Re: [asterisk-dev] Dundi library

Thanks for the feedback. Corey, thanks for the work on the library. Meanwhile I implemented the workaround: let Kamailio forward the request to an Asterisk server which does the lookup and responds with a 302 redirect to (Transfer()) the proper Asterisk server. Using SIP for the query has the

[asterisk-dev] Dundi library

Hi! I have an Asterisk cluster and use Dundi for the internal routing. Now I want to query Dundi also from other systems, e.g. a Kamailio proxy. I thought there will be a Dundi library which can be used, but my web search didn't find any. I also checked Dundi in Asterisk and it seems that Dun

Re: [asterisk-dev] introduce a codec

no. Just take a look at one of the existing codecs. klaus [EMAIL PROTECTED] schrieb: > Hello, > > i have another problem (sorry but i'm totally lost): do i need sockets to > pass the information through the codec to asterisk??? > thank you very much > > > _

Re: [asterisk-dev] Introduce a codec

I also suggest to just "grep" the source code for another codec, e.g. "grep -r -i gsm" Then you see were you have to define the codec and how to add certain codec related functions. regards klaus Moises Silva schrieb: > You have a lot of examples in the codecs/ directory of Asterisk. > Basicall

Re: [asterisk-dev] Asterisk mishandling user busy isdn releases

I would try this: exten => 123,1,Dial(.) exten => 123,2,Gotoif($["${DIALSTATUS}" == "BUSY"]?3:4) exten => 123,3,Busy() exten => 123,4,Hangup() regards klaus Ken Leland III schrieb: > > > Hello All, > > > I have found that Asterisk does not play a busy tone when it > receives a > > > USE

Re: [asterisk-dev] q931 decoding question

Matthew Fredrickson schrieb: > Klaus Darilion wrote: Ext: 0 User information layer 1: Unknown (24) >> >> My questions: Why does it gets parsed? Why is it set to 24? I checked >> libpri source code but could not find the relevant code. Maybe someone >> can

Re: [asterisk-dev] dialog matching

Hi Steve! pedantic in Asterisk is buggy - regardless if it is turned on or off. Thus, please do not just reimplement the old behavior but fix it. AFAIK pedantic controls 2 things: 1. accept short header names and header which are wrapped around multiple lines if pedantic=on 2. use to-tag fo

Re: [asterisk-dev] Kill the user, kill the user!

Johansson Olle E wrote: > 7 dec 2007 kl. 17.45 skrev Igor A. Goncharovsky: > >> Hi! >> >> Johansson Olle E wrote: >>> Incoming calls are handled this way >>> >>> * First, we match on peer object name with the From username >>> * Then we try to match on IP/Port >>> * If we can't match, we send to t

Re: [asterisk-dev] Kill the user, kill the user!

Ryan Mitchell wrote: > > To raise an issue that's come up a few times before, instead of checking > user then ip/port, why not check ip/port first then user? I think this will cause trouble if you have multiple "ex-users" pointing to the same ip:port. Checking ip:port first this will always res

Re: [asterisk-dev] ast_frame allocation/free question

Sergio Garcia Murillo schrieb: > From: Klaus Darilion [mailto:[EMAIL PROTECTED] >> Sergio Garcia Murillo wrote: >>> From: "Simon Perreault" <[EMAIL PROTECTED]> >>>> Either I'm getting it all wrong, or I have an even bigger problem than I >

Re: [asterisk-dev] New branch: calleridutf8

Olle E Johansson schrieb: > 26 nov 2007 kl. 11.14 skrev Klaus Darilion: > >> Johansson Olle E schrieb: >> >>> Feedback, ideas and suggestions is always welcome. >> If you ever need to enter characters on Windows using UTF-8 I can >> recommend http:

Re: [asterisk-dev] New branch: calleridutf8

Johansson Olle E schrieb: > Feedback, ideas and suggestions is always welcome. If you ever need to enter characters on Windows using UTF-8 I can recommend http://www.fileformat.info/tool/unicodeinput/index.htm (e.g. you want to test if you SIP client can handle Czech háček and you have no glue

Re: [asterisk-dev] why 4 UDP ports per SIP call?

Johansson Olle E schrieb: > 22 nov 2007 kl. 10.29 skrev Klaus Darilion: > >> >> Olle E Johansson wrote: >>> 19 nov 2007 kl. 20.53 skrev Klaus Darilion: >>> >>>> Hi! >>>> >>>> Analyzing Asterisk I see 4 new UPD sockets opene

Re: [asterisk-dev] any experience building asterisk under windows ?

