Re: [Asterisk-Users] Why Asterisk can't cope with silence suppression?

Peter Svensson wrote: On Wed, 16 Feb 2005, Rob Scott wrote: Why is it that Asterisk can't cope with silence suppression? All the clients seem to be able to but not Asterisk. What would be needed to get it to work with silence suppression? What is the problem? Asterisk clocks outgoing rtp data to

Re: [Asterisk-Users] chan_sip errors on CVS HEAD

Asterisk wrote: I've got a test * server (hppbx) where I install CVS-HEAD as often as possible, with my extension registered to this, talking through IAX to our production server which then channels out to the PSTN. After completing a call just now, the following appeared on the CLI of hppbx (t

[Asterisk-Users] Asterisk Users in Madrid?

I'm sitting in a hotel close to the Madrid airport... Any Asterisk users in the neighbourhood that wants to meet me for a beer and some Asterisk hacking this evening? Send e-mail to me *off list*, thank you. /O ___ Asterisk-Users mailing list Asterisk-

Re: [Asterisk-Users] SIP proxies & Asterisk ?

Vlasis Hatzistavrou wrote: Hello, We hve been trying to make Asterisk work with SIP proxies with no success. Is there support for SIP proxies in Asterisk in the latest versions? A lot of people use Asterisk with SIP proxys. What is your problem, give us a bit more information. /Olle ___

Re: [Asterisk-Users] breaking friends into users & peers

Andrew Thompson wrote: I am about to start a program that will be generaging sip device configurations for sip.conf. My current sip.conf contains friend entries for each SIP device connected to asterisk. Should I even be attempting to split these in to seperate user/peer devices? Can two entr

Re: [Asterisk-Users] SRV lookups

Robert Spielmann wrote: Hi everyone, I have a question concerning DNS SRV lookups. The situation is like this: - one central Asterisk server - many domains with SRV records, let's say we have bar.com and doe.com Now the question is: if the SRV lookup is done for [EMAIL PROTECTED] the call is mappe

Re: [Asterisk-Users] Sipura SPA-841 auto-answer support [patch]

Geoff Speicher wrote: Sipura has implemented auto-answer in version 0.9.5 of the SPA-841 firmware. However, it is implemented via the Call-Info header, which Asterisk stable doesn't currently support. The attached patch implments a quick hack to support the Call-Info header from the Dial() applica

[Asterisk-Users] * INSTRUCTIONS FOR THE ASTERISK COMMUNITY - PLEASE READ NOW *

Welcome to the Asterisk users community! Asterisk.org is a fast moving project. New code is added every day. Asterisk is the leading Open Source Telephony platform, with support both for classical telephony and IP telephony. Our community is also growing fas

[Asterisk-Users] FastAGi change

Mark just committed a small fix of mine to FastAGI. Previously there was a script option to the URI that wasnt't used. Now, it's sent to the AGI server so that one running server can handle multiple AGI functions. agi://hostname:port/script is the full syntax for the fastagi option to th

Re: [Asterisk-Users] MeetMe

Serge Schumacher wrote: Can it be that the MeetMe application is not installed by default even if there is a meetme.conf ? pbx.c:1280 pbx_extension_helper: No application 'MeetMe' for extension (from-sip, 550, 4) It is not installed if you haven't got a Zaptel timer. See the Wiki docs on

[Asterisk-Users] * INSTRUCTIONS FOR THE ASTERISK COMMUNITY - PLEASE READ NOW *

Welcome to the Asterisk users community! Asterisk.org is a fast moving project. New code is added every day. Asterisk is the leading Open Source Telephony platform, with support both for classical telephony and IP telephony. Our community is also growing fas

Re: [Asterisk-Users] How to connect two Asterisks as secureaspossible without too much additional bandwidth ?

