Re: [Asterisk-Users] CLI SIP Client

2005-03-16 Thread Olle E. Johansson
Klaus Peras wrote: Hey there, does anybody know a CLI SIP Client für Linux? I think you may find one in Vovida.org /O ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE

Re: [Asterisk-Users] Reg Asterisk

2005-03-23 Thread Olle E. Johansson
[EMAIL PROTECTED] wrote: hi, Is asterisk a registrar server. It all depends. If you mean registrar for Inter-Galaxy Travel Permissions, no. If you mean SIP registrar, yes. But we are not a SIP proxy ;-) /O ___ Asterisk-Users mailing list

Re: [Asterisk-Users] help understanding sip header - OPTIONS

2005-03-24 Thread Olle E. Johansson
These OPTIONs packets are what we send for qualification, a scheme that can also be used for NAT keepalives. You turn them on by adding qualify=yes in the [peer] section of sip.conf With qualification on, we regurlarly measure the latency between Asterisk and the client and decide whether the

[Asterisk-Users] Re: [Asterisk-Dev] cvs-head from 3/31/05 fails to load

2005-03-31 Thread Olle E. Johansson
Rich Adamson wrote: Cross posted on purpose FYI, just upgraded from cvs-head from March 23 to this morning (March 31). All compiles and installs completed normal. Loading asterisk via safe_asterisk (or asterisk -cdvvv) fails with the standard oche... message. Piped the output to a text file

[Asterisk-Users] Re: [Asterisk-Dev] cvs-head from 3/31/05 fails to load

2005-03-31 Thread Olle E. Johansson
Rich Adamson wrote: Cross posted on purpose FYI, just upgraded from cvs-head from March 23 to this morning (March 31). All compiles and installs completed normal. Loading asterisk via safe_asterisk (or asterisk -cdvvv) fails with the standard oche... message. Piped the output to a text file

[Asterisk-Users] *** Asterisk 2.0 Stable release out now

2005-03-31 Thread Olle E. Johansson
During the developer's conference call yesterday evening, it was decided that we finally should release the much-awaited Asterisk 2.0 Stable release, also called codename AAFJ. This relaese is based on the hidden cvs that has been in operation for six months by a group of core development members

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

2005-04-03 Thread Olle E. Johansson
Welcome to the Asterisk users community! Asterisk is the leading Open Source Telephony platform, with support both for classical telephony and IP telephony. Asterisk.org is a fast moving project. New code is added every day. Our community is also growing

Re: [Asterisk-Users] How to send email from the dial plan?

2005-04-04 Thread Olle E. Johansson
Ronald Wiplinger wrote: I would like to get a notice by email, if we run out of gateways! exten = _9011Z.,410,Busy exten = _9011Z.,411,EMAIL = How to? -= Info about application 'System' =- [Synopsis]: Execute a system command [Description]: System(command): Executes a command

Re: [Asterisk-Users] Previous sip reload not yet done

2005-04-04 Thread Olle E. Johansson
administrator tootai wrote: Nabeel Jafferali a écrit : Does anyone else have this problem? Is there a workaround? Yeah, I had this problem when I added a lot of SIP register statements and SIP peers. Changing the hostnames (FQDNs) to IP addresses solved the problem. It seem * was getting stuck

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

2004-12-27 Thread Olle E. Johansson
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:

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

2004-12-30 Thread Olle E. Johansson
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

Re: [Asterisk-Users] MeetMe

2005-01-02 Thread Olle E. Johansson
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] FastAGi change

2005-01-08 Thread Olle E. Johansson
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

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

2005-01-26 Thread Olle E. Johansson
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

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

2005-01-30 Thread Olle E. Johansson
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()

Re: [Asterisk-Users] SRV lookups

2005-02-08 Thread Olle E. Johansson
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

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

2005-02-08 Thread Olle E. Johansson
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

Re: [Asterisk-Users] SIP proxies Asterisk ?

2005-02-10 Thread Olle E. Johansson
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

[Asterisk-Users] Asterisk Users in Madrid?

2005-02-15 Thread Olle E. Johansson
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

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

2005-02-17 Thread Olle E. Johansson
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

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

2005-02-17 Thread Olle E. Johansson
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] Sip Notify PAP2-NA?

