so getting loaded i have xten lite, i used the registry patch ( http://bugs.digium.com/bug_view_page.php?bug_id=918 ) but still when i use xten lite i get the following errors Jun 24 10:46:15 WARNING[-1305486416]: codec_speex.c:167 speextolin_framein: Out of buffer space Jun 24 10:46:15 WARNING[-1305486416]: dsp.c:1478 ast_dsp_process: Unable to process inband DTMF on 512 frames am i doing something wrong? any pointers is helpful thanx sriram ___ 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
RE: [Asterisk-Users] Which Linux ?
Ed Brady [EMAIL PROTECTED] wrote: I am about to build my first asterisk box, I want to make it Gentoo based with a 2.4 kernel. I'm on 2.6.7-gentoo-r6, which I installed today (upgraded from r5). I have found the 2.6 kernel to be a lot better, in my unscientific opinion, than the 2.4 kernel I've used in the past. When performing the initial system emerge on the Gentoo bos, are there any special USE flags you would recommend setting to make the asterisk build go smoothly? I don't set the USE variable at all when I built my Asterisk box. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ 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
Re: [Asterisk-Users] IAX2 authentication confusion
Kevin P. Fleming wrote:
Is there a reason why Asterisk allows incoming IAX2 calls without
USERNAME specified at all?
1) host=dynamic makes no sense in a type=user
2) One sends the username to be used to the peer
On the machine you wish to dial out, you have in your iax.conf:
[peer]
type=peer
host=1.2.3.4
secret=foo
and in that same machine's extensions.conf you have something that looks
like:
Dial,IAX2/[EMAIL PROTECTED]/${EXTEN}
Then on the 'peer' (other) machine you need:
[USER]
type=user
context=incoming
auth=md5
which is cAsE SeNsITiVe. Plus you need the appropriate extension(s) in
this (other) machine's extensions.conf.
Have you bothered to study any of the documentation out there? Start
here: http://www.voip-info.org/
Jeremy McNamara
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Re: [Asterisk-Users] Asterisk SIP
for question 4. You need to register with fwd first, then use registry = command in sip.conf. aaron On Fri, 25 Jun 2004 08:32:07 -0700, Fletcher Bonds [EMAIL PROTECTED] wrote: Good morning all, I'm setting up Asterisk for the first time with no prior PBX experience. I'm following Andy Powell's 'Getting Started with Asterisk' (http://www.automated.it/guidetoasterisk.htm). This is my second time through that document - as I did something weird the first time and really upset it somehow - and I wanted to ask a few general questions of the list. First, a little on what I'm trying to do: I need to setup the PBX to answer on multiple 'lines' (I use that word with trepidation as I'm not sure if it's the right term in the absence of modems actual lines) and play a brief message identifying itself as the 'line' connected to. The originator of that call will be a softphone. Before rolling this out to my lab, I'm trying to work out the proper config on my laptop. Therein I have Windows XP w/ VMware - Red Hat 7.3 is running in a VMware session. My connection to the net is NAT'd. My internal IPs for both XP Linux are of the 192.168.x.x private network variety. I have the Xten X-Lite softphone on XP to test with. (I also have another called SJPhone, but haven't done much with that past installing it.) I've configured a number for that through freeworlddialup.com. X-Lite appears to be working fine. At least I can dial their echo test numbers without a problem and get the expected responses. So the questions: 1. A general will this work? (vmware linux, same pc as phone, NAT'd addresses,etc) 2. Has anyone done it this way before and/or followed Andy Powell's doc, and have any suggestions or things to watch out for? 3. Reading the various published SIP documentation (Ubiquity's 'Understanding SIP' for instance), it seems like freeworlddialup is acting as Registrar, Proxy Redirect server. Is that accurate? 4. How do I tell the freeworlddialup registrar 'where' to find my PBX? Should I setup an account from it - like I did with the softphone on XP - so it will have a 'phone number' of its own? Or is the proxy/redirect server expecting to talk to the Asterisk PBX in some other way? I appreciate any and all responses. Please cc my email address directly on replies as I have the list configured in digest mode to stem the flow a bit and don't want to miss any of them in the mix. Thanks to one and all Fletcher Bonds Operations Software Tester TeleCommunication Systems, Inc. (TCS) Enabling Convergent Technologies www.telecomsys.com [EMAIL PROTECTED] ___ 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
Re: [Asterisk-Users] Asterisk SIP
On Fri, 25 Jun 2004, Fletcher Bonds wrote: 1. A general will this work? (vmware linux, same pc as phone, NAT'd addresses,etc) You'll probably be the first person to try it. I'd guess that it will work, but expect call quality to be impacted because of all the extra scheduling and virtualisation. You will need to make sure that UDP streams from your linux side can get through the Windows with NATTing, and importantly that replies get back. Port 5060 needs to be permanently directed through to Linux. You set another range of ports in rtp.conf (rtpstart and rtpend) - these will also need directing through to Linux. 3. Reading the various published SIP documentation (Ubiquity's 'Understanding SIP' for instance), it seems like freeworlddialup is acting as Registrar, Proxy Redirect server. Is that accurate? With SIP there are all these terms - but practically speaking you send SIP messages somewhere, and get replies. FWD is a place which you talk to using SIP. I guess they are a registry (you can send REGISTER packets). They aren't generally a proxy - usually your audio streams end up going directly to the other endpoint. They do have a proxy that you can use if you need to because of NAT. You might need it. And I don't know what a redirect server does. 4. How do I tell the freeworlddialup registrar 'where' to find my PBX? You put a register line in your sip.conf Should I setup an account from it - like I did with the softphone on XP - so it will have a 'phone number' of its own? Yes - you can open another FWD account, or you can use the one you setup for your XP softphone. Or is the proxy/redirect server expecting to talk to the Asterisk PBX in some other way? FWD is happy to talk to you with SIP. They do also have a test connection using IAX, Asterisk's own protocol - but why not leave that for another day. I appreciate any and all responses. Please cc my email address directly on replies as I have the list configured in digest mode to stem the flow a bit and don't want to miss any of them in the mix. Thanks to one and all Fletcher Bonds Operations Software Tester TeleCommunication Systems, Inc. (TCS) Enabling Convergent Technologies www.telecomsys.com [EMAIL PROTECTED] ___ 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
