Re: [Asterisk-Users] oh323 driver and RFC2833
Which version of the driver do you use? Fernando Herrera wrote: Hello, I have installed oh323 channel driver. Outgoing calls to H.323 world do not include RFC2833 in the outgoing TerminalCapabilitiesSet despite that userInputMode=RFC2833 has already been set. Does anyone know how to make RFC 2833 DTMF relay work over oh323 channel? Kind regards, */Fernando Herrera/* *De:* Fernando Herrera [mailto:[EMAIL PROTECTED] *Enviado el:* Miércoles, 21 de Septiembre de 2005 12:51 *Para:* 'asterisk-users@lists.digium.com' *Asunto:* [Asterisk-Users] Help with asterisk-oh323 driver DV, Have you solved this? I am facing the same problem. I am running Asterisk 1.0.9 and outgoing TCS does not show the receiveRTPAudioTelephonyEventCapability. Kind regards, */Fernando Herrera/* Hi all, Sorry if this has been answered previously, but I have not had any luck trying to find it. I am trying to connect my Asterisk server (1.0 stable, Fedora Core 2, kernel 2.6.8-1.521) to connect to a gateway that can only support H323. I have installed the asterisk-oh323 channel driver (version 0.6.3b) using Open H323 1.13.5 (patched as per asterisk-oh323's instructions) and PWLIB 1.6.6. This is all working fine for very basic call setup and tear down, from any of my SCCP, SIP, H323 or POTS (X100P card) phones. NB: The gateway only handles signalling, so all media will flow between the endpoints and the gateway will handle signalling to the receiving gateway, as such (excuse the dodgy diagram :) ): -[Gateway]--- | | (H323)(H323 or MGCP/ISUP) | | V V [Asterisk]---(RTP)--[Terminating gateway] | (Signalling + RTP) | (Zaptel/SIP/H323/SCCP phones) There are some requirements for me to connect to this switch: 1. I must support H245 tunneling and faststart (working fine) 2. I must dynamically negotiate the codecs (i.e. send multiple codecs as part of the faststart and the softswitch will decide which of the codecs to use based on the terminating gateway's capabilities). The codec picked will be passed back in the return faststart from the gateway. 3. It must support RFC2833 for OOB DTMF. The problems I am facing are that my faststart in my setup messages only ever has one codec, regardless of what I have set in the [codecs] section of oh323.conf, and even if I specify userInputMode=RFC2833 in oh323.conf my TCS does not include the capability receiveRTPAudioTelephonyEventCapability hence RFC2833 is never neogitated. I'm sure this is just a minor tweak of the source code, but not being an expert in C I am having problems figuring out what needs to be done and where. Any help on this matter would be appreciated. Cheers DV ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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/listinfo/asterisk-users
[Asterisk-Users] asterisk-oh323: New versions 0.6.7 and 0.7.3
Hello all, Updated versions of asterisk-oh323 are now available both for use with Asterisk v1-0 (version 0.6.7) and Asterisk HEAD/v1-2 (version 0.7.3). Download from the usual location: http://www.inaccessnetworks.com/projects/asterisk-oh323 Regards, Michael. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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/listinfo/asterisk-users
Re: [Asterisk-Users] OH323 for HEAD? 0.7.1 doesn't compile.
