[Asterisk-Users] chan_capi [0.4.0|-cm-0.5.4] and Asterisk 1.2.0-beta1 - early B3 not early enough sometimes
Hi, I've just installed Asterisk 1.2.0-beta1 for the first time and downloaded chan_capi and compiled it in and run it. (For comparison purposes, I've tried this with both chan_capi-0.4.0PRE1 and chan_capi-cm-0.5.4 and chan_capi-cm latest CVS). Whilst most things are fine, it seems that if I specify the 'b' parameter in the dial string before the number, sometimes the early B3 isn't early enough or not there at all. From the limited tests that I did last night, it would appear that it seems to depend on which carrier I use to make the call on my BRI line (I am based in the UK, and have a BT ISDN2e line). If I use my CPS provider to make the call, I get full early B3 including the ringing tone passed through from the exchange. If I route the call through BT by using the 1280 prefix, I do not get ringing tone at all and only get the sound through when either (a) a recorded anouncement is played or (b) the call is answered. What is more strange is that early B3 has been flawless whilst using Asterisk 1.0.x (currently 1.0.9) and chan_capi-0.3.5 (that is, it does the right thing in all cases.) So I'm making the assumption here that it probably isn't the fault of the telephone companies. Has anyone else come across this? Is it a bug in chan_capi? TIA Gary H -- Gary Hawkins MBCS [EMAIL PROTECTED] PGP: 0x6355BF46 (expires 31 Dec 2005) Web: http://www.garyhawkins.me.uk ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 [0.4.0|-cm-0.5.4] and Asterisk 1.2.0-beta1 - early B3 not early enough sometimes
On Sat, 3 Sep 2005, Gary Hawkins wrote: Hi, I've just installed Asterisk 1.2.0-beta1 for the first time and downloaded chan_capi and compiled it in and run it. (For comparison purposes, I've tried this with both chan_capi-0.4.0PRE1 and chan_capi-cm-0.5.4 and chan_capi-cm latest CVS). Whilst most things are fine, it seems that if I specify the 'b' parameter in the dial string before the number, sometimes the early B3 isn't early enough or not there at all. From the limited tests that I did last night, it would appear that it seems to depend on which carrier I use to make the call on my BRI line (I am based in the UK, and have a BT ISDN2e line). If I use my CPS provider to make the call, I get full early B3 including the ringing tone passed through from the exchange. If I route the call through BT by using the 1280 prefix, I do not get ringing tone at all and only get the sound through when either (a) a recorded anouncement is played or (b) the call is answered. What is more strange is that early B3 has been flawless whilst using Asterisk 1.0.x (currently 1.0.9) and chan_capi-0.3.5 (that is, it does the right thing in all cases.) So I'm making the assumption here that it probably isn't the fault of the telephone companies. Has anyone else come across this? Is it a bug in chan_capi? Since the call handling has changed in chan_capi-cm, it is possible that not all cases are handled as expected. To find out what the difference with your 1280 provider is, can you please send me a verbose log of level 5 with 'capi debug' of such an 'not-working' call? If possible, please use latest CVS for that test/log creation. Thanks, Armin ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Multiple ASTCC Cards Configuration
Hi: I need help setting-up multiple calling cards with different prices for the same routes using astcc. All my calling cards' routes now have the same price, but I need to be able to set multiple calling cards with different prices for the same route. I appreciate your feedback of How I can do that. Thanks; Chawki __ Yahoo! Mail Stay connected, organized, and protected. Take the tour: http://tour.mail.yahoo.com/mailtour.html ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] IVR Prompts
Try ENIcommunications 973 828 1625 Joshua Abbott wrote: Anyone know any companies that record IVR prompts and if so how much per prompt? I know Digium does this but any other company? Joshua ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Distortion/crackling/skipping problems on outgoing calls -- please help!!!
So, can no one give me any suggestions? Perhaps I can elaborate upon further testing and attempts to debug this tremendously frustrating problem. My softphone (typically IAXComm, but same results connecting via SIP on Xten Xlite) is installed on a P4 1.6 w/ 256 megs of RAM and an integrated sound chipset (Intel/AC97). I've had some problems with this chipset in Linux, and it doesn't support hardware mixing, so I've had to attempt to get dmix and ALSA running in an acceptable fashion; needless to say, I still have problems, and I don't know if this is related. I can record and playback my own voice and other audio (podcasts, Net radio, and music) fine with this headset (some cheap, Chinese $15 headset). However, when it comes to receiving decoded audio on the other end of a VoIP conversation, it sounds scratchy, distorted, crackly, whatever you want to call it. It's not the clarity so much as it is the other things I just mentioned. It's very hard ot put into words, but I'm hoping *someone* can associate with my problem. To make sure it wasn't my Asterisk box that was mucking things up, I connected directly with my softphone to my outgoing VoIP terminators, voipjet and Voxee. Sure enough, it sounded *exactly* the same as it does going through my * box on the LAN and then out through the public Net to voipjet and/or Voxee. Thus, I know it is my side of the equation that is mucking things up, but I cannot for the life of me pinpoint exactly *WHERE* this is taking place. Actually, I also unplugged my headphones and plugged back in the speakers, and it sounded roughly the same, but it's harder to tell because they're not surrounding your ears, quality isn't as clear, and ambient noise can be more easily heard and is distracting. I'm thinking, at this point, that it's my sound card that's messing things up, or its configuration or something. However, can anyone explain *why* non-VoIP-conversations sound perfect on my speakers and headset, while VoIP calls sound very bad? If I were to try to diagnose the above, I'd be using Ethereal to capture the voip packets coming from your itsp's, and analyzing that captured data to look for unusual things. If your itsp connections are sip based, ethereal has a utility to analyze/summarize some of this for you. If those connections are iax based, then you will need to analyze the packets yourself looking for unusual things. Analyzing the packets (either sip or iax) can consume a lot of time, but you really need to ensure those packets are arriving in a consistent manner, timestamps contained within the packets are consecutive and proper, packets are not arriving out of order, etc. At the same time that you're capturing those packets, use the facilities within asterisk to summarize what it thinks is going on (eg, 'iax2 show netstats'), so _that_ data can be correlated to the info derived from the packet captures. If this is a small soho * system, then run ethereal right on the same asterisk system capturing the data as it arrives at the system. Doing so will help identify any issues that you might have involving your local lan, broadband issues, etc, etc. Assuming the problem that you've described is consistent and happens on a regular basis, you don't need to collect and analyze megabytes of packet captures. Just collect a short duration sample that is assured to contain the representative packets associated with the bad audio (maybe five or ten seconds worth). If the analysis does not indicate a problem at that point, then at least you know the problem is internal to asterisk, etc. If you don't feel you have the skills or knowledge to do that analysis, then hire someone that can. Both your original post and the followup post contain a ton of adjectives and adverbs describing a technical problem, but contain little (or no) technical data (such as the results from above or results from various * show commands) that would allow anyone to comment on your problem. So, doubtful anyone is ever going to reply to such postings with anything helpful. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to tell reason for hangup or busy in SIP or IAX
Hi, using Zap, I have several messages to pass when terminating a successful or unsuccessfull call, indicating the reason e.g., why a call failed. Using SIP or IAX2, I know only Hangup Busy Congestion without passing any more detailed information. Am I right, that I can't tell the caller in SIP or IAX2, whether the call was rejected or whether the called number does not exist. I have to assign those hangupcauses to one of Hangup (which I use for normal termination after successfull call and for No answer after timeout. Busy (which I use, if the called phone is busy) Congestion (which I use, if our gateway or one of the used carriers are busy) Thanks for any hints, if there further means! Roger. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Sipura 3000 setup
Can anybody show me a working Sipura 3000 setup please? I need to setup one to my * box, ... What are the variants you can setup? Advantage - disadvantage. Have you looked at or tried the setup wizard on www.voxilla.com? Having used the spa3k for about a year, I can tell you there is more functionality and flexibility in that tiny box then what the average ata or sip phone has, and understanding how to take advantage of that functionality has a rather steep learning curve. The more one digs into and plays with it, the more one finds in terms of additional functions. Any attempt to summarize those variants for you would be very incomplete. I'd consider the polycom phones as one of the best on the market in terms of quality and configuration flexibility. I'd suggest the spa3k's audio quality is less then a polycom, but its configuration flexibility ranks right up there (if not higher) then a polycom. Its too bad that sipura is targeting the box mostly to itsp's as they do have more documents (eg, administrators guide) that is very useful and rather easy to read with lots of feature examples. Unfortunately, the guide says Any unauthorized disclosure, copying, distribtuion, or use of this information is prohibited. Personally, I'm using the spa3k as a household box configured to: - minimize the need for spouse remedial training ;) - fxs line registered with asterisk - fxo line registered with asterisk - incoming pstn-fxo calls ring-thru to the fxs without * involvement (I'm not concerned whether asterisk is up/down, etc) - all outgoing fxs calls default to using the pstn-fxo unless the called number is prefixed with an 8 - if the outgoing fxs call is prefixed with an 8, the spa3k dialplan stips the 8 and forwards the call to asterisk. Asterisk's dialplan then handles whatever is dialed. - asterisk calls destined for the spa3k fxs port use distinctive ringing, therefore all incoming itsp or asterisk originated calls ring all household phones with two short rings - asterisk calls destined for the spa3k fxo port are totally functional, allowing asterisk full access to the pstn network - E911 (etc) calls from the fxs phones are routed directly to the pstn-fxo port, therefore I'm not at all concerned with FCC and other US regulatory voip rulings The above can be replicated via the voxilla setup wizard, and tweeking done to the box after that. I've not played with the spa3k custom calling features at all, so no knowledge/experience as to what it actually supports, etc. There appears to be a ton of functionality there, just haven't paid any attention to that so far. The only limitations I've observed thus far include things like: - g729 use is limited to one sip session - v3.1.5 firmware has a serious echo issue; v3.1.3 code seems to be the best that sipura has produced in terms of audio quality, functionality, and bug fixes to date - some spa3k configuration options require a reboot that are not documented anywhere that I could find, therefore a power cycle is always included in my changes ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Debug info from txfax - howto?