2005-02-17 Thread Olle E. Johansson
Chris St Denis wrote: I am using mysql sipfriends and can't seem to get the MWI to work. From what I've read it seems this is not supported with that dynamic system, and probably never will be. In the 1.0 stable release, you can not send MWI for database peers. In CVS head, the base for the future

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

2005-02-17 Thread Olle E. Johansson
Welcome to the Asterisk users community! Asterisk is the leading Open Source Telephony platform, with support both for classical telephony and IP telephony. Asterisk.org is a fast moving project. New code is added every day. Our community is also growing

Re: [Asterisk-Users] Accountcode and SIP Peers Part 2

2005-02-17 Thread Olle E. Johansson
Marcello Lupo wrote: Hi, notice that i have Grandstream phones and i have the problem if i activate the Send Anonymous function on them. If i do not activate that option the ACCOUNTCODE is correctly populated. SO i think it may be a bug of asterisk. I'm using Asterisk CVS-HEAD-10/07/04-18:07:25

Re: [Asterisk-Users] functional difference: canreinvite=yes, no, or update

2005-02-17 Thread Olle E. Johansson
Kevin P. Fleming wrote: I have a patch in my local system that allows the canreinvite setting (which I renamed) to actually be based on IP address masking, so that Asterisk can make a more intelligent decision, but even that has problems, because we don't actually _know_ that any given IP route

Re: [Asterisk-Users] Astricon 2004 tutorials available?

2005-02-17 Thread Olle E. Johansson
Spencer Nassar wrote: Does anyone know if the tutorial materials from Atricon 2004 are available for download anywhere? I'm particularly interested in Joachim Vanheuverzwijn's Performance and Scalability tutorial slides (Asterisk - building your system for performance and scalability).

Re: [Asterisk-Users] Is this a bug or by design? Workaround?

2005-02-18 Thread Olle E. Johansson
Stig Andersson wrote: So, I try - SetVar(cid=${CALLERIDNUM:-5:5}) The result is a empty string if CALLERIDNUM is less than 5 digits long, which is NOT the case of SubString. SubString command returns what remains of the variable, that is - if CALLERIDNUM is 4 digits in length, it

[Asterisk-Users] *** Important *** About the bug tracker

2005-02-18 Thread Olle E. Johansson
During the last week, we have had several support issues being reported as bugs on the bug tracker. Since we are going into a final development stage on version 1.1dev (CVS HEAD) in order to complete the 1.2 release we are under pressure to fix bugs and handle a lot of reports in a short time

Re: [Asterisk-Users] Simulated dialtone like in other PBX

2005-02-20 Thread Olle E. Johansson
Anton Krall wrote: I think it would be your last suggestion.. When I pickup the phone I hear a tone, the sip phone box tone... Then I hit 9, no tones :) and enter the whole phone number and it starts to ring on the other side.. So no outside dialtone get heard ever.. I was wondering if it could be

Re: [Asterisk-Users] Bug in SUBSCRIBE handling : running out of RTP ports

2005-02-24 Thread Olle E. Johansson
Sarat Vemuri wrote: While trying to deploy a bunch of Polycom IP 500 phones, I ran in to the following. I limited the RTP ports from 8000-8050 to limit holes in firewall. Pretty soon Asterisk ran out of RTP ports. Traced the problem back to how * is handling SUBSCRIBE. A sip structure is

[Asterisk-Users] Introducing the Asterisk Realtime Architecture - ARA

2005-02-27 Thread Olle E. Johansson
I've added an introduction article about the ARA on my web site http://www.voip-forum.com/ The same text is now also added to CVS head as README.realtime. On the same site, you will also find the news item about how we used Asterisk for a call from an airline jet above Greenland to Stockholm,

[Asterisk-Users] Important :: Please support the development of a new Jitterbuffer for SIP

2005-03-01 Thread Olle E. Johansson
Steve Kann has developed a new jitterbuffer for IAX2, that hopefully will be integrated into Asterisk v1.1 soon, to be part of the 1.2 stable relase. Zoa and his bulgarian team is porting this buffer to SIP/RTP, but needs support in the form of funding in order to take the time to test this

Re: [Asterisk-Users] Important :: Please support the development of a new Jitterbuffer for SIP

2005-03-01 Thread Olle E. Johansson
Steve Underwood wrote: re here: http://www.astertest.com/forum/viewtopic.php?t=13 Thank you for your contribution! The hard work of building the thing was done for free, and now someone brings out the begging bowl for the relatively minor activity or porting into to another home. Frankly, that