Dinesh Nair schrieb: > On Wed, 3 Oct 2007 22:52:39 -0700, Luigi Rizzo wrote: >> I couldn't find a decent video client under windows: > > try playing around with openwengo, a linux/mac/windows qt-based softphone > which is also gpl-licensed. we've tested it with asterisk 1.2.x and video > (using

Re: [asterisk-dev] AST_FRAME_DIGITAL

Steven Critchfield wrote: > On Fri, 2007-09-14 at 17:15 +0200, Klaus Darilion wrote: >> critch schrieb: >>> The addon would then be a transcoder of H223 data to the demuxed >>> audio/video/control frames using the library that is GPL licensed. >>> >>>

Re: [asterisk-dev] AST_FRAME_DIGITAL

Steven Critchfield wrote: > So let us go about this in an interim manner. > > Modify the ISDN channel drivers to hand up > AST_FRAME_VOICE(AST_FORMAT_H223) from them to asterisk. Your application > could request format AST_FORMAT_H223 such that there is no translation > required. From here you con

Re: [asterisk-dev] AST_FRAME_DIGITAL

Matthew Fredrickson wrote: > Klaus Darilion wrote: >> Russell Bryant schrieb: >>> Matthew Fredrickson wrote: >>>>> So, if I get it right - there is no need to introduce AST_FRAME_DIGITAL >>>>> as it is already there (but named AST_FRAME_MODEM)?

Re: [asterisk-dev] AST_FRAME_DIGITAL

Matthew Fredrickson wrote: > Tony Mountifield wrote: >> In article <[EMAIL PROTECTED]>, >> Sergio Garcia Murillo <[EMAIL PROTECTED]> wrote: >>> Just cheking q931 bearer capability: >>> >>> pri debug span 1 >>> Bearer Capability (len= 3) >>> Ext: 1 Q.931 Std: 0 Info transfer capability: >>>

Re: [asterisk-dev] proper way to signal user information layer 1 (ISDN)

Matthew Fredrickson wrote: > Klaus Darilion wrote: > I think that probably making chan_zap so that it is more codec aware > would be the right way to adjust this field. I thought having a generic mechanism which allows changing UL1 without coding would be cool - but I have no p

Re: [asterisk-dev] AST_FRAME_DIGITAL

critch schrieb: > On Thu, 2007-09-13 at 14:55 +0200, Klaus Darilion wrote: >> Tilghman Lesher schrieb: >>> On Thursday 13 September 2007 05:14:40 Klaus Darilion wrote: >>>> Do you transcode IP packets from ulaw to alaw when sent from US to >>>> Europe? N

Re: [asterisk-dev] AST_FRAME_DIGITAL

Tilghman Lesher schrieb: > On Thursday 13 September 2007 04:52:20 Klaus Darilion wrote: >> Russell Bryant schrieb: >>> Klaus Darilion wrote: >>>> Thus, why do we have a AST_FRAME_IMAGE? Why is IMAGE not treated as >>>> VOICE? Obviously because

Re: [asterisk-dev] AST_FRAME_DIGITAL

Russell Bryant schrieb: > Christian wrote: >> I vote for AST_FRAME_DIGITAL too! I think i suggested that about over >> one year already.. It would be very useful to transport Digital Data >> like HDLC Frames which might occur in the media channel of chan_zap and >> chan_misdn. At the moment s

Re: [asterisk-dev] AST_FRAME_DIGITAL

Russell Bryant schrieb: > I can tell you right now that you will not be able to convince me that we > should > add an opaque frame type for this situation. It is completely against the > Asterisk architecture. As you said, the stream you are handling is in fact > voice and video. Asterisk has e

Re: [asterisk-dev] AOC in chan_sip

Hi Wolfgang! Regarding AOC-encoding/decoding: For AOC decoding there is already code in libpri. For AOC encoding you can take a look at http://bugs.digium.com/view.php?id=7494 regards klaus Wolfgang Pichler schrieb: > Hi all, > > as far as i know there is no standard way (no RFC...) to implem

Re: [asterisk-dev] behavior of 'nat=yes' with 'directrtpsetup=yes'

You can make things even complicater - e.g. if you Answer() before Dialing out to the other client (e.g. for some announcements) - then directrtp wont work without reINVITE. regards klaus Adam Gundy wrote: > following on from a suggestion that a bug report I raised (10335) is at > least partly