Re: [Asterisk-Users] X101P on a UK BT line ---- txgain issue
Thanks for the info Rich looks like I'll have to wait for the new FXO module. The impedence in the UK is zcomplex(2) which looks a long way away from a straight 600 ohms. Here is the list of zcomplex impedences Zcomplex(1) = 150 nF // 750 ohms + 270 ohms ( European harmonized, France Telecom Telefonica ) Zcomplex(2) =230 nF // 1050 ohms + 320 ohms ( British Telecom plc ) Zcomplex(3) = 115 nF // 820 ohms + 220 ohms ( Deutsche Telekom AG ) Zcomplex(4) = 310 nF // 620 ohms + 370 ohms ( Telecom New Zealand ) Zcomplex(5) =47 nF // 510 ohms + 150 ohms ( Russian Telecom ) - Original Message - From: Rich Adamson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, June 25, 2004 2:32 PM Subject: RE: [Asterisk-Users] X101P on a UK BT line txgain issue I do get echo, lots of it, I am waiting until the new patch they are all on about on the list gets into a stable release, then I will upgrade and see if that does the trick. Not likely the patch will get applied to the Stable release since its been stated several times that's all but dead. The patch didn't seem to work for me. I am told that some of the echo may be to do with a mismatch in the impedance with the BT line. There are at least several sources of echo, which have been noted several times in the past six months or so: a. echo can in * not functioning correctly in some circumstances (patch) b. mismatched x100p - pstn line c. pstn line problems (eg, imbalance between tip/ring and ground) d. 2-wire to 4-wire conversion along the end-to-end voice path Any single pstn line could have one or more of those happening. Problem is do we really want BT messing with gain there end and impedance cos it might mess our ADSL lines up =) I know im on the limits. The telco isn't going to mess with changing impedance on their stuff as that would require re-engineering their entire outside plant (cables), line interfaces within the CO, etc. You couldn't pay them enough money to do it. Its also highly unlikely they have any technician adjustable transmission level adjustments on ordinary pstn CO line interfaces, as those are engineered to 'standards', and manufacturing engineers typically don't want support technicians to muck with those for lots of very valid reasons. I'm in the US and don't have any real clue what the UK standards are for impedance, transmission levels, etc. (It would be somewhat interesting to here from someone who knows for sure what those are.) The x100p card (from digium) uses the Silicon Labs ( www.silabs.com ) 3012 chip to interface with the pstn line, the 3021 chip to interface the 3012 (analog-to-digital converter) to the Tigerjet PCI controller. The 3012 is responsible for matching pstn line impedance. The spec sheets at their site tend to suggest the 3012 was built to interface to 600 ohm pstn lines and is not adjustable/setable to other values. If the UK pstn lines are not 600 ohm impedance, then its unlikely the x100p is going to properly match up with UK lines from an impedance matching perspective. However, imedance mismatches have to be rather dramatic to cause a lot of echo. The tdm fxo module uses the 3019 and 3050 chipset, where the 3019 pstn line interface chip has many different pstn line impedance settings including 600, 900, 270, 220, 370, 320, 275, 120, 350, etc, ohms. Have no clue which countries use which settings, but obviously Silicon Labs intended this chip set to operate in different countries, whereas the 3012 spec sheet doesn't seem to support those objectives. So, backing into exactly what is causing the echo in the three UK cases noted yesterday on this list... - not likely to be d (2-wire to 4-wire conversion along the end-to-end voice path) as that would impact all telco users, not just * users. - item c (pstn line problems) can contribute to echo depending upon how bad the pstn line actually happens to be. Most telco's have the equipment to measure line quality, however most will stop at the cable entrance to your home/business, leaving you to guess at what's happening inside. - item b (mismatched x100p - pstn line) can contribute, but without knowing the exact specs (and probably more info from Silicon Labs), its impossible to guess at this one. - item a (echo can in *) is still a very real possibility, and Mark is about the only person I know of that has the knowledge of the spec's and * to weight in on this one. One of the methods that I used to help determine whether my tdm echo was a pstn line or * issue was to eval echo on three different pstn lines using the exact same physical * port (x100p). If the lines are clean from an analog phone perspective (eg, no hum, no noise) and the lines cause equal echo when used with *, then its highly likely the issue is either a or b. If the answers to b rule out impedance mismatches, then a is
Re: [Asterisk-Users] Cheap (US$120 or less) SIP Phones
James H. Thompson wrote: Are there any online retailers that carry the Uniden UIP series phones? I did a quick Froogle search to no avail. See: http://www.voip-info.org/wiki-Uniden So you *must* sign up as a reseller to purchase one? What are your opinions/problems on the UIP-200? It looks like a pretty good phone for a reasonable price. -- perl -e 'print $i=pack(c5,(41*2),sqrt(7056),(unpack(c,H)-2),oct(115),10);' http://ccicolorado.org Exceptional Dogs for Exceptional People - Help Out Today!