Hi, I read here in this mailing list about the debug info from txfax. I plaid a lot, but didn't get debug infos. I added the debug argument to txfax, I enabled debug in logger.conf, what else should I do? In which file or medium can I then expect the debug infos from txfax? Thanks for hints! Roger. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] stale nonce?
I'm trying to debug a new spa3k to * config issue and am receiving hundred's of the following: Sep 3 09:38:51 NOTICE[23092] chan_sip.c: stale nonce received from '784-7103 s ip:[EMAIL PROTECTED]' What's the typical issue behing this message? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Current status on _outgoing_ Swedish/Dutch DTMF CLIP for TDM400 FXS interfaces?
Hi all,
I have been looking at the code for both the zaptel driver (wctdm.c/wcfxs.c)
and the asterisk channel driver (chan_zap.c) trying to figure out how much
of this that has been implemented. So far I can see that the current stable
1.0.9.1 zaptel driver don't have the SETPOLARITY ioctl that would be
required to properly signal the Swedish/Dutch CLIP, but the 1.2 beta1 has
this support..
Moving forward from that I then look at the 1.2 beta1 chan_zap code, but
nothing's in there could possibly give us any DTMF tones. I have seen some
bugs being posted and closed (particulary this one:
http://bugs.digium.com/view.php?id=3866). After a bit of experimenting with
that code and the fxstest.c code supplied with zaptel I have something that
actually reverses the polarity of the idle line, waits, sends the digits,
waits again and reverses polarity back to normal - this results in fully
working CallerID using my stand-alone CID display and a Gigaset4010 DECT
phone which unfortunately doesn't understand FSK CLIP/CNIP. I have also
checked the signal with my scope and it looks just like my landline CLIP
procedure (and working equally well.. :))
The code is provided at the end of this mail. After browsing the 1.2beta1
release I also downloaded the latest bleeding edge cvs and verified that
these parts of the code looked exactly like 1.2beta1.
There are a few questions I have though:
Why do I have to continuously set the ProSLIC register 64 to 6 (active
reverse transmission) to get the tones onto the line if I'm onhook? Only
doing it once just after reverse polarity ioctl doesn't do the trick... The
idea with SETPOLARITY in addition with going OFFHOOK with ZT_HOOK should be
the same, right? It works if I take the phone offhook though, then I can
hear the tones even without having to force the register to 6. Next question
would be if there's a better way to know that the dialing is complete,
especially if this is to be embedded into the chan_zap code, it would be
nice to have a cleaner solution?
Is this something for the dev list as well, I guess there's some interest in
getting this to work in the stock Asterisk code, even though we're not that
many people requiring this as it seems..? Should I reopen the bug?
I'm thankful for any input in this matter..
Regards,
Werner
#include stdio.h
#include string.h
#include errno.h
#include fcntl.h
#include stdlib.h
#include unistd.h
#include sys/ioctl.h
#include zaptel.h
#include tonezone.h
#include wctdm.h
static int tones[] = {
ZT_TONE_DIALTONE,
ZT_TONE_BUSY,
ZT_TONE_RINGTONE,
ZT_TONE_CONGESTION,
ZT_TONE_DIALRECALL,
};
int main(int argc, char *argv[])
{
ZT_DIAL_OPERATION dop;
struct wctdm_regop regop;
struct zt_dialparams dps;
int fd,ctlfd,toneduration;
int res;
int x;
if (argc 3) {
fprintf(stderr, Usage: fxstest zap device cmd\n
where cmd is one of:\n
stats - reports voltages\n
regdump - dumps ProSLIC registers\n
tones - plays a series of tones\n
polarity - tests polarity reversal\n
ring - rings phone\n);
exit(1);
}
// Code added by [EMAIL PROTECTED] to set the default tone length to shorten the
time it takes to transmit DTMF CLIP
// 25ms is enough for me but according to specs 40 _should_ be understood
as a valid digit, YMMV.
ctlfd = open(/dev/zap/ctl, O_RDWR);
if (ctlfd == -1) {
fprintf(stderr,Couldn't open zapctl!\n);
}
toneduration = 40;
if (toneduration -1) {
dps.dtmf_tonelen = dps.mfv1_tonelen = toneduration;
res = ioctl(ctlfd, ZT_SET_DIALPARAMS, dps);
if (res 0) {
fprintf(stderr,Couldn't dialparms!\n);
}
}
close(ctlfd);
// End of code added - [EMAIL PROTECTED]
fd = open(argv[1], O_RDWR);
if (fd 0) {
fprintf(stderr, Unable to open %s: %s\n, argv[1],
strerror(errno));
exit(1);
}
if (!strcasecmp(argv[2], ring)) {
fprintf(stderr, Ringing phone...\n);
x = ZT_RING;
res = ioctl(fd, ZT_HOOK, x);
if (res) {
fprintf(stderr, Unable to ring phone...\n);
} else {
fprintf(stderr, Phone is ringing...\n);
sleep(2);
}
} else if (!strcasecmp(argv[2], polarity)) {
// Code modified by [EMAIL PROTECTED] to not only reverse polarity but also send an
example DTMF CLIP
fprintf(stderr, Sending DTMF CLIP...\n);
x = 1;
res=ioctl(fd, ZT_SETPOLARITY, x);
if (res) {
fprintf(stderr, Unable to set
Re: [Asterisk-Users] random beeps in MeetMe
Changing IRQ's seems to fix the problem. The maching is a relatively underpowered Athalon XP 1600 with 128 Mb of Ram. Not a production system, to be sure. But it is adequate for testing before I deploy changes to the production system. Tomas Florian wrote: Depends what the beep sounds like ... but I've been having this system on a busy system which has XP100 and the Ethernet cards or other devices sharing one IRQ. You might need to spread out the IRQs so that XP100 get's its own and Ethernet gets another one of its own. How fast of a system is it? Tomas -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Ben Brown Sent: Friday, September 02, 2005 7:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] random beeps in MeetMe I have 3 users in a meetme conference. 2 of them are monitor only. I get a random beep in the audio during the conference. There appears to be no pattern. The 2 monitors are SIP softphones and the third is a POTS line on an XP100 card. disconnecting either of the monitors does not resolve the situation. This is currently a test box, so I would consider some sort of hardware issue a possibility, but I just want to make certain that there is not an asterisk issue here. Anyone have any thoughts on where I should start to look? Thanks BEN ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] newbie install problem. And I already searched everywhere!
Hi,
When startin * it tells me:
[chan_zap.so] = (Zapata Telephony w/PRI)
== Parsing '/etc/asterisk/zapata.conf': Found
Sep 3 17:21:59 DEBUG[1076873856]: chan_zap.c:1208 update_conf: Updated
conferencing on 1, with 0 conference users
-- Registered channel 1, PRI Signalling signalling
Sep 3 17:21:59 DEBUG[1076873856]: chan_zap.c:1208 update_conf: Updated
conferencing on 2, with 0 conference users
-- Registered channel 2, PRI Signalling signalling
-- Automatically generated pseudo channel
== Starting D-Channel on span 1
== Registered channel type 'Zap' (Zapata Telephony Driver w/PRI)
== Registered channel type 'Tor' (Zapata Telephony Driver w/PRI)
== Registered application 'CallingPres'
== Manager registered action ZapTransfer
== Manager registered action ZapHangup
== Manager registered action ZapDialOffhook
== Manager registered action ZapDNDon
== Manager registered action ZapDNDoff
== Manager registered action ZapShowChannels
[app_sendtext.so] = (Send Text Applications)
==
When I try to use a channel it says:
Executing Dial(SIP/4001-d927, Zap/2/0652456343}) in new stack
Sep 3 17:23:02 NOTICE[1117264816]: app_dial.c:727 dial_exec: Unable to create
channel of type 'Zap'
My relevant extension is:
exten = _97.,1,Dial(Zap/2/${EXTEN:2}})
and my zapata.conf is:
===
[channels]
switchtype = euroisdn
; p2mp TE mode
signalling = bri_cpe_ptmp
; p2p TE mode
;signalling = bri_cpe
; p2mp NT mode
;signalling = bri_net_ptmp
; p2p NT mode
signalling = bri_net
pridialplan = national
prilocaldialplan = local
nationalprefix = 0
internationalprefix = 00
echocancel=yes
echotraining = 100
echocancelwhenbridged=yes
immediate=yes
group = 1
context=incoming
channel = 1-2
can anyone give me a hint how th debug the rest?