Hi Tony, The new packages of asterisk-oh323 (for STABLE HEAD) are ready to be released on inAccess Networks site. Expect them in the following two or three days. Michael. Tony Mountifield wrote: I have successfully been using OH323 v0.6.5 with Asterisk 1.0.x. I now need to move to CVS HEAD in order to use some features that are not in v1.0.x, and am trying to compile OH323 to use with it. On the InaccessNetworks site, it ways that OH323 v0.7.1 is for HEAD. However, when I compile it, it appears that it hasn't been updated since the channel structures were revamped. I get many errors, starting with the following: chan_oh323.c:37:34: asterisk/channel_pvt.h: No such file or directory chan_oh323.c: In function `oh323_exception': chan_oh323.c:1145: error: structure has no member named `pvt' Has anyone updated chan_oh323 to work with the latest HEAD? Cheers Tony ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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/listinfo/asterisk-users
Re: [Asterisk-Users] Register Asterisk with Gatekeeper - oh323
Hi Steve, Your [general] section looks fine. In the [register] section remove everything else and leave these lines. context=incoming-h323-calls alias=HMA0200.10szxn- alias=22xx2912 alias=HMA0200.10szxn- alias=22xx2913 Now all H.323 calls will enter in 'incoming-h323-call' context. Try this and see if it works. Michael. Steve Ducat wrote: I have tried everything. to register with this gatekeeper to make and receive calls These are the details I received from the voip provider: protocol H.323 Gatekeeper Address - [EMAIL PROTECTED] Port - 1719 RAS - 53 Q931 - 80 h245 - 1722 RTP - 1722 Username - H323 I have 2 phone number/accounts with this gatekeeper that I need to register to. ID - HMA0200.10szxn- e.164 - 22xx2912 ID - HMA0200.10szxn- e.164 - 22xx2913 Here is my oh323.conf: [general] listenAddress=0.0.0.0 listenPort=1720 [EMAIL PROTECTED] gatekeeperTTL=600 tcpStart=1 tcpEnd=2 udpStart=1 udpEnd=2 fastStart=no h245Tunnelling=no h245inSetup=no inBandDTMF=no jitterMin=20 jitterMax=100 ipTos=none outboundMax=10 inboundMax=10 simultaneousMax=10 wrapLibTraceLevel=1 libTraceLevel=0 libTraceFile=stdout userInputMode=TONE amaFlags=default accountCode=H323 language=en context=voip-h323 [register] alias=ASTERISK [codecs] codec=G711A frames=20 [22xx2912] type=friend [EMAIL PROTECTED] port=1720 alias=HMA0200.10szxn- e164=22xx2912 context=default disallow=all allow=ulaw dtmfmode=rfc2833 [22xx2913] type=friend [EMAIL PROTECTED] port=1720 alias=HMA0200.10szxn- e164=22xx2913 context=default disallow=all allow=ulaw dtmfmode=rfc2833 All I get from Asterisk is the following: Aug 29 10:00:57 WARNING[9715]: chan_oh323.c:4228 oh323_gk_check: Failed to register with gatekeeper '[EMAIL PROTECTED]'. -- Retrying gatekeeper registration. Am I on the right track or have I missed the point. I do not want Asterisk to be the gatekeeper, I simply want Asterisk to register with the gatekeeper so I can receive calls from it and then use this gatekeeper to make calls to it. Any help would be appreciated. Thanks Steve.. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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/listinfo/asterisk-users
[Asterisk-Users] asterisk-oh323: New version 0.6.6
Hello all, A new bug-fix release of asterisk-oh323 for the *stable version* of Asterisk is available. This version has the option to compile with latest OpenH323/Pwlib libraries but we recommend to stay with the Janus version. The updated version that is compatible with the *CVS HEAD version* of Asterisk will delay for a while. Download from the usual location: http://www.inaccessnetworks.com/projects/asterisk-oh323 Regards, Michael. ___ 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/listinfo/asterisk-users
Re: [Asterisk-Users] 180 Ringing? (BUG?)
Mirko Marghitola wrote: Asterisk don't send the 180 Ringing SIP message to the calling phone when the called party is ringing. How can I force asterisk to send the ringing messages? The option 'r' in the Dial() command or the Ringing() command didn't solve the problem. Mirko Did the sip channel driver sent a progress when the called phone started ringing? In this case the driver does not send the ringing. Anyway, I don't think this behavior is correct because it breaks other protocols. E.g. if two Asterisks use SIP for their interconnection and talk H.323 with foreign gateways, then the H.323 conversation produced by the conversion H.323 - SIP - H.323 is wrong because the ALERTING of the first H.323 leg won't be generated on the second leg. And according to the H.323 recommendation ALERTING is a mandatory message. Michael. ___ 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/listinfo/asterisk-users
Re: [Asterisk-Users] How to use same h323-conf-id in incoming and outgoing legs?
Papadopoulos Georgios wrote:
Hello,
I am pretty new with Asterisk and I am using it as an H323 gateway.I
would like to keep the same h323-conf-id in the outgoing leg as in the
incoming leg.