RE: [Asterisk-Users] SS7 to Pri
That would be great if * could do it. We *have* a switch, but it will not give us a pri by manufacture design :( . It will give us ss7 though. And as far as I can tell, the only way to get callerid etc is by a PRI to *. I can do fx trunks from the switch, but the switch will not include callerid. On Fri, 2004-06-25 at 12:09, Jeremy Jones wrote: Didn't I hear a week or two ago (on this list) that someone had taken it upon themselves to write an asterisk module for the openss7-modified digium t1/e1 cards? Maybe soon asterisk'll do it. Jeremy Jones -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Friday, June 25, 2004 9:28 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] SS7 to Pri On Fri, 2004-06-25 at 09:24, Joseph wrote: Does anyone know of a device that will take an SS7 link and convert it to a PRI? I think it's called an ILEC or CLEC. 8-) -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ 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 -- respectfully, Joseph - (606) 477-2355 x140 --= ___ 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
Re: [Asterisk-Users] SIP extension outside of IP tables firewall
Brian Weaver wrote: I have an Asterisk PBX on the private lan, which is protected from the public Internet with a Linux iptables machine. The firwall is it's own seperate box running NAT with SPI. The only way to make this work well is to run a SIP proxy of some kind on the firewall system and tell the SIP client to use that proxy. Asterisk providing SIP behind a NAT and/or firewall is very difficult to get working. A SIP client behind a NAT, on the other hand, works fine as long as the client supports STUN. ___ 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
Re: [Asterisk-Users] IAX2 authentication confusion
Jeremy McNamara wrote:
On the machine you wish to dial out, you have in your iax.conf:
[peer]
type=peer
host=1.2.3.4
secret=foo
and in that same machine's extensions.conf you have something that looks
like:
Dial,IAX2/[EMAIL PROTECTED]/${EXTEN}
Then on the 'peer' (other) machine you need:
[USER]
type=user
context=incoming
auth=md5
which is cAsE SeNsITiVe. Plus you need the appropriate extension(s) in
this (other) machine's extensions.conf.
I understand that, except that this succeeds even if the calling host's
Dial command does _not_ include the USER name at all!
Have you bothered to study any of the documentation out there? Start
here: http://www.voip-info.org/
Of course :-) I've spent the last month doing exactly that... But I
don't understand how Asterisk can authenticate an incoming IAX2 call
that does not include a USERNAME field (checked with iax2 debug turned
on). I have done it on my machine, and moved the shared secret to a
different entry in the receiving machine's iax.conf file, and the call
still succeeds, with the receiving Asterisk thinking that the caller is
now coming from that different entity.
In other words, somehow Asterisk is using _only_ the secret to identify
_and_ authenticate the caller. I don't have any problem putting all the
needed information on the calling systems (they will be under my
control); my concern is that on my receiving end unless I use IP-based
restrictions for callers anyone at all can connect if they can guess any
secret in my iax.conf file, not a valid username/secret pair.