(off course a clean cut solution would be nice too :-)
kind regards,
--
Jeroen Baten| EMAIL : [EMAIL PROTECTED]
_ __ | web : www.i2rs.nl
| )|_)(_ | tel : +31 (0)499 477 688
_|_/_| \__) | fax : +31 (0)499 476 804
Roerlaan 36, 5691 HJ, Son, the Netherlands
___
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[Asterisk-Users] equipment and network advice
I am planning to get voip for my apartment, and I am getting close to my service (packet 8 or SunRocket). My confiuration is currently like this: Cable Modem to 4 port Netgear Ethernet 10/100 router. I would like to use a hardphone(http://www.voip-info.org/wiki-VOIP+Phones#id323078) , instead of me buying an analog phone or using any type of adapters. My ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] One way echo canceling?
On Thu, Sep 01, 2005 at 12:40:16PM -0400, Doug Lytle wrote: When there is a call on zap 1, from a sip phone on the remote office I have not seen it myself, but I have heard that some people have ahd trouble with overlapdial and echo cancellation. I have not been able to confirm whether or not this is actually a bug. One possible fix is to disable overlapdial and see if echo cancellation is enabled after this. If it is, this might be a bug in chan_zap.c Turning off overlapdial did indeed fix it. It now shows as being enabled. Try updating to latest HEAD. I just fixed it this morning. It should work with overlapdial enabled now. -- Matthew Fredrickson ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] newbie install problem. And I already searchedeverywhere!
When startin * it tells me: [chan_zap.so] = (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found Sep 3 17:21:59 DEBUG[1076873856]: chan_zap.c:1208 update_conf: Updated conferencing on 1, with 0 conference users -- Registered channel 1, PRI Signalling signalling Sep 3 17:21:59 DEBUG[1076873856]: chan_zap.c:1208 update_conf: Updated conferencing on 2, with 0 conference users -- Registered channel 2, PRI Signalling signalling -- Automatically generated pseudo channel == Starting D-Channel on span 1 ..snip.. and my zapata.conf is: === [channels] switchtype = euroisdn ; p2mp TE mode signalling = bri_cpe_ptmp ; p2p TE mode ;signalling = bri_cpe ; p2mp NT mode ;signalling = bri_net_ptmp ; p2p NT mode signalling = bri_net pridialplan = national prilocaldialplan = local nationalprefix = 0 internationalprefix = 00 echocancel=yes echotraining = 100 echocancelwhenbridged=yes immediate=yes group = 1 context=incoming channel = 1-2 This configuration looks strange. You know that the channel word sets those channels to the latest settings above, right? The syntax is a bit weird. That means that your signalling looks odd - first you set it to bri_cpe_ptmp and then to bri_net, meaning that bri_net overrides the previous setting. I'm a bit confused about the PRI signalling here, it looks like you're using a BRI card, so why pridialplan? I haven't played around with BRI equipment together with Asterisk (yet) so maybe the PRI signalling should be there.. What is ztcfg -vv giving as output if you run it? /Werner ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dialing internal extensions with AAH 1.5
Has anyone had any trouble dialing internal extensions with AAH 1.5? The digital receptionist works great and I can dial functions and outside lines, but if I try to dial an internal extension between two phones it always goes to voice mail. anyone have any clues? Regards, Chris ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Overhead Paging Systems...
William Boehlke wrote: Viking makes everything you might need for paging and door control. www.vikingtelecomsolutions.com William Boehlke Signate I have one customer with a nortel meridian pbx and there is viking stuff all over the backboard. I never had to mess with any of it because it all works as intended. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Speed Questiosn
Ok, But what coding are you using? 3072 splitted by half is 1536 kbps. You right that gives you 307.2 kbps by phone, if everything is right. In the works case you shouldn't be needing more than 64 kbps by phone. Carlos -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joshua Abbott Sent: Thursday, September 01, 2005 2:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Speed Questiosn Hi I currently have a 3072kbps line that I'm splitting in half for 5 of my phones. That's 307.2kbps +/- a couple of kpbs. What is the minimum kbps for a phone to maintain clarity and volume? Joshua ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Contact Directory on Polycom IP-501 phones
On Thu, 2005-09-01 at 15:59 -0400, Jeremy Melanson wrote: Hi Jesse. A couple questions.. What firmware version are you using? Bootrom 2.6.2.20032 Sip 1.5.2.0054 How does your phone get it's config (FTP, TFTP, Manual config)? Initially it got the config from TFTP w/ the new boot rom. After that I did manual config on the phone. -- Jesse Keating GameHouse -- Systems Engineer ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ipvolution t1 cards
Has any one used the Ipvolution tdm120 cards i am intrested to know how well it works and how well the on board dsp's work. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Best costs effective solution...
Need help I lost the overview. The situation is the following: I am working in a small office with one Branche office, wich have right now a Telephone configuration like this. Branch1 PSTN --Digital Panasonic SystemPBX 20 Telephones Branch2 PSTN --Analog Panasonic SystemPBX 10 Telephones Now they like to connect the second Branch office over VoIP. Something like this: Branch1 PSTN --Digital Panasonic SystemPBX 20 Telephones | VoIP | Branch2 Analog Panasonic SystemPBX 10 Telephones Some phone calls arriving in Branch 1 should be redirected automatically to Branch 2 and all phone calls made from Branch 2 should going out over Branch 1. Which would be the best costs effective solution. Which hardware Routers, Serves, etc. would you recommend. Thank you in adwance Housi Mueller__Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DNS SRV and new Asterisk install
Heya, Just wondering if anyone has deployed a DNS SRV example that I can call to test my new asterisk install? Just want to listen to an IVR or recorded message to test I can call test@test.com or whatever. Can't find one on google :( Cheers, Chris. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.10.18/89 - Release Date: 02/09/2005 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Multiple ASTCC Cards Configuration
You have to use "Brands" and increase the prices for all routes in %.. Make a new brand. Put the increased % in "Markup" * 100. Example: Brand name: "20% profit" Markup : "2000" All prices on the Brand "20% profit" will now be 20% higher. I dont't know how to do it if you want the price to lower. Hope it helped. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Multiple ASTCC Cards Configuration
If you want the price to be lower use a negative percentage. ie -2000 Darren Wiebe [EMAIL PROTECTED] Insider KT wrote: You have to use Brands and increase the prices for all routes in %.. Make a new brand. Put the increased % in Markup * 100. Example: Brand name: 20% profit Markup : 2000 All prices on the Brand 20% profit will now be 20% higher. I dont't know how to do it if you want the price to lower. Hope it helped. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Distortion/crackling/skipping problems on outgoing calls -- please help!!!
Rich Adamson wrote: So, can no one give me any suggestions? Perhaps I can elaborate upon further testing and attempts to debug this tremendously frustrating problem. My softphone (typically IAXComm, but same results connecting via SIP on Xten Xlite) is installed on a P4 1.6 w/ 256 megs of RAM and an integrated sound chipset (Intel/AC97). I've had some problems with this chipset in Linux, and it doesn't support hardware mixing, so I've had to attempt to get dmix and ALSA running in an acceptable fashion; needless to say, I still have problems, and I don't know if this is related. I can record and playback my own voice and other audio (podcasts, Net radio, and music) fine with this headset (some cheap, Chinese $15 headset). However, when it comes to receiving decoded audio on the other end of a VoIP conversation, it sounds scratchy, distorted, crackly, whatever you want to call it. It's not the clarity so much as it is the other things I just mentioned. It's very hard ot put into words, but I'm hoping *someone* can associate with my problem. To make sure it wasn't my Asterisk box that was mucking things up, I connected directly with my softphone to my outgoing VoIP terminators, voipjet and Voxee. Sure enough, it sounded *exactly* the same as it does going through my * box on the LAN and then out through the public Net to voipjet and/or Voxee. Thus, I know it is my side of the equation that is mucking things up, but I cannot for the life of me pinpoint exactly *WHERE* this is taking place. Actually, I also unplugged my headphones and plugged back in the speakers, and it sounded roughly the same, but it's harder to tell because they're not surrounding your ears, quality isn't as clear, and ambient noise can be more easily heard and is distracting. I'm thinking, at this point, that it's my sound card that's messing things up, or its configuration or something. However, can anyone explain *why* non-VoIP-conversations sound perfect on my speakers and headset, while VoIP calls sound very bad? If I were to try to diagnose the above, I'd be using Ethereal to capture the voip packets coming from your itsp's, and analyzing that captured data to look for unusual things. If your itsp connections are sip based, ethereal has a utility to analyze/summarize some of this for you. If those connections are iax based, then you will need to analyze the packets yourself looking for unusual things. Analyzing the packets (either sip or iax) can consume a lot of time, but you really need to ensure those packets are arriving in a consistent manner, timestamps contained within the packets are consecutive and proper, packets are not arriving out of order, etc. At the same time that you're capturing those packets, use the facilities within asterisk to summarize what it thinks is going on (eg, 'iax2 show netstats'), so _that_ data can be correlated to the info derived from the packet captures. If this is a small soho * system, then run ethereal right on the same asterisk system capturing the data as it arrives at the system. Doing so will help identify any issues that you might have involving your local lan, broadband issues, etc, etc. Assuming the problem that you've described is consistent and happens on a regular basis, you don't need to collect and analyze megabytes of packet captures. Just collect a short duration sample that is assured to contain the representative packets associated with the bad audio (maybe five or ten seconds worth). If the analysis does not indicate a problem at that point, then at least you know the problem is internal to asterisk, etc. If you don't feel you have the skills or knowledge to do that analysis, then hire someone that can. Both your original post and the followup post contain a ton of adjectives and adverbs describing a technical problem, but contain little (or no) technical data (such as the results from above or results from various * show commands) that would allow anyone to comment on your problem. So, doubtful anyone is ever going to reply to such postings with anything helpful. Thank you very much for your response. I do acknowledge that my previous posts did not contain much technical information to speak of, but it was mainly because I wasn't/am not familiar with the Asterisk CLI and troubleshooting Asterisk problems, so I apologize for that. I did get the idea early this morning to try to analyze packets with ethereal, and I captured packets when I was made an internal IAX call to the Asterisk system (voicemail). I don't really know what to look for, but I will learn (again, I'm not very familiar with ethereal). Do you hapeople say ve any suggestions for filters to use, to evaluate possible packet loss or resending of data? Regarding the command that you suggested in the CLI, iax2 show netstats, it doesn't recognize that command or anything similar, and 'help' doesn't return anything similar that I can see
Re: [Asterisk-Users] Distortion/crackling/skipping problems on outgoing calls -- please help!!!