Brian West wrote: OpenVPN What happened to AES in IAX2? /O ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/li

[Asterisk-Users] SIP Multicast Support desperately needed :: Mission critical bug in Asterisk

Friends! I have recently discovered that chan_sip, chan_sip2 and chan_sipx all lack support of SIP multicast. This has a major impact on my network, since I haven't got the bandwidth needed to call all of you and send you this message. With that feature missing, I have to go back to old Interne

Re: [Asterisk-Users] Incoming calls from Sipgate go through the wrong peer

For incoming calls, Asterisk matches peer's on IP, meaning that the first peer it finds will match. This is the *last* one you have in sip.conf. The context given in that peer must have *all* extensions you need for incoming calls, which is the extension at the end of the register= line in the

Re: [Asterisk-Users] hint extension and Snom phones - CVS or stable?

Karl Brose wrote: There is no such thing as "subscribecontext" parameter in SIP. I have updated the wiki with the correct current information to make this work. http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+snom The subscribecontext is part of chan_sip2, not in standard Asterisk ch

[Asterisk-Users] chan_sip errors in CVS stable

*** SIP Channel fixed in CVS stable --- During a few days there's been a buggy SIP channel in CVS STABLE, but not in the 1.0.3 release tarballs on the FTP server and mirrors. We have now removed the patch that was integrated by mistake so CVS should be ok again.

Re: [Asterisk-Users] Cannot transfer with Cisco or Snom

Alexander Lopez wrote: OK with CVS-HEAD-12/22/04-09:47:32 incoming (to ata) sip calls on a ATA0186 now seam to work fine. However transfers still do not work. With CVS-HEAD-12/22/04-12:46:47 transfers still do not work. You need to download the fix from the bugtracker until Markster approve and c

Re: [Asterisk-Users] How to generate a SIP NOTIFY for Cisco 7960 remote reboot?

Mick Hastings wrote: Hi Folks, cheers for all the great info on the list. I need to create a SIP NOTIFY message to reboot my Cisco 7960 phones but I dont know how. The admin guide gives an example of the packet (attached), I have tried a few web searches and found some cool little programs that g

Re: [Asterisk-Users] SIP registrations not staying registered

Doug Reid - Stormcorp wrote: HI I got the same problem that only started lately. I have to do a stop start to get the phones registered again. One site out of 12 with the same spec. Running CVS head or a stable release? Please show us SIP debug on first registration and then failed re-regist

Re: [Asterisk-Users] Confirm MWI doesnt work with SIP RealTime?

Matthew Boehm wrote: Can someone else confirm that your phone does not recieve MWIs when using SIP and RealTime? Yes, confirmed. Is this a problem with SIP or with Voicemail? With the realtime architecture, really. Realtime Peers (or Mysql Friends in 1.0) does not stay in memory, as does the stati

[Asterisk-Users] * INSTRUCTIONS FOR THE ASTERISK COMMUNITY - PLEASE READ NOW *

Welcome to the Asterisk users community! Asterisk.org is a fast moving project. New code is added every day. Asterisk is the leading Open Source Telephony platform, with support both for classical telephony and IP telephony. Our community is also growing fas

Re: [Asterisk-Users] ACK from asterisk not matched to transaction by SER / LCS2005

Public Dump wrote: For reasons unknown to me, SER and subsequently a Microsoft Live Communcations Server 2005 seems to have problems, matching a SIP ACK request from asterisk to the ongoing SIP transaction, I have attached the complete log, but the essential lines are: That's a bug in Asterisk

[Asterisk-Users] Re: [Asterisk-Dev] SIP SECURITY WARNING: v1-0 (cvs today) sip context in general section ignored goes to default instead - allowing unauthorized sip devices to place calls in default context

Andy Reinke wrote: SIP SECURITY WARNING [general] contex=sip-unauthorized If you spell this right, all calls from unknown SIP devices will be sent to the context you set here. If you do not set a context in the general section of sip.conf, "default" will be used. This is the way you configure how t

[Asterisk-Users] Asterisk Newsletter :: Back online!