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Re: [Asterisk-Users] Cheap (US$120 or less) SIP Phones
On Friday 25 June 2004 10:56, Chris Hirsch wrote: James H. Thompson wrote: Are there any online retailers that carry the Uniden UIP series phones? I did a quick Froogle search to no avail. See: http://www.voip-info.org/wiki-Uniden So you *must* sign up as a reseller to purchase one? What are your opinions/problems on the UIP-200? It looks like a pretty good phone for a reasonable price. We buy our UIP200s through AVS Technologies, in Canada (not an online retailer). The phones offer great value for the price, including decent speakerphone, a hands-free jack (stereo-mini), programmable buttons, tilting alpha/numberic display, etc. The only real downsides are: - no 3-way call support (at least not yet) - no way to change your mind and get your caller back after starting to transfer a call (not yet anyway) Uniden Support has also been very responsive - I do recommend Uniden. Ryan ___ 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
Re: [Asterisk-Users] Polycom IP 600 Programmability
John, Mine gives the call waiting beep also, but only when a call comes in on the same extension that is in use. If a call comes in on another extension on my phone, then I get no beep, just a light on the button flashing and the sreen letting me know that there is another incoming call. -Tor -Tor John Baker wrote: My IP 600 gives me a call-waiting tone when another call comes in. I'm quite sure there's a setting for that in the xml. As for the feature buttons, I'll look at that this weekend, but it seems to me that using SIP with these phones precluded alot of the key programming stuff i.e., there was a chart in the admin guide that showed what the phone was capable of with respect to the keys under SIP and it wasn't much. John Tor Roberts wrote: Hi, Speaking of programming the IP 600, does anybody know how to programm any of the feature buttons to send a combination of digits while a call is in progress? The most obvious use would be to send #700 while a call is in progress and label the button Park. If I could do that, I would be very happy. Another option that would be nice would be if the phone would ring on a incoming when you are already on another line, instead of just flashing on the screen. Thanks, -Tor Roberts Erik Barker wrote: I would also be interested in similar functionality. We have agents using Polycom IP 600s that would like some sort of notification that they are logged into Asterisk Queues - either a flashing LED or perhaps some sort of graphic on the display. I know that there are numerous configuration options in the XML files and I've looked at the Admin guide, but i haven't seen any examples yet. Erik On Wed, 2004-06-16 at 04:44, Ray Burkholder wrote: I'm looking to program some sort of web-services function: user presses a button and send some info to a web server or scripting program. The web server or script returns some text and/or imagery for the screen. Lather, rinse, repeat. I saw in section 3.7.1 of the manual referenced below that there is a services function. However, it appears to not be enabled. Yet. Any other way of doing this, or has the 3.7.1 function been enabled yet? Ray. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Baker Sent: Tuesday, June 15, 2004 11:02 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Polycom IP 600 Programmability Polycom IP 600's are fully programmable, much more so than the Cisco phones. Yes, you can program the phone buttons. That and just about everything else you can imagine is programmable via xml configuration files. Goto http://www.polycom.com/common/pw_item_show_doc/0,1276,2545,00. pdf for the admin guide and you can see for yourself how great the difference is. John P.S. Here's the wiki: http://www.voip-info.org/wiki-Polycom+Phones Ray Burkholder wrote: Do the Polycom IP phones have some programmability so you can do some programmable phone buttons like you can on the Cisco phones? If there is programmability, such as for soft-keys and the like, how would you rate Polycom's vs Cisco's capabilities? And where can one find the programming documentation? Thanx. Ray. ___ 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 -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ 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 ___ 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
Re: [Asterisk-Users] Asterisk with PostgreSQL
Hello Neil/Everybody, Yes you are correct, PostgreSQL has to be specifically configured within the server it is hosted on to allow host calls from Asterisk so that the socket connects. Here is how I solved the problem through the help of everybody here: Firstly enable the -i command in the /etc/init.d/postgresql command for starting up postgresql. Secondly, edit pg_hba.conf to add in a host (not local) entry in it so that PostgreSQL server allows host connects, most probably from 127.0.0.1 using the password method. Once you are done, asterisk should be able to connect to postgresql when it starts up. Here is a big THANK YOU to all who helped along the way. Cheers :) On Fri, 25 Jun 2004 10:55:28 -0400, Neil Cherry [EMAIL PROTECTED] wrote: Caleb Kow wrote: Here we go: [EMAIL PROTECTED] root]# netstat -ap Active Internet connections (servers and established) Proto Recv-Q Send-Q Local Address Foreign Address State PID/Program name tcp0 0 *:32768 *:* LISTEN 3221/ tcp0 0 *:imaps *:* I didn't see Postgres running but did notice mysql. They run on different ports so that not a problem unless you are mistaking one for the other. Another poster stated that Postgres runs local socekts by default and that a change in the config is needed to get it working with TCP/IP. I'd investigate that as that's what it looks like. I hope this helps. -- Linux Home Automation Neil Cherry[EMAIL PROTECTED] http://home.comcast.net/~ncherry/ (Text only) http://linuxha.sourceforge.net/ (SourceForge) http://hcs.sourceforge.net/ (HCS II) ___ 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
[Asterisk-Users] Termination Provider
I've been looking for a good iax or sip ==> ptsn provider. Someone with very low cost usa calling and can offer incoming ptsn connections in most markets. The only decent providers I could find were iconnecthere and nufone. Has anyone found someone that really stood out? Matt Hohman New Heights Church http://www.newheights.org 7913 NE 58th Ave. Vancouver, WA 98665 Office: 360.694.4985 Fax: 360.694.0219 Email: [EMAIL PROTECTED]
