Re: [asterisk-users] Very bad TDMF tone !
OK, tel me put my share of experience. I have a PRI TE212P which had onboard VMP echo cancellation. I am using Asterisk-1.2-18. the DTMF issue was really bad. It rang the wrong extension some tings rang invalid extension. In the begining I didnt knew that it was an DTMF issue. Then to resolve this I enbled hardware DTMF in my TE212P card which worked fine, but still had issue with detecting DTMF Down 'f' signals during voice calls, due to which calls got dropped ? was the calls dropped... this was when during conversation DTMF Down 'f' signal was detected then a FAX line was initiated bcos it saw a 'f' signal, this was bcos in zapata.conf I had allowed FAX. I had to disable FAX in zapata.conf to resolve that issue. But daily I see at least more than 50 counts of DTMF Down 'f' signal getting detected in voice calls. So my question is the common issue which I see in this posting and mine is Asterisk 1.2-18 version, do anyone have same issue. Its bcos of hardware DTMF I am able to use Astersik, else its not worthy of that version to detect DTMF properly changed the DMF relaxed many other options, recompiled the sources to decrease the DTMF threashold value for DTMF(this is for hardware DTMF). -- Deepak Noah Miller [EMAIL PROTECTED] wrote: i am using tdm400P in my office. i tested that TDMF generated by asterisk is so bad. the sound is very soft and quality is so bad. i am using asterisk 1.2.18. most of time, the # key can not be detected correctly. Does anyone has that problem? please give me a hit for that problem! The only time I've heard of that problem is when VoIP is involved. I've never heard of this problem when the call is all analog, like with a TDM400P. 1. If asterisk is detecting DTMF, the parameter relaxdtmf= can affect DTMF detection. 2. Have you checked your handsets on both ends of the call? Some handsets try to filter out DTMF tones. 3. Is voice quiet on your calls, too, or is it just DTMF? It's possible to affect overall signal levels in zapata.conf. Can you post the relevant portion of your zapata.conf? - Noah ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - New Yahoo! Mail is the ultimate force in competitive emailing. Find out more at the Yahoo! Mail Championships. Plus: play games and win prizes.___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-users Digest, Vol 36, Issue 25
Hi Stephane, Solutions remove all the Asterisk modules as well as the addon modules, and re-install Asterisk and Addon again. # rm -rf /usr/lib/asterisk/modules/* # rm -rf /usr/include/asterisk Then reinstall it again. It works for me. Have you change the addons Makefile so that the ooh-323 file will load without problem. Good luck. 2. Re: fail to load modules (St?phane Kamga) Message: 2 Date: Sun, 8 Jul 2007 20:06:14 +0100 From: St?phane Kamga [EMAIL PROTECTED] Subject: Re: [asterisk-users] fail to load modules To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 Hi All I?ve got the same message after installing asterisk addons [res_convert.so]Jul 8 20:51:10 WARNING[4685]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/res_convert.so: undefined symbol: ast_module_unregister Jul 8 20:51:10 WARNING[4685]: loader.c:554 load_modules: Loading module res_convert.so failed! Your help please ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] List delays
Most of the users using this list do not experience the issue you are having, rather than insult the admins, please trouble shoot and if you cannot, at least post headers so others can. -- Original Message -- From: Dimitri Volski [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Date: Tue, 10 Jul 2007 11:32:49 +1000 There is definitely something wrong with this list. I have my emails sorted by date, and every day, the emails do not just come on top, but get slotted in. Today (10 July 2007), I received about 6 emails from 29th of June, couple from 30th, up until the 5th of July, nothing of today's, or, well, for the last 5 days. Admin, get your act together ! ;) -- This message has been scanned for viruses and dangerous content by Mail Call antivirus software, and is believed to be clean. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Sent via the WebMail system at rockynet.com Sent via the WebMail system at rockynet.com Sent via the WebMail system at rockynet.com Sent via the WebMail system at rockynet.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Very bad TDMF tone !
Hi, Yup, I agree. You have to post out your Zapata.conf for discussions, and Have you tried out the cable terminations and also the incoming DTMF signal voltage level? It has so many possibilities for this case. i am using tdm400P in my office. i tested that TDMF generated by asterisk is so bad. the sound is very soft and quality is so bad. i am using asterisk 1.2.18. most of time, the # key can not be detected correctly. Does anyone has that problem? please give me a hit for that problem! The only time I've heard of that problem is when VoIP is involved. I've never heard of this problem when the call is all analog, like with a TDM400P. 1. If asterisk is detecting DTMF, the parameter relaxdtmf= can affect DTMF detection. 2. Have you checked your handsets on both ends of the call? Some handsets try to filter out DTMF tones. 3. Is voice quiet on your calls, too, or is it just DTMF? It's possible to affect overall signal levels in zapata.conf. Can you post the relevant portion of your zapata.conf? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-users Digest, Vol 36, Issue 25
Hi Thanks for your help. What do I need to change in Makefile for ooh-323 to be loaded? Regards -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de clive.chan(Alpha Trilogies Networks) Envoyé : mardi 10 juillet 2007 07:17 À : asterisk-users@lists.digium.com Objet : Re: [asterisk-users] asterisk-users Digest, Vol 36, Issue 25 Hi Stephane, Solutions remove all the Asterisk modules as well as the addon modules, and re-install Asterisk and Addon again. # rm -rf /usr/lib/asterisk/modules/* # rm -rf /usr/include/asterisk Then reinstall it again. It works for me. Have you change the addons Makefile so that the ooh-323 file will load without problem. Good luck. 2. Re: fail to load modules (St?phane Kamga) Message: 2 Date: Sun, 8 Jul 2007 20:06:14 +0100 From: St?phane Kamga [EMAIL PROTECTED] Subject: Re: [asterisk-users] fail to load modules To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 Hi All I?ve got the same message after installing asterisk addons [res_convert.so]Jul 8 20:51:10 WARNING[4685]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/res_convert.so: undefined symbol: ast_module_unregister Jul 8 20:51:10 WARNING[4685]: loader.c:554 load_modules: Loading module res_convert.so failed! Your help please ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Information NOD32 2301 (20070531) __ Ce message a ete verifie par NOD32 Antivirus System. http://www.nod32.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] awful list delays: 4 days!