So far I have only tried inaccessnetworks' oh323 module, but I think
this is a more generic issue. My extensions.conf is pretty simple:
[oh323_context]
exten = _XX,1,Dial(oh323/${EXTEN},30,tr)
So my question is the following. It seems to me like Dial means start a
new call. Is there some way to just forward an incoming call, in my case
to some other H323 gateway?
asterisk-oh323 does not implement this. The Dial app will spawn
a new H.323 channel with different call-id and conf-id.
thank you
Michael.
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Re: [Asterisk-Users] Chan OH323 and overlapping digits
There is nothing wrong with your config, it is just unimplemented functionality. Michael. Alexander Topolanek wrote: Hi, Perhaps there's something wrong in my config... I did some tests connecting Asterisk to an Ericsson MD110 PBX by setting up an h323 trunk. When dialling into asterisk I got some problems when the entire number is not in the setup message, i.e. I'm dialling digit by digit on the ericsson phone. Lets say I have 4001 in my extensions, and dial that from the Ericsson PBX, then the Ericsson switch is sending a h.225 setup message with a called party number 4. The oh323 channel replies with a h.225 callProceeding Message, which makes the MD110 stop sending further digits. I commented out already the s extension, so no matching pattern is found for a 4. I would have expected the channel to collect digits until a matching pattern is fount or until a timeout. Best regards Alexander ___ 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/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Audio flutter on OH323 output?
Tony Mountifield wrote: In article [EMAIL PROTECTED], I wrote: In article [EMAIL PROTECTED], Michael Manousos [EMAIL PROTECTED] wrote: Can you get an ethereal trace on a call with that problem? Run an RTP analysis on the captured stream (Tools Menu) and save the contents of the RTP packets in audio files. Then check if the playback of these files is normal or not. Aha, sounds like Ethereal has even more clever features - I didn't know it could do that. Fortunately, I've been running a continuous tcpdump capture on the Asterisk box of the traffic between it and the switch. I'll see what Ethereal thinks of it. Hi Tony, Hi Michael, I've now done some RTP analysis of affected streams, and have found that the times when we get the audio flutter correspond with parts of the analysis showing repeated packets in the RTP stream. This only happens on the outgoing stream; the incoming stream from the switch is perfect. When a packet is repeated, it is identical to the previous and occurs about 7 to 14 microsec after, as if something in the software has decided to send the packet twice in immediate succession. This seems as a bug in OpenH323, but also as a bug in the switch's RTP implementation. Normally, the remote RTP stack should reject the duplicate packet. Of course this does not cancel the need to fix the bug in OpenH323. I'll prepare a version which compiles with the latest code of OpenH323/Pwlib. You can try it and see if the problem is still there. I've pasted below an extract from the CVS file saved from the RTP stream analysis. I also saved