Re: [asterisk-users] zonedata.c
On Sat, May 12, 2007 at 08:27:23PM -0500, Jadrien Wauthier wrote:
Hi,
Could anyone tell me how to read the values in the zonedata.c file?
I am looking at the zt_tone_ringtone field mainly.
{ ZT_TONE_RINGTONE, 425/1000,0/4000 }
means a tone of frequency 425 Hz for 1000 ms and then silense (0 Hz) for
4000 ms.
{ ZT_TONE_RINGTONE, 400+450/400,0/200,400+450/400,0/2000 }
Similar notation. Though in this case the frequency is a bit more complex:
400+450 mean means a sound composed of a 400Hz and 450Hz frequencies.
(right?)
{ ZT_TONE_INFO, !950/330,!1440/330,!1800/330,0/1000 }
The '!' tells that those sounds should not be repeated on the second
time this sound is played.
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Re: [asterisk-users] HPEC audio clipping
I didn't know that ! Thanks for the tip ! 2007/5/11, Noah Miller [EMAIL PROTECTED]: Our last trial dates from 1.2.17 days (3 weeks ago). My question is : are those HPEC audio clipping issues fixed with 1.2.17.1 ? It's not about the Asterisk version, it's about the HPEC version. According to other posts on the list, HPEC version 8.2 does not have the clipping issues, but the 9.x versions do. The echo updates to Asterisk 1.2.17.1 are to allow troubleshooting of the 9.x versions of HPEC so we can locate where the issues are. The updates do not fix the clipping issues themselves. If you have the clipping issue, make sure you get HPEC version 8.2 from Digium. - Noah 2007/5/9, Matthew Fredrickson [EMAIL PROTECTED]: If you contact Digium tech support directly they will provide you with the previous version of the echo canceler until the fix is made to the current version. Matthew Fredrickson On May 9, 2007, at 7:27 AM, Olivier wrote: Any field return on this ? Our last field trial of HPEC concluded we shouldn't use it at all, due to audio clipping. Is it now fixed ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Snom 320 voicemail key MWI
Stephen Bosch wrote: Ariel Monaco wrote: Dear List, I'm having a blinking MWI light on the snom 320 even when there's no message waiting in Asterisk. We've managed to make the voicemail button work using fromdomain=192.168.0.1 in sip.conf vmexten=2500 (our VoicemailMain application extension in extensions.conf). We also added notifymimetype=application/simple-message-summary also in sip.conf to allow SIP simple MWI notifications. But the light is still blinking and there are no voicemail messages, any ideas about how to address this issue will be welcome. You've mentioned how your Asterisk server is configured, but how is the *phone* configured? If the MWI light on the phone is set to use the wrong mailbox, you would see a blinking light, even if you've erased all the messages in the mailbox that is accessed from the voicemail button. Two things are happening here: 1. You've got a button that you configure for retrieving messages 2. You've got a Message Waiting Indicator light that blinks when there are messages in the specified mailbox. Those are separate things -- you can have a button that retrieves from one box and a light that indicates messages in another box. Check your phone configuration again. By default the Snom phones also use that light for missed calls. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk High-Capacity Stability
Atlanticnynex wrote: whether Asterisk could handle roughly one DS3's worth of calls (672 calls) just doing the LCR (I've seen some pre-built LCR apps, looks like they all do on-the-fly MySQL queries- I think I'd write my own AGI that would use a cache). When appropriately configured, MySQL does a pretty good job of caching results too. [129 lines snipped] /Per Jessen, Zürich -- ENIDAN Technologies GmbH - managed email security. Starting at SFr1/month/user - http://www.spamchek.ch/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Zapateller and IAX2
Hi, I have been using Zapateller with a TDM400 no problems at all, but recently I have ported our BT number to a VoIP provider, and have a strange problem. When I phone our number I first get the BT unavailable three tone sound, and then it actually connects the call via IAX2. So, I disabled zapateller in the dialplan and tried again. Would you believe it worked fine. Has anybody else come across this ? I am using * 1.4.4. Regards, -- --[ UxBoD ]-- // PGP Key: curl -s http://www.splatnix.net/uxbod.asc | gpg --import // Fingerprint: 543A E778 7F2D 98F1 3E50 9C1F F190 93E0 E8E8 0CF8 // Keyserver: www.keyserver.net Key-ID: 0xE8E80CF8 // Phone: +44 845 869 2749 // SIP Phone: [EMAIL PROTECTED] -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] List of telemarketers??