Am Freitag, den 29.06.2007, 14:23 -0600 schrieb Anthony Francis: Andres Paglayan wrote: On Jun 29, 2007, at 12:50 PM, Lenz wrote: Hello list, I am getting the list with days of delay, take for example this message: As you can see, the message was posted on June 25th and was sent to my email on June 29th! am I the only one who is getting such an awful message turn-around time? l. I'll let you know next week, ;^) ROFL, yeah its you. I see posts within a few hours. This one just arrived here. From the mail headers: Delivery-date: Tue, 10 Jul 2007 08:21:49 +0200 Received: from lists.digium.com ([216.207.245.17]) by server2.hoffmeister.be with esmtps (TLS-1.0:RSA_AES_256_CBC_SHA1:32) (Exim 4.63) (envelope-from [EMAIL PROTECTED]) id 1I896X-0001I0-U1 for anselm (a)hoffmeister-online.de; Tue, 10 Jul 2007 08:21:49 +0200 Received: from localhost ([127.0.0.1] helo=INXS.digium.internal) by lists.digium.com with esmtp (Exim 4.63) (envelope-from [EMAIL PROTECTED]) id 1I4N0F-0002TB-Hz; Fri, 29 Jun 2007 15:23:39 -0500 Received: from exprod8mx3.postini.com ([64.18.3.103] helo=psmtp.com) by lists.digium.com with smtp (Exim 4.63) (envelope-from anthonyf (a)rockynet.com) id 1I4N01-0002SD-UZ for asterisk-users@lists.digium.com; Fri, 29 Jun 2007 15:23:26 -0500 So it seems to be trouble between lists.digium.com and my mailserver. Judging from what I know about other people's trouble with mail delays, probably the earlier. This becomes rather unnerving, as a regular discussion cannot take place. 11 days delays is just incredible (but some messages take only 5 days ;-/ ) Perhaps someone at the server management team knows something about all this, I have forwarded this mail over there. Thanks for input how to get around this. I do not assume it is a problem on my part, but if it is, I would like to know. Anselm ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ZAP TDM and DTMF issue
Are you using SIP phones ? The issue can be from your phone to asterisk. - Original Message - From: AL Daei [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, July 10, 2007 8:42 AM Subject: [asterisk-users] ZAP TDM and DTMF issue Hi, I'm curious if there is any other option beside relaxdtmf in zapata , or any where else to tune dtmf detection on TDM400 fxo boards. in one of our sites provider is giving us 4 analog lines out of Adtran router and Asterisk often recognize DTMF wrong. Obviously playing with relaxdtmf was not helpfull. What do we know anout 1.2 and 1.4 DTMF handling diffrences? At this time i'm using 1.2 but i can change to 1.4 if i see a motivation. _ Local listings, incredible imagery, and driving directions - all in one place! Find it! http://maps.live.com/?wip=69FORM=MGAC01 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ipv6 patch
Hi 2007/7/10, Russell Bryant [EMAIL PROTECTED]: As far as I know, the patch is ready for use. It has not yet been merged into asterisk trunk, but I don't think there are technical reasons for that. It's just a matter of someone else taking a final look over it, and merging it in. The patch is available on this branch: $ svn co http://svn.digium.com/svn/asterisk/team/blanchet/v6 asterisk-ipv6 I have tried it and it builds and installs without problems. On another thread (Asterisk and IPv6) I have reported a problem related to pthread. I would like to contact the developer for this branch. Any idea who that might be? Marc Blanchet? Bent ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Session Border Controller time...
On 7/8/07, Dovid B [EMAIL PROTECTED] wrote: What does the NexTone run for ? - Original Message - From: Andy Brezinsky [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, July 03, 2007 8:17 PM Subject: Re: [asterisk-users] Session Border Controller time... We use NexTone for our SBC's on our network. We like: - 10,000 concurrent calls with media routing - SIP H.323 signaling with ability to take care of odd vendor specific issues - Basic routing engine allows you to create calling plans for individual end points - Limits by bandwidth or concurrent calls (or egress/ingress) for either discrete endpoints or via an iEdge group. - Easy GUI for those less tech savvy to do work on the machines. - Reasonable pricing on a per-port basis - Amazing Sales/Support teams. We've had some super funky requests we've thought about on a Friday night, they've got their teams together to walk us through every part of the configuration. Very knowledgeable and fun staff. (Seriously, best vendor support we've ever had, Hi Dan!) If you upgrade your SBC's to their RSM product you get basically a full Class 4 soft switch with a full LCR routing engine, reporting system and analytics engine. It's pretty powerful. Right now we're using just the SBC component and sending all ingress traffic to a egress trunk group (pointed to our OpenSER routers) but we're running a few thousand concurrent calls throught it. -- ~Andy Brezinsky On Tue, 2007-07-03 at 12:14 -0400, J. Oquendo wrote: Come on you carriers on the list... Give up the dibs what are you using and why? About to sledgehammer these SELECT * FROM GARBAGE WHERE SBC = 'nCite' Don't bother shooting me off Newport Networks stuff... Too pricey I agree with J.Oquendo! Maybe the story with 4.2 ver is different but their 3.1x line is horrible at the subscriber/access/line side, and they admit to it and have personally asked/recommended 'off-the-record' for me to go somewhere else for providing feature rich line-side features. A load of SIP METHODs/Messages aren't supported, no support for geographical redundancy (both SBCs must be placed physically in the same CoLo alongside with a x-over cable between them), Registration throttling doesn't work for me, neither does session-refresh, NAT traversal isn't adaptive (i.e. you can either media route everything or nothing, it doesn't detect that two endpoints might be behind the same NAT so don't bother media-routing them all the way to the PoP and back), doesn't load-balancing multiple application and/or call/proxy-servers (manually must assign priorities to each server) and many more but the worst and absolute worst is the support! I have solved more problems for them that I should be charging them for support instead of the other way around. I've found bugs, security holes, and incorrect implementation of the SIP RFCs. If the bug is obvious and they can figure out a solution for fast, they will work on it. If it involves investigation and/or major change/fix, they let it lie there. I had about 2 bugs that lay there in their system for almost 6 mths. Luckily I found workarounds for them and my service is running on those workarounds and will forever till we upgrade to 4.2 as we've been told that these might've (still no guarantee) been fixed in the newer releases. On the +ve side, their carrier side is good (but then, carrier/peering/interconnect