Robert Geller wrote: At this point, I'm thinking that it could be a matter of bad cabling or something. The Cat5 cable that's running the 8 or so feet from my PC to my router is homemade by me, and many people do report problems with homemade cables. I may not have made it exactly right, or the untwisted segment may be longer than 1/2, which supposedly causes distortion and interference. Perhaps I ought to run out and buy a couple factory-made cables to test the difference, if any, between them. I had a problem with the precise behavior you describe in your earlier mail. For me, the solution came when I discovered that my Ethernet card and my sound card were sharing an interrupt. I moved the Ethernet card around until it got a different IRQ, and the problem vanished instantly. YMMV. B. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Linux-HA Heartbeat2 and Asterisk
On 9/2/05, Matt Riddell [EMAIL PROTECTED] wrote: Geoff Karl wrote: The new version of heartbeat (http://linux-ha.org/GettingStartedV2) supports up to 16nodes. I was wondering if anyone has tried it with Asterisk. The biggest hurdle would be to configure multiple instances of Asterisk on the same box. Anyone configure more than one copy of asterisk on the same machine? Each instance would need their own set of configuration files. This would be a VOIP only box with a zaptel interface for timing. Um...doesn't running the multiple copies on the same machine kinda make the aim of redundancy redundant? :) Use different contexts if you want to split by companies, then failover to other PCs if you have hardware/network problems. -- Cheers, Matt Riddell When you are building out a HA clustering solution you fail-over an instance to a another machine (node). You also tie a virtual IP address to the asterisk instance. Pretty much everything moves from one machine to another. Lots of times people design a system like Active/Active/Active/Passive; where the Active nodes are running an instance of Asterisk and you have a Passive node to accept any other failed nodes instance. This means that Passive nodes needs to be able to run any Asterisk instance. Really what you are protecting here is hardware failure and file system issues. If you had an Active/Passive design you could run different version of Asterisk. This would allow you to try out a new version and if it failed you could easily switch back to the working version. Geoff ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] newbie install problem. And I already searchedeverywhere!
Op zaterdag 3 september 2005 18:04, schreef Werner Johansson: This configuration looks strange. You know that the channel word sets those channels to the latest settings above, right? The syntax is a bit weird. That means that your signalling looks odd - first you set it to bri_cpe_ptmp and then to bri_net, meaning that bri_net overrides the previous setting. I'm a bit confused about the PRI signalling here, it looks like you're using a BRI card, so why pridialplan? I haven't played around with BRI equipment together with Asterisk (yet) so maybe the PRI signalling should be there.. As per your suggestion I commented the odd things out. No joy still :-) ; ; Zapata telephony interface ; ; Configuration file [channels] ; ; Default language ; ;language=en ; ; Default context ; ; switchtype = euroisdn ; p2mp TE mode ;signalling = bri_cpe_ptmp ; p2p TE mode ;signalling = bri_cpe ; p2mp NT mode ;signalling = bri_net_ptmp ; p2p NT mode signalling = bri_net ;pridialplan = national ;prilocaldialplan = local ;nationalprefix = 0 ;internationalprefix = 00 ;echocancel=yes ;echotraining = 100 ;echocancelwhenbridged=yes ;immediate=yes group = 1 context = default channel = 1-2 What is ztcfg -vv giving as output if you run it? server:/var/log/asterisk # ztcfg -vv Zaptel Configuration == SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1) Channel map: Channel 01: Individual Clear channel (Default) (Slaves: 01) Channel 02: Individual Clear channel (Default) (Slaves: 02) Channel 03: D-channel (Default) (Slaves: 03) 3 channels configured. server:/var/log/asterisk # BTW: thanks for looking in to this. kind regards, -- Jeroen Baten| EMAIL : [EMAIL PROTECTED] _ __ | web : www.i2rs.nl | )|_)(_ | tel : +31 (0)499 477 688 _|_/_| \__) | fax : +31 (0)499 476 804 Roerlaan 36, 5691 HJ, Son, the Netherlands ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] newbie install problem. And I already searchedeverywhere!
Op zaterdag 3 september 2005 18:04, schreef Werner Johansson: This configuration looks strange. You know that the channel word sets those channels to the latest settings above, right? The syntax is a bit weird. That means that your signalling looks odd - first you set it to bri_cpe_ptmp and then to bri_net, meaning that bri_net overrides the previous setting. I'm a bit confused about the PRI signalling here, it looks like you're using a BRI card, so why pridialplan? I haven't played around with BRI equipment together with Asterisk (yet) so maybe the PRI signalling should be there.. I just changed in zapata.conf the signalling to: signalling = bri_net_ptmp and this changes the output considerably: *CLI -- Executing Dial(SIP/4001-2fea, Zap/2/mycellphonenumber}) in new stack -- Called mycellphonenumber} -- Channel 0/2, span 1 got hangup -- Hungup 'Zap/2-1' == No one is available to answer at this time *CLI Cool. Too bad I seem to be unavailable :-) -- Jeroen Baten| EMAIL : [EMAIL PROTECTED] _ __ | web : www.i2rs.nl | )|_)(_ | tel : +31 (0)499 477 688 _|_/_| \__) | fax : +31 (0)499 476 804 Roerlaan 36, 5691 HJ, Son, the Netherlands ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] *81, block CID, using ATA
I searched the wiki for a solution to allow a user on an analog ATA to send *81 to block Asterisk CID (or any other * code). The ATA has *81 built in to block the CID the ATA generates. Any examples would be appreciated. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] *81, block CID, using ATA
I searched the wiki for a solution to allow a user on an analog ATA to send *81 to block Asterisk CID (or any other * code). The ATA has *81 built in to block the CID the ATA generates. Any examples would be appreciated. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] unicall deploy
Hi every one . There are any out there that have a unicall deploy working without problem ? Can give me some tips or referenece about his config ? Regards ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] newbie install problem. And I already searchedeverywhere!
On Sat, 2005-09-03 at 20:47 +0200, Jeroen Baten wrote: I just changed in zapata.conf the signalling to: signalling = bri_net_ptmp and this changes the output considerably: *CLI -- Executing Dial(SIP/4001-2fea, Zap/2/mycellphonenumber}) in new stack -- Called mycellphonenumber} -- Channel 0/2, span 1 got hangup -- Hungup 'Zap/2-1' == No one is available to answer at this time *CLI That's exactly the same as I got on Friday. I'd just changed from a Fritz! with chan_capi to a 4 way BRI, reverting to the Fritz! restored everything. -- Dave Cotton [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_iax2.c:7672 iax2_poke_noanswer
I have two units at customer locations in the Caribbean registering to a server in the US. Both units are connected to the Cable TV company's internet feed. If I run mtr to the units I see clean internet and low latency, but when I watch the CLI, I see constant problems. The audio quality is terrible, but I can't see why. Sep 3 16:42:46 NOTICE[20423]: chan_iax2.c:7014 socket_read: Peer 'islanddream' is now REACHABLE! Time: 96 -- Registered IAX2 'islanddream' (AUTHENTICATED) at 204.14.xxx.139:4569 Sep 3 16:45:50 NOTICE[20423]: chan_iax2.c:7672 iax2_poke_noanswer: Peer 'islanddream' is now UNREACHABLE! Time: 0 Sep 3 16:46:00 NOTICE[20423]: chan_iax2.c:7014 socket_read: Peer 'islanddream' is now REACHABLE! Time: 103 -- Registered IAX2 'islanddream' (AUTHENTICATED) at 204.14.xxx.139:4569 Sep 3 16:50:25 NOTICE[20421]: chan_sip.c:8102 handle_response: Peer '832' is now REACHABLE! (1179ms / 2000ms) -- Registered IAX2 'islanddream' (AUTHENTICATED) at 204.14.xxx.139:4569 Sep 3 16:54:18 NOTICE[20423]: chan_iax2.c:7672 iax2_poke_noanswer: Peer 'islanddream' is now UNREACHABLE! Time: 96 Sep 3 16:54:28 NOTICE[20423]: chan_iax2.c:7014 socket_read: Peer 'islanddream' is now REACHABLE! Time: 95 Sep 3 16:55:32 NOTICE[20423]: chan_iax2.c:7672 iax2_poke_noanswer: Peer 'islanddream' is now UNREACHABLE! Time: 95 Sep 3 16:56:24 NOTICE[20423]: chan_iax2.c:7014 socket_read: Peer 'islanddream' is now REACHABLE! Time: 95 Ping results: 64 bytes from 204.14.xxx.139: icmp_seq=33 ttl=47 time=95.7 ms 34 packets transmitted, 34 received, 0% packet loss -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Distortion/crackling/skipping problems on outgoing calls -- please help!!!