On Sun, May 13, 2007 at 01:54:08AM +0100, Chris Bagnall wrote: 3. a list of bogus entries..so when you look at it, you know it's a fake phone number...one that recently came in that got me thinking this was 407 111 . I don't know much about the legal position over the other side of the pond, but I'm pretty sure that in the UK caller ID spoofing is illegal. There's nothing to stop you withholding your CLI of course, but to deliberately fake someone else's CLI (whether it exists or otherwise) pushes you over the line. Is the same not the case in the US? I don't know if it's illegal (it would fall under the Comms Act if it was), but I know it's discouraged. There are legal reasons you might spoof CLI. Most telcos will have agreements that end-users can't do nasty things with CLI (withholding doesn't actually block anything, just flags the CLI should be withheld, so telcos, law enforcement etc still get it). Quite a few SS7 providers will allow customers to do what they like with CLI, just have an agreement they wont do anything they shouldn't. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TC400B load problem
Hi Im trying to install my TC400B trans coder card when I do: modprobe wctc4xxp tail -f /var/log/messages says: May 13 14:56:36 pbx2 kernel: Registered codec translator 'DTE Encoder' with 92 transcoders (srcs=000c, dsts=0101) May 13 14:56:36 pbx2 kernel: Registered codec translator 'DTE Decoder' with 92 transcoders (srcs=0101, dsts=000c) May 13 14:56:36 pbx2 kernel: Zaptel DTE (g.729a / g.723.1 5.3kbps) Transcoder support LOADED (firm ver = 56) May 13 14:56:49 pbx2 kernel: wctc4xxp: probe of :03:01.0 failed with error -5 please help thanks arun ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Snom 320 voicemail key MWI
Or you can specify vmexten = *97 in sip.conf and your VM button will work. Regards, Nitesh Nick Adams wrote: Stephen Bosch wrote: Ariel Monaco wrote: Dear List, I'm having a blinking MWI light on the snom 320 even when there's no message waiting in Asterisk. We've managed to make the voicemail button work using fromdomain=192.168.0.1 in sip.conf vmexten=2500 (our VoicemailMain application extension in extensions.conf). We also added notifymimetype=application/simple-message-summary also in sip.conf to allow SIP simple MWI notifications. But the light is still blinking and there are no voicemail messages, any ideas about how to address this issue will be welcome. You've mentioned how your Asterisk server is configured, but how is the *phone* configured? If the MWI light on the phone is set to use the wrong mailbox, you would see a blinking light, even if you've erased all the messages in the mailbox that is accessed from the voicemail button. Two things are happening here: 1. You've got a button that you configure for retrieving messages 2. You've got a Message Waiting Indicator light that blinks when there are messages in the specified mailbox. Those are separate things -- you can have a button that retrieves from one box and a light that indicates messages in another box. Check your phone configuration again. By default the Snom phones also use that light for missed calls. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dry Copper Pair
Quoting Stephen Bosch [EMAIL PROTECTED]: C F wrote: Stephen i disagree. growing up in new work city i can say its quite easy to get away with it in the city. where i live now in new jersey (population of around 6) i wouldnt be able to pull that off. The world is a big place, and I suppose there's room for all kinds. In these parts, the vigilance is pretty high. The pillars are padlocked now; they didn't use to be, and the COs are locked down like Fort Knox. Anyway, I know enough more than one person who has landed in the clink for treating the telco like a personal lab. what exactly was the charge ? - trespass - no its public land for the most part this stuff is on so that doesn't apply - vandalism/mischief - if no other customer was impacted I don't see how this charge would stick since there is no measurable damages. - theft of service ? Going rate for dry copper is under $20/month/pr so to get up into the 5-10k level that might justify a higher level theft charge with jail time that would take some time to add up. Stealing cable TV/satellite probably works out to about 3x the monthly rate of dry copper and I have never heard of anyone being told anything more than disconnect it when they get caught. I am not trying at all to justify the moral aspect of theft, I am just making a point that I have never heard of anyone even getting in trouble, let alone jail. The other issue is what crime