is prob 25% as complex as the line-side) and robust, the quality is good and the pricing is very modularised, so you can cherry pick modules u want depending on what services you want to offer. Other SBC vendors sell you everything whether you ever use it or not! although I've heard now that Netrake has wised up and modularised their pricing after Audiocodes acquisition and having fired most of the original execs from Netrake. Anyone here heard of Covergence? I saw them at VON and had a LONG chat with them with a demo of their product. VERY neat, and am sifting through wads of their whitepapers before contacting them for inter-op for the Next PoP. Apparently the V-Dawg (Vonage) uses them not that that gives any credibility to anything but if anyone knows more than I do about them, please share! What about Acme Packet? Or Metaswitch SBCs, Juniper, Cisco, Sansay? Anyone written their own on Stacks provided by companies like Data Connection? oh BTW Dovid, You should be able to get very minimal config Nextones for about $30K/piece for about 2000 media routed calls and 20,000 registrations. This might've increased with the 4.2 train as you now HAVE to get the media-processor/DSP card which I believe is $6K extra HTH \R ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit:
Re: [asterisk-users] Gigaset 450IP loses registration
gincantalupo ha scritto: Hi, somtimes my Gigaset 450IP loses its registration. Is there anybody who knows why and how to solve it? TIA Giorgio Incantalupo I try some trick and i found: maxexpiry=120 defaultexpiry=120 in sip.conf I put this in a production env and all things are ok now. Think also to upgrade the phone to the most recent firmware (one recent change seems to solve this kind of problem). Ciao. begin:vcard fn:Massimo Nuvoli n:Nuvoli;Massimo org:Progetto Archivio SRL adr:;;Via Giustetto 75;Abbadia Alpina Pinerolo;TO;10060;Italia email;internet:[EMAIL PROTECTED] title:Amministratore Delegato tel;work:0121303544 tel;fax:0121040601 x-mozilla-html:FALSE url:www.progettoarchivio.com version:2.1 end:vcard signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] List delays
No, please don't post headers. Headers tell us nothing. They don't tell us such things such as DNS resolution problems, routing problems, if the recipient's server is tempfailing, etc. The ONLY thing really useful are the mail logs from the list server. The only people that have access are Digium employees. I encourage everyone who is having delays to visit http://lists.digium.com/mailman/listinfo/asterisk-users and look at the bottom of the page. The link to [EMAIL PROTECTED] and the other two addresses listed are the ONLY addresses that should be notified of list problems / delivery issues. Considering that so Very Very few subscribers are having delays, there is a 99% chance that you have something messed up on your side - DNS reliability, your network, one or more of your MX servers, some goofy anti-spam scheme, etc. In this case: dig mx mailcall.com.au ;; ANSWER SECTION: mailcall.com.au.60 IN MX 100 mx2.zoneedit.com. mailcall.com.au.60 IN MX 0 email.mailcall.com.au. Connection attempts to mx2.zoneedit.com were taking well over a minute to get the 220 mx2.zoneedit.com ESMTP Postfix response. Most high-volume list servers won't wait that long. I strongly suggest you find a better backup mail relay service, or don't even list a second MX. Once ALL your MX servers (no matter what priority they are listed at) are working quickly and correctly, THEN contact the list admins for further help. On Tue, Jul 10, 2007 at 12:18:32AM -0600, Anthony Francis said: Most of the users using this list do not experience the issue you are having, rather than insult the admins, please trouble shoot and if you cannot, at least post headers so others can. -- Original Message -- From: Dimitri Volski [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Date: Tue, 10 Jul 2007 11:32:49 +1000 There is definitely something wrong with this list. I have my emails sorted by date, and every day, the emails do not just come on top, but get slotted in. Today (10 July 2007), I received about 6 emails from 29th of June, couple from 30th, up until the 5th of July, nothing of today's, or, well, for the last 5 days. Admin, get your act together ! ;) -- This message has been scanned for viruses and dangerous content by Mail Call antivirus software, and is believed to be clean. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Sent via the WebMail system at rockynet.com Sent via the WebMail system at rockynet.com Sent via the WebMail system at rockynet.com Sent via the WebMail system at rockynet.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] List delays
On Tue, Jul 10, 2007 at 11:32:49AM +1000, Dimitri Volski wrote: There is definitely something wrong with this list. I have my emails sorted by date, and every day, the emails do not just come on top, but get slotted in. Today (10 July 2007), I received about 6 emails from 29th of June, couple from 30th, up until the 5th of July, nothing of today's, or, well, for the last 5 days. Admin, get your act together ! http://lists.digium.com/mailman/listinfo/asterisk-users shows: asterisk-users list run by malcolmd at digium.com, kpfleming at digium.com . I saw a comment on #asterisk (from russelb?) that the problem is basically known and is being tracked or whatever. So Admin: what kind of information would you find useful from those users who notice a delayed delivery? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DUNDI behind NAT?
Hi, i'm having asterisk with sip working fine, including dundi lookups. The only problem i'm having is that the dundi answer allways contains my internal, private ip. Is there any way to set the targeting ip that is sent out in the dundi answer (to my public ip or any other where i want to receive the call)? Regards, Andreas. _ Live Search delivers results the way you like it. Try live.com now! http://www.live.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sharing phone registration information between asterisk servers
Is it possible to share SIP phones registration information between two different asterisk servers, that share the same realtime MySQL DB? Regards, Ricardo. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax Throughput
On Wed, 27 Jun 2007 09:08:21 -0500, Matthew Fredrickson wrote: You fixed your clocking then. That was what I was thinking of. Make sure that your Dialogic card is also pulling timing from the Digium card as well. What version of zaptel drivers are you running? on a related issue, using asterisk 1.2.21 and spandsp 0.0.4 as well as the relevant rxfax and txfax, a loopback fax over an E1 PRI always goes thru at 9600bps. is there a way to increase this, or is it due to the line itself ? -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DUNDI behind NAT?
