Re: [Asterisk-Users] PRI numbering plan
On 12/07/04 11:11, Michael Sandee wrote:
pridialplan=unknown
prilocaldialplan=national
Not only is this that undocumented, but the string prilocaldialplan
doesn't even show up in the latest CVS HEAD source code, so that's not
going to work...
On 12/07/04 13:36, Thomas wrote:
I have an E100P connected to our partner's PBX. They want the
following:
Called number must have numbering plan/type set as:
unknown/unknown and calling number in: ISDN/national.
Our telco requires exactly this same thing - different TON for the calling
and called numbers. You want to apply a patch I wrote that allows you to
configure them separately.
It swaps the single setting pridialplan for two settings that take the
same values as pridialplan: calledpridialplan and callerpridialplan.
I attach the patch (although it is against a pretty old version of
chan_zap.c). I will also clean this up soon and add it to the bug tracker.
Best regards,
Al
--
Alastair Maw
Systems Analyst
Tel: +44 (0) 845 666 7778
http://www.mxtelecom.com
--- chan_zap.c.org 2004-02-20 16:53:31.0 +
+++ chan_zap.c 2004-03-05 12:03:53.0 +
@@ -282,7 +282,8 @@
int minidle;/* Min # of idling calls to keep active */
int nodetype;/* Node type */
int switchtype;/* Type of switch to emulate */
- int dialplan; /* Dialing plan */
+ int callerdialplan; /* Caller dialing plan */
+ int calleddialplan; /* Called dialing plan */
int dchannel; /* What channel the dchannel is on */
int channels; /* Num of chans in span (31 or 24) */
int overlapdial; /* In overlap dialing mode */
@@ -317,7 +318,8 @@
}
static int switchtype = PRI_SWITCH_NI2;
-static int dialplan = PRI_NATIONAL_ISDN + 1;
+static int callerdialplan = PRI_NATIONAL_ISDN + 1;
+static int calleddialplan = PRI_NATIONAL_ISDN + 1;
#endif
@@ -1595,9 +1597,9 @@
}
p-digital = ast_test_flag(ast,AST_FLAG_DIGITAL);
if (pri_call(p-pri-pri, p-call, p-digital ? PRI_TRANS_CAP_DIGITAL : PRI_TRANS_CAP_SPEECH,
- p-prioffset, p-pri-nodetype == PRI_NETWORK ? 0 : 1, 1, l, p-pri-dialplan - 1, n,
+ p-prioffset, p-pri-nodetype == PRI_NETWORK ? 0 : 1, 1, l, p-pri-callerdialplan - 1, n,
l ? (ast-restrictcid ? PRES_PROHIB_USER_NUMBER_PASSED_SCREEN : (p-use_callingpres ? ast-callingpres : PRES_ALLOWED_USER_NUMBER_PASSED_SCREEN)) : PRES_NUMBER_NOT_AVAILABLE,
- c + p-stripmsd, p-pri-dialplan - 1,
+ c + p-stripmsd, p-pri-calleddialplan - 1,
((p-law == ZT_LAW_ALAW) ? PRI_LAYER_1_ALAW : PRI_LAYER_1_ULAW))) {
ast_log(LOG_WARNING, Unable to setup call to %s\n, c + p-stripmsd);
return -1;
@@ -5364,8 +5366,13 @@
free(tmp);
return NULL;
}
- if ((pris[span].dialplan) (pris[span].dialplan != dialplan)) {
- ast_log(LOG_ERROR, Span %d is already a %s dialing plan\n, span + 1, pri_plan2str(pris[span].dialplan));
+ if ((pris[span].calleddialplan) (pris[span].calleddialplan != calleddialplan)) {
+ ast_log(LOG_ERROR, Span %d is already a %s called dialing plan\n, span + 1, pri_plan2str(pris[span].calleddialplan));
+ free(tmp);
+ return NULL;
+ }
+ if ((pris[span].callerdialplan) (pris[span].callerdialplan != callerdialplan)) {
+ ast_log(LOG_ERROR, Span %d is already a %s caller dialing plan\n, span + 1, pri_plan2str(pris[span].callerdialplan));
free(tmp);
return NULL;
}
@@ -5391,7 +5398,8 @@
}
pris[span].nodetype = pritype;
pris[span].switchtype = switchtype;
- pris[span].dialplan = dialplan;
+ pris[span].calleddialplan = calleddialplan;
+ pris[span].callerdialplan = callerdialplan;
pris[span].chanmask[offset] |= MASK_AVAIL;
pris[span].pvt[offset] = tmp;
pris[span].channels = numchans;
@@ -7556,19 +7564,33 @@
}
#endif
#ifdef ZAPATA_PRI
- } else if (!strcasecmp(v-name, pridialplan)) {
+ } else if (!strcasecmp(v-name, calledpridialplan)) {
+ if (!strcasecmp(v-value, national)) {
+calleddialplan = PRI_NATIONAL_ISDN + 1;
+ } else if (!strcasecmp(v-value, unknown)) {
+calleddialplan = PRI_UNKNOWN + 1;
+ } else if (!strcasecmp(v-value, private)) {
+calleddialplan = PRI_PRIVATE + 1;
+ } else if (!strcasecmp(v-value, international)) {
+calleddialplan = PRI_INTERNATIONAL_ISDN + 1;
+ } else if (!strcasecmp(v-value, local)) {
+calleddialplan = PRI_LOCAL_ISDN + 1;
+ } else {
+ast_log(LOG_WARNING, Unknown called PRI dialplan '%s' at line %d.\n, v-value, v-lineno);
+ }
+ } else if (!strcasecmp(v-name, callerpridialplan)) {
if (!strcasecmp(v-value, national)) {
-dialplan = PRI_NATIONAL_ISDN + 1;
+callerdialplan = PRI_NATIONAL_ISDN + 1;
} else if (!strcasecmp(v-value, unknown)) {
-dialplan = PRI_UNKNOWN + 1;
+callerdialplan = PRI_UNKNOWN + 1;
} else if (!strcasecmp(v-value, private)) {
-dialplan = PRI_PRIVATE + 1;
+callerdialplan = PRI_PRIVATE + 1;
} else if (!strcasecmp(v-value, international)) {
-dialplan = PRI_INTERNATIONAL_ISDN + 1
Re: [Asterisk-Users] Problems with asterisk and gnophone on Gentoo box
Mine: snd-pcm60960 0 [snd-via82xx snd-pcm-oss] Yours: snd-pcm65828 0 [snd-pcm-oss] Note that you don't actually have a sound driver loaded there! You should have snd-nvaudio listed. If OSS is working under xmms, it looks to me like your kernel has OSS support built in. You need to disable this, otherwise ALSA will get terribly confused and won't work. You can use either ALSA or OSS, not both. If you use ALSA you can then put an OSS compatibility layer on top of it. But get ALSA working first, then worry about the OSS layer. Check your dmesg output for ALSA failing to load due to this. Alternatively, get rid of ALSA entirely and just keep OSS (although this isn't recommended - ALSA is much nicer). I'm running asterisk 0.7.2 from the portage tree. Should I upgrade to v1-0_stable from CVS or is that unlikely to be the issue here? That's not the issue. Regards, Alastair ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems with asterisk and gnophone on Gentoo box
On 18/03/04 15:40, Kevin wrote: I seem to be having problems using my sound card with asterisk and gnophone in a Gentoo system (not sure if it being Gentoo is important or not, but thought I'd mention it just in case). I have the following errors when starting gnophone: Looks to me like you're probably using ALSA but you don't have its OSS compatibility layer enabled. emerge alsa-oss Check out: - http://www.gentoo.org/doc/en/alsa-guide.xml Regards, Alastair ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems with asterisk and gnophone on Gentoo box
On 18/03/04 18:46, Kevin wrote: Thanks for your reply, Alastair. I did use that guide in getting myself set-up with sound, and do have alsa-oss installed: You need to have it all insmod'ed as well (which I guess it will be): [EMAIL PROTECTED] almaw # lsmod | grep oss snd-seq-oss29216 0 snd-seq-midi-event 3584 0 [snd-seq-oss] snd-seq37584 2 [snd-seq-oss snd-seq-midi-event] snd-seq-device 4304 0 [snd-rawmidi snd-seq-oss snd-seq] snd-pcm-oss38436 0 snd-pcm60960 0 [snd-via82xx snd-pcm-oss] snd-mixer-oss 13680 0 [snd-pcm-oss] snd33636 1 [...snip...] Also make sure your dsp device is accessible for the user running OSS: [EMAIL PROTECTED] almaw # ls -l /dev/dsp lr-xr-xr-x 1 root root 9 Mar 9 10:02 /dev/dsp - sound/dsp [EMAIL PROTECTED] almaw # ls -l /dev/sound/dsp crw-rw 1 almaw audio 14,3 Jan 1 1970 /dev/sound/dsp But I suspect that your real problem is that in addition to the lines you specified in modules.d/alsa, you must have the following: alias snd-card-0 snd-via82xx -- replace with your ALSA driver alias snd-slot-0 snd-card-0-- required for OSS support under ALSA Regards, Alastair ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE410P cards E1 ports not working!