Brian Capouch wrote: Robert Geller wrote: At this point, I'm thinking that it could be a matter of bad cabling or something. The Cat5 cable that's running the 8 or so feet from my PC to my router is homemade by me, and many people do report problems with homemade cables. I may not have made it exactly right, or the untwisted segment may be longer than 1/2, which supposedly causes distortion and interference. Perhaps I ought to run out and buy a couple factory-made cables to test the difference, if any, between them. I had a problem with the precise behavior you describe in your earlier mail. For me, the solution came when I discovered that my Ethernet card and my sound card were sharing an interrupt. I moved the Ethernet card around until it got a different IRQ, and the problem vanished instantly. YMMV. B. Wow, very interesting. Thank you so much! BTW, regarding YMMV, did you have a separate, dedicated sound card? I don't -- it's integrated into my motherboard. Would this still apply? Of course, there are still ports in the back for in, out, and a mic, so it may still apply, but my Ethernet card is way down on the second-to-last PCI port, so would this still apply? Again, thanks very much for your support, and if you think this may still apply to my setup, I will definitely try what you recommended. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Distortion/crackling/skipping problems on outgoing calls -- please help!!!
Robert Geller wrote: Wow, very interesting. Thank you so much! BTW, regarding YMMV, did you have a separate, dedicated sound card? I don't -- it's integrated into my motherboard. Would this still apply? Of course, there are still ports in the back for in, out, and a mic, so it may still apply, but my Ethernet card is way down on the second-to-last PCI port, so would this still apply? Again, thanks very much for your support, and if you think this may still apply to my setup, I will definitely try what you recommended. Check what you get when you cat /proc/interrupts If you can get hold of a hardphone to play with, and it works just fine when it's on the same switch as the computer, then I would be tempted even further to suspect an interrupt-related problem as your culprit. B. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] I connected my quicknet phonejack to the wall phone outlet and .......
waited for the aroma of burnt electronics. There wasn't any so I went to the next room and plugged in a telephone on the wall outlet. I picked up the handset and it rang my asterisk box. I dont have any phone service. The phone is working fine and have used it for hours listening to podcasts. And I can plug it to any phone in the house. Now I am thinking if I can plug another phone somewhere else in the house so I can have one in the kitchen too. Do you think this is asking for trouble? Assuming nothing gets fried, what happens if both handsets are picked up? Will the one phone just work as an extension. Thanks. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Distortion/crackling/skipping problems on outgoing calls -- please help!!!
Brian Capouch wrote: Robert Geller wrote: Wow, very interesting. Thank you so much! BTW, regarding YMMV, did you have a separate, dedicated sound card? I don't -- it's integrated into my motherboard. Would this still apply? Of course, there are still ports in the back for in, out, and a mic, so it may still apply, but my Ethernet card is way down on the second-to-last PCI port, so would this still apply? Again, thanks very much for your support, and if you think this may still apply to my setup, I will definitely try what you recommended. Check what you get when you cat /proc/interrupts If you can get hold of a hardphone to play with, and it works just fine when it's on the same switch as the computer, then I would be tempted even further to suspect an interrupt-related problem as your culprit. B. What should I be looking for in /proc/interrupts? If the first field in each row is the IRQ, I don't see any of the same numbers listed, so would that mean there are no conflicts? Earlier, I ordered a GXP-2000, which will hopefully ship from Voxilla on Tuesday, so that will be a very good indicator of just what my problem is, I think. However, in the meantime, I *really* would like to get this figured out! Leaving problems unfixed, to me, is simply a non-option. Thank you very much, again! Regards, Robert ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Distortion/crackling/skipping problems on outgoing calls -- please help!!!
Thank you very much for your response. I do acknowledge that my previous posts did not contain much technical information to speak of, but it was mainly because I wasn't/am not familiar with the Asterisk CLI and troubleshooting Asterisk problems, so I apologize for that. I did get the idea early this morning to try to analyze packets with ethereal, and I captured packets when I was made an internal IAX call to the Asterisk system (voicemail). I don't really know what to look for, but I will learn (again, I'm not very familiar with ethereal). Do you hapeople say ve any suggestions for filters to use, to evaluate possible packet loss or resending of data? An important item to look at in each packet is the timestamp. In sip packets, the timestamp should be increasing by 160 for each conseq pkt. In iax packets, the timestamp should be increasing by 20 for each pkt. So if you see a timestamp of 3290 in one pkt and 3310 in the next (a diff of 20), that's good. Notice the increasing timestamp value and the diff. If pkt 3310 arrives before 3290, then something in the network is delaying the delivery of packets so as to cause them to not arrive in the proper order. If there are missing packets, then you'll see timestamps jumping by 40, 60, 80 or some other value (diff) for iax packets, or, similar for sip packets. Regarding the command that you suggested in the CLI, iax2 show netstats, it doesn't recognize that command or anything similar, and 'help' doesn't return anything similar that I can see (I'm using 1.0.7 if that helps). Since 1.0.7 is rather old (in the scheme of things), I'd suggest you install something newer to play with. There has been a ton of stuff that has changed since 1.0.7, but I don't recall if those changes would have anything to do with your problem. (I use nothing but cvs head, but I kind of keep an eye on how many changes are happening (and for what reason), and upgrade when the number of problems seem to be at a low. The 'iax2 show netstats' would have been added in a later version. At this point, I'm thinking that it could be a matter of bad cabling or something. The Cat5 cable that's running the 8 or so feet from my PC to my router is homemade by me, and many people do report problems with homemade cables. I may not have made it exactly right, or the untwisted segment may be longer than 1/2, which supposedly causes distortion and interference. Perhaps I ought to run out and buy a couple factory-made cables to test the difference, if any, between them. Replacing the cable would probably be a good start since they are relatively cheap. Go buy a new one so there's no question about its quality. Also, keep the cable at least a little distance away from transformers, ballasts, and other things that tend to generate tons of electical noise. (Some desk lamps even have extremely noisy transformers or ballasts in them.) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Distortion/crackling/skipping problems on outgoing calls -- please help!!!
Robert Geller wrote: What should I be looking for in /proc/interrupts? If the first field in each row is the IRQ, I don't see any of the same numbers listed, so would that mean there are no conflicts? Why don't you include the output in your mail? B. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Distortion/crackling/skipping problems on outgoing calls -- please help!!!
Wow, very interesting. Thank you so much! BTW, regarding YMMV, did you have a separate, dedicated sound card? I don't -- it's integrated into my motherboard. Would this still apply? Of course, there are still ports in the back for in, out, and a mic, so it may still apply, but my Ethernet card is way down on the second-to-last PCI port, so would this still apply? Again, thanks very much for your support, and if you think this may still apply to my setup, I will definitely try what you recommended. Check what you get when you cat /proc/interrupts If you can get hold of a hardphone to play with, and it works just fine when it's on the same switch as the computer, then I would be tempted even further to suspect an interrupt-related problem as your culprit. B. What should I be looking for in /proc/interrupts? If the first field in each row is the IRQ, I don't see any of the same numbers listed, so would that mean there are no conflicts? Earlier, I ordered a GXP-2000, which will hopefully ship from Voxilla on Tuesday, so that will be a very good indicator of just what my problem is, I think. However, in the meantime, I *really* would like to get this figured out! Leaving problems unfixed, to me, is simply a non-option. You should see something like: [EMAIL PROTECTED] asterisk]# cat /proc/interrupts CPU0 0: 201066464 XT-PIC timer 1: 5421 XT-PIC i8042 2: 0 XT-PIC cascade 8: 1 XT-PIC rtc 9: 0 XT-PIC acpi 10:1646446 XT-PIC eth0 11: 200987949 XT-PIC libata, wctdm 12: 148358 XT-PIC i8042 14: 208303 XT-PIC ide0 15:1808413 XT-PIC ide1 NMI: 0 ERR: 0 [EMAIL PROTECTED] asterisk]# In the above, the ethernet card is on interrupt 10 and my digium TDM card is on 11. Look for your ethernet in the list and see if anything is listed next to it. (In the above libata and wctdm are sharing an interrupt. You really don't want anything sharing an interupt with your ethernet card.) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Argentina - zapata.conf switchtype for Argentina
Just to receive a recommendation on switchtype for Argentina, Buenos Aires, 114816. Thanks a lot-- Leandro Rzezak[EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How Separate a few channels from the others on a PRI
Okay, here is the background. I have a PRI with 15 active channels on it. I originally setup all of them in group=1 and all outgoing and incoming calls used this group. The phone number that I have associated with these channels ends with 750 and that is how I direct the calls. i.e. In my extensions.conf I have: exten = 750,1,Dial(SIP/120,20) All this works fine. Now I have the need to separate out three of the channels (13-15). I am using the associated phone number ending in 767 for this purpose. I have currently changed the zapata.conf to look like this: signalling=pri_cpe switchtype=national echocancel=yes echocancelwhenbridged=yes echotraining=yes callerid=asreceived group=1 context=default channel=1-12 group=2 context=fax channel=13-15 To my understanding this is supposed to separate channels 1-12 into group 1 and channels 13-15 into group 2. If that is true, that's fine, but it doesn't help me with my current issue. This is what I would like... When a call comes into the phone number ending in 750 it uses up the 12 channels in the first group, and when a call comes into the phone number ending in 767 it uses the 3 channels in group 2 AND USES NO MORE CHANNELS, only those three. The reason for all of this is that I have faxing through asterisk working and want those last three channels used for faxing but I do not want the fax lines eating up all my channels and leaving none for voice calls. Please help. Thanks. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How To Separate incoming channels from the others on a PRI
Okay, here is the background. I have a PRI with 15 active channels on it. I originally setup all of them in group=1 and all outgoing and incoming calls used this group. The phone number that I have associated with these channels ends with 750 and that is how I direct the calls. i.e. In my extensions.conf I have: exten = 750,1,Dial(SIP/120,20) All this works fine. Now I have the need to separate out three of the channels (13-15). I am using the associated phone number ending in 767 for this purpose. I have currently changed the zapata.conf to look like this: signalling=pri_cpe switchtype=national echocancel=yes echocancelwhenbridged=yes echotraining=yes callerid=asreceived group=1 context=default channel=1-12 group=2 context=fax channel=13-15 To my understanding this is supposed to separate channels 1-12 into group 1 and channels 13-15 into group 2. If that is true, that's fine, but it doesn't help me with my current issue. This is what I would like... When a call comes into the phone number ending in 750 it uses up the 12 channels in the first group, and when a call comes into the phone number ending in 767 it uses the 3 channels in group 2 AND USES NO MORE CHANNELS, only those three. The reason for all of this is that I have faxing through asterisk working and want those last three channels used for faxing but I do not want the fax lines eating up all my channels and leaving none for voice calls. Please help. Thanks. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] newbie install problem. And I alreadysearchedeverywhere!