the payload (pity it converts it to uLaw instead of saving it in the stream's native format) and listened to it to confirm the periods of distortion. I would assume the problem is somewhere in the depths of openh323, but any pointers in the right direction would be appreciated! Cheers Tony Regards, Michael. Here is the CSV extract: Packet,Sequence,Delay (s),Jitter (s),Marker,Status,Date,Length 144209,17353,0.02,0.03,,[ Ok ],05/17/2005 17:05:48.420,214 144225,17354,0.020006,0.03,,[ Ok ],05/17/2005 17:05:48.440,214 144226,17354,0.07,0.03,,Wrong sequence nr.,05/17/2005 17:05:48.440,214 144244,17355,0.019987,0.04,,[ Ok ],05/17/2005 17:05:48.460,214 144260,17356,0.019998,0.04,,[ Ok ],05/17/2005 17:05:48.480,214 144261,17356,0.08,0.04,,Wrong sequence nr.,05/17/2005 17:05:48.480,214 144276,17357,0.019991,0.04,,[ Ok ],05/17/2005 17:05:48.500,214 144296,17358,0.019992,0.04,,[ Ok ],05/17/2005 17:05:48.520,214 144298,17358,0.09,0.05,,Wrong sequence nr.,05/17/2005 17:05:48.520,214 144313,17359,0.019990,0.05,,[ Ok ],05/17/2005 17:05:48.540,214 144329,17360,0.02,0.05,,[ Ok ],05/17/2005 17:05:48.560,214 144349,17361,0.019992,0.05,,[ Ok ],05/17/2005 17:05:48.580,214 144368,17362,0.020003,0.05,,[ Ok ],05/17/2005 17:05:48.600,214 144369,17362,0.06,0.05,,Wrong sequence nr.,05/17/2005 17:05:48.600,214 144383,17363,0.019990,0.05,,[ Ok ],05/17/2005 17:05:48.620,214 144401,17364,0.019995,0.05,,[ Ok ],05/17/2005 17:05:48.640,214 144417,17365,0.020001,0.05,,[ Ok ],05/17/2005 17:05:48.660,214 144432,17366,0.019994,0.05,,[ Ok ],05/17/2005 17:05:48.680,214 144452,17367,0.019997,0.05,,[ Ok ],05/17/2005 17:05:48.700,214 144469,17368,0.020002,0.05,,[ Ok ],05/17/2005 17:05:48.720,214 144486,17369,0.019998,0.05,,[ Ok ],05/17/2005 17:05:48.740,214 144487,17369,0.08,0.05,,Wrong sequence nr.,05/17/2005 17:05:48.740,214 144506,17370,0.019985,0.05,,[ Ok ],05/17/2005 17:05:48.760,214 144523,17371,0.01,0.05,,[ Ok ],05/17/2005 17:05:48.780,214 144540,17372,0.020005,0.05,,[ Ok ],05/17/2005 17:05:48.800,214 144559,17373,0.019992,0.05,,[ Ok ],05/17/2005 17:05:48.820,214 144560,17373,0.08,0.05,,Wrong sequence nr.,05/17/2005 17:05:48.820,214 144576,17374,0.019988,0.06,,[ Ok ],05/17/2005 17:05:48.840,214 144577,17374,0.07,0.06,,Wrong sequence nr.,05/17/2005 17:05:48.840,214 144592,17375,0.019992,0.06,,[ Ok ],05/17/2005 17:05:48.860,214 144593,17375,0.08,0.06,,Wrong sequence nr.,05/17/2005 17:05:48.860,214 144610,17376,0.019987,0.07,,[ Ok ],05/17/2005 17:05:48.880,214 144627,17377,0.019997,0.06,,[ Ok ],05/17/2005 17:05:48.900,214 144628,17377,0.08,0.06,,Wrong sequence nr.,05/17/2005 17:05:48.900,214 144642,17378,0.019989,0.07,,[ Ok ],05/17/2005 17:05:48.920,214 144661,17379,0.019997,0.07,,[ Ok ],05/17/2005 17:05:48.940,214 144678,17380,0.020004,0.06,,[ Ok ],05/17/2005 17:05:48.960,214 144679,17380,0.09,0.07,,Wrong sequence nr.,05/17/2005 17:05:48.960,214 144695,17381,0.019988,0.07,,[ Ok ],05/17/2005 17:05:48.980,214 144696,17381,0.07,0.07,,Wrong sequence nr.,05/17/2005 17:05:48.980,214 144713,17382,0.019982,0.08,,[ Ok ],05/17/2005 17:05:49.000,214 144732,17383,0.020002,0.07,,[ Ok ],05/17/2005 17:05:49.020,214 144733,17383,0.08,0.07,,Wrong sequence nr.,05
Re: [Asterisk-Users] asterisk-oh323: Max simultaneous calls ?