would be involved in assisting the telco to deliver a better level of service by doing work yourself ? For example I often do as much work on their side of the demarc as possible when I have an order pending, then I know its done the way I would have wanted it. I have never got anything other than a thank you when the installer shows up and I just tell them where to make the final connection. Here is another what if - we had an adsl service in Mississauga at one point that would never quite work properly, finally got the answer - the line has a bridge tap on it somewhere but we won't remove it. WTF ? they contract to supply a service, now have an explanation why its substandard yet won't fix it ? The point became moot as we cancelled the service eventually, but say I had stuck my tone generator on it, and walked back down the road to the CO and poked around till I found the bridge tap and removed it. What would they charge me with ? I'm only helping them fulfil a contractural obligation they don't seem to want to meet. The way it ended was they preferred to lose the account entirely rather than fix a problem they knew about. The other issue that hasn't even been touched in this thread is how easy it is to just tap someone's line when everything is so exposed like this. The tap might get found, but if it was a line powered radio transmitter, chances of tracing back to the installer are minimal unless someone saw it get installed. -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Double DTMF digits
I am actually getting DTMF over SIP when people call in to a clients system that is running a2billing. They are using RFC2833. - Original Message - From: Remi Quezada [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, May 09, 2007 6:15 PM Subject: Re: [asterisk-users] Double DTMF digits I wonder if your hardware is doing the actual DTMF detecting. What hardware are you using? I'm using the TE205P and I believe that the DTMF detection is being done in the software in my case. Remi Steve Davies wrote: On 5/3/07, Ken Leland III [EMAIL PROTECTED] wrote: When dtmfmode is set to inband for SIP, and i originate a call from sip out to the PSTN, I can hear the DTMF digit twice in the audio stream. Once very briefly and once for normal duration. Our Theory: While Asterisk is parsing the DTMF, for a fraction of a second, while the end user generated DTMF is being detected, the DTMF is passed inband. Once the DTMF is detected Asterisk silences it and regenerates it. Sensitive machines like auto attendants pick up both the brief end user generated tone as well as the full length asterisk generated tone and ultimately perceive each digit twice. Is anyone else experiencing this? I have reproduced this in an environment * with one asterisk server that is both the feature server and the media gateway, and is timing off of network T1s * with two servers, one feature server (timing off of ztdummy) and one media gateway (timing off of network T1s) using IAX as the inter asterisk protocol It is pretty easy to reproduce: -Dial a PSTN number(like your cell) from a sip phone using inband DTMF, and configured in asterisk sip.conf with dtmfmode=inband. -Answer the PSTN end. -Press and hold a digit on the sip phone. On the PSTN phone you will hear a very brief, end user generated, tone. -Let go of the digit on the sip phone. On the PSTN phone you will hear the asterisk generated tone. Can anyone else hear the brief initial tone? Any help is greatly appreciated! Yes, we have a similar issue, but do not normally use inband DTMF because SIP phones very cleanly generate rfc2833 RTP packets directly and remove this issue. On the other hand, asterisk is not alone dealing with this issue in SIP. The Linksys ATAs have exactly the same issue. Strangely, I do not have a problem receiving inband DTMF through Zaptel, which I believe uses the same DSP code for DTMF detection... Or does it? Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call to Skype network
yeah that would be great! Aren't there any open-source projects out there who handle this? greetz 2007/5/13, Dave Bour [EMAIL PROTECTED]: On x86 asterisk systems, there's 3 options out there, of which the Chanskype one I've found to be the best. It's $20 US for a single channel personal license or $99 / per channel on a business license. On the FreePBX systems/Trixbox, Tim Hunt wrote an excellent script to configure it. I've made a couple notes too if you want, I can send offlist (unless it's generally wanted here onlist as I don't like taking credit for others work). D -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Verscheure Sent: Saturday, May 12, 2007 9:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Call to Skype network Hi everyone, Is it possible to call from your Asterisk server to the Skype network? i.e., let's say I would like to call from an extension from my Asterisk PBX machine to a Skype account, is this possible? I did a little bit of searching and they were talking about that's only possible with windows machines, is this true? greetz, Tim ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisknow b5 - trouble registering at voip provider