Time to reboot and re-start Asterisk, well, hrrm, monthly, news. It's been a hectic fall with a lot to do, both before and after Astricon. At this time, we're preparing for two Astricon shows in 2005. And no, we haven't made a decision on where to run the European Astricon, not yet. I am preparing

Re: [Asterisk-Users] make asterisk accept Register messages

Reid A. Forrest wrote: I don't know why but my * is not accepting Register Messages. Have you seen this kind of problem before?? I need help! Thank in advance. *CLI> Nov 19 15:42:11 NOTICE[12893]: chan_sip.c:4869 register_verify: Peer '1

[Asterisk-Users] Broadvoice update

http://edvina.net/broadvoice/ We now have an update to the patch and new configuration instructions. Apply the patch to a an unpatched copy of chan_sip.c, recompile and reconfigure. Note that if you followed the earlier instructions on adding a host entry to /etc/hosts, you will have to remove that

Re: [Asterisk-Users] SIP register problem

Karl Brose wrote: Is the SIPquest server sending the 401 Unauthorized message verbatim as you printed it here? I.e. is the WWW-Authentcate header broken up into several lines like that? If so, how man spaces are actually at the beginning of each new line? Continuation lines are allowed in SIP, but

Re: [Asterisk-Users] _ALERT_INFO (new subject)

Brian West wrote: Ok to cut confusion here Its: Variable: _ALERT_INFO In CVS head. In stable, it's still ALERT_INFO. The same applies to VMXL_URL that is now _VXML_URL in CVS head. This is due to a change in app_dial where you now are able to set any variable in the new call leg created by dial() b

Re: [Asterisk-Users] Asterisk using the wrong peer in sip.conf

First, please do not use "friends". It will confuse you. We receive calls from type=user and send calls to type=peer (that is the general idea, anyway... :-) For an incoming call, we first match on users based on the username part of the From: sip address. So if the call comes from sip:[EMAIL PRO

Re: [Asterisk-Users] Broadvoice asterisk patch

Dinesh Nair wrote: would this patch help those who're not using broadvoice, i.e. does it fix an issue with the way asterisk does not handle SIP registrations correctly ? No. The actual registration is still handled the same way. What problems do you have, when does Asterisk not handle registrat

[Asterisk-Users] The BV patch: Some notes

Some notes on the Broadvoice patch * It makes your asterisk send less packets to Broadvoice by re-using the authentication when re-registering * It, by mistake, adds extra logging that is simple to remove from your chan_sip.c. This was added to help debug the code, but should not have been in

Re: [Asterisk-Users] Sip clients not longer registering

David Filion wrote: Hi, We have been using Asterisk since version 0.9x with little or no problems. However, for an unknow reasons, our sip clients can nolonger register. We updated to Asterisk 1.0.2 hoping that would solve the problem, but no luck. There is a chance that our change NAT logic

Re: [Asterisk-Users] Sip clients not longer registering

David Filion wrote: Hi, We have been using Asterisk since version 0.9x with little or no problems. However, for an unknow reasons, our sip clients can nolonger register. We updated to Asterisk 1.0.2 hoping that would solve the problem, but no luck. Seems to me that the HT486 never receives th

Re: [Asterisk-Users] chan_sip CallerPres support?

Race Vanderdecken wrote: Roy et All, If someone could expand on CallerPres requirements in chan_sip I can do the work. I have added numerous extras to chan_sip already, RADIUS, new CDRs, Dynamic Dial plans, Find-Me, Follow-Me and such. Do you implement dial plan features in chan_sip or do

Re: [Asterisk-Users] chan_sip changes affecting ACK? - Bug?

Chad, I need a more complete SIP debug than just one packet to try to look into this issue. If the device registers, both a REGISTER transaction and a subsequent call with the ACK - THank you! /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://li

Re: [Asterisk-Users] asterisk & ipv6

Miroslav Nachev wrote: Dear Olle, I can say that Emil Ivov has very good knowledge on IPv6 too. You can use it. Great - the more IPv6 experts that can help us with coding advice, code review and patches - the better! /O ___ Asterisk-Users mailin

Re: [Asterisk-Users] asterisk & ipv6

Marc Blanchet wrote: - no asterisk does not work over ipv6. - ipv6 port won't be as easy as I would like to... - I'm currently working on it. Had a short discussion with Mark during astricon on it. - no release date promised... - will certainly be available through a different source tree. Too man

Re: [Asterisk-Users] Running as non-root user ( was: Vmail.cgi Bahhh!!)