Vamsi Pottangi wrote: Hi All, There is a parameter simultaneousMax=10 in oh323.conf. Had anybody tried out what is the maximum value that can be achieved ? What is the maximum number of simultaneous h323 calls can the oh323 driver can handle. I tried to get it only till 30 to 40 simultaneous calls. Anybody achieved better figures than this ? or have any idea how the oh323 can be tuned to get better values ? Something around 100 is currently the upper bound. Thanks, ~Vamsi Michael. ___ 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/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk-oh323 build problems
What versions of OpenH323/Pwlib/asterisk-oh323 are you trying to install? Michael. FaberK wrote: Hello Guys, first of all, I'm very new with asterisk. I'm trying to set it up. I've already compiled and installed Asterisk-1.0.7 Now I'm trying with asterisk-oh323 I've already installed pwlib, oh323 and I've already set the variables. Now, when I try to make asterisk-oh323 I receive this error messagge: for x in wrapper asterisk-driver; do make -C $x all || exit 1 ; done make[1]: Entering directory `/root/voip/asterisk/asterisk-oh323/wrapper' g++ -Wall -mcpu=i586 -DP_LINUX -D_REENTRANT -DP_HAS_SEMAPHORES -DP_SSL -DP_PTHREADS -DPBYTE_ORDER=PLITTLE_ENDIAN -DPHAS_TEMPLATES -O3 -DNDEBUG -I/usr/include -I/usr/include/crypto -I/usr/lib/pwlib/include/ptlib/unix -I/usr/lib/pwlib/include -I/usr/lib/openh323/include -I../asterisk-driver -g -c wrapper.cxx -o wrapper.o wrapper.cxx: In constructor `WrapH323Connection::WrapH323Connection(WrapH323EndPoint, unsigned int, int, int, short unsigned int)': wrapper.cxx:563: `SetMaxAudioDelayJitter' undeclared (first use this function) wrapper.cxx:563: (Each undeclared identifier is reported only once for each function it appears in.) wrapper.cxx: In function `call_ret_val_t h323_clear_call(const char*)': wrapper.cxx:1230: warning: unused variable `ClearCallThread*clearCallThread' make[1]: *** [wrapper.o] Error 1 make[1]: Leaving directory `/root/voip/asterisk/asterisk-oh323/wrapper' make: *** [subdirs_all] Error 1 What's wrong? Thanks ___ 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/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk H323 Trunk Zone
Mahmoud Badran wrote: AVE! i am trying to register h323 asterisk to the gatekeeper as i installed asterisk, libpri, zaptel from CVS, and pwlib, openh323, asterisk-oh323 on fedora core3 on a cisco mcs 7800 server problem is i want the asterisk to register with gatekeeper endpoint with specific zone name and type... i searched the web, mail list but there weren't any helpful ones could anyone plz tell me how to specify the zone name and type?? You can specify the gatekeeper to use by the zone name using the following in oh323.conf: gatekeeper=GKID:plase-here-the-zonename e.g. if the zone name is MyInternalZone gatekeeper=GKID:MyInternalZone I'm not sure about the type. Michael. ___ 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/listinfo/asterisk-users
Re: [Asterisk-Users] Audio flutter on OH323 output?
Tony Mountifield wrote: Hi, I'm using OH323, mostly with success, to interface Asterisk to a provider's switch (World Telecom INX). I have noticed a particular effect, and I wonder whether anyone else has seen the same? The effect is audio flutter (almost like the flutter one gets on MF or HF radio sometimes) which only happens intermittently. Audio coming into Asterisk is unaffected, as proved by using the Monitor app as follows: Phone1-PSTN-Switch-(via H.323)-Asterisk(Monitor+DISA)-Switch-PSTN-Phone2. Intermittently, each party hears the other party's audio flutter for a few seconds. Reviewing the recordings made by Monitor, no flutter is present, so the incoming audio is fine. Note that this is a direct call. I've also noticed it on MeetMe, where it seems again that the flutter is on the audio leaving Asterisk. Different participants may hear the flutter at different times. The system is a dual-Xeon 3GHz running Fedora Core 3 with the STABLE branch of Asterisk from CVS, together with oh323 0.6.5, openh323 1.13.5.3 and pwlib 1.6.6.3. Any suggestions would be appreciated! Can you get an ethereal trace on a call with that problem? Run an RTP analysis on the captured stream (Tools Menu) and save the contents of the RTP packets in audio files. Then check if the playback of these files is normal or not. Cheers Tony Michael. ___ 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/listinfo/asterisk-users
Re: [Asterisk-Users] Scalability of chan_oh323