Hi, there. I have asterisknow beta 5 with the following data: Ip 192.168.0.60 mask 255.255.255.0 gw 192.168.0.1 the router (a linksys) has port forwarded the port udp 5060 and from 16384 to 16482 udp-tcp from the internet to the asterisk machine. the only protocol allowed is g729. Which work fine for the ip phones I already have setup in the LAN. My problem is trying to register to a voip provider. in the asterisknow gui I provide: protocol sip register (checked) host sf2.clarocom.net username (my phone number) password (assigned password) While executing sip show claro91 asterisk*CLI sip show peer claro91 asterisk*CLI * Name : claro91 Secret : Set MD5Secret: Not set Context : DID_ Subscr.Cont. : Not set Language : AMA flags: Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup: 1 Pickupgroup : 1 Mailbox : VM Extension : asterisk LastMsgsSent : 32767/65535 Call limit : 0 Dynamic : No Callerid : 2029191 MaxCallBR: 384 kbps Expire : -1 Insecure : no Nat : RFC3581 ACL : No T38 pt UDPTL : No CanReinvite : No PromiscRedir : No User=Phone : No Video Support: No Trust RPID : No Send RPID: No Subscriptions: No Overlap dial : No DTMFmode : auto LastMsg : 0 ToHost : sf2.clarocom.net Addr-IP : 200.105.69.132 Port 5060 Defaddr-IP : 0.0.0.0 Port 5060 Def. Username: 2029191 SIP Options : (none) Codecs : 0x80100 (g729|h263) Codec Order : (g729:20) Auto-Framing: No Status : Unmonitored Useragent: Reg. Contact : asterisk*CLI asterisk*CLI and when i try to call with my lan phones to the outside via the claro91 trunk, I get asterisk*CLI -- Executing [EMAIL PROTECTED]:1] Macro(SIP/6000-0820e870, trunkdial|SIP/claro91/66944780) in new stack -- Executing [EMAIL PROTECTED]:1] Dial(SIP/6000-0820e870, SIP/claro91/66944780) in new stack -- Called claro91/66944780 [May 13 17:37:40] WARNING[5522]: chan_sip.c:11860 handle_response_invite: Received response: Forbidden from 'Erick Perez sip:[EMAIL PROTECTED];tag=as7eabcb2e' -- SIP/claro91-082127d8 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing [EMAIL PROTECTED]:2] Goto(SIP/6000-0820e870, s-CONGESTION|1) in new stack -- Goto (macro-trunkdial,s-CONGESTION,1) -- Executing [EMAIL PROTECTED]:1] NoOp(SIP/6000-0820e870, ) in new stack == Auto fallthrough, channel 'SIP/6000-0820e870' status is 'CONGESTION' asterisk*CLI If I switch from my asterisknow box to the linksys box (that has two rj11 ports) then the registration is fine. I would like some guidance as to how to properly format the registration string for my provider. thanks, -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need a RTP/SIP Proxy to be used as SBC (Session Border Controller)
Check rtpproxy from portone for media proxy and nat traversal. http://www.voip-info.org/wiki/view/Portaone+rtpproxy another option is the MediaProxy from AG projects: http://www.voip-info.org/wiki-MediaProxy Joss. On 5/11/07, Jean-Marc Salsa [EMAIL PROTECTED] wrote: Hi all, I have been using asterisk to do such kind of thing, But I must admitt, this is not 100 % conveniant (Mainly because Asterisk isn't a SIP Proxy). I just wanted to know if you knew/used some kind of SBC or packages which would deal both with SIP AND RTP ! SER/OpenSER woulc be a good SIP Proxy ... but then how to deal with RTP ? Any tip, info greatly welcome ! Thanks, JM ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Dealing with 2 SIP providers
Wow, thanks for the detailed response. For your last question, on why I
want to put a forced delay of 30 seconds, it's because the example I gave
was quite simplified to isolate what my real issue was. In reality there
are cases where I want to force 30 seconds before hanging up, and other
cases where I will let the caller decide when he's giving up.
Chris, I really do appreciate your help. I haven't used AEL yet, if is
ready for production at this point? (a while ago it was recommended not to
use it). I ask this because I find this method of writing code MUCH easier
then a sort of
programming with every line numbered like BASIC
Can you easily mix and match AEL and standard Asterisk (i.e. my old code
with new code I would put in?)