Justin wrote: Olle, That's a great start but as the documentation states: NOTE: this requires substantial work to be sure that Asterisk's environment has permission to write the files required for its operation, including logs, its comm socket, the asterisk database, etc. Can that be made ea

Re: [Asterisk-Users] Running as non-root user ( was: Vmail.cgi Bahhh!!)

Justin wrote: It is great that this documentation is out there, and that * supports this. However I think in an ideal world this would be inherently supported by * and ideally setup via config file like with apache: User www Group www From the Asterisk man page: asterisk [ -hfdvVqpRgcin ] [ -

Re: [Asterisk-Users] Attempt at country tones

Garry Taylor wrote: 2. How to get my country tones included into zonedata.c, who would I send them to for inclusion? Open a bug report and add a patch to the bug tracker, http://bugs.digium.com /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://l

Re: [Asterisk-Users] DUNDi in stable? (New subject)

Ryan Courtnage wrote: http://www.dundi.com Yet, another great idea!! Thanks Mark!! I wish it was in v1.0, but I guess I'll have to update to head. I wish it were in v1.0 as well. Would creating a patch for 1.0 be pretty simple, or do the code changes run deep? It will eventuall get into a release

Re: [Asterisk-Users] res_odbc app_realtime

Race Vanderdecken wrote: Hmmm, I have been working on similar. I thought that while some of the SIP information is in the database there are still parameters Asterisk is looking for in the sip.conf. ...but with res_config, sip.conf may be loaded from a database. Do not mix res_config database sta

Re: [Asterisk-Users] New Realtime config and MWI

Karl H. Putz wrote: Currently setting mailbox= in sip.conf along with appropriate additional info is required to set and clear MWI for sip clients. MySQL peer table does not include the mailbox variable and, while ast_data does include the mailbox variable, the polling architecture of chan_sip does

Re: [Asterisk-Users] Getting Call-ID w/o CDR platform

Darren Sessions wrote: Call-ID as in SIP Call-ID *not* Caller ID. In chan_sip2: ${SIPCALLID} Very useful, indeed. And looking at the chan_sip source code, I've obviously ported it to standard Asterisk as well... :-) /Olle ___ Asterisk-Users mailing list

Re: [Asterisk-Users] Presentations

Wojciech Tryc wrote: At AstriCON Steven Sokol said that copy of the presentations should be available on-line within 2 weeks. Did anyone got their user name and password to access them? No. Did we get all the presentations from the speakers? NO! We are reminding them and hope to have something r

Re: [Asterisk-Users] Sip Outbound Proxy

Chad Brown wrote: How do you configure an outbound proxy for Asterisk? If the sip call is not local I want everything to go to a designated sip proxy. In the standard chan_sip, there's no support for outbound proxy. In my chan_sip2 test channel, I have that support. Please test! If I get enough po

[Asterisk-Users] Astricon tutorials :: Open for registration again

We're now opening up registrations for the Astricon tutorials again. We've been able to move to new conference rooms within the same hotel. Register on line at http://www.astricon.net We're sorry for the inconvienience our recent closing of the tutorials may have caused you. You are welcome to con

[Asterisk-Users] Astricon News :: Tutorials are now fully booked

*** Astricon 2004: Over 250 Asterisk professionals! Astricon, the first Asterisk user's and developer's conference is a success and we now have over 250 people registred. Thank you for all your support of this event and please have patience with us as we're trying to handle all details with payment

Re: [Asterisk-Users] WellGate 3504A with Asterisk SIP authentication and config

Dinesh Nair wrote: Proxy-Authenticate: Digest realm="asterisk", nonce="514a6d7a" Don't forget to change your realm to something that is unique for your setup... realm= in sip.conf :-) Regards, /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http:/