On 10/03/04 08:06, Augustine Olaifa wrote: cntext=internal [...] sgnalling=fxo_ls [...] signallng=fxs_ls [...] i do not know what i am doing wrong? I know English probably isn't your first language, but try learning how to spell. ;) Alastair ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Max # of callers in a conference...
On 08/03/04 08:12, Tracy R Reed wrote: On Sun, Mar 07, 2004 at 01:20:36PM -0600, C. Johnson spake thusly: I have anywhere from 15, to a peak max of 30 traders all using a meetme conf during the day. My Thanks for the info. I'm really look for the hard maximum. I doubt cpu issues will be a problem. I am just wondering if there is a limit in the code somewhere. You could have a cursory glance at the source code then... there is there no hard limit. Alastair ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk.org webpage
On 15/01/04 13:12, Roy Sigurd Karlsbakk wrote: hi all for new users, finding asterisk info is unneccesary troublesome. the asterisk.org page has very little information about the product and using google for 'asterisk' is like using google for 'linux'. you get all too many hits that has nothing to do with the product. perhaps the asterisk.org page at least should point to voip-info.org? or perhaps it's time someone rewrote the page? I can't say that I've found this a problem - it *does* point to voip-info.org http://asterisk.org/index.php?menu=support Perhaps support should be broken down into documentation and support to make this more obvious? Alastair ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] capacity testing
On 15/01/04 19:39, Jesse Peterson wrote: #0 ast_smoother_feed (s=0xcbf90080, f=0x5de5c4a8) at frame.c:72 #1 0x41eb00b1 in oh323_write (c=0x8214488, f=0x5de5c4a8) at chan_oh323.c:1504 Do you experience the same problems when you use the other (bundled) h323 driver? (asterisk/channels/h323/README for instructions) Alastair ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How can I get support about Dialogic hardware
On 14/01/04 02:23, wrote: I know that Dialogic hardware is not supported by standard Asterisk, if I want use it ,I must pay for it. But I don't know how to get these pathes and what kind of board is suppoted by Asterisk. http://www.mail-archive.com/[EMAIL PROTECTED]/msg01563.html Any Digium people out there - it'd be useful if this information was available on the asterisk.org web site somewhere. Regards, Alastair Maw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple phonenumbers on one E1 PRI with Digium TE410P ?
On 14/01/04 15:09, Jan Baumann wrote: one short question: Is it possible for the zaptel driver to deal with multiple phone numbers on one single E1 PRI line? I could make my carrier route +49 xxx a-zzz and +49 xxx b-zzz and others down one single PRI trunk to our asterisk box terminating in a Digium TE410P. Does the driver handle this and can I put calls coming in all on the same physical interface put into different contexts based on the dialed prefix? Yes, it's very easy, all that will work out-of-the-box. For example: [default] exten = _496667XXX,Goto(one,s,1) exten = _496668XXX,Goto(two,s,1) [one] exten = s,1,Playback(hello) [two] exten = s,1,Playback(bonjour) Regards, Alastair ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best Linux Distribution
On 13/01/04 11:48, [EMAIL PROTECTED] wrote: which is the best distribution to work with asterisk? They're all just Linux. There is no best. This question is asked so frequently it almost looks like a troll to me. :) I've therefore updated the FAQ on the wiki: - http://www.voip-info.org/tiki-index.php?page=Asterisk+FAQ Which Linux distribution should I choose for Asterisk? -- There is no best distribution. There are no fundamental differences in functionality or behaviour between Linux distributions like there are between versions of Windows. Pick whichever one you feel most comfortable with. M'kay? Alastair ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] E100P works with PCI 3.3V and 5V?
On 13/01/04 17:19, Roger Schreiter wrote: I just bought the E100P from digium. It has both keys: 3.3V and 5V, so it would fit both, in a 5V-PCI slot and in a 3.3V PCI slot. Is it true, that I can plug it without destroying it in an ordenary 5V PCI slot? Yes. If you can plug it in, it'll be fine. It's only the TE410P which is special. Regards, Alastair ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Broken DNS makes Asterisk whacky!
On 09/01/04 23:18, Matt Lawson wrote: When DNS (or outside connection to the network, not sure which) is broken and you have register= lines in iax.conf, Asterisk gets whacky. [...] Does this sound similar to anyone else's experience? Anyone else care to verfiy? Our Asterisk version is pretty close to CVS, maybe a few weeks out but I didn't see any bugs listed that seemed to address this. Yes, it does. I noticed this last week. When this happens, Asterisk forks lots of threads (hundreds over time), none of which die. This puts very high load on the box if the outage in DNS happens for a while. Alastair ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] More words for Allison
On 12/01/04 10:17, Senad Jordanovic wrote: Well... $20 from me as well has just gone to [EMAIL PROTECTED] Hmmm... I think John's turning a profit... :) Al ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX clients
On 08/12/03 13:29, Rattana BIV wrote: Is there IAX client in Applet JAVA which can be embeded in a web page On 08/12/03 14:03, Alastair Maw wrote: Nope. But I'm working on a Java IAX2 library that would let you easily build one. It'll be a little while yet, though. :) On 09/01/04 15:32, [EMAIL PROTECTED] wrote: Has there been any additional work on this yet? I haven't found anything in my searching for an iax client in java and I would really like to find one. I now have a workable Java IAX2 library that can handle uLaw/aLaw calls (not GSM yet, but I'm working on that) and play back a variety of WAVE files (aLaw, uLaw, linear in 8/16 bit at various sample rates in mono/stereo). It also does rudimentary text-to-speech (using diphone or mbrola voices). It can handle about 800 simultaneous calls on an Athlon XP 2200+ (provided it doesn't have to do sample rate conversion, stereo mixdown, etc.), running in only six threads. It can record calls out to RAW aLaw/uLaw/linear files, and I'm working on WAVE file recording this afternoon. Unfortunately, there are no current plans to release this stack under the GPL, although my company might look into that as an option in the future. Regards, Alastair ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone, ideas for incoming call management for CRM system
On 01/01/04 10:19, Olle E. Johansson wrote: What I am looking for is a solution like this: * Call comes in * XXX on Line YYY answers * A URL to a web page is transmitten on some channel, preferably the VoIP channel * The web page opens in a web window´ You're best off writing a separate application to do that - you probably don't want it tied into the VoIP communications layer because you're then tying yourself into using software VoIP clients. Most people prefer hardphones, so keep yourself flexible. This would be really quite easy (couple of hours' coding, tops) to construct using a simple Perl script on the server listening to the Manager stream (telnet) for incoming calls, and a small C++/VB/whatever systray application on each Windows PC. The Perl script would connected to the appropriate PC whenever an incoming call appeared to tell it what URL to pop up. There are security issues with people being able to pop up arbitrary web pages on each other's desktops and the like, but it basically wouldn't be very hard. You can find information on the Manager interface here: http://www.voip-info.org/tiki-index.php?page=Asterisk%20manager%20API Regards, Alastair ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] * Stresstool Help required
On 02/01/04 14:24, Girish Gopinath wrote: I gave the sip debug command, and one of the lines showed:Ignoring this request Can you log the SIP debug messages to a file and put it up on the web somewhere? Or do an ethereal capture or similar. It's very hard to say what the problem might be without a full SIP trace. It's likely that you're generating the same transaction ID for each SIP INVITE or something silly. Regards, Alastair ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FAX, IAX and *....Maybe I'm dreaming...:-)
On 12/12/03 13:56, Dan wrote: This is because the fax is transmitted using the audio stream. It is not related to the signaling protocol (SIP/IAX etc.) but to the audio codec used. Fax uses FSK modulation to transmit the data. If you compress this in a lossy way (GSM, MP3, whatever) then the integrity of the data is affected (more or less seriously depending on the codec used). Fax machines are generally quite picky, so compressing faxes is unlikely to work. I'm wondering why on earth you want to push fax data over a VoIP link at all. Fax compression isn't very efficient. It would be much less bandwidth intensive to decode the fax and send it over as proper data rather than audio, compressed using gzip/gif/png/something else. Alastair ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sendmail not on localhost
On 10/12/03 20:07, Steven Critchfield wrote: postfix and exim should provide a sendmail link or binary that should be command line compatible as the original for sending mail. I don't know about ssmtp. SSMTP does indeed provide command line sendmail compatibility (within reason). Regards, Alastair ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_iax2.c Warnings
On 10/12/03 05:05, Isamar Maia wrote: I'm setting up my Iaxtel connection now and I'm getting some annoying warnings What means: WARNING[7176]: File chan_iax2.c, Line 436 (iax_error_output): Ignoring unknown information element 'Unknown IE' (31) of length 4 ? And how can I fix it? Don't worry too much about it. IAX2 has various information elements with which to set up a call. These include caller ID, destination context/extension, etc. Element 31 is defined as DATETIME. This is sent to tell the remote Asterisk server what time you think it is. This can be useful for working out timezone differences, etc. If you want to remove the message, upgrade your Asterisk to a more recent version (the DATETIME element was added a couple of months ago, IIRC). Regards, Alastair Alastair Maw MXTelecom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sendmail not on localhost
On 10/12/03 07:41, Chris Albertson wrote: I'd prefer to run a local sendmail. Ths means you have a local queue and the mail gets handed off quikly even if your other server is down or slow. A better solution would be an SMTP fowarding agent, such as ssmtp. I'd prefer *not* to have to patch/configure/nurse multiple sendmails in my organization unless I really need to. Regards, Alastair Alastair Maw MXTelecom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX clients
On 08/12/03 13:29, Rattana BIV wrote: Is there IAX client in Applet JAVA which can be embeded in a web page ? Nope. But I'm working on a Java IAX2 library that would let you easily build one. It'll be a little while yet, though. :) Regards, Alastair ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Iax Client Library Issues? (DIAX, iaxComm, etc.)