Hi Andreas,
In dundi.conf, look for the line of yours that is similar to this:
e164 = dundi-e164-canonical,0,IAX2,dundi:[EMAIL PROTECTED]/${NUMBER}
Change ${IPADDR} to your external IP address or hostname, like so:
e164 = dundi-e164-canonical,0,IAX2,dundi:[EMAIL PROTECTED]
/${NUMBER}
Cheers,
AR
On 7/10/07, Andreas Anderson [EMAIL PROTECTED] wrote:
Hi,
i'm having asterisk with sip working fine, including dundi lookups. The
only
problem i'm having is that the dundi answer allways contains my internal,
private ip. Is there any way to set the targeting ip that is sent out in
the
dundi answer (to my public ip or any other where i want to receive the
call)?
Regards,
Andreas.
_
Live Search delivers results the way you like it. Try live.com now!
http://www.live.com
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Alex Robar
[EMAIL PROTECTED]
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [EUQR] Re: North American voice BRI - Informal survey
Since many CLECs (Competitve Local Exchange Carriers in NA) offer fractional PRI, combined with Internet/Data, I haven't seen any demand for ISDN BRIs for voice or data since early 90s. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen Bosch Sent: Wednesday, July 04, 2007 9:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [EUQR] Re: [asterisk-users] North American voice BRI - Informal survey Jeff Davis wrote: Jon Pounder wrote: If someone already has a customer relationship with them, ask straight out does it work in US/Canada with the BRI available here with asterisk. I just got off the phone with my sales rep. It appears I'm the third person today to ask about this. (I wonder why?) Your rep at Sangoma? Or your reseller? The answer is no it will not work in NA. Their reasoning being that with limited resources they went after the biggest market. I get the impression that there are no plans to write a North American driver as the demand seems to be very low. This is a real chicken-and-egg problem. More people would get BRI if there were affordable hardware for it. I would like to see them write a NAm driver for it. To get them to take the chance, there have to be enough people willing to purchase the card to make them consider it seriously. The other option is a bounty or community support to get it done. The hardware already exists. The more people make noise about this, the better the chances of that happening. -Stephen- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] VAD/CNG
I know that asterisk does not support VAD. I poked around and saw some reference to asterisk supporting CNG. Will CNG work without VAD ? If yes is there any way to set this on asterisk 1.2.X for SIP using G729 ? Thanks. Dovid___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting Appearance on Outbound Calls?
Right... you dial *67 to block, however WE are the phone provider and need to set the appearance value so that when our customers dial *67 we correctly block their caller-id from going out. On 7/9/07, Noah Miller [EMAIL PROTECTED] wrote: Hi Matt - What do I need to do to set the outbound appearance on a call so that it shows up as Unavailable or Private? In most cases, I think you'd need to arrange this with your provider. If you want to do it on a call-by-call basis (in the US), dial *67 before you dial the number. If you have Caller ID blocked permanently, dial *82 to unblock for a given call. There may be other ways to do it, though. One of my clients has a Verizon PRI. If I set the CallerID to an invalid number and call another Verizon landline, it will show up as unavailable. If I do the same and call just about any cell phone, the receiving phone will show the invalid number even if it's something like 000. On this PRI, I think it always works to set it to a validly formatted, but fake number. - Noah ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] video phones on 1.4.7
I have 3 phones P1 is a non video phone - grandstream P2 is a Grandstream GXV3000 P3 is a Grandstream GXV3000 Using P1 to place a call to P2 I get audio only (as expected). Then on P1 I transfer the call to P3 and I still only get audio. At this point shouldn't the two video phones P2 and P3 say to each other we are video and so startup the video stream??? This is not working at this time? OR is there something I am missing. Thanks, Jerry ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] video phones on 1.4.7
Hi think that once SIP/SDP invite/reinvite is sent you can not change to video stream. On 7/10/07, Jerry Geis [EMAIL PROTECTED] wrote: I have 3 phones P1 is a non video phone - grandstream P2 is a Grandstream GXV3000 P3 is a Grandstream GXV3000 Using P1 to place a call to P2 I get audio only (as expected). Then on P1 I transfer the call to P3 and I still only get audio. At this point shouldn't the two video phones P2 and P3 say to each other we are video and so startup the video stream??? This is not working at this time? OR is there something I am missing. Thanks, Jerry ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ZAP TDM and DTMF issue
No there is no SIP involved in this issue, its at IVR and ZAP incoming channel.Are you using SIP phones ? The issue can be from your phone to asterisk.- Original Message - From: AL Daei ar_daei at hotmail.comTo: asterisk-users at lists.digium.comSent: Tuesday, July 10, 2007 8:42 AMSubject: [asterisk-users] ZAP TDM and DTMF issue Hi, I'm curious if there is any other option beside relaxdtmf in zapata , or any where else to tune dtmf detection on TDM400 fxo boards. in one of our sites provider is giving us 4 analog lines out of Adtran router and Asterisk often recognize DTMF wrong. Obviously playing with relaxdtmf was not helpfull. What do we know anout 1.2 and 1.4 DTMF handling diffrences? At this time i'm using 1.2 but i can change to 1.4 if i see a motivation. _ See what you’re getting into…before you go there. http://newlivehotmail.com___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Edit ulaw file
I recorded some sound files using Asterisk record() app as ulaw file. I need to edit these sound files. What kind of audio editor can I use to edit these files? Gary Chen___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ZAP TDM and DTMF issue
On Mon, Jul 09, 2007 at 11:42:19PM -0600, AL Daei wrote: Hi, I'm curious if there is any other option beside relaxdtmf in zapata , or any where else to tune dtmf detection on TDM400 fxo boards. in one of our sites provider is giving us 4 analog lines out of Adtran router and Asterisk often recognize DTMF wrong. Obviously playing with relaxdtmf was not helpfull. What do we know anout 1.2 and 1.4 DTMF handling diffrences? One thing to do: Record the audio (with e.g. ztmonitor) and make sure Asterisk gets valid DTMFs in the audion stream. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Edit ulaw file
What operating system are we talking ? Darrell S. Long BestWeb Corporation Gary Chen wrote: I recorded some sound files using Asterisk record() app as ulaw file. I need to edit these sound files. What kind of audio editor can I use to edit these files? Gary Chen ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] video phones on 1.4.7