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris Bagnall
Sent: Friday, May 11, 2007 22:22
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] Dealing with 2 SIP providers
What I mean is I want a call to go out on ProviderA, UNLESS it's down
and then go to ProviderB.
I want it to ring 30 seconds and then Hangup if nobody has answers.
This one's actually a bit more complicated than it first seems, since you
need to know how each provider reports status when it's unavailable. We run
the following AEL macros to achieve something similar:
(apologies to the list for the big chunk of code below - I'm not sure how
well/if the list handles attachments)
// DIAL NUMBER (with a range of routing options) macro outbound (number,
route1, route2, route3, route4, route5) {
// set correct outbound caller id
if (${LEN(${CALLERID(number)})} 10 ${LEN(${CALLERID(number)})}
0) {
if (${LEN(${DB(callerid/${CDR(accountcode)})})} 9) {
CALLERID(number)=${DB(callerid/${CDR(accountcode)})};
} else
Set(CALLERID(number)=);
};
dialstart:
switch (${route1}) {
case dundi:
if (${number:0:2} = 00) {
dundi-e164 (${number:2});
} else if (${number:0:1} = 0) {
dundi-e164 (44${number:1});
} else
dundi-e164 (${number});
break;
case provider1:
dialout (IAX2/provider1/${number});
break;
case provider2:
dialout (IAX2/provider2/${number});
break;
case provider3:
dialout (IAX2/provider3/${number});
break;
case pstn:
dialout (Zap/g1/${number});
break;
default:
NoOp (invalid route: ${route1});
};
if (${LEN(${route2})} 0) {
route1=${route2};
} else {
Playtones (congestion);
Congestion ();
};
if (${LEN(${route3})} 0)
route2=${route3};
if (${LEN(${route4})} 0)
route3=${route4};
if (${LEN(${route5})} 0)
route4=${route5};
goto dialstart;
};
// DIAL NUMBER (ignoring anything except busy) macro dialout (dialstring) {
Dial (${dialstring},,TW);
switch (${DIALSTATUS}) {
case BUSY:
Playtones (busy);
Busy ();
break;
case CONGESTION:
Playtones (busy);
Busy ();
break;
};
};
You can then dial from your main dialplan something like this for UK
landlines:
exten = _0[12]X,1,Macro(outbound,${EXTEN},provider1,provider2,pstn)
The dialout macro ignores any responses from the SIP/IAX provider except
Busy or Congestion (we have a provider which provides congestion when the
dialled number is busy, that's why it's there). So, if the provider's server
is unavailable (through qualify=yes or whatever), it'll fall through as
channel status unknown and loop onto the next provider.
On an outbound call made from one of your users, why would you want a 30
second timeout? Surely you'd want to keep ringing the callee until the
caller (i.e. your user) loses interest and hangs up their device? The length
of time for a device to be rung before doing something else is usually
determined by the recipient, not the initiator.
Hope that helps.
Regards,
Chris
--
C.M. Bagnall, Director, Minotaur I.T. Limited For full contact details visit
http://www.minotaur.it/chris.html This email is made from 100% recycled
electrons
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RE: [asterisk-users] Dealing with 2 SIP providers
I haven't used AEL yet, if is ready for production at this point? (a while ago it was recommended not to use it). I think AEL's been replaced by AEL2 in 1.4, but I've yet to migrate our asterisk deployments to 1.4, so cannot confirm. I've not found any stability issue using AEL - essentially it's converted to regular dialplan syntax on load anyway, so there shouldn't be any issues. Can you easily mix and match AEL and standard Asterisk (i.e. my old code with new code I would put in?) Yes, I tend to program macros and the like which'll be common across a number of asterisk boxes in AEL, then have deployment-specific stuff in extensions.conf. That way I can fire a common extensions.ael file at all the boxes we maintain whilst not overwriting custom dialplan logic related to an individual install. As I said in my earlier post, you'll need to do some testing to see what your 2 providers report when they're unavailable (and likewise when a number's genuinely engaged - does the provider report congestion or busy back to asterisk?) Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited For full contact details visit http://www.minotaur.it/chris.html This email is made from 100% recycled electrons ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RE: zonedata.c
Thank you very much. This helps a lot.