RE: [Asterisk-Users] H.323 - NO AUDIO IN BOTH DIRECTIONS
FYI I am experiencing the same problem. I have complied asterisk from the latest CVS The call connects with no audio or DTMF to either end. I tested with ulaw and g729 with no success. -Michael On Fri, 2004-06-25 at 10:55, Scott Stingel wrote: Just checking that you have installed the proper versions of both OpenH323 and PWLib, as mentioned in ~/asterisk/channels/h232/README, and have rebuilt asterisk after those installations as specified? If so, then you are having the same problem I'm experiencing: no audio on H.323. I'm also connecting through a Cisco 5300. I'm just generating audio in one direction: outbound from asterisk - I hear nothing. This used to work I'm pretty sure... There is an outstanding bug report covering H.323 problems (#1334), not sure what the current status is. Cheers Scott Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com http://www.evtmedia.com/ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sebastian Nocetti Sent: Friday, June 25, 2004 7:55 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] NO AUDIO IN BOTH DIRECTIONS hello all, I am having a trouble with Audio using h.323 channel... I am doing this Call comes into cisco 5300 and is sent to Asterisk, asterisk catch call with h.323 driver and send call to a SoftSwitch that routes the call, I can see log debug telling me, CALLED XXX, and then RINGING, and I can hear ring tones... but when call is answered, I DONT HEAR ANYTHING... I am using lastest ASTERISK download somebody can help me to solve this problem thanks..!! ___ 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 Michael K. Rodriguez Dialmex LLC Director of Network Operations 200 S. 10th Suite 1209 McAllen, TX 78501 (956) 994-0014 x107 office (956) 682-8521 fax (956) 239-0627 mobile
Re: [Asterisk-Users] NO AUDIO IN BOTH DIRECTIONS
- I'm here with you on this one. I've not been able to figure this out - I triple quadruple checked that I have the right versions of pwlib openh323, I've followed all recommendations in the README, yet I still do not have audio in both directions. I'm also using a cisco 5300, there is no firewall. Tcpdump has revealed the following: when calls are made from the 5300 to asterisk, the 5300 sends continual udp packets, but asterisk doesn't seem to be responding. when calls are made from asterisk to the 5300, no udp packets are sent. It should be noted that when the calls are made using sip, everything works just fine. -g On Fri, 2004-06-25 at 10:54, Sebastian Nocetti wrote: hello all, I am having a trouble with Audio using h.323 channel... I am doing this Call comes into cisco 5300 and is sent to Asterisk, asterisk catch call with h.323 driver and send call to a SoftSwitch that routes the call, I can see log debug telling me, CALLED XXX, and then RINGING, and I can hear ring tones... but when call is answered, I DONT HEAR ANYTHING... I am using lastest ASTERISK download somebody can help me to solve this problem thanks..!! ___ 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] 503 Unavailable
I'm having troubles... I am new to Asterisk and SIP. I was just given this setup and it was running fine. And somehow it stopped. I thought it was the DID(again) But it wasn't. All calls are getting rejected. Called 1403(Phone#)@###.###.###.### -- SIP/###.###.###.###-1c69 answered Zap/83-1 -- Got SIP response 503 Unavailable back ___ 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
RE: [Asterisk-Users] Cheap (US$120 or less) SIP Phones
FWIW, I just ordered a prototype for a single-line sip speakerphone from China. If I get this to work on asterisk, I may import it -- I should be able to sell it for about $85 to endusers and $70-$75 in quantities. The way I look at it, $75/port is right on the dot for cheapest possible endpoint. FXS ports are between $50 (Sipura) to $150 (Channel Bank) each, so getting a programmable speakerphone screen for $75 compares to a Sipura + Phone solution. I'll also be working with the manufacturer to implement additional features. - Jay -Original Message- From: Chris Hirsch [mailto:[EMAIL PROTECTED] Sent: Friday, June 25, 2004 11:57 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Cheap (US$120 or less) SIP Phones James H. Thompson wrote: Are there any online retailers that carry the Uniden UIP series phones? I did a quick Froogle search to no avail. See: http://www.voip-info.org/wiki-Uniden So you *must* sign up as a reseller to purchase one? What are your opinions/problems on the UIP-200? It looks like a pretty good phone for a reasonable price. -- perl -e 'print $i=pack(c5,(41*2),sqrt(7056),(unpack(c,H)-2),oct(115),10);' http://ccicolorado.org Exceptional Dogs for Exceptional People - Help Out Today! ___ 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
RE: [Asterisk-Users] SS7 to Pri
And as far as I can tell, the only way to get callerid etc is by a PRI to *. I have EM trunks into * out of our switch. We tell the switch to pass digits in Feature Group D DTMF format, and we are able to get ANI and DNIS. Of course this does not allow to get the calling party's name though. ___ 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
Re: [Asterisk-Users] Polycom IP 600 Programmability