Hi think that once SIP/SDP invite/reinvite is sent you can not change to video stream. On 7/10/07, Jerry Geis geisj at pagestation.com http://lists.digium.com/mailman/listinfo/asterisk-users wrote: / // I have 3 phones // // P1 is a non video phone - grandstream // P2 is a Grandstream GXV3000 // P3 is a Grandstream GXV3000 // // Using P1 to place a call to P2 I get audio only (as expected). // Then on P1 I transfer the call to P3 and I still only get audio. // // At this point shouldn't the two video phones P2 and P3 say // to each other we are video and so startup the video stream??? // // This is not working at this time? // OR is there something I am missing. // // Thanks, // // Jerry This is NOT good from my point of view. The above situation is the smallest reproducable situation that I could find. That did not involve a lot of detail. What I am trying to do is use call files (which have NO video just like P1 above) and call a video phone, then call another video phone. The result is NO VIDEO. Since the first session (call file to video phone) did not have video the second session video phone to video phone does not either. How can this get fixed? Jerry / ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Edit ulaw file
sox On 7/10/07, Gary Chen [EMAIL PROTECTED] wrote: I recorded some sound files using Asterisk record() app as ulaw file. I need to edit these sound files. What kind of audio editor can I use to edit these files? Gary Chen ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- /* Andrew Latham LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] [EMAIL PROTECTED] */ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] installing * from source
The page has been wiki-fied and looks more usable, thank Mat Kovach of NOOSS for the suggestion and enhancements. http://nooss.org/wiki/Installing_Asterisk_From_Source thnx, -baji. -- On 7/7/07, Baji Panchumarti wrote: Just a quick listing of tested, and updated, steps from my notes. Enjoy ! http://asterisk-notes.blogspot.com/2007/07/installing-asterisk-from-source-on.html -baji. -- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Edit ulaw file
Gary Chen wrote: I recorded some sound files using Asterisk record() app as ulaw file. I need to edit these sound files. What kind of audio editor can I use to edit these files? Gary Chen You could try Audacity. They have both a windows and linux version. http://audacity.sourceforge.net/ It should handle ulaw files. I don't have it installed at the moment to tell you for certain. -Dave ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Edit ulaw file
Sox will convert them to a different format. If you want to edit them, you will need something more sophisticated than that. Audacity should be able to do it for most OS's. Darrell S. Long BestWeb Corporation Andrew Latham wrote: sox On 7/10/07, Gary Chen [EMAIL PROTECTED] wrote: I recorded some sound files using Asterisk record() app as ulaw file. I need to edit these sound files. What kind of audio editor can I use to edit these files? Gary Chen ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Edit ulaw file
For interactive GUI-based editing, I have used audacity on linux workstations. I use command line sox for things such as format conversions. Andrew Latham wrote: sox On 7/10/07, Gary Chen [EMAIL PROTECTED] wrote: I recorded some sound files using Asterisk record() app as ulaw file. I need to edit these sound files. What kind of audio editor can I use to edit these files? Gary Chen ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call still in queue after Reject Signal
Hi kevin, My problem it's i can't remove a caller from queue. I defined a queue wich can accept only one call, and i have agent on this queue, my agent send a busy signal if he is on communication, and i want that asterisk remove new incoming call from the queue if the agent is busy (execute the next extension of dial plan). dialplan: exten=888,1,Queue(mqueue) exten=888,2,Voicemail(888) If the agent takes a first call, he send a busy signal to the other callers, Asterisk recieves the busy but he doesn't execute the next extension of dial plan. Have you eny idea to say to Asterisk to execute the next extesion if he recieve a busy signal?. Thanks. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Edit ulaw file
On Tue, 2007-07-10 at 10:24 -0400, Gary Chen wrote: I recorded some sound files using Asterisk record() app as ulaw file. I need to edit these sound files. What kind of audio editor can I use to edit these files? You can use audacity, works on GNU/Linux and windows and is free software (free as in freedom). Regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] List delays
At 02:22 AM 7/10/2007, you wrote: Considering that so Very Very few subscribers are having delays, there is a 99% chance that you have something messed up on your side - DNS reliability, your network, one or more of your MX servers, some goofy anti-spam scheme, etc. Or maybe it's just a VERY VERY few who are complaining. I posted a message on the 6th that I'm still waiting to see, but others show up in minutes. Ira ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PlayDTMF and Asterisk Manager
Hi sorry to bother, but I wasted a lot of time on this question, contact several forum (as much english as french), and still no answer :-(. In order to simulate an xfert, I thought that the PlayDTMF manager command will be the right one. All seems to be perfect : - I have the sound associated to the simulated key - I have the message : DTMF successfully queued But, nothing happen (no xfert, no voice). Did somebody use this command before, or have an idea ? P.S. : - asterisk 1.4.6 (make, make install) - tests performed with agents or standard communications (Peer to Peer) _ Personnalisez votre Messenger avec Live.com http://www.windowslive.fr/livecom/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk, AudioCodes, Caller ID
Hello all, I'm working on a little project right now and have ran into a snag. Was hoping someone would be kind enough to give me a few pointers to help me get past the current issue... I have an AudioCodes MediaPack MP-114 (2FXS and 2FXO... SIP firmware...) that I'm trying to get to play nice with Asterisk 1.4. I've got it to the point where the AudioCodes box picks up calls coming in on the FXO ports and routes them to a predefined extension on the Asterisk server. The problem I am having is that I cannot seem to get the AudioCodes box to pass the Caller ID data to Asterisk. I have tested the setup with both my Teltone TLS-5C line simulator and the local telco's POTS lines... Both the teltone and the telco are passing caller ID data onto the line and it is being displayed properly on a standalone CID display box that I hooked up for testing. Yet, that data seems to disappear somewhere in the MP-114... As far as I can tell, I have the AudioCodes box setup to accept caller ID... (Enable Caller ID = yes, type = Bellcore). The AudioCodes documentation is somewhat lack-luster when it comes to real examples, but I did my best to interpret all the various settings throughout the box that seem to affect Caller ID presentation... Does anyone have any experience with getting an AudioCodes MediaPack to work nicely with Asterisk? If so, some advice or sample config snips or anything really would be very helpful... I'd appreciate it... Thanks in advance, Brad ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting Appearance on Outbound Calls?