On Sat, 2005-09-03 at 20:47 +0200, Jeroen Baten wrote: I just changed in zapata.conf the signalling to: signalling = bri_net_ptmp and this changes the output considerably: *CLI -- Executing Dial(SIP/4001-2fea, Zap/2/mycellphonenumber}) in new stack -- Called mycellphonenumber} -- Channel 0/2, span 1 got hangup -- Hungup 'Zap/2-1' == No one is available to answer at this time *CLI Ok - what the heck is this?v Dial(SIP/4001-2fea, Zap/2/mycellphonenumber}) I've never seen a } used in a dial statement... I saw that in your first message and (too) quickly deleted it. Brett ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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: equipment configuration help
Erick Perez wrote: So, with this i solve the issue on main office. But what about the two remote? they are so little that they will not let me place another * box there. The phones will be SIP and they are like this INTERNET--PIX--LAN(machines and sip phones). The pixes in those two offices have an ipsec tunnel with the main office via internet. I was thinking of placing the asterisk with a public IP so the remote phones can NAT outside to the public asterisk located in the main office. What do you think? On 9/2/05, asterisk groups [EMAIL PROTECTED] wrote: That is correct. Normally the layer 3 switches include advanced features such as QoS but they may be available on simpler layer 2 switches. I think the key words to look for are 'Managed, QoS (802.1p) with priority queues, VLAN, (802.1q)'...maybe even PoE if you go with some SIP phones in the future that can be powered by Power Over Ethernet. Something else to keep in mind. best of luck. On Thu, 2005-09-01 at 22:03 -0500, Erick Perez wrote: Why an L3? just for the QoS part? I checked the alliedtelesyn 8624T at $1000.00 http://www.cdw.com/shop/products/default.aspx?EDC=772793 but i also looked at the 8550T which has 48 port 10-100 but L2 http://www.cdw.com/shop/products/default.aspx?EDC=773964RecommendedForEDC=772793RecoType=upsell at 900.00 is the QoS different? sorry for the question but i keep reading that asterisk needs qos to function better. Thanks, On 9/1/05, asterisk groups [EMAIL PROTECTED] wrote: Erick- Can't say if they will or not. In theory they should respect all outgoing traffic unless being filtered by another device such as your PIX. You might want to check with the ADSL router manufacturer just to be safe. On Thu, 2005-09-01 at 09:25 -0500, Erick Perez wrote: Do i have to change the adsl routers? or just do QoS with the Layer 3 switches? Will my ADSL router respect the QoS setting when sending the packet to the Internet? On 9/1/05, asterisk groups [EMAIL PROTECTED] wrote: Erick, After reviewing your original message a little closer it occurs to me that you may be able to trunk Asterisk---Meridian with 2 Digium TDM400 cards. These are Quad FXS or FXO cards that could receive the lines from your 8 analog line card. You'll still need an E1 card (Digium or Sangoma) to terminate your PRI line, but maybe with those TDM400 cards you can avoid the added cost of a channel bank. Regarding your WAN and branch offices; 1. I've seen comments that tunneling VoIP traffic through IPSec can add overhead/delay that could impact voice quality. Something to keep in mind. I have trunked Asterisk boxes in MIA,BUE,SAO, etc. but trunk with IAX over the Internet not tunneled or encrypted and performance is fine. 2. In your two locations with 15 50 users you should consider installing Asterisk boxes in those locations and trunking them together with IAX over the Internet. Perhaps go ahead and do the same thing with the smaller office. You can justify a small Asterisk implementation in an office with 5 phones. 3. For QoS look for L3 managed switches that can do QoS and/or bandwidth allocation. Cisco, Dell, Nortel, HP can all do this, maybe even more economical D-Links. Put these behind your PIX. It is also recommended to do separate VLANs for any SIP hard phones you deploy. This adds another layer of security and reliability. Hope this helps. On Wed, 2005-08-31 at 21:43 -0500, Erick Perez wrote: -M, The norstar has no E1 card, i will have to ask the nortel provider for the cost of it and configuration prices. I might end up paying the same as the channel bank. I was also thinking of using a Citel SIP-N-NORSTAR converter but its priced at around 3k. Too expensive because its only 24 ports and i have 32 nortel phones. According to this wiki http://www.voip-info.org/tiki-index.php?page=Asterisk+Nortel One problem with this approach is that in a Norstar system, it isn't easy to forward an extension to an outside line, which means Norstar phone users will have to remember to do something different when they want to call a user who has been switched to an IP phone for example. I guess that can be sorted out. Any manuals out there for configuration like [Telephone Company] ---E1--- [Asterisk with sangoma s102] ---T1 channel bank--- [Norstar]? (only the asterisk-t1-norstar part) Now another section, networking. The 3 offices are linked via VPNs like this Internet---ADSL Router-Cisco PIX Firewall---LAN doin ip tunneling will solve all communication problems internally, but what about QoS and SIP phones being taken to the public internet? one office has 5 users, the other 15, the other 50. ADSL Router recommendiations? and as for the phones being taken to the outside? what kind of configuration do i use? IAX is not an option. On 8/31/05, asterisk groups [EMAIL PROTECTED] wrote: Erick, Consider trunking your Meridian to the Asterisk via an E1 card on the Nortel.
Re: [Asterisk-Users] Polycom 301 second line registration,
Hi, Thanks Jeremy for replying, the extensions that matches the Polycom extension is set to listen on 5061, I don't know if Asterisk needs other setting besides that in orther to listen port 5061, but I tryied before with port 5060 too, but same negative result, I don't what else to try, any one any idea about the registration string in the Polycom? Thanks again, Andres Jeremy wrote: Is your Asterisk server listening on port 5061? If not, just change the entry to 5060. Hi, I am having problems on getting the second line to work on a Polycom 301, this is the phone.cfg file, the * box is 192.168.1.8 http://192.168.1.8 and the phone is 192.168.1.18 http://192.168.1.18 I am not 100% sure about what the reg.x.address should be, with this setting I only get the line number to work, the second just gives me busy signal, and its extension is not available. I also tried [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] and 203 as the reg.2.address parameter but without success, the 203 extension setting in Asterisk is a clon of the 200 except for the id and the port, (that matches this conf file) PHONE_CONFIG OVERRIDES reg.1.displayName=FD1 reg.1.label=L1 reg.1.address=192.168.1.18 http://192.168.1.18 reg.1.server.1.address=192.168.1.8 http://192.168.1.8 reg.1.server.1.port=5060 reg.1.auth.userId=200 reg.1.auth.password=123 reg.2.displayName=FD2 reg.2.label=L2 reg.2.address=[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] reg.2.auth.userId=203 reg.2.auth.password=123 reg.2.server.1.address=192.168.1.8 http://192.168.1.8 reg.2.server.1.port=5061/ /PHONE_CONFIG Thanks for any help, Andres ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Any suggestions for an IP phone?