Jad
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Date: Sun, 13 May 2007 10:58:44 +0300
From: Tzafrir Cohen [EMAIL PROTECTED]
Subject: Re: [asterisk-users] zonedata.c
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
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On Sat, May 12, 2007 at 08:27:23PM -0500, Jadrien Wauthier wrote:
Hi,
Could anyone tell me how to read the values in the zonedata.c file?
I am looking at the zt_tone_ringtone field mainly.
{ ZT_TONE_RINGTONE, 425/1000,0/4000 }
means a tone of frequency 425 Hz for 1000 ms and then silense (0 Hz) for
4000 ms.
{ ZT_TONE_RINGTONE, 400+450/400,0/200,400+450/400,0/2000 }
Similar notation. Though in this case the frequency is a bit more complex:
400+450 mean means a sound composed of a 400Hz and 450Hz frequencies.
(right?)
{ ZT_TONE_INFO, !950/330,!1440/330,!1800/330,0/1000 }
The '!' tells that those sounds should not be repeated on the second
time this sound is played.
--
Tzafrir Cohen
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Message: 6
Date: Sun, 13 May 2007 10:40:37 +0200
From: Olivier [EMAIL PROTECTED]
Subject: Re: [asterisk-users] HPEC audio clipping
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID:
[EMAIL PROTECTED]
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I didn't know that !
Thanks for the tip !
2007/5/11, Noah Miller [EMAIL PROTECTED]:
Our last trial dates from 1.2.17 days (3 weeks ago).
My question is : are those HPEC audio clipping issues fixed with
1.2.17.1 ?
It's not about the Asterisk version, it's about the HPEC version.
According to other posts on the list, HPEC version 8.2 does not have
the clipping issues, but the 9.x versions do.
The echo updates to Asterisk 1.2.17.1 are to allow troubleshooting
of the 9.x versions of HPEC so we can locate where the issues are.
The updates do not fix the clipping issues themselves.
If you have the clipping issue, make sure you get HPEC version 8.2 from
Digium.
- Noah
2007/5/9, Matthew Fredrickson [EMAIL PROTECTED]:
If you contact Digium tech support directly they will provide you with
the previous version of the echo canceler until the fix is made to the
current version.
Matthew Fredrickson
On May 9, 2007, at 7:27 AM, Olivier wrote:
Any field return on this ?
Our last field trial of HPEC concluded we shouldn't use it at all,
due
to audio clipping.
Is it now fixed ?
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Message: 7
Date: Sun, 13 May 2007 19:21:06 +1000
From: Nick Adams [EMAIL PROTECTED]
Subject: [asterisk-users] Re: Snom 320 voicemail key MWI
To: asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Stephen Bosch wrote:
Ariel Monaco wrote:
Dear List,
I'm having a blinking MWI light on the snom 320 even when there's no
message waiting in Asterisk.
We've managed to make the voicemail button work using
fromdomain=192.168.0.1 in sip.conf
vmexten=2500 (our VoicemailMain application extension in
extensions.conf). We also added
notifymimetype=application/simple-message-summary also in sip.conf to
allow SIP simple MWI
notifications.
But the light is still blinking and there are no voicemail messages, any
ideas about how to address this
issue will be welcome.
You've mentioned how your Asterisk server is configured, but how is the
*phone* configured?
If the MWI light on the phone is set to use the wrong mailbox, you would
see a blinking light
[asterisk-users] Sudden appearance of SIP/2.0 401 Unauthorized
Yesterday we moved one of our servers to a new IP. We updated DNS and various adapters configured to register to that server registered to the new IP correctly. All seemed to be well. This evening I discovered that with one exception, all of the adapters are getting a SIP/2.0 401 Unauthorized message back from asterisk. The exception is an Innomedia adapter -- Linksys PAP2's and (I believe) one Cisco ATA-188 are getting the Unauthorized. I have stopped, restarted, unloaded loaded sip, and erased astdb to start from scratch... no dice. None of the config files have changed, and, as I said, they all appeared to work last night. Can anyone give me a clue here? Yours, Yaakov Menken -- Yaakov Menken Capalon Communications, Inc. Ask us about Voice over IP for Business! http://www.capalon.com 888-CAPALON (227-2566) 410-358-9800 x120 410-510-1053 fax 443-413-1042 cell [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Now way OT: [asterisk-users] Need some help with a very simple Queestion..