Re: [Asterisk-Users] WellGate 3504A with Asterisk SIP authentication and config

Dinesh Nair wrote: hey * folk, am trying to configure a WellGate 3504A FXS SIP ATA (http://www.welltech.com.tw/products_ea01.htm) with asterisk. i've set up two SIP clients in sip.conf as follows: The "Unathorized" really suggests that the password is wrong for 1235. Check that you have the same

Re: [Asterisk-Users] SIP authentication problem

Kurt Bauer wrote: Hi, I have the following setup: E100P SER <> * <-> PBX This works just fine, except when there are users on both boxes (ie. SER and asterisk), whose usernames are the same, although the realm is different. At this point, Asterisk doesn't care about the re

[Asterisk-Users] ** ASTRICON * LAST CALL FOR REGISTRATION

er limit in number of attendees. Make sure you register now to get a hotel room and an entrance ticket. http://www.astricon.net Thank you for all your support! Steven Sokol & Olle E. Johansson Astricon Organizers PS: And if we haven't been as a

Re: [Asterisk-Users] Asterisk PBX Functions via SIP phone

Chris Shaw wrote: I am suprised that one would have to create a dialplan since its an already built in function that works with regular POTS phones. Or is it because of the way DTMF is sent via SIP? I don't know digium's long range plans, but looking through chan_sip.c NONE of the vertical service

Re: [Asterisk-Users] Re: Searchable Archive

Muiz Motani wrote: Google search does not work very well. For example, a search of the keywords "opencall.org" and "asterisk-users" on Google turned up nothing useful. http://search.voip-forum.com/cgi-bin/htsearch?words=opencall.org+asterisk-users&config=voipsearch /O

[Asterisk-Users] Mpg123 clarification

This was recently added as README.mp3: * Asterisk MP3 Support == Asterisk supports mp3 playback for music on hold via the mpg123 program, available from www.mpg123.de. The latest release of mpg123 is mpg123 0.59r. The latest development release of mpg123 is mpg123 pre0.59s. Plea

[Asterisk-Users] List-etiquette * AGAIN *

* Please DO NOT post the same message to two lists. We have divided the lists to be able to stay focused and lesser the burden. You are not raising the chances of getting a reply, you are instead annoying a lot of people. Most of the people on the -dev list are reading all other lists. * Please mov

Re: [Asterisk-Users] DID Terminations

Andrew Yager wrote: Hi, Since DID's are a topic of conversation at the moment... (and I'm in the market..) Please take this kind of business questions to the asterisk-biz list. /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/

Re: [Asterisk-Users] CVS Number after Update

Martin Keding wrote: Whenever I do a "Show version", I get CVS-02/10/04. I have updated and recompiled Asterisk a number of times and it comes up clean. Should'nt this show the lastest version number? I have been using CVS download, then "make clean;make install". I have also tried "make clean; cvs

[Asterisk-Users] CPC on Zaptel

A newbie on Zaptel asks: Is there any way I can force CPC on a Zap channel - Digium TDM four port FXS card? /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit

Re: [Asterisk-Users] Formatting in sip.conf...can you have 2 @ signs for register?

Chris A. Icide wrote: On 07:12 PM 8/16/2004, John Todd wrote: > stupid top-posting confuses the hell out of these threads, >but I'll continue the insanity. > >I _swear_ I already brought this problem up and it got resolved, like >1.5 years ago. I explicitly remember talking with kram about a

Re: [Asterisk-Users] Formatting in sip.conf...can you have 2 @ signs for register?

Greg Hill wrote: On Mon, 16 Aug 2004, Olle E. Johansson wrote: James Freire wrote: Hi All, I am trying to setup another sip trunk in addition to what I am already using. The sip provider we are using right now gives you your username as your email address. So IE. If my email is [EMAIL PROTECTED

Re: [Asterisk-Users] Formatting in sip.conf...can you have 2 @ signs for register?