Hi Matt - Right... you dial *67 to block, however WE are the phone provider and need to set the appearance value so that when our customers dial *67 we correctly block their caller-id from going out. Have you tried explicitly setting the CID variables to NULL strings? - Noah ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [EUQR] Re: North American voice BRI - Informal survey
Jason Aarons (US) wrote: Since many CLECs (Competitve Local Exchange Carriers in NA) offer fractional PRI, combined with Internet/Data, I haven't seen any demand for ISDN BRIs for voice or data since early 90s. I don't know what it's like in your area, but here, fractional PRI is just not cost competitive if you need fewer than 10 channels. -Stephen- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] G722 and Polycom 550
Has anyone found a way to enable the g722 codec as a prefered codec in the Polycom provisioning files for the 550's? I couldn't find a pref for voice.codecPref.IP_550. What needs to be put into the allow field (sip.conf) for asterisk to allow the codec? -- *** Forrest Beck IAXTEL: 17002871718 [EMAIL PROTECTED] http://www.shift8.biz ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting Appearance on Outbound Calls?
No I haven't. Shouldn't I be able to set the appearance to like '4' or '5', etc? On 7/10/07, Noah Miller [EMAIL PROTECTED] wrote: Hi Matt - Right... you dial *67 to block, however WE are the phone provider and need to set the appearance value so that when our customers dial *67 we correctly block their caller-id from going out. Have you tried explicitly setting the CID variables to NULL strings? - Noah ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.2.21.1 and 1.4.7.1 released
The Asterisk development team has released Asterisk version 1.2.21.1 and 1.4.7.1. These releases are minor updates to the releases that were made yesterday to fix a couple of introduced issues. One issue was related to the ODBC realtime driver. Another was related to music on hold. Thank you for your support! ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] North American voice BRI - Informal survey
Quoting Benny Amorsen [EMAIL PROTECTED]: SB == Stephen Bosch [EMAIL PROTECTED] writes: SB Hi, folks: I remain intrigued by the gap in BRI implementation SB between North America and Europe, and I wanted to get feedback SB from the list members on the matter. I'm seriously considering SB making the leap in our office. BRI is being phased out in some parts of Europe. Try ordering a new BRI line from Telia in Sweden... I wish here in Canada it was more widely used, and configuration info and what hardware works was well known. Cost difference is not really there at all when you compare the actual feature packages vs analog lines. The cost is the unknown of getting it all installed and then will it actually all work as intended. 10years ago we had a ton of bri's for our isp business and it would have been no trouble to experiment at that time, but today is a different story, we dumped that part of the business when adsl took over. you would think the telcos would be more interested in selling this to small/medium businesses that are not ready for a voice pri but it seems to be a forgotten technology. (bell canada calls it microlink if you ever need to order it, and be prepared to have to argue that yes it does actually exist) last time I checked it was $88/month for 2B+D which is actually cheaper than 2 business lines with the same feature set. /Benny ___ --Bandwidth and Colocation Provided by http://www.api-digital.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 http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Odd AGI Issue - STREAM FILE, GET DATA not playing file
Apologies if this has been brought up before, but extensive googling and digging through my list archive didn't turn anything up. Basically, I'm working on an AGI web app and need to read some digit input. I'm having multiple issues with asterisk interpreting agi commands at the moment, but I figured I'd start with this one. when I call GET DATA or STREAM FILE I don't get any audio, and timeouts aren't respected. The command is received, the console says it's playing, then the script goes straight to the next line, nothing actually gets done. For reference, here's the relevant perl snippet (I know I'm calling a file of silence, but this occurs with other files as well) print exec swift \Caller I D not recognized, please enter your pin coade followed by the pound key\ \n; print GET DATA silence1 \n; print exec swift \Thank You\\n; Console Dump: AGI Tx agi_request: webrelay.agi AGI Tx agi_channel: SIP/207.103.129.146-08243870 AGI Tx agi_language: en AGI Tx agi_type: SIP AGI Tx agi_uniqueid: 1184095513.2 AGI Tx agi_callerid: 2152592100 AGI Tx agi_calleridname: unknown AGI Tx agi_callingpres: 0 AGI Tx agi_callingani2: 0 AGI Tx agi_callington: 0 AGI Tx agi_callingtns: 0 AGI Tx agi_dnid: 2157011063 AGI Tx agi_rdnis: unknown AGI Tx agi_context: default AGI Tx agi_extension: 2157011063 AGI Tx agi_priority: 2 AGI Tx agi_enhanced: 0.0 AGI Tx agi_accountcode: AGI Tx AGI Rx exec swift Caller I D not recognized, please enter your pin coade followed by the pound key -- AGI Script Executing Application: (swift) Options: (Caller I D not recognized, please enter your pin coade followed by the pound key) [Jul 10 15:25:17] DEBUG[19758]: app_swift.c:168 swift_cb: got END callback; done generating audio AGI Tx 200 result=0 AGI Rx GET DATA silence1 5000 8 -- SIP/207.103.129.146-08243870 Playing 'silence1' (language 'en') AGI Rx exec swift Thank You -- AGI Script Executing Application: (swift) Options: (Thank You) At no point does it even provide time to enter digits (silence1.gsm is 10sec long). It goes immediatey from receipt of the GET DATA command to executing the next command, not even a return value. Ha anyone seen this before? What can I do to get this working? Thanks, Wayne ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue Status
Lee Jenkins wrote: Arun Kumar wrote: Hi I already tried asterisk manager but Im not able to get status for each queue member. thanks That must be a problem with your configuration. I get QueueMemberStatus on my AMI interface (1.2): Event: QueueMemberStatus Privilege: agent,all Queue: support Location: SIP/112 Membership: dynamic Penalty: 0 CallsTaken: 2 LastCall: 1184016974 Status: 1 Paused: 0 If you want all queue information in realtime I wrote a patch for app_queue.c that sends an event to the AMI everywhere it currently writes a queue log entry. The patch is for 1.2.17, will be making an 1.2.20 version soon but with 24 production softswitches one doesn't rollout a new version without extensive testing. Anthony ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Macro Goofiness