Re: [Asterisk-Users] SIP registration problem

2005-03-02 Thread Olle E. Johansson
In the Grandstream setup, turn off subscribe to message waiting indication. ...or upgrade to CVS head, where I've fixed this problem with SUBSCRIBE. Best regards, /Olle ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com

Re: [Asterisk-Users] Asterisk not relaying back the SIP response messages

2005-03-03 Thread Olle E. Johansson
Atif Rasheed wrote: HI all, I have the following setup running: EP---Calling Asterisk---Relaying Asterisk---Softswitch--- PSTN The Endpoint EP is registered with the Calling Asterisk. Calls are forwarded from this machine to Relaying Asterisk which in turn forwards it to the Softswitch. In

Re: [Asterisk-Users] Asterisk on FreeBSD 4.9?

2004-01-15 Thread Olle E. Johansson
Jason T. Nelson wrote: I have already started playing with trying to figure out why Asterisk runs so badly under FreeBSD, such as eating 100% of the CPU without warning unload pbx-wilcalu.so, see http://www.voip-info.org/wiki-Asterisk+freebsd /O ___

Re: [Asterisk-Users] QoS anyone?

2004-01-16 Thread Olle E. Johansson
Rich Adamson wrote: Has anyone played around with QoS or TOS relative to * and sip phones? I was just doing a little real-time research and noticed our C7960's mark IP packets with low delay and high throughput (presumably due to tos_media: 5 in the SIPDefault config file), and rtp packets

Re: [Asterisk-Users] CDR problem with macros

2004-01-17 Thread Olle E. Johansson
Philipp von Klitzing wrote: Hi there, whenever I use a macro to dial out I see only s recorded in the dst field of the CDR. Is there anyway to get around that problem except for not using a macro? Example: ) Try to match every extension before dialing out instead, using s is a bad thing for

Re: [Asterisk-Users] Codec matching weirdness

2004-01-17 Thread Olle E. Johansson
Dustin Goodwin wrote: I did find something interesting. If you set reinvite=yes then * can setup the RTP stream so that it avoids the media proxy in the * box completely. I haven't tested to see if it changes anything. Can we please kill reinvite - it does not exist in the SIP channel as an

Re: [Asterisk-Users] New sounds also now in CVS

2004-01-18 Thread Olle E. Johansson
John Todd wrote: The soundfiles I submitted earlier today have been cleaned up, and added to the Digium CVS server in a more formal manner. Also, some of the really bad formatting in my .txt description file has been rectified. All of the sounds on my website are now on the Digium site, and

Re: [Asterisk-Users] Asterisk as SIP Redirect Server -- Implemented - Not Working - Plz Help

2004-01-18 Thread Olle E. Johansson
Could you please explain what you want to do, why you want asterisk to register but not take the calls? You could take the calls into the dialplan (extensions.conf) and dial out from there with an agi script that performed the same thing. If you have canreinvite=yes, asterisk will leave the

Re: [Asterisk-Users] Search engine for this list

2004-01-19 Thread Olle E. Johansson
Steven Critchfield wrote: On Mon, 2004-01-19 at 05:19, Kim Hendrikse wrote: Is there a search engine for this list? Google Use site:lists.digium.com to limit the search to just the list server. ...or http://search.voip-forum.com Indexes our lists, the Wiki, asterisk.org and some related

Re: [Asterisk-Users] Some SIP Setup problems

2004-01-25 Thread Olle E. Johansson
[EMAIL PROTECTED] wrote: Are you using the 0.7.1 tar distribution or CVS? I was able to compile the 0.7.1 Asterisk program/sample config's to get a working system on a PC with no sound device and no phone interfaces. This system is about as simple as it can get (except for the 3 fixed disks in

Re: [Asterisk-Users] Is it possible to push the media processing off to a gateway for processing?

2004-01-25 Thread Olle E. Johansson
T. Chan wrote: I think what Todd was referring to was to JUST do the signaling proxy on the Asterisk but not proxying the media. This is the definition of a SIP proxy. Asterisk is a PBX that supports SIP, but not really a SIP proxy. As a PBX, it wants to be in the middle of a call. As an

Re: [Asterisk-Users] iax1.conf ??