On 03/12/03 16:43, Steven Sokol wrote: Thanks, but I already have the clients configured as IAX2 rather than IAX. The failure is not universal (not ALL calls are missed). Rather the client seems to go to sleep for some reason -- almost always after handling a call. I have been monitoring the process from both the Asterisk CLI (with IAX2 debug and IAX debug turned on), from Gastman (monitoring call activity), and from a packet sniffer (unfortunately not Ethereal with the new plugin). Trust me on this one - you *really* want to take the time to install Etheral with the plugin. It makes debugging problems like this much easier - you'll be able to see whether the client sees the packet, whether it sends a response, if there's version skew causing INVALID packets to be sent for certain challenge/responses, etc. I'd only stick trace code in the iax-client library when you've sniffed what's going on so you know where to add it. :) I can, I suppose, add some trace code to the iaxClient library, but I don't really know where to go in the code to get it to trace/log. I would like to place it as low as possible -- in the listener function, then perhaps in the parser. If anybody knows how to do this, please let me know. My C coding skills are fairly rusty. Just point out the proper file and function(s) and I will be on my way. iaxclient/lib/libiax2/src/iax.c is probably where you'd want to look. Which functions depends on what's happening. iax_do_event() might be relevant for outbound packets, for example. You'll have to delve. Alastair ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2 Ethereal plugin v0.3 is out
On 28/11/03 07:39, Olle E. Johansson wrote: The latest version of my Ethereal plugin for IAX2 is now available here: - http://almaw.com/ethereal-iax2-plugin-0.3.zip Could you please create a URL that is a bit more non-version-specific? http://almaw.com/etheral-iax2/ It now, inevitably, has a web site. :) Alastair ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX port numbers?
On 02/12/03 16:32, Matt Lawson wrote: I see that when an Asterisk connects to another one via IAX, it seems to use port 4569 for the first one. But if it has multiple IAX connections the additional ports seem to be chosen at random. Is there anyway to predict, or specify which ports or range of ports to use, for the sake of setting up a firewall? Thanks. Asterisk binds to port 4569 to listen for IAX2 packets. It should send from that same socket on port 4569, whether it's sending to a single machine or many. If it's sending from random ports, that means Asterisk can't bind to port 4569 initially. You should see this in your logs. Are you sure you're not trying to run multiple asterisks on one machine? Alastair ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX2 Ethereal plugin v0.3 is out
Hi people. The latest version of my Ethereal plugin for IAX2 is now available here: - http://almaw.com/ethereal-iax2-plugin-0.3.zip A screenshot showing what you're missing is here: - http://almaw.com/ethereal.png The new version adds the following features/bugfixes: - Decomposes the CODEC fields for supported CODECs, complete with nice English descriptions. This gives you a list of both supported and unsupported CODECs during initial negotiation (see screenshot). - Understands CONTROL packets better. - Decomposes mini-voice packets properly. - Now uses the INFO column to display packet type, etc. - Better categorization for colour filtering, etc. - Fixed the timestamps. Still to-do: - Prevent nastiness if someone sends malformed packets down the wire (better bounds error checking). - Understand TRANSFER stuff. - Understand DIALPLAN status updates. Regards, -- Alastair Maw Systems Analyst http://www.mxtelecom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Ring requested on channel 1 already in use...
I'm running an E400P. Every now and then Asterisk stops receiving incoming calls. This turns up in the messages log: Nov 25 10:49:12 WARNING[65541]: File chan_zap.c, Line 5793 (pri_dchannel): Ring requested on channel 1 already in use on span 1. Hanging up owner. Nov 25 10:49:15 WARNING[81926]: File chan_zap.c, Line 5793 (pri_dchannel): Ring requested on channel 1 already in use on span 2. Hanging up owner. Nov 25 10:49:25 WARNING[98311]: File chan_zap.c, Line 5793 (pri_dchannel): Ring requested on channel 1 already in use on span 3. Hanging up owner. Nov 25 10:49:25 WARNING[114696]: File chan_zap.c, Line 5793 (pri_dchannel): Ring requested on channel 1 already in use on span 4. Hanging up owner. A little while back I also had this in my logs: Nov 15 17:25:21 WARNING[114696]: File chan_zap.c, Line 5790 (pri_dchannel): Duplicate setup requested on channel 11 already in use on span 4 Nov 15 17:25:21 WARNING[65541]: File chan_zap.c, Line 5790 (pri_dchannel): Duplicate setup requested on channel 4 already in use on span 1 Nov 15 17:25:21 WARNING[114696]: File chan_zap.c, Line 5790 (pri_dchannel): Duplicate setup requested on channel 12 already in use on span 4 Nov 15 17:25:21 WARNING[65541]: File chan_zap.c, Line 5790 (pri_dchannel): Duplicate setup requested on channel 5 already in use on span 1 Nov 15 17:25:22 WARNING[65541]: File chan_zap.c, Line 5790 (pri_dchannel): Duplicate setup requested on channel 3 already in use on span 1 Nov 15 17:25:22 WARNING[65541]: File chan_zap.c, Line 5790 (pri_dchannel): Duplicate setup requested on channel 2 already in use on span 1 Nov 15 17:25:24 WARNING[114696]: File chan_zap.c, Line 5790 (pri_dchannel): Duplicate setup requested on channel 13 already in use on span 4 FWIW, my libpri/zaptel/asterisk installs are all about two months old. Might whatever causes this have been fixed by now? (I don't want to upgrade otherwise as this problem is quite intermittent). Anyone have any ideas? Alastair ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PCI 3.3 V
On 25/11/03 16:58, Cristian Vasiliu wrote: Why PCI 3.3V for E1/T1 card!? I can not use it because I can not find any motherboard with PCI 3.3 . Any sugestions!? Wait for the TE405P to appear, which is a 5V version of the TE410P. It should be shipping in the next week or two. Alastair ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Ethereal plugin for IAX2
On 24/11/03 17:51, Bob Knight wrote: This is way cool stuff. Thanks Is there any way to put this under the same * cvs control tree? One stop update. Well, I'm not sure there's much point. The Ethereal folk are generally very happy about having yet more packet filters hit their CVS. Once this hits v0.3 I'm going to ask everyone to test it lots and then throw it at them. I doubt it'll change much after that. Alastair ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Ethereal plugin for IAX2
The Etheral plugin is now actually workable. A new version is available at: - http://almaw.com/ethereal-iax2-plugin-0.2.zip I think some stuff might still be slightly off - unsigned/signed stuff for timestamps, etc. but it basically works. Expect a pretty much final v0.3 some time soon that decomposes voice mini-header packets properly so you can filter all the traffic for a single call by source call number. Alastair ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is Asterisk suitable for this use?
On 21/11/03 10:06, Richard Bennett wrote: Are you aware of any motherboards with 4 x 3.3 volt PCI slots, or will there be a 5 volt version of the card available soon? AFAIK, Digium is testing the TE405P (or whatever they're going to call it) right now. There seem to have been some delays - it should have been finished by now. Whenever I ask anyone, they always say 4 weeks. :) Any chance of an update from Mark or someone about this? Alastair ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX2 Ethereal Plugin initial release
Lots of people seem to want this, so I've stuck it up here: - http://almaw.com/ethereal-iax2-plugin-0.1.zip Note that it currently only does IAX-2. I might expand it to cope with IAX-1 at a later date, but no promises. It's fairly basic - unzip the file and follow the README instructions. Regards, Alastair ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk Lists
On 20/11/03 14:58, WipeOut wrote: I am not sure a newbies list would help all that much, all that would happen is that they would cross post to both lists and we would get everything twice.. To a certain extent this is true. Newbie lists also inevitably become filled with people with less experience telling each other things that are wrong or sub-optimal, which can confuse people even more. I think the best things would be a much more prominent link to the wiki, which IMHO is the best place to find answers to newbie questions. It's more up to date and contains more information than the handbook, for example. -- Alastair ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] tunnel iax via gnophone with ssh?