I am trying to use a macro to screen calls by calling several different
phones at the same time in different groups. Find me will not work and
queues will not work either. Trust me, I've tried them both and they
don't work like they should. Here is what I have:
A call comes into 6084 and does the following (in default context):
exten = 6084,1,NoOp(test)
exten = 6084,2,SetMusicOnHold(default)
exten = 6084,3,Dial(LOCAL/office,40,m)
exten = 6084,4,Voicemail(6084|u)
exten = 6084,5,Hangup
That calls the following, also in default context:
exten = office,1,Dial(SIP/6080,30,M(screen))
exten = office,2,Hangup
That calls the screening macro:
[macro-screen]
exten = s,1,Wait(1)
exten = s,2,Background(testmessage)
exten = s,3,WaitExten(5)
exten = s,4,NoOp(${MACRO_RESULT})
exten = h,1,Set(MACRO_RESULT=CONTINUE)
exten = h,2,NoOp(${MACRO_RESULT})
exten = i,1,Set(MACRO_RESULT=CONTINUE)
exten = i,2,NoOp(${MACRO_RESULT})
exten = t,1,Set(MACRO_RESULT=CONTINUE)
exten = t,2,NoOp(${MACRO_RESULT})
exten = 1,1,NoOp(Caller accepted)
exten = 2,1,NoOp(Pressed 2)
A call comes in, plays music on hold and calls the 6080 phone. It plays
the message and waits for a key. According to everything I've read, if
I do anything but hit 1, it should fall through and NOT bridge the
call, which should make the call go to voicemail. What happens is that
no matter what I do, it does bridge the call. If I hit 1, 2, 7 or
just wait, the call is bridged together. I've seen several samples that
are the same as this and say it should work. I can't figure it out.
I've tried it on 1.4.7 and 1.4.5 and both have the same issue.
Any ideas? If I hit a key during the WaitExten, I do NOT see the NoOp
in line 4. It's like anything just pauses execution and bridges the
call. Thanks for any tips (except people that tell me to use find
me/follow me, that won't work).
Peder
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom multiple registrations
Noah Miller wrote: The 430's have two line appearances. I'm trying to get the second line registered to a different extension but for some reason it's not allowing me to do this. The first line will register fine but the second line never seems to register no matter how I swap the device ID's and permissions around. I've played with the lineKeys and callsperlinekey settings to no avail. For what you want to do, you'll have to set lineKeys to 1 for both of your registrations. callsPerLineKey can be anything from 1 to up to (I think) 6, your preference. Can you share the reg ... / statement from your phone.cfg file? Also your sip.conf? - Noah ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Noah, Thanks for your reply. It was just me not being careful. When I looked at the reg.x.server.y.address setting I noticed I had server.2 instead of server.1. Thanks for your help. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] mixing video and non-video clients
It's not clear to me what happens when video and non-video clients are using the same PBX together. Let's say, from a video softphone I place a call through Asterisk, through the analog phone line and my phone provider, to a regular analog phone. Or the other way round - from an analog phone, through Asterisk, to a video softphone. What happens then? I imagine that only the audio channel is routed, and the video screen goes blank on the softphone, but the audio communication works fine. Is that correct? -- Florin Andrei http://florin.myip.org/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [EUQR] Re: North American voice BRI - Informal survey
On 7/10/07, Stephen Bosch [EMAIL PROTECTED] wrote: Jason Aarons (US) wrote: Since many CLECs (Competitve Local Exchange Carriers in NA) offer fractional PRI, combined with Internet/Data, I haven't seen any demand for ISDN BRIs for voice or data since early 90s. I don't know what it's like in your area, but here, fractional PRI is just not cost competitive if you need fewer than 10 channels. -Stephen- That's because, as Jason says, it's often offered with internet in the US. In Canada we see those offered as two discrete services. It adds a bit more money to the pot in terms of the economics of displacing analog. Dave ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting Appearance on Outbound Calls?
Hi Matt - No I haven't. Shouldn't I be able to set the appearance to like '4' or '5', etc? I can do that on the PRI's I've had experience with. I found that on most landlines, this will show up as Unavailable or something similar, but on most cell phones it will show that number. - Noah ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] video calls - Windows / Linux interoperability ?
I will install Asterisk on my home server, I want to be able to route video calls, but I need the Windows and Linux clients to be interoperable. On Linux, it looks like Ekiga is a good candidate. But how about Windows? Anyone using Kapanga in an Asterisk network that includes Ekiga? Are these two interoperable? I'm not necessarily looking for open source software, free for personal use is enough. -- Florin Andrei http://florin.myip.org/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting Appearance on Outbound Calls?
Noah Miller wrote: Hi Matt - No I haven't. Shouldn't I be able to set the appearance to like '4' or '5', etc? I can do that on the PRI's I've had experience with. I found that on most landlines, this will show up as Unavailable or something similar, but on most cell phones it will show that number. pbx-1*CLI show application setcallerpres pbx-1*CLI -= Info about application 'SetCallerPres' =- [Synopsis] Set CallerID Presentation [Description] SetCallerPres(presentation): Set Caller*ID presentation on a call. Valid presentations are: allowed_not_screened: Presentation Allowed, Not Screened allowed_passed_screen : Presentation Allowed, Passed Screen allowed_failed_screen : Presentation Allowed, Failed Screen allowed : Presentation Allowed, Network Number prohib_not_screened : Presentation Prohibited, Not Screened prohib_passed_screen: Presentation Prohibited, Passed Screen prohib_failed_screen: Presentation Prohibited, Failed Screen prohib : Presentation Prohibited, Network Number unavailable : Number Unavailable pbx-1*CLI ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Monitor events?