John, Oops! I was wrong, I do get a call waiting beep even if the call comes in on another extension. I still would prefer the phone to ring when another call comes in. If anybody knows how to do this, that would be great. -Tor Tor Roberts wrote: John, Mine gives the call waiting beep also, but only when a call comes in on the same extension that is in use. If a call comes in on another extension on my phone, then I get no beep, just a light on the button flashing and the sreen letting me know that there is another incoming call. -Tor -Tor John Baker wrote: My IP 600 gives me a call-waiting tone when another call comes in. I'm quite sure there's a setting for that in the xml. As for the feature buttons, I'll look at that this weekend, but it seems to me that using SIP with these phones precluded alot of the key programming stuff i.e., there was a chart in the admin guide that showed what the phone was capable of with respect to the keys under SIP and it wasn't much. John Tor Roberts wrote: Hi, Speaking of programming the IP 600, does anybody know how to programm any of the feature buttons to send a combination of digits while a call is in progress? The most obvious use would be to send #700 while a call is in progress and label the button Park. If I could do that, I would be very happy. Another option that would be nice would be if the phone would ring on a incoming when you are already on another line, instead of just flashing on the screen. Thanks, -Tor Roberts Erik Barker wrote: I would also be interested in similar functionality. We have agents using Polycom IP 600s that would like some sort of notification that they are logged into Asterisk Queues - either a flashing LED or perhaps some sort of graphic on the display. I know that there are numerous configuration options in the XML files and I've looked at the Admin guide, but i haven't seen any examples yet. Erik On Wed, 2004-06-16 at 04:44, Ray Burkholder wrote: I'm looking to program some sort of web-services function: user presses a button and send some info to a web server or scripting program. The web server or script returns some text and/or imagery for the screen. Lather, rinse, repeat. I saw in section 3.7.1 of the manual referenced below that there is a services function. However, it appears to not be enabled. Yet. Any other way of doing this, or has the 3.7.1 function been enabled yet? Ray. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Baker Sent: Tuesday, June 15, 2004 11:02 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Polycom IP 600 Programmability Polycom IP 600's are fully programmable, much more so than the Cisco phones. Yes, you can program the phone buttons. That and just about everything else you can imagine is programmable via xml configuration files. Goto http://www.polycom.com/common/pw_item_show_doc/0,1276,2545,00. pdf for the admin guide and you can see for yourself how great the difference is. John P.S. Here's the wiki: http://www.voip-info.org/wiki-Polycom+Phones Ray Burkholder wrote: Do the Polycom IP phones have some programmability so you can do some programmable phone buttons like you can on the Cisco phones? If there is programmability, such as for soft-keys and the like, how would you rate Polycom's vs Cisco's capabilities? And where can one find the programming documentation? Thanx. Ray. ___ 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 -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ 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 ___ 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 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update
Re: [Asterisk-Users] SS7 to Pri
Joseph schrieb: Does anyone know of a device that will take an SS7 link and convert it to a PRI? ... Hi, your question is not very asterisk related. So you can use any signal converter on the telephone device market. I you are looking for something below 10 kEUR, there is not much choice, e.g. - SP201 (SignalPath) - Aculab's Groomer II in smallest version But just wait two or three weeks, and we will know, whether there will be some SS7 support for asterisk in the near future. Roger. ___ 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] Stable branch usable? Development branch better?
Is the stable branch usable? Is there ever going to be a 1.0 release? Should I be using the "stable" branch or the development branch? The development branch seems to have more fixes than the stable branch. It looks like fixes going into the release branch aren't going into the stable branch. TKS Paul Paul Mahler [EMAIL PROTECTED] Signate, LLC665 Third StreetSuite 100San Francisco, CA94107-1901Asterisk Services and Training signate small logo.gif
RE: [Asterisk-Users] SS7 to Pri
What switch do you have? On Fri, 2004-06-25 at 11:56, Joseph wrote: That would be great if * could do it. We *have* a switch, but it will not give us a pri by manufacture design :( . It will give us ss7 though. And as far as I can tell, the only way to get callerid etc is by a PRI to *. I can do fx trunks from the switch, but the switch will not include callerid. On Fri, 2004-06-25 at 12:09, Jeremy Jones wrote: Didn't I hear a week or two ago (on this list) that someone had taken it upon themselves to write an asterisk module for the openss7-modified digium t1/e1 cards? Maybe soon asterisk'll do it. Jeremy Jones -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Friday, June 25, 2004 9:28 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] SS7 to Pri On Fri, 2004-06-25 at 09:24, Joseph wrote: Does anyone know of a device that will take an SS7 link and convert it to a PRI? I think it's called an ILEC or CLEC. 8-) -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ 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 -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ 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] SS7 status report 2
Hi, there are still some questions to be answered by OpenSS7.com in order to decide, whether E400P-SS7 is a good choice for the asterisk SS7 support. In the meanwhile I'm also in negotiations with another manufacturer (whose name I currently may not tell due to a NDA) of SS7 hardware, who gets likely persuaded to offer a cheap SS7-PCI-card which would be suitable for asterisk. Asterisk users seems to be a market not neglectible. I would prefer the coorperation with OpenSS7, if it will become clear, that it would be a technically good solution. My plans are to order one of theese two cards by the end of next week. In the meanwhile I got offered support by list members: - a programmer - one who could probably deliver SS7 testing equipment - one who could perform tests inside a telco network Roger. ___ 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
RE: [Asterisk-Users] H.323 - NO AUDIO IN BOTH DIRECTIONS