2004-01-25 Thread Olle E. Johansson
Tilghman Lesher wrote: On Sunday 25 January 2004 11:19, Philipp von Klitzing wrote: Jan 25 17:30:02 ERROR[40979]: chan_iax.c:4826 set_config: Unable to load config iax1.conf As a matter of chan_iax slowly moving towards the deprecated pile, to be replaced everywhere with chan_iax2, chan_iax

Re: [Asterisk-Users] Does anyone manage the wiki?

2004-01-29 Thread Olle E. Johansson
mattf wrote: Go ahead and edit the page. I've fixed several little errors on pages that I didn't create. The voip-info.org Wiki is like the total-open-source Asterisk manual. Although the total-access may be a problem in the future because all someone has to do to delete everything is just to

Re: [Asterisk-Users] Running Asterisk on FreeBSD.

2004-01-29 Thread Olle E. Johansson
Dmitry Mishchenko wrote: Now major problems comes: After starting asterisk it is trying to get all available CPU time. I'm using standard config files. Turning off modules h323 and oss didn't help. Read http://www.voip-info.org/tiki-index.php?page=Asterisk%20freebsd for advice on how to get rid

Re: [Asterisk-Users] Calls dropping off

2004-01-30 Thread Olle E. Johansson
Steve Foy wrote: Hi, I've got a fairly working Asterisk setup, with a few minor glitches, one of which is very very irritating. Sometimes, during a call, the remote end just drops off. We're using software SIP phones (SJPhone) connecting to * then out through analogue lines with X100P cards.

Re: [Asterisk-Users] P2P RTP without SIP re-invites

2004-01-31 Thread Olle E. Johansson
Let's go through how SIP works in Asterisk compared with a SIP Proxy. Remember that Asterisk is not designed to be a SIP Proxy, it's designed to be a Multi-VOIP and PSTN PBX, a quite complicated task. (I'm not going into all details (ACK, TRYING, RINGING etc)) We have two SIP users, Alice and

Re: [Asterisk-Users] P2P RTP without SIP re-invites

2004-01-31 Thread Olle E. Johansson
Brancaleoni Matteo wrote: Hi. If both Alice and Bob are connected without NAT, have the same codec support and have canreinvite=yes * Asterisk send (re-)INVITEs to both, trying to get the RTP stream transferred so it goes directly from Alice to Bob Not all UAs support a re-INVITE and in

Re: [Asterisk-Users] SUBSCRIBE in chan_sip - anyone?

2004-02-01 Thread Olle E. Johansson
Rich Adamson wrote: So, what hardware or use is the SUBSCRIBE method used for in chan_sip.c? I asked this question a while ago, and got resounding silence. Maybe someone who is better at de-tangling C code than I am could take a peek. Not sure, but seems to me it came in about the time Olle

Re: [Asterisk-Users] SUBSCRIBE in chan_sip - anyone?

2004-02-01 Thread Olle E. Johansson
Florian Overkamp wrote: Hi, -Original Message- So, what hardware or use is the SUBSCRIBE method used for in chan_sip.c? I asked this question a while ago, and got resounding silence. Maybe someone who is better at de-tangling C code than I am could take a peek. Hmm, dunno. Could

Re: [Asterisk-Users] SUBSCRIBE in chan_sip - anyone?

2004-02-01 Thread Olle E. Johansson
Florian Overkamp wrote: Hi, -Original Message- So, what hardware or use is the SUBSCRIBE method used for in chan_sip.c? I asked this question a while ago, and got resounding silence. Maybe someone who is better at de-tangling C code than I am could take a peek. Hmm, dunno. Could

Re: [Asterisk-Users] Dial via sip gateway?

2004-02-01 Thread Olle E. Johansson
Rich Adamson wrote: What's the proper syntax for dialing out via a sip g/w (Mediatrix)? Been trying stuff similar to: exten = _6X.,1,Dial(SIP/[EMAIL PROTECTED]/${EXTEN-1}) where 3091 is alias for the port on the Mediatrix. Sniffer indicates * did even try the IP. Rich from my extensions.conf:

Re: [Asterisk-Users] SIP debug logs

2004-02-04 Thread Olle E. Johansson
Debuuging SIP to a file: asterisk -c | tee /tmp/sipdebug.log then turn on 'sip debug' at the CLI /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:

Re: [Asterisk-Users] Caller-ID is being sent wrong. How to fix it?