James Freire wrote: Hi All, I am trying to setup another sip trunk in addition to what I am already using. The sip provider we are using right now gives you your username as your email address. So IE. If my email is [EMAIL PROTECTED] that is my username . Now... When I put this in the sip.conf

Re: [Asterisk-Users] world incoming calls

Sergey Lapin wrote: Hi, all!!! We have one big Asterisk server, supporting LAN calls inside and gatewaying calls to voipexchange.ru for real phone calls. Now we need to recieve world calls. Start with reading the latest issue of my Asterisk Newsletter, http://www.asterisknews.com /Olle :-)

[Asterisk-Users] * INSTRUCTIONS FOR THE ASTERISK COMMUNITY - PLEASE READ NOW *

Welcome to the Asterisk users community! Asterisk.org is a fast moving project. New code is added every day. Asterisk is the leading Open Source Telephony platform, with support both for classical telephony and IP telephony. Our community is also growing fas

Re: [Asterisk-Users] Asterisk MIBS

Alagalah wrote: Hi, I was wondering if there are any Asterisk MIBS (specifically regarding call information) ? I noticed a post citing ___www.faino.org_ , but this site doesn’t seem to exist anymore, and The Book v2 doesn’t have any references to MIBS. Any pointers greatl

[Asterisk-Users] Do you speak Czech?

Continuing the work done on Internationalization of Asterisk, we've begun working on Czech. If you speak Czech, we need you help in continuing the work that has been started. Roll up your sleaves and visit http://bugs.digium.com/bug_view_page.php?bug_id=0002013 Thank you! /O ___

[Asterisk-Users] *** Asterisk Summer News: Forget numbers, dial by domain!

Welcome to a new issue of Asterisk Summer News! The holiday season is coming to an end here in Sweden, people are getting back to work and the kids will start going to school next week. Life is slowly adopting to normal and I have to start dressing more towards a businessman than a beach bum. Guess

[Asterisk-Users] ::::: Pssst. Rc2! :::::::

If you look into the download areas, you'll find Asterisk 1.0 release candidate two... Find the mirrors on the link below, they'll update during the day if they don't already have RC2. Please don't hit the Digium FTP-server, since development need that connection for the bug tracker and the CVS. (

Re: [Asterisk-Users] SNOM 200 and Asterisk Woes

Dan Mahoney, System Admin wrote: You start up the phones, they register, all is good. They show up in sip show peers like thus: danm/danm65.125.237.91D N 255.255.255.255 5060 OK (29 ms) We pass a few calls in and out, and asterisk "deadlocks" (not a true deadlock, see

[Asterisk-Users] Astricon Dev Meeting On Line

Friends, Please send all offers for help *off list* to us at [EMAIL PROTECTED] Do not disturb the list with offers of your services, please. I repeat: Only the Developer's Meeting will be considered for broadcast at this time. In order to enjoy the conference, you will simply have to be there. It's

Re: [Asterisk-Users] Astricon Conference Call?

If possible, we will broadcast the Asterisk Developer's meeting on the friday. Internet connections in conference hotels is a complex and utterly commercial story, where it is easy to reach sales, but not easy to reach someone that has a clue. We don't know what we can do, what the specifics are in

Re: [Asterisk-Users] Asterisk GUIs at Astricon * REMINDER *

Scott Stingel wrote: Hi Olle- I wonder of you could please post the most recent agenda for each day, even if it's not finalized. Some of us can't attend the whole conference, and so need to pick the best days/times to come. (I'm scheduling a trip, and a stop at astricon could be on the way there)

[Asterisk-Users] *** Asterisk Summer News: The heat is on!

Another issue of Asterisk Summer News, delivered right to your mailbox! Back here in Sweden, it's finally summer weather. Sunshine and some heat. It's good for our ice bears and the snow houses to get some sunshine :-) Asterisk development and IRC chat has gone into a lazy summer mode, but the mail

[Asterisk-Users] Asterisk GUIs at Astricon * REMINDER *

I'm working with the final details of the Astricon agenda. I haven't got anything so far on Asterisk GUI's and there are plenty of projects out there. I would like to invite developer's of Asterisk GUI's, both open source and commercial, to participate. What I'm thinking of is giving each GUI a slo

[Asterisk-Users] SIP Outbound Proxy Support

In the latest release of chan_sip2 I've added support for SIP Outbound Proxy. I've seen a lot of requests for that lately, so if you can test this and confirm wheather it works for you or not, I'll be grateful. If I get positive reports, we'll try to add this to chan_sip in CVS. It works like this

Re: [Asterisk-Users] Outgoing works, incoming doesn't...