I have the same problem here. I have to servers working with identical (same) configurations, the old one is working just perfect and the new one I got, is not working (no voice in both directions). Im trying to fix this problem with digium, we are exchanging emails so if I get a solution Im gona reply it here. Best Regards Hekuran Doli FYI I am experiencing the same problem. I have complied asterisk from the latest CVS The call connects with no audio or DTMF to either end. I tested with ulaw and g729 with no success. -Michael On Fri, 2004-06-25 at 10:55, Scott Stingel wrote: Just checking that you have installed the proper versions of both OpenH323 and PWLib, as mentioned in ~/asterisk/channels/h232/README, and have rebuilt asterisk after those installations as specified? If so, then you are having the same problem I'm experiencing: no audio on H.323. I'm also connecting through a Cisco 5300. I'm just generating audio in one direction: outbound from asterisk - I hear nothing. This used to work I'm pretty sure... There is an outstanding bug report covering H.323 problems (#1334), not sure what the current status is. Cheers Scott Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com http://www.evtmedia.com/ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sebastian Nocetti Sent: Friday, June 25, 2004 7:55 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] NO AUDIO IN BOTH DIRECTIONS hello all, I am having a trouble with Audio using h.323 channel... I am doing this Call comes into cisco 5300 and is sent to Asterisk, asterisk catch call with h.323 driver and send call to a SoftSwitch that routes the call, I can see log debug telling me, CALLED XXX, and then RINGING, and I can hear ring tones... but when call is answered, I DONT HEAR ANYTHING... I am using lastest ASTERISK download somebody can help me to solve this problem thanks..!! ___ 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 Michael K. Rodriguez Dialmex LLC Director of Network Operations 200 S. 10th Suite 1209 McAllen, TX 78501 (956) 994-0014 x107 office (956) 682-8521 fax (956) 239-0627 mobile ___ 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
RE: [Asterisk-Users] Problems with faxing via TE405P/Asterisk
Ok, may have got to the bottom of this. The te405P was sharing interrupts with a via82cxxx audio chip, that was being used to generate the music on hold for our existing pbx. Having now shut down the music on hold the system will now run correctly with telewest as the master and the gdk as the slave. This came about due to the high number of missed IRQ's, it seems as though the 2.4.22 via driver is none to selective about what it grabs. I'll leave it running over the weekend and see how it holds up. Steve -Original Message- From: Storer, Darren [mailto:[EMAIL PROTECTED] Sent: 20 June 2004 18:23 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Problems with faxing via TE405P/Asterisk Hi Steve, How bizarre, your config doesn't look like it should work too well and certainly doesn't look like it should improve your fax problem! I assume that pri_cpe is set for span1 and pri_net for span2 ? Maybe, just maybe, Telewest reconfigured your PRI to look for clocking from your CPE but I've not encountered that configuration before. Try and leave the current config up for as long as you can before you return it to production mode and watch the CLI/logs to see if you get any sporadic clock slips (within a couple of hours I'd expect at least one episode of messages). One last thought, did you bounce the system after you made the changes to zaptel.conf or did you just reload * ? HTH Darren -- Comgate TelcoInternetBroadcast -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Steve Hanselman Sent: 20 June 2004 16:48 To: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] Problems with faxing via TE405P/Asterisk They look odd to me for sure, I'm certain (99.9%) that Telewest would not clock off of us, but as far as I can see, the current config (which allows the GDK to send and receive faxes) has no external clocking??? Here's the current config: span=1,0,0,ccs,hdb3,crc4 span=2,0,0,ccs,hdb3,crc4 Here's the original config which I took to mean that Telewest provided clock and span2 clocked off span1? span=1,1,0,ccs,hdb3,crc4 span=2,0,0,ccs,hdb3,crc4 (Span1 goes to Telewest - our Telco provider, span2 goes to our current PBX, a GDK-186) Steve -Original Message- From: Storer, Darren [mailto:[EMAIL PROTECTED] Sent: 20 June 2004 16:34 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Problems with faxing via TE405P/Asterisk Steve, your config description (timing) does sound odd. Could you re-post your revised config files? Thanks Darren -- Comgate TelcoInternetBroadcast -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Steve Hanselman Sent: 20 June 2004 16:18 To: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] Problems with faxing via TE405P/Asterisk I've changed the zaptel.conf to set both as internal, and it now seems to work, which is backwards to the config I thought it should have been, I would have thought that the Telewest PRI would have been 1 and the GDK 0? Can somebody confirm that this is the correct definition for timing, if it's a +ve number then it's external clocking with the lowest 1 being the highest priority. All spans are clocked relative to the external source and the external source selected is the lowest priority number that is currently being clocked? I'll experiment some more. -Original Message- From: Yifang Dai [mailto:[EMAIL PROTECTED] Sent: 19 June 2004 03:22 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Problems with faxing via TE405P/Asterisk Let's try again, missed a line in the last reply... On Fri, Jun 18, 2004 at 09:55:24PM -0400, Yifang Dai wrote: On Fri, Jun 18, 2004 at 09:27:52PM +0100, Steve Hanselman wrote: LG GDK-186 PBX --PRI--- TE405P/Asterisk ---PRI--- Telewest (Telco provider) --snip--- Any ideas on where to start? This is most likely to be a timing issue. You need to make sure your asterisk is get timing from your telco, and provide timing for you gdk pbx. /etc/asterisk/zaptel.conf is the place to look. -- Yifang Dai Senior System Administrator Yarde Metals Inc 45 Newell St, Southington, CT 06010 (Phone) 860-406-6107; (FAX) 860-406-4060 ___ 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 The information contained in this email is intended for the personal and confidential use of the addressee only. It may also be privileged information. If you are not the intended recipient then you are hereby notified that you have received this document in error and that any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) Ltd Nevendon Hall, Nevendon Road,
Re: [Asterisk-Users] Leave one call to pick up another