On 20/11/03 15:44, Chris Hirsch wrote: Hey all...I'm trying to use gnophone to connect to my asterisk box behind my firewall..I thought I could just setup a tunnel with something like ssh host.com -L5036:asteriskserver:5036 and just change my gnophone to connect to localhost:5036 but I never see anything happen on the asterisk server. I'm even trying this on the same network just in case there is something funky with NAT. Anybody have any ideas? Yes, IIRC SSH only tunnels TCP. IAX is UDP based. You'll need to find something that will tunnel UDP over TCP, so you can tunnel that over SSH (!). Good luck. :) Alastair ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Ethereal plugin for IAX2
As mentioned on the devel list earlier today, I'm interested in writing an IAX2 plugin for Ethereal to make debugging IAX protocol implementation and simultaneous calls on normal networks easier. Anyway, I started work on it this evening, so it's not complete yet, but it's starting to look quite sensible: - http://raq626.uk2net.com/~al/ethereal.png A couple of people have e-mailed me to say that they're in a hurry for such things. If you'd like me to e-mail you a copy, give me a shout. Otherwise, I'll polish it up and give it to the Ethereal guys for inclusion in Ethereal 0.9.17 or something. :) Regards, -- Alastair Maw Systems Analyst http://www.mxtelecom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] http://www.skype.com/
On 05/11/03 10:14, Olle E. Johansson wrote: As I understand it they must not be fully peer-to-peer even if they use your bandwidth, there has to be media servers in their network, handling calls. Or? No, the whole point is that it's completely decentralized. More interesting to end users is that the calls are encrypted and can traverse NAT. The way Skype can bounce between peers effectively enables it to provide a few different routes for the traffic, from which it picks the least latency one. Add a nice UI, and it's not surprising that it's gathering speed rapidly. -- Alastair Maw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: IAX2 Java library
On 04/11/03 04:59, Steven Critchfield wrote: But C isn't as maintainable as nice Java apps, and it's as simple as that. Basically, I'm after the most powerful interface possible to Asterisk, but trying to make it as friendly as possible to code things against. As far as our organization is concerned, that pretty much means Java objects. So you bought that line of Marketecture didn't you. I think there are several large open source projects that prove that C is maintainable. Maintainability is really a function of organization. If you can't be organized, you will not produce very maintainable C code. Stop it right there. This mailing list isn't the place for language holy wars. Java is the tool of choice for our organization, and therefore Java code is more maintainable within it. Everyone here talks Java. Not everyone talks C. If we hire less good programmers, Java hold their hand and enforces structure in a way that C doesn't. I'm not arguing that decent C code isn't maintainable. I'm stating that in our particular case, Java fits the bill better. It's that simple. As for this: It all comes down to the number of CPU cycles needed to perform a given function. When doing real time processing, a few cycles here and a few there can add up to make a real difference. Object Oriented is nice for ease of writing/maintaining code but all of those objects have blocks of code behind them. A slight inefficiency there can really impact performance. Sure we have faster processors and lower cost memory every 6 months but thats no excuse for not writing the most efficient code possible. You're entirely wrong. Provided it's fast enough, it's all about maintainability, extensibility and scalability. The bottom line is that we're in it to make money. If it runs on hardware that's $100 cheaper, but takes four times as long to develop, then total cost is much higher. If it's easy to repurpose it at a later date, or easier to have someone unfamiliar with the codebase to come along and customize it, that's worth a lot. Both are significant, but maintainability is much more important than performance, as that's where the real cost lies. -- Alastair ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: IAX2 Java library
On 03/11/03 23:31, Jeremy McNamara wrote: - The documentation for AGI is very poor. I know it is for IAX, too, but I can see a Java IAX library being useful for client development too, and I'd like to give a little back to the * community, you know? You have the source code. What more do you need? Seeing as you're asking, an RFC would be nice. But then, this has already been discussed. Maybe you had your blinkers on so tight you didn't notice. :) For those who are interested in such things, if I were to write an RFC proposal, would anyone actually take the time to check it and do some proof-reading (who is qualified to do this for IAX2? Is it only really Mark? Or are there others out there who are experts?). If people are interested in this, I'll try to document things as I go along... Regards, -- Alastair ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] high system load running asterisk
On 04/11/03 11:33, Christoph Loibl wrote: however while running asterisk without any clients online my system-load is about 1 - and asterisks seems to run all the time. is this the default behavior? No. Load average on an idle Asterisk box should be very close to zero. I don't think there's an easy way to find which module is using all the CPU time. You can add a noload command to the modules.conf file for each module in turn until you've pinned it down. Regards, -- Alastair Maw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] High Availability and Mass Deployment for Asterisk
On 31/10/03 12:11, Senad Jordanovic wrote: You are right, but what if each * server had a single source for all of its configuration files from a file server over NFS or similar. Single point of failure at the file server. Better to rsynch all the machines config files or similar. -- Alastair Maw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
IAX2 Java library (was Re: [Asterisk-Users] New IAX software phone (for WIndows platform))
On 03/11/03 00:25, Mark Spencer wrote: As a side note, I strongly would like to see someone implement a client using libiax2 which implements IAX2 instead of the (now obsolescent) IAX version 1. I'm implementing a Java-based IVR server (and yes, I know Asterisk does IVR, and no, it's not flexible enough to do what I want and no, it doesn't integrate well with the Java systems we have, etc. hence my doing this). Currently it uses SIP (using the NIST JAIN-SIP stack) and JMF to handle RTP/audio stuff. I've found that JMF/RTP doesn't scale very well, as it spawns a *lot* of threads, and can't reliably handle more than 20 simultaneous calls. So, I'm investigating the possibility of writing an IAX library for Java. Searching the archives, it seems various other people would be interested in this. So, my questions are: - Should I implement IAX or IAX2? What's the main difference, other than IAX2 supporting trunking (which according to the docs needs a Zaptel timing source). - Has anyone else made any headway with this? - Is anyone else interested in making this an LGPL or even a GPL project and helping me with it? I'm likely to implement just the call management/DTMF/audio type stuff required for IVR initially (i.e. not worry about call xfer, etc.). It'll also be geared towards handling the hundreds of simultaneous calls required in a server environment, although there'll be no reason not to use it for IAX clients too. Obviously such a library would enable a nice GUI cross-platform IAX(2?) client to be easily created, which would be a nice by-product. -- Alastair Maw MX Telecom http://www.mxtelecom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: IAX2 Java library (was Re: [Asterisk-Users] New IAX software phone (for WIndows platform))
On 03/11/03 16:35, Jeremy McNamara wrote: I'm implementing a Java-based IVR server (and yes, I know Asterisk does IVR, and no, it's not flexible enough to do what I want and no, it doesn't integrate well with the Java systems we have, etc. hence my doing this). Are you mad? What is not flexable enough for you? Java knows what STDIN and STDOUT is, right? What more do you need? Not wanting to start a flamewar, but... - I can't possibly fork a whole JVM process for each caller. It's much too inefficient. This needs to support hundreds of simultaneous calls, and the GNU Java compiler just isn't good enough for our needs. I guess I could write an AGI wrapper script which connected to the Java server over a TCP connection or something and piped the stdin/out down the line to it. - We'd like to use Java because: - Need to do RMI to existing systems. Can't be bothered with all the CORBA nonsense. - It's more maintainable within our organization. - We have lots of existing components to support. - It does all the interoperability stuff we need very nicely, so we save time once the system is built (XML, etc.). - We like it. :) - I need access to the raw audio streams in realtime for various reasons (need to do DSP stuff for some clients, etc). Can I get this easily with AGI? Along with this, I need to be able to play audio from a URL. I don't want to have to download the whole file from the URL in order to play it - it wants to be streamed. Is this possible with AGI? The docs aren't very good for AGI, so I don't really know... - I need to be able to generate large amounts of audio in realtime, conference people together but then only play an audio file to one person within the conference, etc. I don't think AGI is flexible enough to do this. - I'd like to be able to move from Asterisk to something else if I need to. This is why originally I was doing things using SIP/RTP. - The documentation for AGI is very poor. I know it is for IAX, too, but I can see a Java IAX library being useful for client development too, and I'd like to give a little back to the * community, you know? There are other reasons, but I haven't the time to explain right now. The above are the most important. -- Alastair Maw MX Telecom http://www.mxtelecom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: IAX2 Java library (was Re: [Asterisk-Users] New IAX software phone (for WIndows platform))
On 03/11/03 18:02, Alastair Maw wrote: I'm implementing a Java-based IVR server (and yes, I know Asterisk does IVR, and no, it's not flexible enough to do what I want and no, it doesn't integrate well with the Java systems we have, etc. hence my doing this). Are you mad? What is not flexable enough for you? Java knows what STDIN and STDOUT is, right? What more do you need? snip There are other reasons, but I haven't the time to explain right now. The above are the most important. Additionally, I'd like to spread the load across two machines - one for the PSTN/SIP/IAX routing and one for the IVR software. -- Alastair Maw MX Telecom http://www.mxtelecom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: IAX2 Java library (was Re: [Asterisk-Users] New IAX software phone (for WIndows platform))
On 03/11/03 20:03, Steven Critchfield wrote: Sounds like you really need a C programmer and get into the guts of asterisk. Can't get more flexible than having the source code yourself to do anything you want. You could add your DSP routines into the dsp.c file and call them when needed. You can also write a asterisk application and have direct access to all the audio in every direction just as you want it. But C isn't as maintainable as nice Java apps, and it's as simple as that. Basically, I'm after the most powerful interface possible to Asterisk, but trying to make it as friendly as possible to code things against. As far as our organization is concerned, that pretty much means Java objects. - I'd like to be able to move from Asterisk to something else if I need to. This is why originally I was doing things using SIP/RTP. What else is there worth using??? Are you one of those people who always develops apps thinking, what if someone buys this and wants to run it on Oracle? or atleast something to that idea. There isn't anything else worth using. :) But there might be in six months' time. Or we might want to plug the IVR into something with far more channels than an E1 that has a Cisco badge on the front and talks SS7 (although I guess we can always break that into a T1 channel bank or whatever and plug T1s into the TE405Ps we have on order). - The documentation for AGI is very poor. Actually they documentation is just programmer oriented. The documentation is included as example scripts and the section of apps/app_agi.c that contains a nice description of each function that is available. snip I shall look into that. Thanks. :) -- Alastair Maw MX Telecom http://www.mxtelecom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Info on UK ISDN30e?