yes, a regular ftp server will do it fine, I have another issue with a Poly300 but not ftp, bear in mind that when you set up the Polycom user, -PlcmIp (or something like that) you have to use the --force-badname option to allow your server a username with capital letters, since that is case sensitive, Chris Mason (Lists) wrote: [EMAIL PROTECTED] wrote: Polycom does not support Asterisk. Thsi does not mean phones do not work with it. Rudolf P.S. I am having troubles setting up Polycom 300 with tftp server. By some reason phones always say can not contact boot server. Phones are set to use tftp and correct boot server IP is set via dhcp. I will investigate further, but any suggestions are appreciated. I always use FTP instead, it works famously. Make sure you configure the ftp server in DHCP or in the ftp servers settings, as an IP of course, and that you change the ftp password to the password for the user PlcmSpIp on the server. After that it's flawless. Polycom does not support Asterisk. Polycom, the company, does not support the use of the phones with Asterisk. Who cares? SIP is a standard, we don't need any help from them and we don't need their blessing. The phones are excellent quality and work very well with Asterisk, there's no support issue. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MWI - message waiting indication
hello, I read http://www.voip-info.org/tiki-index.php?page=Asterisk+at+large anybody could tell me more about this ? Is it available with ARA ? Regards Harry Method 3 Q: If you have your SIP phones registered with SER but your voicemail is handled by asterisk, how do you get the MWI (Message Waiting Indicator) light to function on the phone? A: In sip.conf create a section pointing at your SER router. [ser] type=friend; We allow incoming and outgoing calls. Use peer if you are only doing MWI context=ser; This is the context incoming calls land in host=ser.server.tld; This is the hostname or IP address of your SER server fromdomain=ser.server.rld ; This is your SER_DOMAIN insecure=very ; This allows incoming calls from the phones routing through ser to be passed into asterisk [EMAIL PROTECTED] ; This is where you list the voicemail boxes to monitor This tells asterisk that if a voicemail comes in to user then it needs to send a SIP NOTIFY message to the ser.server.tld phone. Well this is all well and good except how does SER deliver this NOTIFY to the phones? First thing is that you need to make a tiny change to the asterisk code to pass the mailbox user in the SIP NOTIFY packet. --- channels/chan_sip.c.origThu Jul 14 12:03:18 2005 +++ channels/chan_sip.c Thu Jul 14 12:05:26 2005 @@ -9710,6 +9710,7 @@ /* Called with peerl lock, but releases it */ struct sip_pvt *p; int newmsgs, oldmsgs; + char *s; /* Check for messages */ ast_app_messagecount(peer-mailbox, newmsgs, oldmsgs); @@ -9735,6 +9736,10 @@ /* Recalculate our side, and recalculate Call ID */ if (ast_sip_ouraddrfor(p-sa.sin_addr,p-ourip)) memcpy(p-ourip, __ourip, sizeof(p-ourip)); + strcpy(p - username, peer - mailbox); /* Username = Mailbox name */ + s = strchr(p - username, '@'); /* Remove the context part */ + if (s != NULL) +*s = 0; build_via(p, p-via, sizeof(p-via)); build_callid(p-callid, sizeof(p-callid), p-ourip, p-fromdomain); /* Send MWI */ After this patch is applied, the MWI NOTIFY messages coming from asterisk will have the URI [EMAIL PROTECTED] This can be then routed with ser to the correct phone with normal SER routing rules. ie. SER does a lookup(location) and then a t_relay(). I don't believe this patch should effect any non-ser controlled sip phones. For me, this method was a lot easier then Method 2 listed above. You can add as may mailbox's as you like into the mailbox= line in the asterisk sip.conf file. One possible problem is if you have a mailbox called [EMAIL PROTECTED] and another called [EMAIL PROTECTED], this patch will make the MWI indicator light up for phone [EMAIL PROTECTED] when either mailbox gets a message. A simple modification to the patch and SER could be used to handle multiple contexts if required however this simplification is sufficient for me. ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Mplayer as replacement to mgp123 in MP3Player cmd
mp3 doesn't work, I recoded everything to gsm using a batch script with mplayer and sox. It took a couple of days but everything is working fine. Thanks. On Mon, 2005-08-29 at 16:00, Kris Edwards wrote: cmisip wrote: Controlplayback with the wealth of codecs supported by mplayer would be nice though as one of my future plans is sending a tv audio source through asterisk. I've done that w/ a radio, allbeit a bit convoluted setup. I was just running the live audio on an mp3 stream and then using the mp3 stream as an onhold class. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sipura spa841 problems
Guys. I just unpacked on of the new spa841 I orderd and I was changing the ringtone (and listening to the options) when suddently the phone stopped playing back the tones and now the phone doesn't ring, speaker doesn't work and no ringtone play can be heard. Has anybody had this kind of problems? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Distortion/crackling/skipping problems on outgoing calls -- please help!!!
Brian Capouch wrote: Robert Geller wrote: What should I be looking for in /proc/interrupts? If the first field in each row is the IRQ, I don't see any of the same numbers listed, so would that mean there are no conflicts? Why don't you include the output in your mail? B. CPU0 0: 79766966IO-APIC-edge timer 1: 41185IO-APIC-edge i8042 7: 2IO-APIC-edge parport0 8: 1IO-APIC-edge rtc 9: 0 IO-APIC-level acpi 12: 72IO-APIC-edge i8042 14: 213361IO-APIC-edge ide0 15: 876060IO-APIC-edge ide1 169: 0 IO-APIC-level uhci_hcd 177: 61956 IO-APIC-level Intel 82801BA-ICH2, eth2 185: 0 IO-APIC-level uhci_hcd 193: 2 IO-APIC-level ohci1394 201:2008201 IO-APIC-level eth0 NMI: 0 LOC: 79777609 ERR: 0 MIS: 0 Interesting. Eth0 is my wired LAN interface and eth2 is my wireless LAN interface. Usually, I disable the wireless for obvious reasons (it's installed only because I previously had a wireless solution when I couldn't wire the house, but I recently did so I now have wired), but now that it /and/ eth0 are enabled, I don't know which one applications are using! It's quite possible that the softphone is using eth2, which would probably cause problems as you previously mentioned. Well, I disabled eth2 -- since I don't use/want to use it anyway -- with ifconfig eth2 down, but /proc/interrupts still shows the same entry for eth2 sharing the interrupt, which, as you and Rich said, isn't good. Perhaps I should disable eth2, reboot, and see what /proc/interrupts is then, along with how the softphone sounds? Note that I can listen to streaming music and other files perfectly fine--does this still apply? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Distortion/crackling/skipping problems on outgoing calls -- please help!!!
Rich Adamson wrote: Thank you very much for your response. I do acknowledge that my previous posts did not contain much technical information to speak of, but it was mainly because I wasn't/am not familiar with the Asterisk CLI and troubleshooting Asterisk problems, so I apologize for that. I did get the idea early this morning to try to analyze packets with ethereal, and I captured packets when I was made an internal IAX call to the Asterisk system (voicemail). I don't really know what to look for, but I will learn (again, I'm not very familiar with ethereal). Do you hapeople say ve any suggestions for filters to use, to evaluate possible packet loss or resending of data? An important item to look at in each packet is the timestamp. In sip packets, the timestamp should be increasing by 160 for each conseq pkt. In iax packets, the timestamp should be increasing by 20 for each pkt. So if you see a timestamp of 3290 in one pkt and 3310 in the next (a diff of 20), that's good. Notice the increasing timestamp value and the diff. If pkt 3310 arrives before 3290, then something in the network is delaying the delivery of packets so as to cause them to not arrive in the proper order. If there are missing packets, then you'll see timestamps jumping by 40, 60, 80 or some other value (diff) for iax packets, or, similar for sip packets. Regarding the command that you suggested in the CLI, iax2 show netstats, it doesn't recognize that command or anything similar, and 'help' doesn't return anything similar that I can see (I'm using 1.0.7 if that helps). Since 1.0.7 is rather old (in the scheme of things), I'd suggest you install something newer to play with. There has been a ton of stuff that has changed since 1.0.7, but I don't recall if those changes would have anything to do with your problem. (I use nothing but cvs head, but I kind of keep an eye on how many changes are happening (and for what reason), and upgrade when the number of problems seem to be at a low. The 'iax2 show netstats' would have been added in a later version. At this point, I'm thinking that it could be a matter of bad cabling or something. The Cat5 cable that's running the 8 or so feet from my PC to my router is homemade by me, and many people do report problems with homemade cables. I may not have made it exactly right, or the untwisted segment may be longer than 1/2, which supposedly causes distortion and interference. Perhaps I ought to run out and buy a couple factory-made cables to test the difference, if any, between them. Replacing the cable would probably be a good start since they are relatively cheap. Go buy a new one so there's no question about its quality. Also, keep the cable at least a little distance away from transformers, ballasts, and other things that tend to generate tons of electical noise. (Some desk lamps even have extremely noisy transformers or ballasts in them.) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Your advice was *extremely* helpful. It seems that I learn something new each time I read you all's posts. To me, it looked like each packet was correctly sent at the right interval each time, but I didn't evaluate each one. However, the general trend is that there seems to be no packet loss or resending. I could buy another cable as well, just to be safe, but it seems to me the potential IRQ conflict is more the more likely problem--of course, even when I ifconfig eth2 downed the interface, it still showed up in /proc/interrupts -- does bringing the interface down not completely bring it down? Should I permanently disable it and reboot? Again, thank you very much for your ongoing help; I feel like I'm paying (or ought to) for professional support here. :-) Regards, Robert Geller ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Argentina - zapata.conf switchtype for Argentina
Probably you need to use unicall+mfcr2 support instead of zapata, as Argentina uses R2. Guillermo From: Leandro Rzezak [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED],Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Argentina - zapata.conf switchtype for Argentina Date: Sat, 3 Sep 2005 18:54:59 -0300 Just to receive a recommendation on switchtype for Argentina, Buenos Aires, 114816. Thanks a lot -- Leandro Rzezak [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] unicall deploy
I´m using an unicall box with 4 E1 lines getting between 6000-15000 calls per day, and between 15-30 operators using AgentLogin, all using R2 signaling to the telco and a local PBX. I´m using the Argentina variant, and using the last version of unicall 0.0.2 and asterisk 1.0.7 Guillermo From: acriollo [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED],Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] unicall deploy Date: Sat, 3 Sep 2005 15:04:20 -0500 Hi every one . There are any out there that have a unicall deploy working without problem ? Can give me some tips or referenece about his config ? Regards ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Distortion/crackling/skipping problems on outgoing calls -- please help!!!