On Sat, 2007-05-12 at 22:35 -0600, Stephen Bosch wrote: [EMAIL PROTECTED] wrote: Stephen Bosch wrote: Is Marmite also available in Ontario, or only Out West? As far as I know, Marmite is available all across this land, from sea to sea to sea. Three cheers for Marmite. IMO most Americans have never even *heard* of Marmite, much less tasted it. And it's quite a hoot to watch someone ingest it for the first time. Always causes a surprised look. Someone should write a book about it--or maybe someone already has :-) b. Yep, chocolate/hazelnut it aint. FWIW, Vegemite (the better, Auzzie variety of the same brew) is now illegal in the US. Something to do with its containing folate and t Is that ban still in place? My understanding is that it was temporary (insane) and lifted... If anyone needs a fix, let me know. First jar is always free. PaulH ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] h323 problem with asterisk 1.2.18
Instead of using those H323. chan drivers try using the ones in asterisk-addons-1.2.16. They seemed to work a lot better for me than the ones that came with the main asterisk package. - Original Message - From: nik600 [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, May 07, 2007 8:40 PM Subject: [asterisk-users] h323 problem with asterisk 1.2.18 i am experiencing problem with asterisk 1.2.18 I've downloaded and installed pwlib and openh323 with the following commands: cd /path/to/pwlib ./configure make clean opt cd /path/to/openh323 ./configure make clean opt then 'ive set the corresponding PATH PWLIBDIR=/data/programmi/asterisk_1.2.18/pwlib_v1_10_0/ export PWLIBDIR OPENH323DIR=/data/programmi/asterisk_1.2.18/openh323_v1_18_0/ export OPENH323DIR LD_LIBRARY_PATH=$PWLIBDIR/lib:$OPENH323DIR/lib export LD_LIBRARY_PATH but when i go to: cd asterisk-1.2.18/channels/h323/ and do a make opt: [EMAIL PROTECTED]:/data/programmi/asterisk_1.2.18/asterisk-1.2.18/channels/h323# make opt make: *** No rule to make target `opt'. Stop. why? where am i wrong? i've also tried the last version of pwlib and openh323, but without fixing the problem thanks -- /*/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/nikstresser ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] load modules
I have this in /etc/rc.d/rc.local #Modprobe Zaptel - Loads Zaptel modprobe zaptel modprobe wctdm #Start Asterisk with screen detatched screen -A -d -m -S asterisk asterisk -vvp -c - Original Message - From: Josu Lazkano Lete To: asterisk-users@lists.digium.com Sent: Tuesday, May 08, 2007 4:23 PM Subject: [asterisk-users] load modules Hello again, I have a little problem, every time I switch on the Asterisk server I must load two modules: modprobe zaptel and modprobe wctdm Is there any way to load there automatically when the server start? I have a Debian Etch. One more cuestion, it's posible to start Asterisk (asterisk -vvvc)as well? What metod do you prefer? asterisk or asterisk -vvvc? Thanks very much to all of you. Bye. -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP registration problem
I have seen this issue where there were internet connectivity issues. Asterisk registers every so often with the ITS. For some reason or another (it can be many reasons such as DNS, internet, ISP has issue etc). asterisk cant re-register so it keeps trying. As far as the so context if you have a simple register line in sip.conf (such as register= axe:[EMAIL PROTECTED]) then asterisk will tell the server that it is registering it with to send all calls to the s extension in your default context. - Original Message - From: Michelle Dupuis To: asterisk-users@lists.digium.com Sent: Saturday, May 05, 2007 4:08 PM Subject: [asterisk-users] SIP registration problem I've reposted with a more meaningful subject - hopefully someone will replyWe have an Asterisk v1.2.16 box registering with an ITSP using SIP. The registration succeeds, and is confirmed with SIP SHOW REGISTER. However, we frequently (every few minutes) see this on our console: REGISTER attempt 1 to [EMAIL PROTECTED] REGISTER attempt 2 to [EMAIL PROTECTED] Any ideas what is going on? In particular 1. What causes the two register attempt messages above? 