2004-02-07 Thread Olle E. Johansson
Would like to see a SIP debug * The invite from the caller phone to Asterisk * The invite from Asterisk to the called phone As well as the configs (extensions.conf and sip.conf) Can't reproduce in my servers. /O ___ Asterisk-Users mailing list [EMAIL

Re: [Asterisk-Users] Caller-ID is being sent wrong. How to fix it?

2004-02-08 Thread Olle E. Johansson
Vic Cross wrote: On Sat, 7 Feb 2004, John Fraizer wrote: snip all the trace data Here are the configs: ; ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to bindaddr = 66.35.64.38 ; Address to bind to context = default ;

Re: [Asterisk-Users] Snom 200 MWI Button

2004-02-08 Thread Olle E. Johansson
Dustin Knuttgen wrote: Anyone have success in getting the MWI button to work on Snoms? If so I would LOVE to hear from you. Read http://www.voip-info.org/tiki-index.php?page=Asterisk%20phone%20snom The problem is well-known. /Olle ___

Re: [Asterisk-Users] Calling SIP

2004-02-09 Thread Olle E. Johansson
Tim Sailer wrote: On Mon, Feb 09, 2004 at 01:37:55PM -0600, Eric Wieling wrote: That's just the way Asterisk's dial command works. Hmm. I see. If it can't create the channel for either reason (busy or not registered), it's handled the same. I think I'll kludge up a perl script to watch the SIP

Re: [Asterisk-Users] Calling SIP

2004-02-09 Thread Olle E. Johansson
Tim Sailer wrote: I've looked, poked, and hoped, but I can't seem to make * understand the difference between a SIP channel being busy or not being there. Both come up as 'busy'. I would expect the unregistered SIP to be seen as unavailable. Am I just missing something obvious, again? I've heard

Re: [Asterisk-Users] Anybody going to the Spring VON converence [OT]

2004-02-13 Thread Olle E. Johansson
I'm going. Would be great to have an Asterisk gathering. /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:

Re: [Asterisk-Users] two UA with the same usr/pwd

2004-02-17 Thread Olle E. Johansson
Jeff wrote: 1. Two SIP phones can login to FWD at the same time with the same username/pwd , is this normal? Ed explained this. 2. can Two SIP phones login to * at the same time with the same username/pwd ? how to prevent this? Well, a SIP proxy normally allows this. Asterisk is not a SIP proxy

Re: [Asterisk-Users] SIP config documentation

2004-02-18 Thread Olle E. Johansson
Costa Tsaousis wrote: I was trying to figure out all the valid options for a sip.conf and I believe I found a few weird things (or just a few things that are weird to me :) Anyway, I decided to post this here together with my questions and notes in case other people need this info too or have

Re: [Asterisk-Users] SIP config documentation

2004-02-21 Thread Olle E. Johansson
Costa Tsaousis wrote: context= ; UP, the context name for placing calls Q1: Why is there a context for peers? We use peers in some other situations as well. This is strange and rather undocumented, but an incoming call is first matched by username with the defined users (including 'friends').

[Asterisk-Users] Re: incominglimits and outgoinglimits (New subject)

2004-02-21 Thread Olle E. Johansson
Fran Boon wrote: On Sat, 2004-02-21 at 09:06, Costa Tsaousis wrote: incominglimit= ; U- concurrent call limitations ( = 0 ) outgoinglimit= ; U- concurrent call limitations ( = 0 ) Q6: How is it possible for a type=user phone to have BOTH incoming and outgoing limits? Interesting question. Anyone

Re: [Asterisk-Users] Re: [Asterisk-Users] SIP config documentation

2004-02-21 Thread Olle E. Johansson
Costa Tsaousis wrote: Sorry, I was on the wrong topic, canreinvite has yes|no|update as keywords. with UPDATE a SIP method UPDATE is initiatied to change the media path. with YES, a new INVITE is issued within the current call. (a re-invite) with NO, the call stays within asterisk. Sorry for

Re: [Asterisk-Users] SIP extension busy when not available ??