On 30/10/03 14:38, Gavin Hamill wrote: Has anyone used ISDN30e in the UK with the Digium E1 cards? Many people. What options are there to stick on a couple of ISDN2's on top of that should we require some 'backup lines'. It's more a question of how to implement the backup lines - they're fine for outbound calls, but it's backup for inbound lines that you really want. This is difficult to achieve, but you might be able to get BT to give you a hunt group that hits a pair of ISDN2s after looking through the E1 bank and failing to connect. It's only worth doing if you're going to route them directly to some other kit, though, so Asterisk support for ISDN2 hardware is largely irrelevant here. Do BT terminate the ISDN30e in a format that I can literally just plug into the Digium cards, or will I need some kind of adapter (whether electronics or even just a simple socket/plug changer)? You will be able to just plug it straight in (standard RJ45 termination). I'm trying to gather some tangibility for the project - I see the first mailing list post in November 1999... when did the project start, and when was it first usable as a simple PBX? Can't answer this one, others? Many people/organizations have successfully deployed it, though. Be aware that it's currently not as easy to configure as many commercial PBXs, but it tends to be cheaper and more flexible. FAX support is also coming soon. :) Finally, are my options for handsets limited to IP phones via Ethernet, or analogue phones via a channel bank (and then to another Digium E1/T1 card), or is there the possibilty to re-use proprietary handsets from a previous PBX? I doubt you can reuse proprietary handsets. Please provide more details (model/make). -- Alastair Maw MX Telecom www.mxtelecom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Answering Machine Detection
On 27/10/03 21:57, DUSTIN WILDES wrote: Does anyone have any recommendations on implementing Answering Machine detection for call generation programs? There's obviously no nice way of doing this. If you're doing telemarketing, and you're playing pre-recorded audio, which of course is a nasty thing to do, the algorithm is something like: 1. Dial out. 2. Wait for answer. 3. Start playing audio. 4. If you hear something that sounds like a beep, either hang up and try again later, or stop the audio, pause for two seconds and start playing it again. 5. Hang up when finished playing audio. Step 4 is accomplished by doing a FFT on the incoming audio into frequency buckets and taking a rolling average of the mean and standard deviation, such that you can detect when a fixed monotone beep occurs at the other end. If you don't want to play audio files and wait for beeps, and want to connect real humans to each other, then there's no decent way to do this, as the only difference between humans and arbitrary answering machines is that the answering machines give you a beep prompt to record your message. Regards, -- Alastair Maw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PRI Asterisk Redundancy/Fail-Over
On 28/10/03 08:17, Florian Overkamp wrote: look for a T1 failover switch. Nice tool. Anyone know of an E1 equivalent ? :-)) Most people who'd want this sort of thing probably have multiple incoming E1 lines. If you have multiple lines, you can set up a hunt group to range over the lines, with each line plugged into a different Asterisk box, which gets you your failover for free. I'd therefore expect any E1 failover boxes to have a low sales-volume and thus be somewhat expensive. -- Alastair Maw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk with Gentoo
On 22/10/03 08:37, WipeOut wrote: You should build you Asterisk box as a dedicated system not as a desktop with and Asterisk server built in.. :) This is certainly true for production use. However, I'm also happily running Asterisk for development and testing on my desktop machine under Gentoo, running Eclipse, Mozilla, X, OpenOffice, etc. There's no reason you shouldn't do this. If you want it to be able to handle higher loads and be less prone to breaking up audio when the box is under load, just nice the Asterisk process to some low value when you start it so it gets prioritized over X, etc. I'm also running some custom Java-based SIP IVR software on the same production server as the Asterisk+E400P that routes PSTN calls to it, which is working well. -- Alastair Maw Systems Analyst - MX Telecom http://www.mxtelecom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems with TE410P and E1 line -- Unable to open D-channel 24 (No such device or address)
You have the card jumpered as a T1 card, not an E1 card. Look in the middle of the card for the jumpers. -- Alastair Maw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NAT, SIP (was: No sound with SIP Phones on the Internet)
On 15/10/03 00:15, Uriel Carrasquilla wrote: Does anybody else have a strong opinion one way or the other? If it is left to John and myself we have a 1:1 vote. See how much easier it is to follow the thread of conversation if you quote just enough of the e-mail you're responding to so people know what's going on without having to read through pages of text? Please see RFC 1855: - http://www.faqs.org/rfcs/rfc1855.html Decent mail clients that behave sensibly regarding quoting are easy to come by. You can even set up Outlook to behave vaguely properly and quote using . As a matter of fact, I am of the opinion that the response to E-mails should go at the top to save time. So, it's not worth *your* time organizing your e-mail sensibly, but it's worth everyone else's time having to dig through lines of text to work out what the context is? I find that selfish, at best. Please see the following page (strong words warning). It pretty much sums it all up nicely: - http://thegestalt.org/simon/quoterant.html -- Al Maw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] outbound caller ID problem on PRI
I can't seem to hide and/or set my caller ID from *.
I'm using a quite recent (three weeks or so) CVS with an E400P card.
I have pridialplan=unknown in zapata.conf and I'm based in the UK.
The relevant bit of pri debug looks like this (reformatted to fit 80
char width):
Calling Number (len= 4) [ Ext: 0
TON: Unknown Number Type (0)
NPI: Unknown Number Plan (0)
Presentation: Unknown (67) '' ]
I'm dialing in from SIP outbound to Zap with a context like this:
exten = _X.,1,SetCallerID(mxtelecom 0845123456)
exten = _X.,2,Dial(Zap/1/${EXTEN})
Although:
exten = _X.,1,SetCallerID(0845123456|a)
exten = _X.,2,Dial(Zap/1/${EXTEN})
Equally doesn't work.
I've tried setting these in zapata.conf:
callerid=foo 0845123456
hidecallerid=yes
No matter what I do, I get a default caller ID provided by my telco. If
I prefix the number dialed with 141 (standard UK hide caller ID) the
caller ID isn't presented to the end user, but this is an ugly kludge
that I don't want to have to do. Ideally, I'd like to be able to set my
callerID to an arbitrary number.
If I set pridialplan=national/international I can't work out what format
the outbound calls numbers should take and get denied messages back.
Anyone have any ideas?
--
Alastair Maw
MX Telecom - Systems Analyst
www.mxtelecom.com
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Re: [Asterisk-Users] bare-bone config
On 13/10/03 14:05, Conrad Braun wrote: Why do you want to remove some of the conf files? Just leave them all there.. its not like they use up a lot of space or anything.. :) I am just starting to use asterisk as well as VoIP in general, and it's a bit confusing finding out what goes where... in my eyes it seems to be a lot easier to start with a bare minimum, thereby eliminating as many causes for error as possible. when I feel comfortable, I can always expand on top of it. If you just bit the bullet and removed them all, you'll discover all sorts of interesting dependencies on musiconhold, etc. On my production boxes I have autoload=no in modules.conf and then load everything in manually, as reducing the number of modules that are loaded that you don't actually use is obviously a good idea for memory footprint, stability, etc. -- Alastair Maw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] multiple SIP users on one phone?
Interesting problem: An organization has departments. Each department has a single phone. Each department has multiple people. Each person within the organization has a direct dial incoming number. It's easy to set * up so that multiple DDIs get mapped to the same extension. What I'm wondering is if there's any way, with reasonably priced hardware, to notify the person who's about to pick up the phone who the call is for. Maybe change the caller ID field or something? In an ideal world, the hardware would have two lines, one with caller ID, one with the name of the person they're trying to call. Additionally, a person should be able to field a call, decide they can't deal with it, and push a button that redirects the caller to the appropriate person's voicemail (based on the incoming DNID). Is any of this possible? Phone cost per handset should be as low as possible, as per usual. :) I have no experience of SIP hardphones, so don't know how much/what information about the call they're capable of displaying. Regards, -- Alastair Maw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone else use Audacity for prompts?