2. Why is our asterisk box being associated with the entryunauthorized context, not the entryinternal context? (See below) 3. Why is the contact sip:[EMAIL PROTECTED]:5060 in our SIP messages, why s@ anything? Thanks MD -- Contents of sip.conf at ITSP: [999] context=entryinternal ; I know this context exists! This is the right context. type=friend username=999 secret= callerid=Test 999 host=dynamic nat=no canreinvite=no allow=ulaw allow=alaw dtmfmode=rfc2833 --- Console log from ITSP show strange SIP traffic: --- Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms pbx*CLI pbx*CLI -- SIP read from 123.183.86.231:5060: REGISTER sip:pbx.itsp.com SIP/2.0 Via: SIP/2.0/UDP 123.183.86.231:5060;branch=z9hG4bK1a8cee82;rport From: sip:[EMAIL PROTECTED];tag=as3218ff14 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 103 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Authorization: Digest username=999, realm=pbx.itsp.com, algorithm=MD5, uri=sip:pbx.itsp.com, nonce=5cec66c0, response=6451967016fc38f896efeb7247523fe1, opaque= Expires: 120 Contact: sip:[EMAIL PROTECTED]:5060 Event: registration Content-Length: 0 --- (13 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 123.183.86.231 : 5060 (NAT) Transmitting (no NAT) to 123.183.86.231:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 123.183.86.231:5060;branch=z9hG4bK1a8cee82;received=123.183.86.231;rport=5060 From: sip:[EMAIL PROTECTED];tag=as3218ff14 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 103 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- Transmitting (no NAT) to 123.183.86.231:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 123.183.86.231:5060;branch=z9hG4bK1a8cee82;received=123.183.86.231;rport=5060 From: sip:[EMAIL PROTECTED];tag=as3218ff14 To: sip:[EMAIL PROTECTED];tag=as7d680d48 Call-ID: [EMAIL PROTECTED] CSeq: 103 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Expires: 120 Contact: sip:[EMAIL PROTECTED]:5060;expires=120 Date: Fri, 04 May 2007 19:27:58 GMT ontent-Length: 0 -- SIP read from 123.183.86.231:5060: OPTIONS sip:pbx.itsp.com SIP/2.0 Via: SIP/2.0/UDP 123.183.86.231:5060;branch=z9hG4bK36c1df86;rport From: asterisk sip:[EMAIL PROTECTED];tag=as6e5334cf To: sip:pbx.itsp.com Contact: sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 04 May 2007 19:38:36 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- (12 headers 0 lines) --- Looking for s in entryunauthorized (domain pbx.itsp.com) Transmitting (no NAT) to 123.183.86.231:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 123.183.86.231:5060;branch=z9hG4bK36c1df86;received=123.183.86.231;rport=5060 From: asterisk sip:[EMAIL PROTECTED];tag=as6e5334cf To: sip:pbx.itsp.com;tag=as51d476cd Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:74.110.57.25 Accept: application/sdp Content-Length: 0 -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:
Re: [asterisk-users] Swissvoice IP10s setup
Are you using trixbos ? If yes have a look at the forums on www.trixbox.org - Original Message - From: Paul A Brown To: asterisk-users@lists.digium.com Sent: Friday, May 11, 2007 9:18 PM Subject: [asterisk-users] Swissvoice IP10s setup Hi Does anyone have a howto on how to set one of these up on Asterisk or Trix box please? I can make it SIP or MGCP so whatever you have ;-) I have found one page but it isn't really a howto setup Thanks in advance Paul -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need a RTP/SIP Proxy to be used as SBC (Session Border Controller)
Thanks, I already found these names, but maybe I missed some ! Thanks again, JM On 5/14/07, Yossi Ben Hagai [EMAIL PROTECTED] wrote: Check rtpproxy from portone for media proxy and nat traversal. http://www.voip-info.org/wiki/view/Portaone+rtpproxy another option is the MediaProxy from AG projects: http://www.voip-info.org/wiki-MediaProxy Joss. On 5/11/07, Jean-Marc Salsa [EMAIL PROTECTED] wrote: Hi all, I have been using asterisk to do such kind of thing, But I must admitt, this is not 100 % conveniant (Mainly because Asterisk isn't a SIP Proxy). I just wanted to know if you knew/used some kind of SBC or packages which would deal both with SIP AND RTP ! SER/OpenSER woulc be a good SIP Proxy ... but then how to deal with RTP ? Any tip, info greatly welcome ! Thanks, JM ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