Evert Meulie wrote: Hi! Problem with my 7960. Outgoing calls work, but incoming don't. A 'sip show peers' gives: Name/usernameHostDyn Nat ACL Mask Port Status 105/105 192.168.2.175D 255.255.255.255 5060 UNREACHABLE Is there something wrong with the

Re: [Asterisk-Users] shan:Needed help

Shanmuganathan Kumaravel wrote: Hi, I'm dialling 1234 in the softphone or Grandstream Phone and without disconnecting the phone > i want to dial "10" after dialling "1234". Is it possible to do? Shan, Welcome to the Asterisk community! The questions you ask indicate that you need to do your ho

Re: [Asterisk-Users] Re: Echo in asterisk phones.

Randy Bush wrote: Also beware if checking it in debug mode (like asterisk -vc) Took me awhile to notice it was going away when I started asterisk normally ! If you add a "d" for debug, like "asterisk -vvdc" you will get even more output. /O _

Re: [Asterisk-Users] Broadvoice problems again

Gregory Youngblood wrote: It appears * doesn't use both IP addresses returned by DNS for sip.broadvoice.com as a failover method. Depending on how * works (I'm still learning, and have been battling bad hardware it turns out...) you may be able to use a work around. This can help in several situati

Re: [Asterisk-Users] VoiceMail Group Broadcasting

Seth Remington wrote: This bug report patch will do what you want with the "cc" and "delete" options to the mailbox. It was supposedly added to CVS but I have not seen it come through yet. Very confusing. I guess you could apply the patch manually. The patch is in the CVS, but if you read bug repor

Re: [Asterisk-Users] Feature question

James Jones wrote: Does asterisk support outbound proxies? No, we do not support outbound SIP proxies ...yet. /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options vis

[Asterisk-Users] Astricon news :: The conference agenda now published

ng to participate in Astricon. If you have any questions regarding Astricon, please mail us on [EMAIL PROTECTED] . See you all in Atlanta! Best regards, Steven Sokol & Olle E. Johansson http://www.astricon.net ___ Asterisk-Users mailing lis

Re: [Asterisk-Users] NAT + iConnectHere Broken in 1.0RC1

There has been a number of changes in how SIP handles outbound registrations. That is registration with Asterisk as a SIP client to another SIP proxy, propably with a service provider. To be able to document a new "how to" I would like those of you that have problems with this mail me a SIP DEBUG *

Re: [Asterisk-Users] Doublehash transfers

John Todd wrote: I hate being a "me too" poster, but the double-hash patch I have implemented four times now, and I know at least three other people who have also gone well out of their way to put that patch into their system. Making this an "official" modification would be ideal, in my opinio

Re: [Asterisk-Users] SIP register and unregister events via Manager API

Holger Schurig wrote: And while I was at this patch, I also changed the Event: SIPRegistry Domain: ... Status: ... to Event: Register Channel: SIP Domain: ... Status: ... I still believe it would be better to call this "Registry" since that's a common term across IAX and SIP for outbound registrat

Re: [Asterisk-Users] SIP register and unregister events via Manager API

Holger Schurig wrote: I think we have several problems here. Once it's "Peer:", the other time it's "Peername". That's clearly a bug. Also, I don't like the name of the event. It should just be "PeerStatus" and "PeerRegistration", because we might add something to IAX2 as well. So I'd suggest

Re: [Asterisk-Users] SIP register and unregister events via Manager API

ith PeerStatus: IAX2/qtiax. I guess I'll redo the path and re-submit it to bugs.digium.com. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: h