2004-02-22 Thread Olle E. Johansson
Rich Adamson wrote: How come * says 1010 is BUSY in the trace below? I would have guessed UNAVAILABLE since 1010 is not logged on/registered. Sounds right to me. That's what has been programmed in the asterisk code and has been that way since the beginning of time. Is that right? I was afraid it

Re: [Asterisk-Users] How to best debug SIP registration failure

2004-02-22 Thread Olle E. Johansson
George Pajari wrote: I am having trouble getting SIP phones to register with Asterisk. I know that the phone can register with FWD and I have used tcpdump to see the registration packets arrive at the Asterisk server, but nothing goes back. How should I attack the problem? What debugging tools

Re: [Asterisk-Users] Re: How to best debug SIP registration failure

2004-02-23 Thread Olle E. Johansson
If you see nothing with full verbosity and SIP debug turned on, the Asterisk SIP channel gets nothing. The reason why we always mix in NAT with questions like yours is that in 90% of the cases, NAT is the problem. It's just a standard response, like when Microsoft support tells you to reinstall

Re: [Asterisk-Users] VM: Multilanguage and digits

2004-02-23 Thread Olle E. Johansson
Lars Fredriksson wrote: Hi! I'm trying to record som voiveprompts, and I've created a directory se in /var/lib/asterisk/sounds - in that directory I've put files like vm-intro.gsm, vm-the-person.gsm and do on. And if I use SetLanguage(se) I hear my own voice prompts! But wehre should I place the

Re: [Asterisk-Users] Changes in capi.conf

2004-02-24 Thread Olle E. Johansson
I noted that I have to put a load=res_parking.so before chan_capi.so in modules.conf, since chan_capi 3.1 uses some parking group stuff. Otherwise startup failed with error on symbol ast_get_group Worth to notice in the README! /O ___ Asterisk-Users

Re: [Asterisk-Users] Re: calls dropped with grandstream

2004-02-24 Thread Olle E. Johansson
Stephen R. Besch wrote: Sean Rodger wrote: I'm using a grandstream phone with asterisk. Everything seems to be working fine, but every once in a while talking to someone, the call is dropped. A loud busy signal immediately interrupts the call for the grandstream user, while the other person

Re: [Asterisk-Users] Problem with SIP 407

2004-02-25 Thread Olle E. Johansson
Vic Cross wrote: G'day Marc, On Wed, 25 Feb 2004, Marc Fargas wrote: Im in trouble with SIP. Ive got a SIP FXS gateway from www.micronet.info (SP5002/S) and traed to register to asterisk, It seems to autentcate but sniffing the net it shows a 407 proxy authen required error message and I

Re: [Asterisk-Users] Problem with SIP 407

2004-02-25 Thread Olle E. Johansson
Seems like republica registers ok, but not republica2. Republica2 failes to authenticate. You have a normal registration sequense here: -Client sends a REGISTER without authentication -Server sends trying... -Server sends 407 Proxy auth (should be WWW auth) with challenge -Clients ACK -Client

Re: [Asterisk-Users] Message waiting light not coming on

2004-02-26 Thread Olle E. Johansson
Iain Stevenson wrote: Works perfectly fine for me - but I'm not using rfc2683 - my Grandstream uses the latest firmware and SIP INFO. Iain --On Thursday, February 26, 2004 12:55 am -0500 Rana Dutt [EMAIL PROTECTED] wrote: I cannot get the Message Waiting Light (MWL) on my Grandstream phone

Re: [Asterisk-Users] Re: Grandstream transfer into outer space

2004-02-26 Thread Olle E. Johansson
Stephen R. Besch wrote: Olle E. Johansson wrote: Going back to the subject, what does the grandstream really do, SIP-wise, when you press the transfer button? 4.3.7 Call Transfer The user can transfer an active call to a third phone by using the Transfer button. The sequence is like

Re: [Asterisk-Users] Grandstream transfer into outer space

2004-02-26 Thread Olle E. Johansson
Going back to the subject, what does the grandstream really do, SIP-wise, when you press the transfer button? /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options

Re: [Asterisk-Users] chan_sip support for SIP:Remote-Party-ID, specifically CLID priva cy

2004-02-26 Thread Olle E. Johansson
Low, Adam wrote: Hey All, I have a Cisco AS5300 running SIP against an Asterisk server with multiple C7940 phones. My issue is that from what I see in chan_sip.c there is no support for the Remote-Party-ID field in relation to withholding the calling partys number. This is a legal