[asterisk-dev] common functions for int2string conversion

Hi! Are somewhere common functions for string/integer conversion defined? thanks klaus ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com

Re: [asterisk-dev] Fixing Asterisk DNS - bug 9152, asynchronous DNS, etc

Olle E Johansson wrote: > If dnsmgr is not asynch DNS, I suggest that someone takes a look into > the C-ares library. > It has a license that Kevin/Mark has approved of and will help us to > proper asynch > DNS. It's used by curl, so in some installations it's already used by > asterisk in

Re: [asterisk-dev] Adding Octastic Soft-Echo to external SIP adapters

Vazir wrote: How than my CISCO AS5350 gets H323 VOIP and echo-cancels so exelently :)) If it is a plain H323-H323 call then there should not be any echo at all. If you have a PSTN<->VoIP call and the PSTN leg uses analog lines then the PSTN side will create an echo which will be cancell

Re: [asterisk-dev] TLS/SSL futures

d as part of the Linux distribution. It's somehow crazy - it is legal to offer debian packages for an TLS enabled openser, but it is illegal to include these packages to debian itself. Thus, I suspect Asterisk may have the same problems.

Re: [asterisk-dev] SIP Address port remove suggestion

Alexandre Almeida wrote: I have a peer with this configuration, it's used as trunk for outgoing calls. [xxyy] username=123456 type=peer secret=1423241 port=5070 outboundproxy=my.proxy.com host=gateway.server.com workaround: make SRV records which point to port 5070 and remove the port= optio

Re: [asterisk-dev] shutting down zaptel spans

Kevin P. Fleming wrote: - Tzafrir Cohen <[EMAIL PROTECTED]> wrote: When is there a need to shut down zaptel spans from userspace? I'm not aware of any benefit of doing so. What if I want to "reset" an E1 link remotely without stopping asterisk? Sometimes connected PBXs behaves strange a

Re: [asterisk-dev] Re: Advice of charge

Tomislav Parčina wrote: -Original Message- From: Klaus Darilion [mailto:[EMAIL PROTECTED] Sent: 28. rujan 2006 10:29 To: Asterisk Developers Mailing List; Tomislav Parčina Subject: Re: [asterisk-dev] Re: Advice of charge The patch is on the bug tracker. It is not a nice solution

Re: [asterisk-dev] Advice of charge

Only ZAPTEL PRI regards klaus Kai Ober wrote: does this patches provide AOC to mISDN as well? or just PRI CARDS? Kai Klaus Darilion schrieb: http://bugs.digium.com/view.php?id=7495 http://bugs.digium.com/view.php?id=7494 It is in use since 2 month without problems (1.2) regards klaus

Re: [asterisk-dev] Advice of charge

http://bugs.digium.com/view.php?id=7495 http://bugs.digium.com/view.php?id=7494 It is in use since 2 month without problems (1.2) regards klaus Tomislav Parčina wrote: What is current status of advice of charge messages in Asterisk? Will it be full supported in next releases? I would apprecia

Re: [asterisk-dev] Zaptel-1.2.8 compile problem

Vidura Senadeera wrote: Please guide me on this Sir .. How do I get rid of this? Update to Centos 4.4 (or RedHat 4.4) or fix the kernel sources. Google is your friend. regards klaus On 9/4/06, *Kevin P. Fleming* <[EMAIL PROTECTED] > wrote: - Vidu

Re: [asterisk-dev] Asterisk as voice gw to the IMS world

Raphael Jacquot wrote: George Thanos wrote: SIP signalling from IMS is a little bit different from the Asterisk SIP stack. 3GPP has defined some extentions to the protocol. yet another case of "lets break the RFCs for profit" ? AFAIK all 3GPP extensions are RFCs (or on the way to an RFC)

Re: [asterisk-dev] chan_zap questions

Armin Schindler wrote: On Wed, 12 Jul 2006, Klaus Darilion wrote: Maybe an additional subclass AST_CONTROL_AOC should be introduced. Thats what I did and it works fine. But ast_indicate_data requires a ast_channel as first parameter. If FACILITY is received on leg2, I have to find out the

Re: [asterisk-dev] chan_zap questions

Armin Schindler wrote: On Wed, 12 Jul 2006, Klaus Darilion wrote: Armin Schindler wrote: On Tue, 11 Jul 2006, Kevin P. Fleming wrote: - Armin Schindler <[EMAIL PROTECTED]> wrote: I think a defined indication via ast_indicate() (which is forwarded by Asterisk even in app_dial()) wo