Re: [Asterisk-Users] SIP register and unregister events via Manager API

Matthias Endler wrote: As promised yesterday: Anybody interrested can download the patch for Asterisk 0.9.1 at http://matthiasendler.net/asterisk/patch/. Great! Please add it to the bugtracker in a .txt file created with cvs diff -u channels/chan_sip.c The diff has to be for CVS HEAD, that is 1.0r

[Asterisk-Users] * INSTRUCTIONS FOR THE ASTERISK COMMUNITY - PLEASE READ NOW *

Welcome to the Asterisk users community! Asterisk.org is a fast moving project. New code is added every day. Asterisk is the leading Open Source Telephony platform, with support both for classical telephony and IP telephony. Our community is also growing fas

Re: [Asterisk-Users] * CLASS codes

muralikrishnan lakshmanan wrote: Hello friends, I got one page from net "http://www.voip-info.org/wiki-CLASS"; In that page I saw lot of *xx codes for asterisk feautres. I don't know how to use these codes. If anyone used these codes can you teach me. This is just a list of

[Asterisk-Users] *** Asterisk Sun/Monday News: Time to download, Scotty!

This week starts with the exciting news: We're getting close to Asterisk 1.0 again. After the failed attempt earlier this year, we've been able to remove a lot of the MAJOR/CRASH bugs from the bug tracker and Mark feel's it's time to target 1.0 again. At this point, the community needs to work as a

Re: [Asterisk-Users] SIP register and unregister events via Manager API

Nicolas Gudino wrote: On Fri, 2004-07-16 at 18:28, Matthias Endler wrote: is it possible to receive SIP/IAX register and unregister events via the manager API (like in CLI)? I do receive all kinds of call events (Hangup|Join|Leave|Link|Newchannel|Newexten|Newstate|Rename|Unlink). chan_sip2 support

[Asterisk-Users] Patch to test: Mailbox path changes

http://bugs.digium.com/bug_view_page.php?bug_id=0001188 This patch unifies the code that decides on the location of a mailbox and stores voicemail in a tree-like structure, to be prepared for very large volumes of voicemailboxes in one file system. It's disputed whether this affects performance on

[Asterisk-Users] Patch to test: Never say goodbye to an agent :-)

http://bugs.digium.com/bug_view_page.php?bug_id=0001693 This patch adds a lot of options for AgentLogin/AgentCallbackLogin Please test and respond in the bug tracker! /O - "This patch adds quite a few new features i

[Asterisk-Users] Path to test: Sending HTML virus, no, VOICEMAIL!

http://bugs.digium.com/bug_view_page.php?bug_id=0002055 This patch adds the ability to send text and HTML messages as voicemail notificiations. Please test and respond to the bug tracker! /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.d

[Asterisk-Users] Path to test: Czech localization

http://bugs.digium.com/bug_view_page.php?bug_id=0002013 If you use the Czech language, please test this and add your opinion, good or bad, to the bug tracker. /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/as

[Asterisk-Users] Patch to test: Dynamic queues

http://bugs.digium.com/bug_view_page.php?bug_id=0001858 Constfilin writes: "The attached patch allows dynamic configuration of asterisk queues. Queue information is re-read from the configurable database in real time. Additional Information Right now implemented only for postgres, no mysql" Please

Re: [Asterisk-Users] Anyone experience with early dial?

Holger Schurig wrote: I keep replying to myself quite often. As it turned out, this is a problem with incrementing CSEQ on the Grandstream. I don't have the clue if the SIP specification says that you have to increment it, but the GS sometimes sends a different SIP message with the same CSEQ. He

Re: [Asterisk-Users] freenode #asterisk IRC channel identd problem

Nathan Alpert wrote: Sorry to ask this question here since it's related to IRC and not Asterisk, but I am having trouble logging into the #asterisk IRC channel on freenode and was wondering if anyone else has had this problem and solved it. So here's the situation: Whenever I try to login to the #a

Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous

Sunrise Ltd wrote: Olle E. Johansson wrote: Well, I have users that get an account on my PBX. I really don't care how many phones they want to use, hardware phones on their desktop or soft phones on their laptop while travelling. It's still a user with one account. Two words: self pr

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