Re: [Asterisk-Users] chan_sip support for SIP:Remote-Party-ID, sp ecifically CLID priva cy

2004-02-27 Thread Olle E. Johansson
Low, Adam wrote: Could you please point me in direction of standard documents, drafts or documentation of this? IETF specification, draft-ietf-privacy-.02.txt, SIP Extensions for Caller Identity and Privacy. Thank you for the pointer, as this is still a draft (a lot of SIP things are), it's

Re: [Asterisk-Users] retrieve_sip_conf_from_mysql.pl data format

2004-02-29 Thread Olle E. Johansson
Fran Boon wrote: Olle's chan_sip2 introduces a 3rd possibility: Using templates autocreate peers for the majority of user options storing just the passwords in the MYSQL database. Combining this with MYSQL_FRIENDS, storing template= settings in a database would be very powerful. /O

Re: [Asterisk-Users] retrieve_sip_conf_from_mysql.pl data format

2004-02-29 Thread Olle E. Johansson
Fran Boon wrote: I guess I need to implement this with astdb instead of MySQL, since this can be queried direct within the dialplan. Would be lovely to have dbget/dbput routines for MySQL as well as just db1! Brian was working on odbcget/put. I think there's a beta uploaded on his web site. /O

Re: [Asterisk-Users] Asterisk as a SIP server behind nat, clients on the outside connecting to Asterisk

2004-02-29 Thread Olle E. Johansson
Steve Beaumont wrote: On the wiki pages it suggests that clients on the outside of NAT can connect to an Asterisk server behind nat. (option no 3). The note suggests that this can work with port forwarding and some 'header mangling magic'. I have the port forwarding configured however, when I try

Re: [Asterisk-Users] peer is UNREACHABLE when using XLITE

2004-03-07 Thread Olle E. Johansson
Senad Jordanovic wrote: Hans-Henrik Andresen wrote: Hi, I have 3 friends trying to connect to my Asterisk using x-lite, all of them are using 3 dif. adsl-provider. For each of them I got this in sip.conf: disallow=all allow=ulaw allow=alaw allow=ilbc allow=g729 allow=g723.1 [seholm]

Re: [Asterisk-Users] Grandstream Budgetone SIP registration fails

2004-03-08 Thread Olle E. Johansson
Jean-Marc V. Liotier wrote: On Sun, 2004-03-07 at 13:03, Philipp von Klitzing answered off-list: [6040] defaultip=192.168.1.40 Replace this with host=dynamic and see what happens. That's it ! Thinking it was going to make things easier to diagnose, I had chosen to set the phones with a static

Re: [Asterisk-Users] Grandstream Budgetone SIP registration fails

2004-03-09 Thread Olle E. Johansson
Jean-Marc V. Liotier wrote: On Mon, 2004-03-08 at 18:50, Olle E. Johansson wrote: If you configure a static address, the PBX already know how to reach the client and no registration is therefore needed (and not allowed in asterisk). Enabling registration makes the SIP device mobile across

Re: [Asterisk-Users] logic problem with GotoIf?

2004-03-09 Thread Olle E. Johansson
Rich Adamson wrote: exten = s,1,GotoIf(DBget(FEAT/ivron) == yes?bus-ivr-main|s|1) Rich, I haven't seen GotoIF calling another application, only $[] constructs. See http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20gotoif for examples. Take the output of dbget into a variable and test

Re: [Asterisk-Users] ast_rtp_raw_write errors distorting sound on G729 passthrough

2004-03-14 Thread Olle E. Johansson
Check out the latest CVS, Mark applied changes to the code in this area tonight. The rtp.c is changed, so the old patch in bugs.digium.com may not be necessary any more. /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED]

Re: [Asterisk-Users] extensions problem (SIP)

2004-03-15 Thread Olle E. Johansson
Jon Lawrence wrote: Hi, I've got 2 x100p's installed in my system. Both execute the same incoming contexts as follows: [inboundA] include = dialjon [inboundB] include = dialjon|09:00-16:30|Mon-Fri|*|* [dialjon] exten = s,1,answer exten = s,2,Dial(SIP/2000,15) exten = s,3,Playback(noone) exten =

Re: [Asterisk-Users] extensions problem (SIP)