Re: [asterisk-dev] chan_zap questions

Armin Schindler wrote: On Tue, 11 Jul 2006, Kevin P. Fleming wrote: - Armin Schindler <[EMAIL PROTECTED]> wrote: I think a defined indication via ast_indicate() (which is forwarded by Asterisk even in app_dial()) would be more appropriate. We've already added the ability to queue control f

Re: [asterisk-dev] Bridging two H324M calls

(), chan_zap.c). BTW, I did the H324M call briding in a v-1.2.4 Asterisk in the UK. regards, Zhuoqun Li Date: Tue, 02 May 2006 11:17:14 +0200 From: Klaus Darilion < [EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>>

Re: [asterisk-dev] Bridging two H324M calls

Hi Sergio! I've done this once and it worked (relaying). But I was not able to record the sessions. When I tried the various "recording" applications the video call setup did not worked anymore. Relaying was only successful when the bridging was done directly on the ISDN card. I did this onc

Re: [asterisk-dev] H324M Gateway

Sergio García Murillo wrote: Just ftp and decode FTP-DATA as h223. That does nto work. My ethereal only offers H245, no h223. I'm using ethereal 0.10.13 on windows. Are you using a special version of ethereal? Ethereal 0.10.14 on windows also.. check in help->about->plugins Protocol field n

Re: [asterisk-dev] H324M Gateway

I've look also for the guys that make the plugin and their company offer 3G video gateway services, so I don't expect to get much help from them.. :( -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Klaus Darilion Sent: martes, 25 de abril d

[asterisk-dev] Zaptel echo canceler as runtime option

Hi! Currently, the echo canceler in zaptel modules is a compile time option. This makes echo tweaking complicated. HAve there ever been thoughts about having all of them available and choose them as zapata.conf option? btw: Why is the echo cannceler not a generic PBX feature? Instead it is

Re: [asterisk-dev] H324M Gateway

Sergio García Murillo wrote: Is anyone currently interested or doing some work about that? I have been investigating a bit and find that the capture files that are in the wiky are correctly parser by ethereal (just send it over tcp, capture and decode as h223, it will decode perfectly).

Re: [asterisk-dev] ENUM changes: discussion

On Wed, February 15, 2006 11:00, Mark Elkins said: > On Tue, 2006-02-14 at 20:44 -0800, John Todd wrote: >> I saw the changes to the ENUMLOOKUP function below, and one of them >> raises a question in my mind. It seems to be going backwards to >> continue using the "enum.conf" file when this is eas

Re: [asterisk-dev] ENUM changes: discussion

Hi! I agree with John. regards Klaus On Wed, February 15, 2006 5:44, John Todd said: > > I saw the changes to the ENUMLOOKUP function below, and one of them > raises a question in my mind. It seems to be going backwards to > continue using the "enum.conf" file when this is easily specified in >

Re: [Asterisk-Dev] startup

Hi! Take a look at the briStuff from junghanns.net. This patch includes res_watchdog.c, which sends keep alive to an ISDN failover switch (via serial connection). This sounds similar to the application you need. klaus himanshoo kumar saxena wrote: Hi all, I am new to asterisk and am lookin

Re: [Asterisk-Dev] Q.956 Advice of Charge (AOC) - basic implementation now in CVS

Hi! Are there any news on Advice of Charge inegration into asterisk? I understand the problem of relaying the AOC at teh end of the call in a bridged scenario. But is it possible to relay AOC-D messages during the call? regards, klaus Frank Sautter wrote: hi, my patches for basic support

Re: [Asterisk-Dev] ENUM multiple records handling

Conroy, Lawrence (SMTP) wrote: Hi Klaus, folks, I beg to differ. Your SIP proxy is no damn good if one of the NAPTRs is voice:X-iax2 (or voice:X-skype :), so it's a dangerous to assume that SIP is the answer, now what was the question? In particular, I *strongly* disagree with the argument that on

[Asterisk-Dev] help needed implementing H324M (UMTS Video)

Hi all! I investigae implementing a UMTS Video <-> SIP Video gateway in astersik. Currently I'm analyzing the asterisk code and reading the specifications (H.324M, H.223). I have several questions how to incorporate this into asterisk and hope that some of you can give me some answers. (If some