On 06/10/03 08:25, Shaun Ewing wrote: The .wav files I'm producing are of stellar quality. However, once I turn them into .gsm, they sound buzzy and muffled. An example line to convert: sox file.wav -r 8000 -c 1 file.gsm It'll sound much better if you go: sox file.wav -r 8000 -c 1 file.gsm resample Of course, there's only so much you can do to make 8kHz prompts sound any good. Doing the original recording at 8kHz is a good start. -- Alastair Maw System Analyst @ MX Telcom www.mxtelecom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VIA vs Intel
Sean P. Robertson wrote: I have. Heads up on the built-in sound. Like everything else on the motherboard, it uses a VIA chipset and chan_oss will not work with it. To the best of my knowledge, there's nothing special about how chan_oss uses the OSS drivers that would make it fail on this chipset in particular. I can only assume therefore that OSS isn't configured properly on your machine. Is this the case? Do other OSS applications work properly? I used to have VIA82C686 support built into the kernel, and chan_oss worked fine with it. I now use the snd-via82xx ALSA driver with snd-pcm-oss module for OSS emulation/compatibility layer and that also works fine. -- Alastair Maw MX Telecom - Systems Analyst http://www.mxtelecom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cannot compile channel h323 from yesterday's CVS?
Garry, Please don't cross-post. On the basis that this has nothing to do with development, and everything to do with user deployment, it should be in the users list. When I try to compile channel h323, I get multiple compile errors. Can someone help? asterisk, ptlib, openh323 all are fresh from CVS. Please read the README - it has changed. If you'd read it you wouldn't be trying to do a make install in h323. Note that various people have also been unsuccessful trying to get chan_h323 working with the latest openh323/pwlib builds. Again, read the README. Can someone (JerJer?) please update the install target for the Makefile in channels/h323 to spit out an error telling you to read the README/INSTALL instructions? This is getting tired. -- Alastair Maw MX Telecom - Systems Analyst http://www.mxtelecom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] configuring TE410P for four E1 PRI lines
Rahul Arvind Jadhav wrote: This is obviously because Quad T1 has only 96 channels. The question is how do i get the zaptel module to recognize the card as Quad E1 and not Quad T1. You have to change the jumpers on the card. There are four in the middle of the card. Your zaptel.conf looks good, by the way. -- Alastair Maw MX Telecom - Systems Analyst http://www.mxtelecom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CPU Optimisations For asterisk
Robert Boardman wrote: How would I compile asterisk for the Athlon XP arch, would there be any advantage doing this? CHOST=i686-pc-linux-gnu CFLAGS=-mcpu=athlon-xp -O3 -pipe Well, it might run slightly faster, but you probably won't really notice the difference. You might well be better off with -O2 rather than -O3, as -O3 tends to agressively unroll branches to inlines which reduces the amount of code that fits on the chip's cache, resulting in slowness. It's swings and roundabouts, really. If you're using echo cancelling, it should be quicker if you enable the MMX stuff for that (see the Asterisk Makefile). Why do you need the extra speed? If you're desperately trying to optimise things like this to gain extra performance, you must have a pretty big system. Pretty big system should mean you have the cash to upgrade your CPU a bit, which will make much more difference. -- Alastair Maw MX Telecom - Systems Analyst http://www.mxtelecom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Prebuilt Asterisk
Mike Hjorleifsson wrote: Does anyone sell a preinstalled asterisk server ? I believe TelAppliant.co.uk will sell you a system called mypbx, which is basically just that. -- Alastair Maw MX Telecom - Systems Analyst http://www.mxtelecom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Recommended OS
Michael A. Miller wrote: Other than the generalized tools, most drivers and such are specific to the company that released that version as well as the hardware. This is quite incorrect. But here is not really the place to discuss such things. Suffice to say that the vast majority of hardware support is provided by the Linux kernel. You'll only use distribution specific drivers if RedHat or SuSE or whoever deem them necessary (support for 802.11g springs to mind). What makes the distributions different is merely how they package their files (if at all) and how they update things, find patches automatically, what extra tools they have, what logo they've put on the start screen, etc. It's all academic. Provided you have recent enough versions of the libraries and compiler tools Asterisk needs installed, it will all work, no matter what distribution you're using. That's all that matters. I have gotten Asterisk to work. It was a knowledge issue on my end. My main issue is getting a Netjet ISDN and voice modem installed correctly. Both of which I am having issues with the drivers at the OS level and have not attempted any use with Asterisk at this point. If someone has experience with a Netjet install under RH9 or Lucent/Agere linmodem, I would sincerely appreciate a bit of help. This page may be of use: - http://www.traverse.com.au/downloads/drivers/ Welcome to the Asterisk community. :) -- Alastair Maw MX Telecom - Systems Analyst http://www.mxtelecom.com -- Alastair Maw [EMAIL PROTECTED] MX Telecom - Systems Analyst http://www.mxtelecom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
List ettiquette (was Re: [Asterisk-Users] Grandstream Source?)
PJ Welsh wrote: This */IVR/VOIP/Telephony stuff is only easy when you get to *REALY* know it. I am not there! I know my GNU/Linux systems... I don't know this... please be nice to me atleast ;) I am nice. :) The point of that tongue-in-cheek e-mail was that hopefully Senad will type the single obvious word into Google next time before he wastes hundreds of people's time (albeit only 5 seconds each) with questions he can answer for himself very very easily. VoIP is complex. PSTN systems are complex. But using Google isn't. If someone points out that Company Xyzzy sells a product/service, I can't imagine why anybody would even bother asking a mailing list about it, rather than just going straight to Google and searching for Xyzzy. If you have a genuine problem, the list is friendly and nice. If someone can't be bothered to type a single and specific word into Google, and it's very obvious they haven't made an attempt to think/look for themselves, then it's hardly surprising that most people have little patience for them. So, as a reference for all you people who get burnt when posting to the list, here is a guide: - Ask a new question by clicking the new/compose button in your mail client. Only hit reply if you are actually replying. In particular, don't hit reply, delete the whole of the subject line, and attempt to start a new thread this way. Stephen will flame you, and the rest of us with threaded mail readers will silently sit and seethe quietly in a corner (or miss it altogether, having marked that thread as uninteresting/irrelevant/don't know anything about it). - Don't post in HTML/RTF. Basically, it holds no advantage over plain text, and has many disadvantages (size, accessibility, etc, etc.) - Use Google if you think the question might be obvious. In particular, search like so to look in the list archives (e.g.): site:lists.digium.com SIP H323 gateway - If you can't find it after five minutes of looking, but still worry that it's quite an easy obvious question, everyone will like you lots if you say things like It's probably quite easy, but I can't find anything on Google about it unless I'm being blind... And that's about it, really. Simple, see? -- Alastair Maw MX Telecom - Systems Analyst http://www.mxtelecom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream Source?
Senad Jordanovic wrote: have you more info on this free phone offer? please send it to me off the lest? Just as a totally wild guess, and call me crazy and amazingly intelligent for thinking of it, but how about looking at www.nikotel.com? I remain astonished by how many people need constant spoon feeding... -- Alastair Maw [EMAIL PROTECTED] MX Telecom - Systems Analyst http://www.mxtelecom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] using pci modem cards as fxs/fxo ports in *
Bryan Nolen wrote: forgive the question but is it possible to use PCI modem cards (aka winmodem's) as FXO/FXS ports in * ? what about external modems like the USR Sportsters? Methinks this needs to go in an FAQ, and the FAQ needs to be linked to from the mailing list signup page/confirmation e-mails. Long answer: Search the list archives/Google. Short answer: Nope. Lack of voice duplexing tends to scupper you. -- Alastair Maw [EMAIL PROTECTED] MX Telecom - Systems Analyst http://www.mxtelecom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dialogic Hardware (Take 2)
[EMAIL PROTECTED] wrote: All I want to know is how, where. And is there any other third party channel for Dialogic is available. Now I dont see anything wrong with my question!!. Congratulations on learning how to start a new thread properly. :) As Stephen helpfully stated (and you seem to have missed because you didn't read the rest of his post past the disappointed comment), Digium sell Dialogic support for $15 per channel. Please contact [EMAIL PROTECTED] for information. You'll notice that this pricing is deliberately set such that it's about the same price to buy a Digium board as it is to license a Dialogic one. -- Alastair Maw [EMAIL PROTECTED] MX Telecom - Systems Analyst http://www.mxtelecom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Legal Interception - tapping
Tais M. Hansen wrote: I know it is possible to record calls, it will record them to a directory you define on the server. But are you required to provide archives/recordings of the calls or permit real-time tapping? You should ask some kind of justice department. We did here in Denmark and were told that there currently wasn't officially defined a set of rules about phonetapping but that we should be able to do so if legally requested by law enforcement. These rules will be clearly defined eventually when the government figures out their version of the anti-terror-package. We're required by ICSTIS to record any live entertainment services that run over our gear. Unfortunately they've not been very helpful regarding requirements - there's merely a single-page document that mentions things like timestamping recordings every 1/2 second to ensure they haven't been tampered with. That doesn't make much sense to me, even in an analogue world, let alone a digital one. FWIW, they require real-time monitoring capability (max 3 second delay) and archives for I think six months (the document doesn't actually specify, would you believe). If anyone has any more information about this, or could point me towards any, I'd be most grateful - ICSTIS' consultant doesn't seem to reply very promptly (if at all) to his e-mails. Regarding the real-time monitoring, it'd be great if we could develop an extension to zapbarge/scan that let you tap in the callerid of the person you're wanting to monitor. It's a little hard to find things in a 120 channel bank sometimes... -- Alastair Maw [EMAIL PROTECTED] MX Telecom - Systems Analyst http://www.mxtelecom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] how to make sip uri work
Lee Goodman wrote: Lets say I have an * at my business, with 7960 SIP phones. All the sip phones are registered using their extension number (like 305), but I would also like to put my SIP URI on my business card and in a name format, not an extension number (like lee.goodman), so that the SIP URI would read [EMAIL PROTECTED] How would I set this up in extensions.conf? I got [EMAIL PROTECTED] to work If I have this extension exten = 305,1,Dial(SIP/305,16) ;Lee Goodman exten = 305,2,Voicemail2,su305 exten = 305,3,Hangup would I have the following as well exten = lee.goodman,1,Dial(SIP/305,16) ;Lee Goodman exten = lee.goodman,2,Voicemail2,su305 exten = lee.goodman,3,Hangup when I try this, it looks like it trys to go to voicemail (my 305 phone is not registered) but it hangs up first No, no, no. :) You have to define the users in sip.conf. There are examples and comments in there by default. There's also some information about this in the PDF handbook. -- Alastair Maw [EMAIL PROTECTED] MX Telecom - Systems Analyst http://www.mxtelecom.com -- Alastair Maw [EMAIL PROTECTED] MX Telecom - Systems Analyst http://www.mxtelecom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP to SIP monitor and record?