Hey Daniel, I think adding the events would be a good idea. Just open an issue on http://bugs.digium.com/ and attach your patch there. Be sure to send a disclaimer to digium so your patch can be included in the distribution (see http://asterisk.org/developers/bug-guidelines for details). actually you probably know i am using your java-asterisk :) and of course if you already have patches for Asterisk-Java that support your new events post it to http://jira.reucon.org referencing the digium issue. =Stefan -- reuter network consulting Neusser Str. 110 50760 Koeln Germany Telefon: +49 221 1305699-0 Telefax: +49 221 1305699-90 E-Mail: [EMAIL PROTECTED] Jabber: [EMAIL PROTECTED] Steuernummern 215/5140/1791 USt-IdNr. DE220701760 signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call queues
Floyd wrote:
Hi everyone:
I've searching for a while and haven't found what i
need.
The thing is that i have a tdm422p with the two fxo
ports connected to the pstn. I want my sip users to
be
able to call other numbers(any number) in the pstn
through my zap fxo channels. I have a big number of
sip users so as you can imagine there will be
congestion when some of them(more than two!!) want
to
call outside, that is why i want to be able to put
those outgoing calls in a queue. For example if i
want
to call someone in the pstn and the fxo port is
already in use, i want to be placed in a queue and
when the channel is free my call is routed to the
aproppiated destination. As far as i have read the
queues are not for this kind of stuffs, there are
just agents or extensions that attend the calls in
the
queue and nothing more. am i wrong???
Any help will be useful.
thanks in advance!!
Hi,
first of all i would like to thanks to C. Chad Wallace
Noah and Rob Schall. I just solve the problem of the
outgoing call queue. Following the instructions from
Chad
i did something like this in my extensions.conf:
exten = _9XXX,1,Answer
exten = _9XXX,2,Set(_number=${EXTEN:1})
exten = _9XXX,3,Wait(2)
exten = _9XXX,4,NoOP(${number})
exten = _9XXX,5,Queue(Myqueue)
exten = _9XXX,6,Hangup
I also have a context like this
[outbound]
exten = 1,1,NoOP(${number})
exten = 1,2,Dial(Zap/G2/${number},30,t)
exten = 2,1,NoOP(${number})
exten = 2,2,Dial(Zap/G2/${number},30,t)
Finally in queues.conf
[myqueue]
member = Local/[EMAIL PROTECTED]/n
member = Local/[EMAIL PROTECTED]/n
And it worked perfect!!
I have my outgoing calls routed ok and the variable
travells throgh the queue without problems..
thanks!!
Eve
__
Correo Yahoo!
Espacio para todos tus mensajes, antivirus y antispam ¡gratis!
Regístrate ya - http://correo.espanol.yahoo.com/
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] extensions reload -- what impact?
I am putting some scripts together to allow a local admin to add extensions, then to reload the extensions, something like: asterisk -r -x extensions reload Are registered extensions forced to reauth? Are active calls disrupted? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] video calls - Windows / Linux interoperability ?
For windows you can't really go wrong with X-lite 3.0, the free version of eye-beam. http://www.counterpath.com/index.php?menu=Productssmenu=xlite On 7/10/07, Florin Andrei [EMAIL PROTECTED] wrote: I will install Asterisk on my home server, I want to be able to route video calls, but I need the Windows and Linux clients to be interoperable. On Linux, it looks like Ekiga is a good candidate. But how about Windows? Anyone using Kapanga in an Asterisk network that includes Ekiga? Are these two interoperable? I'm not necessarily looking for open source software, free for personal use is enough. -- Florin Andrei http://florin.myip.org/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Edit ulaw file
Garry, Google around. I use wavepad. It works for me. Thank you 1. Edit ulaw file (Gary Chen) -- Message: 1 Date: Tue, 10 Jul 2007 10:24:58 -0400 From: Gary Chen [EMAIL PROTECTED] Subject: [asterisk-users] Edit ulaw file To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 I recorded some sound files using Asterisk record() app as ulaw file. I need to edit these sound files. What kind of audio editor can I use to edit these files? Gary Chen -- next part -- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070710/8320ad 04/attachment-0001.htm -- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] video calls - Windows / Linux interoperability ?
Hi Florin - On Linux, it looks like Ekiga is a good candidate. But how about Windows? There's a windows version of Ekiga, too. It's at the very bottom of the download page. - Noah ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] extensions reload -- what impact?
Hi Robert - I am putting some scripts together to allow a local admin to add extensions, then to reload the extensions, something like: asterisk -r -x extensions reload Are registered extensions forced to reauth? Nope. Are active calls disrupted? Nope. Reloads are safe to do in the middle of any normal asterisk operation. - Noah ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Looking for a USB phone handset or headset
Hello, I would like people to use soft phone, but they are used to have the standard phone handset in their hands... Is there a USB handset or a handset that connects to the audio card? Thanks! __Yehavi: ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looking for a USB phone handset or headset
From: Yehavi Bourvine +972-8-9489444 [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Date: Wed, 11 Jul 2007 7:37 +0300 Hello, I would like people to use soft phone, but they are used to have the standard phone handset in their hands... Is there a USB handset or a handset that connects to the audio card? Thanks! __Yehavi: ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Google is your friend http://www.google.com/search?hl=enq=usb+handsetbtnG=Google+Search Sent via the WebMail system at rockynet.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looking for a USB phone handset or headset
Hi Yehavi You should consider USB Phone just as any sound device that is added to your systems. Once the configuration has been done, it should appear as audio device. Then you can configure your soft phone (ie Idefisk or any) to use the new audio device as input, output and may be ringing device. But don't forget that it would always be the softphone which controls the communications. Hope this would help Regards -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Anthony Francis Envoyé : mercredi 11 juillet 2007 05:43 À : Objet : Re: [asterisk-users] Looking for a USB phone handset or headset From: Yehavi Bourvine +972-8-9489444 [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Date: Wed, 11 Jul 2007 7:37 +0300 Hello, I would like people to use soft phone, but they are used to have the standard phone handset in their hands... Is there a USB handset or a handset that connects to the audio card? Thanks! __Yehavi: ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Google is your friend http://www.google.com/search?hl=enq=usb+handsetbtnG=Google+Search Sent via the WebMail system at rockynet.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Information NOD32 2301 (20070531) __ Ce message a ete verifie par NOD32 Antivirus System. http://www.nod32.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