2004-03-15 Thread Olle E. Johansson
Jon Lawrence wrote: Surely * should know if a phone is in use ? After all it initiated/took part in the call in the first place ;) Again, the SIP device is not a slave device. It could receive a call from somewhere else and be busy without Asterisk having a clue. A lot of SIP UAs, like Xten

Re: [Asterisk-Users] extensions problem (SIP)

2004-03-15 Thread Olle E. Johansson
Walker Haddock wrote: On Mon, Mar 15, 2004 at 09:28:00PM +0100, Olle E. Johansson wrote: The incominglimit limits how many simultaneous calls a UA may place to Asterisk. I'm pretty sure that the incominglimit specifies how many calls that * can send to the SIP device. If you set incominglimit=1

Re: [Asterisk-Users] Asterisk USR

2004-03-16 Thread Olle E. Johansson
Ignace CARIA wrote: Hi, How can I use my external Modem (US Robotics Sportster Flash - RS 232) as a voice client connected to Asterisk. You can't. /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED]

Re: [Asterisk-Users] Re: Calling one local SIP user from another (using X-Lite)

2004-03-16 Thread Olle E. Johansson
Ross Finlayson wrote: voice mail. However, if I try to call user2 from user1's X-Lite - or vice-versa - I get a 404 Not Found error. Is there anything obvious that I'm doing wrong? (In particular, do I also need to add entries to extensions.conf for user1 and user2??) Ross. Try

Re: [Asterisk-Users] Asterix Sip Stack

2004-03-18 Thread Olle E. Johansson
Mark Phillips wrote: [EMAIL PROTECTED] said: Is it Sip Registry Server ?. Could it work as Proxy Server ? Hello Ahmet, Asterisk is more than a proxy. Its an entire PBX. At a basic level it can be used as a proxy though. My favourite subject... :-) No, Asterisk is not even close to a SIP

Re: [Asterisk-Users] Important: The Asterisk Mailing list (new subject)

2004-03-19 Thread Olle E. Johansson
Mark Spencer wrote: The Asterisk community is growing at a remarkable pace. I know there are thousands of you out there -- in fact there are over eight *thousand* subscribers to asterisk-users alone, and almost one *thousand* registered users on the bug tracker. This means that everything

Re: [Asterisk-Users] Important: The Asterisk Mailing list (new subject)

2004-03-20 Thread Olle E. Johansson
As I started this trend I take the right to end it. I just want us to follow John Postel's rule for how to act on the Internet (I think he defined it for TCP/IP software, but it can be applied here too.) Be strict in what you send Be generous in what you accept Sending a reply to

Re: [Asterisk-Users] Festival

2004-03-20 Thread Olle E. Johansson
Justin Carlson wrote: I am sorry if this is a silly question but I can not seem to locate the festival binaries. does this come with asterisk or is it another project? No question is silly. This is a good time to remind the list of the FAQ

Re: [Asterisk-Users] Registration from xxx failed for 'xxx'

2004-03-20 Thread Olle E. Johansson
Thomas Gallaway wrote: Here is my problem. I have 2 phones (Grandstream Budge Tone-100) loosing the sip registration every 4 hours. I can not find out why. It seems like the registration fails, then a few minutes after registers sucessfull. Mar 19 14:06:14 NOTICE[147466]: Registration from

Re: [Asterisk-Users] Store caller IP in CDR

2004-03-20 Thread Olle E. Johansson
Fritz Müller wrote: How can I configure * to store the caller and called Party IP Address in the CDR file. Depends on the channel, not all channels are IP based. Check the CDRuserfield - it's a free field in the CDR you set in the dialplan or from a script. Without knowing why you want this, I

Re: [Asterisk-Users] Qualify statement

2004-03-20 Thread Olle E. Johansson
Senad Jordanovic wrote: Does anyone know if qualify=XXX should be used ONLY for user agents behind NAT. No, you can use it to qualify any address. Qualification means that Asterisk regurlarly sends SIP messages with the OPTION method and the UA answers. We clock the time and if the client takes

Re: [Asterisk-Users] Basic authentication

2004-03-20 Thread Olle E. Johansson
Joao Carlos Moura wrote: How can I settup a way for Asterisk doesn´t make any use of DIGEST AUTHENTICATION method? I don t want ASTERISK to check out any username or password of my users. Set no secret in sip.conf our use autocreatepeer /Olle ___

  1   2   3   4   5   6   7   8   9   10   >