Timothy Soos wrote: Good. Since I do not have the VoIP system set up yet, and I control the IP network (which is still small), please tell me what I need to do to monitor and record SIP to SIP calls. In SIP.conf: canreinvite=no Then use the Monitor application. -- Alastair Maw [EMAIL PROTECTED] MX Telecom - Systems Analyst http://www.mxtelecom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] E400P woes
We've changed E1 providers and I'm trying to reconfigure an E400P to make it work with the new lines. They're supposedly standard EuroISDN lines (in the UK). I'm initially just trying to get a single line up. I have the following in /etc/zaptel.conf: span=1,0,0,ccs,hdb3 bchan=1-15 dchan=16 bchan=17-31 loadzone=uk defaultzone=uk The LED on the back of the card shows red/green simultaneously when you ztcfg it. Asterisk itself spews messages to the console like so: == D-Channel on span 1 up == D-Channel on span 1 up == D-Channel on span 1 up == D-Channel on span 1 up == D-Channel on span 1 down == D-Channel on span 1 up == D-Channel on span 1 up == D-Channel on span 1 up == D-Channel on span 1 up == D-Channel on span 1 down (Always four ups and then a down a bit later, followed by a pause, then the same again.) Changing span=1,0,0,css,hdb3 to use crc4 causes Asterisk to throw lots of yellow alarms when it starts. Putting a 1 in the timing field makes no discernable difference. The guy who tested the lines isn't very clueful - he just plugs in his gear and puts a tick in the box, but he's pretty sure it uses hdb3 coding. CAS framing makes the LED go RED. Anyone give me a clue here? -- Alastair Maw [EMAIL PROTECTED] MX Telecom - Systems Analyst http://www.mxtelecom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] E400P woes
OK, so I've done this: *CLI pri intense debug span 1 Enabled EXTENSIVE debugging on span 1 Sending Set Asynchronous Balanced Mode Extended [00 01 7f ] Unnumbered frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 M3: 3 P/F: 1 M2: 3 11: 3 [ SABME ] 0 bytes of data Sending Set Asynchronous Balanced Mode Extended The above is repeated about once a second. Our provider does indeed have a System-X switch. Is this a known problem then? Is there no way to resolve it? I couldn't find anything on Google about it... I'd put a TE410P card in instead, but the 1u servers we have are all P4s and don't have 3.3V PCI slots. The official Word from Digium is that they'll have a 5V version of the TE410P out in about six weeks' time, but we have some services that need to go live on these new E1 lines in about three weeks time, plus we need to do some testing, etc. Time to buy a Xeon with 3.3V slots, I guess. :( -- Alastair Maw [EMAIL PROTECTED] MX Telecom - Systems Analyst http://www.mxtelecom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Some Question of extension.conf
Alvaro Parres wrote: ¿How do you make that some user who is in a menu, can dial any extension that is define in other context ? Example.. [office] 100,1.. 200,1.. 300,1.. [menu] s,1 - When the user is here.. can dial 200 and it takes 1,1the 200 extension of office context. 2,1 3,1 Please read the PDF document linked to on the asterisk web site on the documentation page. Look for the use of the include command to include contexts. -- Alastair Maw [EMAIL PROTECTED] MX Telecom - Systems Analyst http://www.mxtelecom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP client-NAT-Asterisk-NAT-SIP client. only works with canreinvite=no.
WipeOut . wrote: Any ideas on the client A to C (same LAN, same NAT box, unique outside IP, same * server)? Only thing that springs to mind is to install another * box internally and then use IAX to connect the internal * box to the external one.. then the internal phone will call each other without crossing the NAT.. It shouldn't be *too* hard to change Asterisk such that it allows reinvites for particular netmasks. If you can ensure that your NAT clients are on different subnets, for example, this might be possible. -- Alastair Maw [EMAIL PROTECTED] MX Telecom - Systems Analyst http://www.mxtelecom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sound error during launch
Klaus-Peter Junghanns wrote: i had the same problem (and also asterisk wont share your sound device with anything else). try my chan_oss patch: www.junghanns.net/asterisk/downloads Please don't use this patch - commenting out log messages is a deeply silly thing to do. If you want to use ALSA rather than OSS, put the following in your modules.conf file: noload = chan_oss.so load = chan_alsa.so Note that if you don't have a sound card at all, you should noload both of these. -- Alastair Maw [EMAIL PROTECTED] MX Telecom - Systems Analyst http://www.mxtelecom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE410 - 3.3v?
Mark Spencer wrote: If it fits, it'll work. Generally motherboards with 64-bit slots and most DELL's including the 600SC work fine. Look closely at the two edge connectors in these pics: - http://digium.com/images/wildcard_te410p.jpg - http://digium.com/images/wildcard_t400p.gif You'll notice that the T400P has two gaps in the edge connector, whereas the TE410P only has one (nearest the ports). I.e. 3.3V PCI slots have the longer part of them closest to the rear of the machine, and 5V slots have the shorter part closest to the rear. The following motherboard has both types of slots, with the 5V ones to the right, and the longer 3.3V 64-bit PCI-X slots to the left. - http://www.supermicro.com/PRODUCT/MotherBoards/E7505/X5DA8.htm Note that it seems to be possible to get non-64-bit 3.3V slots, like on this board, which looks like it has one: - http://www.supermicro.com/PRODUCT/MotherBoards/VIA/P3TDDE.htm Annoyingly, no-one seems to make 3.3V PCI motherboards for P4s, it's pretty much Xeon-only. When I spoke to TelAppliant when ordering one of their last E400Ps, they seemed to think that Digium has plans to produce a 5V version of the TE410P, but that it won't be ready for a few months. Can anyone confirm or deny this? -- Alastair Maw [EMAIL PROTECTED] MX Telecom - Systems Analyst http://www.mxtelecom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help configuring E400P cards
Carlos Fernández Puente wrote:
We have a problem with the configuration of the card, the cards work
and we receive incoming calls but asterisk don't receive dnid. We
have 5 servers with 1 E400P with the same problem and the telco told
us that we need to configure the card to request it, how we can do
this?
Hi Carlos. When you say don't receive dnid, what exactly do you mean?
If you start Asterisk with -vgc, and make a call into the box,
does it give you any information?
For example, on my E400P, I get:
Connected to Asterisk CVS-08 currently running on vampire (pid = 23767)
-- Accepting call from 'x' to 'y' on channel 31, span 1
(Where x and y have been changed from the actual numbers,
obviously, with y being the DNID.)
Or are you referring to the ${DNID} variable not being set?
I've patched my installation to set ${DNID}. I'm currently using it to
route calls via an external AGI database lookup. I will submit a patch
to the dev-list later today for consideration for CVS if you like.
--
Alastair Maw [EMAIL PROTECTED]
MX Telecom - Systems Analyst
http://www.mxtelecom.com
___
Asterisk-Users mailing list
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Re: [Asterisk-Users] Call script after hangup
Frank N. wrote: I believe the porblem is that, since the incoming call is not closed before the outgoing call is created, the outgoing call does not work. I was hoping the delay would solve this problem... but obviously it doesn't. No - it still won't relinquish the call until the hangup handler has completed. What you need to do is to have the AGI script return, such that the call exits. Then five seconds later, copy the file. You could do this by setting up a BASH script which executed the Perl in the background. I.e. #!/bin/sh /path/to/script/foo.pl Make sense? -- Alastair Maw [EMAIL PROTECTED] MX Telecom - Systems Analyst http://www.mxtelecom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip Software from Nero Folk?