Andrew Thompson wrote: Can the original poster open a bug at bugs.digium.com (if they haven't already)? Original poster emits heartfelt, Bwaahh I have posted about this problem at least three times previously, and been scoffed at each time. I'll be glad to do a bug report, and will head thither anon. . . B. ___ 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
RE: [Asterisk-Users] Speex
Does it work if you try ULAW or some other codec? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sola 2000 Sent: Friday, June 25, 2004 12:08 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Speex thanx that works here is wat i want to do sip (xlite)-*(iax2)(codec speex)-internet*(iax2)--pstn when i do that all i get is noise in the other end..am sure am doing something stupid... any help would be appreciated ( i want to meassure the bandwidth will be using the program called rate) sriram - Original Message - From: Jon Radon [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, June 24, 2004 12:55 PM Subject: RE: [Asterisk-Users] Speex Inband dtmf does not work with speex(only ulaw). Switch your dtmf mode to rfc2833. :) From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sola 2000 Sent: Thursday, June 24, 2004 11:16 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Speex I am a new to asterisk, i wanted to test the opensource codec speex i have installed speex, and recompiled asterisk i can see the speex_codec.so getting loaded i have xten lite, i used the registry patch ( http://bugs.digium.com/bug_view_page.php?bug_id=918 ) but still when i use xten lite i get the following errors Jun 24 10:46:15 WARNING[-1305486416]: codec_speex.c:167 speextolin_framein: Out of buffer space Jun 24 10:46:15 WARNING[-1305486416]: dsp.c:1478 ast_dsp_process: Unable to process inband DTMF on 512 frames am i doing something wrong? any pointers is helpful thanx sriram ___ 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 ___ 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
RE: [Asterisk-Users] SS7 to Pri
Hi Joseph, J Does anyone know of a device that will take an SS7 link J and convert it to a PRI? Telesoft Technologies make the Okeford range of protocol converters and baby switches that I have used for this purpose. Have a look at: http://tinyurl.com/3drjp If you are converting a number of SS7/PRI circuits at the same time the cost per conversion comes down dramatically. HTH Darren -- Comgate TelcoInternetBroadcast -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Joseph Sent: 25 June 2004 15:25 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] SS7 to Pri Does anyone know of a device that will take an SS7 link and convert it to a PRI? -- respectfully, Joseph - (606) 477-2355 x140 --= ___ 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
Re: [Asterisk-Users] Termination Provider
I've been using BroadVoice (sip based) for about 2 weeks now with no problems at all. 19.95/month Unlimited USA plan... -Begin Sarcasm--- And amazingly... you can use your own device with their service.. what a novel idea! Other sip providers should get in on this!! -End Sarcasm www.broadvoice.com - Original Message - From: Matt Hohman To: [EMAIL PROTECTED] Sent: Friday, June 25, 2004 10:18 AM Subject: [Asterisk-Users] Termination Provider I've been looking for a good iax or sip == ptsn provider. Someone with very low cost usa calling and can offer incoming ptsn connections in most markets. The only decent providers I could find were iconnecthere and nufone. Has anyone found someone that really stood out? Matt Hohman New Heights Church http://www.newheights.org 7913 NE 58th Ave. Vancouver, WA 98665 Office: 360.694.4985 Fax: 360.694.0219 Email: [EMAIL PROTECTED] ___ 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
RE: [Asterisk-Users] SS7 to Pri
On Fri, 2004-06-25 at 13:57, Mike Machado wrote: And as far as I can tell, the only way to get callerid etc is by a PRI to *. I have EM trunks into * out of our switch. We tell the switch to pass digits in Feature Group D DTMF format, and we are able to get ANI and DNIS. Of course this does not allow to get the calling party's name though. Well, this sounds interesting. Feature Group D DTMF must be the magic words? respectfully, Joseph - (606) 477-2355 x140 --= ___ 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
RE: [Asterisk-Users] Polycom IP 500 - Quality Issues
We have seen this on one of our polycom phones and it had something to do with the power adapter being fried. This was the only phone on an unprotected (non-UPS) outlet and we basically had to throw the adapter away and use a new one then the interferance went away and all was good. MATT--- -Original Message- From: Brent Franks [mailto:[EMAIL PROTECTED] Sent: Friday, June 25, 2004 12:50 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Polycom IP 500 - Quality Issues Hello, We have 15 Soundpoint IP 500 phones and recently call quality has deteriorated. On some calls there is a static-buzzing of sorts that occurs when users talk. It can be picked up on SIP-SIP calls and SIP-ZAPTEL (Channel BANK-T100P). It basically sounds really weird whenever someone talks, it sounds like a bee buzzing or something. Very hard to explain. Also, there will be echo on internal calls. We were not having any call quality issues with internal calls until recently. Upon inspection of the network, there isn't a virus or anything bogging down traffic. Additionally, the network is switched 100Mbps with a Gig port to our Asterisk Server. The Gig Ethernet card in the server is connected to the switch at 1000mbps. Any one know what other areas I can pursue? Thanks, - Brent ___ 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
Re: [Asterisk-Users] Cheap (US$120 or less) SIP Phones
- Original Message - From: Chris Hirsch To: [EMAIL PROTECTED] Sent: Friday, June 25, 2004 6:56 AM Subject: Re: [Asterisk-Users] Cheap (US$120 or less) SIP Phones James H. Thompson wrote: Are there any online retailers that carry the Uniden UIP series phones? I did a quick Froogle search to no avail. See: http://www.voip-info.org/wiki-Uniden So you *must* sign up as a reseller to purchase one? What are your opinions/problems on the UIP-200? It looks like a pretty good phone for a reasonable price.Todd at Teledynamics (see wiki page mentioned above) has been very responsive to email, and we did not need to sign up as a reseller to purchase the Uniden phones.