Robert Geller wrote: Rich Adamson wrote: Thank you very much for your response. I do acknowledge that my previous posts did not contain much technical information to speak of, but it was mainly because I wasn't/am not familiar with the Asterisk CLI and troubleshooting Asterisk problems, so I apologize for that. I did get the idea early this morning to try to analyze packets with ethereal, and I captured packets when I was made an internal IAX call to the Asterisk system (voicemail). I don't really know what to look for, but I will learn (again, I'm not very familiar with ethereal). Do you hapeople say ve any suggestions for filters to use, to evaluate possible packet loss or resending of data? An important item to look at in each packet is the timestamp. In sip packets, the timestamp should be increasing by 160 for each conseq pkt. In iax packets, the timestamp should be increasing by 20 for each pkt. So if you see a timestamp of 3290 in one pkt and 3310 in the next (a diff of 20), that's good. Notice the increasing timestamp value and the diff. If pkt 3310 arrives before 3290, then something in the network is delaying the delivery of packets so as to cause them to not arrive in the proper order. If there are missing packets, then you'll see timestamps jumping by 40, 60, 80 or some other value (diff) for iax packets, or, similar for sip packets. Regarding the command that you suggested in the CLI, iax2 show netstats, it doesn't recognize that command or anything similar, and 'help' doesn't return anything similar that I can see (I'm using 1.0.7 if that helps). Since 1.0.7 is rather old (in the scheme of things), I'd suggest you install something newer to play with. There has been a ton of stuff that has changed since 1.0.7, but I don't recall if those changes would have anything to do with your problem. (I use nothing but cvs head, but I kind of keep an eye on how many changes are happening (and for what reason), and upgrade when the number of problems seem to be at a low. The 'iax2 show netstats' would have been added in a later version. At this point, I'm thinking that it could be a matter of bad cabling or something. The Cat5 cable that's running the 8 or so feet from my PC to my router is homemade by me, and many people do report problems with homemade cables. I may not have made it exactly right, or the untwisted segment may be longer than 1/2, which supposedly causes distortion and interference. Perhaps I ought to run out and buy a couple factory-made cables to test the difference, if any, between them. Replacing the cable would probably be a good start since they are relatively cheap. Go buy a new one so there's no question about its quality. Also, keep the cable at least a little distance away from transformers, ballasts, and other things that tend to generate tons of electical noise. (Some desk lamps even have extremely noisy transformers or ballasts in them.) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Your advice was *extremely* helpful. It seems that I learn something new each time I read you all's posts. To me, it looked like each packet was correctly sent at the right interval each time, but I didn't evaluate each one. However, the general trend is that there seems to be no packet loss or resending. I could buy another cable as well, just to be safe, but it seems to me the potential IRQ conflict is more the more likely problem--of course, even when I ifconfig eth2 downed the interface, it still showed up in /proc/interrupts -- does bringing the interface down not completely bring it down? Should I permanently disable it and reboot? Again, thank you very much for your ongoing help; I feel like I'm paying (or ought to) for professional support here. :-) Regards, Robert Geller ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I modprobe -r'ed prism54, the wireless kernel modules for my card, and here's what /proc/interrupts looks like: [EMAIL PROTECTED]:~/torrents$ cat /proc/interrupts CPU0 0: 81981003IO-APIC-edge timer 1: 49755IO-APIC-edge i8042 7: 2IO-APIC-edge parport0 8: 1IO-APIC-edge rtc 9: 0 IO-APIC-level acpi 12: 499429IO-APIC-edge i8042 14: 224679IO-APIC-edge ide0 15: 900392IO-APIC-edge ide1 169: 0 IO-APIC-level uhci_hcd 177: 67917
Re: [Asterisk-Users] Distortion/crackling/skipping problems on outgoing calls -- please help!!!
Robert Geller wrote: Robert Geller wrote: Rich Adamson wrote: Thank you very much for your response. I do acknowledge that my previous posts did not contain much technical information to speak of, but it was mainly because I wasn't/am not familiar with the Asterisk CLI and troubleshooting Asterisk problems, so I apologize for that. I did get the idea early this morning to try to analyze packets with ethereal, and I captured packets when I was made an internal IAX call to the Asterisk system (voicemail). I don't really know what to look for, but I will learn (again, I'm not very familiar with ethereal). Do you hapeople say ve any suggestions for filters to use, to evaluate possible packet loss or resending of data? An important item to look at in each packet is the timestamp. In sip packets, the timestamp should be increasing by 160 for each conseq pkt. In iax packets, the timestamp should be increasing by 20 for each pkt. So if you see a timestamp of 3290 in one pkt and 3310 in the next (a diff of 20), that's good. Notice the increasing timestamp value and the diff. If pkt 3310 arrives before 3290, then something in the network is delaying the delivery of packets so as to cause them to not arrive in the proper order. If there are missing packets, then you'll see timestamps jumping by 40, 60, 80 or some other value (diff) for iax packets, or, similar for sip packets. Regarding the command that you suggested in the CLI, iax2 show netstats, it doesn't recognize that command or anything similar, and 'help' doesn't return anything similar that I can see (I'm using 1.0.7 if that helps). Since 1.0.7 is rather old (in the scheme of things), I'd suggest you install something newer to play with. There has been a ton of stuff that has changed since 1.0.7, but I don't recall if those changes would have anything to do with your problem. (I use nothing but cvs head, but I kind of keep an eye on how many changes are happening (and for what reason), and upgrade when the number of problems seem to be at a low. The 'iax2 show netstats' would have been added in a later version. At this point, I'm thinking that it could be a matter of bad cabling or something. The Cat5 cable that's running the 8 or so feet from my PC to my router is homemade by me, and many people do report problems with homemade cables. I may not have made it exactly right, or the untwisted segment may be longer than 1/2, which supposedly causes distortion and interference. Perhaps I ought to run out and buy a couple factory-made cables to test the difference, if any, between them. Replacing the cable would probably be a good start since they are relatively cheap. Go buy a new one so there's no question about its quality. Also, keep the cable at least a little distance away from transformers, ballasts, and other things that tend to generate tons of electical noise. (Some desk lamps even have extremely noisy transformers or ballasts in them.) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Your advice was *extremely* helpful. It seems that I learn something new each time I read you all's posts. To me, it looked like each packet was correctly sent at the right interval each time, but I didn't evaluate each one. However, the general trend is that there seems to be no packet loss or resending. I could buy another cable as well, just to be safe, but it seems to me the potential IRQ conflict is more the more likely problem--of course, even when I ifconfig eth2 downed the interface, it still showed up in /proc/interrupts -- does bringing the interface down not completely bring it down? Should I permanently disable it and reboot? Again, thank you very much for your ongoing help; I feel like I'm paying (or ought to) for professional support here. :-) Regards, Robert Geller ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I modprobe -r'ed prism54, the wireless kernel modules for my card, and here's what /proc/interrupts looks like: [EMAIL PROTECTED]:~/torrents$ cat /proc/interrupts CPU0 0: 81981003IO-APIC-edge timer 1: 49755IO-APIC-edge i8042 7: 2IO-APIC-edge parport0 8: 1IO-APIC-edge rtc 9: 0 IO-APIC-level acpi 12: 499429IO-APIC-edge i8042 14: 224679IO-APIC-edge ide0 15: 900392IO-APIC-edge ide1 169: 0 IO-APIC-level uhci_hcd
[Asterisk-Users] Asterisk Community Participant; Katrina Refugee
Hi All, My family and I are doing well. Thank you all for your prayers. We are using this as an opportunity to rebuild. I didn't think I really needed to but God knows best and we will obey. My family and I will temporarily be in Lafayette, Louisiana for a while but will probably relocate to Houston, TX in the future. We already have my Daughter registered in school here. Lafayette is my old stomping ground so I'm already at home. My Wife is having a time with directions though. She went half way to Lake Charles (wrong direction) yesterday when she was coming back home from shopping. My house, office, lab and 2 vehicles back in Chalmette, LA, St Bernard Parish are swimming with the fishes, snakes and alligators along with all my computers and Asterisk application development. 100% loss, but hey, we have our health. I have both homeowners and flood insurance so I should recoup most of my losses, it will take a while to get back on track. Insurance adjusters will not be able to enter the Parish till the water is out which could take several weeks if not a few months. I was planning on speaking at this years Astricon conference in Anaheim, CA on Embedded Asterisk Systems but have to resend the invitation at this time. As you can imagine, I have other priorities. I will miss this opportunity to collaborate and share my work with this community. My FTP server is 8 feet under Lake Ponchatrain at this time and foreseeable future. My Internet provider is not online anyway but I am committed and will get my work on-line as soon as possible. I will keep up with Asterisk development as I can and will jump back into the community when available to contribute with focus and vigor. I have bought and collected equipment since being in Telecommunications, VoIP and Internet Technologies for 15 years that are irreplaceable but I will re-build my VoIP laboratory bigger and better than ever. If anyone has any trade secrets on successfully recovering waterlogged electronic equipment, please let me know. God Bless. JR Richardson ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