Gavin Hollinger wrote: SIPPS generates DTMF based on the standard set-op for DTMF for PSTN telephones. SIPPS transmits DTMF as tones and not as events. Hence, any application awaiting an event instead of a tone will not be able to work with SIPPS Thereby rendering DTMF unreliable with non G711 codecs. :( It'd sure be nice if the SIP phone makers would abandon inband DTMF... Regards, -- Alastair Maw [EMAIL PROTECTED] MX Telecom - Systems Analyst http://www.mxtelecom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Conference without zaptel??
WipeOut . wrote: Am I correct in saying that conferencing does not work on a system that does not have a Digium board installed?? It doesn't work on a system that doesn't have a zaptel driver installed. If you don't have any zaptel hardware, you can compile a dummy zaptel driver which will let you do conferencing. Download the zaptel module from ftp.asterisk.org (or cvs.digium.org). Edit the Makefile - uncomment the ztdummy.o part in the MODULES line by removing the hash in front of it. make clean make install the zaptel driver. You might need to make sure you have the appropriate stuff in your kernel, such as the USB UHCI support (not the alternative JE version). depmod -a should give no unresolved symbols. Install the ztdummy module with modprobe ztdummy. You can check it all worked by going lsmod. HTH, -- Alastair Maw [EMAIL PROTECTED] MX Telecom - Systems Analyst http://www.mxtelecom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Conference call
Jeroen wrote: Conference call problem - do not have any special hardware added to the system yet. Did the following: * Uncommented the ztdummy.c in the Makefile (/zaptel) - recompiled all [...] Any ideas? When you do an lsmod, is ztdummy listed? If you do a depmod -a is there any output, and if so, what is it? -- Alastair Maw [EMAIL PROTECTED] MX Telecom - Systems Analyst http://www.mxtelecom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP agent logging into queue?
Jamie Carl wrote: And what does your post have to do with SIP? Where did u get H323 from? Dave was referring to the way in which Sebastian included a References: header in his e-mail, generated by clicking reply to and changing the subject, rather than creating a fresh and shiny new mail. This breaks the threading on any mail client that has a clue (Mozilla, Evolution, numerous others) and is therefore to be avoided. If you don't use a thread-view capable mail client on high volume mailing lists, you don't know what you're missing. In your defence, I'd say to Dave that one exclamation mark is generally sufficient. Additionally, making aggressive remarks regarding netiquette isn't terribly useful. Pointers saying what exactly someone has done wrong and suggesting a solution are much more useful. Everybody play nice, m'kay? Learn to read! J (aka mailing list etiquette nazi) For someone who claims to be a mailing list etiquette nazi, you're not doing terribly well. :) I believe, for example, that most people prefer you not to include the whole of the previous post within your own, but to quote in context. I'd also prefer not to have the CommuniGate(tm) Pro spam taglines. But enough of these petty squablings... -- Alastair Maw [EMAIL PROTECTED] MX Telecom - Systems Analyst http://www.mxtelecom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Java SIP Client
Stuart Hirst wrote: Does anyone know of a Java based SIP client and if so have has anyone used it. I found JAIN at https://sip-communicator.dev.java.net/ but have not tried it yet. The NIST JAIN implementation is quite mature, and the soft-phone demo app that it used to ship with has now been farmed off into the separate project that you mention above. It's good for debugging (you can get all the SDP headers and things out), but it's not a terribly good bit of software UI/features-wise. In particular, IIRC it lacks the ability to send DTMF. -- Alastair Maw [EMAIL PROTECTED] MX Telecom - Systems Analyst http://www.mxtelecom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] reload
Chee Foong wrote: I wonder is there a way where I reload asterisk on CLI without disconnect any call that is currently taken place. Type help into the console and read. canopy*CLI help restart gracefully Usage: restart gracefully Causes Asterisk to stop accepting new calls and exec() itself performing a cold. restart when all active calls have ended. canopy*CLI help restart when convenient Usage: restart when convenient Causes Asterisk to perform a cold restart when all active calls have ended. canopy*CLI help reload Usage: reload Reloads configuration files for all modules which support reloading. -- Alastair Maw [EMAIL PROTECTED] MX Telecom - Systems Analyst http://www.mxtelecom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] h extension seems to wipe variables?
Brian West wrote: Correct me if i'm wrong but doesn't the cdr modules log the call duration? If you look at the last sentence of my post: Storing stuff using the cdr isn't really an option. This is because I want to add other things to my call log that CDR doesn't support (for custom IVR apps and the like), and I'd rather not have to write scripts to pull stuff from the CDR database and sync it with a table in another database, which would be really ugly. -- Alastair Maw [EMAIL PROTECTED] MX Telecom - Systems Analyst http://www.mxtelecom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How do i configure so an incoming call triggersan http request?
Dave Wilson wrote:
I basically want asterisk to request an uri on our intranet, which will pass
call details to our application and consequently store call details in a MS
SQL Server database. Of course, if there are tools available for directly
interacting with the database via odbc, then that would be excellent,
however we still want to be able to trigger http requests for various other
features.
It's not terribly efficient (starts up a Perl interpreter for every
incoming call), but you can do this with AGI, something like:
extensions.conf
---
[default]
s,1,AGI(log-call.pl)
log-call.pl
---
#!/usr/bin/perl
use DBI;
use Asterisk::AGI;
my $agi = new Asterisk::AGI;
my $dbh = DBI-connect('DBI:mssql:database:hostname', 'user', 'pass') or
die %DBI::errstr;
my $clid = $agi-get_variable('CALLERID');
my $dnid = $agi-get_variable('DNID');
my $add_date = time();
$sql = INSERT INTO call (call_from, call_to, add_date) values
('$myclid', '$mydnid', FROM_UNIXTIME($add_date));;
$sth = $dbh-prepare($sql) or die $sth-errstr();
$sth-execute() or die $sth-errstr();
$dbh-disconnect();
Sort of thing. Obviously you could use LWP or similar in Perl to do HTTP
GETs, or just write a bash script which used wget to do an HTTP GET,
which would be much more efficient.
--
Alastair Maw [EMAIL PROTECTED]
MX Telecom - Systems Analyst
http://www.mxtelecom.com
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Re: [Asterisk-Users] How do i configure so an incoming call triggersan http request?
Alastair Maw wrote:
log-call.pl
[...]
oops. I meant:
$sql = INSERT INTO call (call_from, call_to, add_date) values
('$clid', '$dnid', FROM_UNIXTIME($add_date));;
not:
$sql = INSERT INTO call (call_from, call_to, add_date) values
('$myclid', '$mydnid', FROM_UNIXTIME($add_date));;
Also note that you'll need the DBI PEAR module, along with DBD::mssql
(if it exists, maybe you should use sybase for MS-SQL instead). You'll
also want the Asterisk Perl module (use Asterisk::AGI) , which you can
download from here:
- http://asterisk.gnuinter.net/
This link might also be useful:
- http://home.cogeco.ca/~camstuff/agi.html
--
Alastair Maw [EMAIL PROTECTED]
MX Telecom - Systems Analyst
http://www.mxtelecom.com
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Re: [Asterisk-Users] How do i configure so an incoming call triggersan http request?
Dave Wilson wrote: [...] [default] s,1,AGI(bash-scriptname.sh) To call my script from asterisk? That should work fine. You need to put the shell/perl script in the agi-bin directory specified in /etc/asterisk/asterisk.conf (typically /var/lib/asterisk/agi-bin/). Make sure you remember to chmod +x your bash script, or it won't execute. :) Something like this should work fine: #!/bin/sh # uncomment one of these: #/usr/bin/lynx -source http://your.site.com/foo /dev/null 21 /usr/bin/wget -O - http://your.site.com/foo /dev/null 21 You can write AGI scripts in anything you like - it uses stdin and stdout to communicate with Asterisk (set variables, send commands, etc.). Check out the last link I sent in my previous mail for details. -- Alastair Maw [EMAIL PROTECTED] MX Telecom - Systems Analyst http://www.mxtelecom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] h extension seems to wipe variables?
Hi. I'm trying to do some custom call logging, and I want to call an AGI script from a hangup handler to log call durations and things. Although the script executes, it isn't retrieving variables from the AGI interface. Looking closer, I realised the variables are actually getting unset before the h extension is reached. [foo] s,1,SetVar,foo=bar s,2,Play(audio/a-long-prompt) h,1,AGI(log-call-duration.pl) When I do an $agi-get_variable(foo) from the perl, I get the string noresponse returned. This all works fine if I don't call the AGI from the hangup extension, but from a normal one instead. Does anyone have any idea how I might fix or work around this? It's important for us to log call durations (and other things), which obviously needs to be done when the users hangs up. Storing stuff using the cdr isn't really an option. -- Alastair Maw [EMAIL PROTECTED] MX Telecom - Systems Analyst http://www.mxtelecom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] dialogic drivers
Where do I get a Dialogic driver for Asterisk from? The handbook mentions it in passing as a paid-for option. How much does this cost, and how does one go about obtaining it? -- Alastair Maw MX Telecom http://www.mxtelecom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