Alistair Cunningham wrote: Michael Manousos wrote: Alistair Cunningham wrote: I have a customer who wants to do large volumes of H.323 to H.323 hairpinning. We haven't tested this scenario for large volumes before; maybe someone on asterisk-users has. If they buy a top of the line PC, how many concurrent calls are we likely to get? Routing logic will be simple, the machine won't be doing anything else, and let's assume no transcoding for now. We're not looking for an exact figure at this point, just a rough estimate for cost / benefit of Asterisk versus a proprietary system. Currently, without transcoding, you can get maximum 100 simultaneous H.323 channels per box. With the next release of asterisk-oh323 this number will be raised to ~180 channels. After that, major optimizations at the OpenH323 RTP/jitter buffer code are required to push this number up. Michael. That's a shame; my customer probably needs 400 to 500 channels (200 to 250 calls). I know but the performance bound is set by OpenH323. Also, an additional note, these numbers are almost the same even when transcoding (G.729 - G.711) is used. Does anyone have experience of GNU Gatekeeper in proxy mode? Any idea of what load it can handle? GNUGK in full proxy mode (all signaling and RTP through it) has a similar upper limit. Michael. ___ 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/listinfo/asterisk-users
Re: [Asterisk-Users] oh323 Zone
Sebastian Atala wrote: Hi, Someone knows how can I register my Asterisk to a gatekeeper using zone parameters? I'm using asterisk 1.0.7 and oh323 0.6.5. I'm trying to register to a gatekeeper in another network and I can't reach this with a broadcast. Zone is the name who Cisco call the GK identification. In oh323.conf set: gatekeeper=GKID:zone_name e.g. if lala is the zone name: gatekeeper=GKID:lala Thank in advance SA Michael. ___ 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/listinfo/asterisk-users
Re: [Asterisk-Users] ASterisk OH323.CONF Gateway Gatekeeper
Jorge Alayon wrote:
Hi,
Does anybody knows how to konfigure oh323.conf to allow calls comming from a
peering gateway (i.e.: Cisco 5300) which is not connected to a gatekeeper,
and also from the gatekeeper to which Asterisk is registered ?
Nothing special here. Configure the channel driver with the gatekeeper
selected. For incoming calls there are no other things to check. For
outgoing calls, if they are heading to the GK dial:
exten = _.,1,Dial(OH323/${EXTEN})
If they are heading to the gateway dial:
exten = _.,1,Dial(OH323/${EXTEN}@gw_ip)
Something like:
GK(Carrier1)Registered to:-AS5300(carrier 1)-peer
GW2GW--Asterisk---registered to:---GK(Carrier2)
I Would like to receive calls from both carriers. Registering AS5300 to GK
from carrier 2 is not an option.
Regards,
Jorge A.
Michael.
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Re: [Asterisk-Users] OH323 incoming audio stutter
Hi Tony, Can you get an ethereal trace of the RTP packets on both directions? A short analysis of those streams (from within the ethereal tools) would help us find the problem. Michael. Tony Mountifield wrote: I'm using asterisk-oh323-0.6.5 with the Janus patch 4 versions of pwlib (v1.6.6.3) and openh323 (v1.13.5.3), and using it to connect to my provider's switch. The effect that I am seeing is that a call starts off fine, but suddenly after a few minutes the audio coming into Asterisk via OH323 gets very broken up to the point of being about 90% silence with occasional brief snippets of audio getting through. When this happens, the audio going out from Asterisk to the other end is still fine, with no disturbances. I have observed this both when using SIP for the local leg of the call and when using IAX. I'm not really sure where to look to diagnose this, not whether it is likely to be an Asterisk problem or something in the switch. Any advice would be appreciated! Cheers Tony ___ 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/listinfo/asterisk-users
Re: [Asterisk-Users] OH323 and Asterisk CVS-HEAD-03/21/05-15:32:10
Try the 0.7.2-pre1 version of asterisk-oh323. It can be found at the Download section on the home page of asterisk-oh323. Michael. Jose R. Ortiz Ubarri wrote: I have problems compiling the OH323 channel with Asterisk CVS-HEAD-03/21/05-15:32:10. I have the following errors. chan_oh323.c:4895: warning: passing arg 1 of `ast_channel_unregister' from incompatible pointer type chan_oh323.c: In function `load_module': chan_oh323.c:5192: warning: passing arg 1 of `ast_channel_register' from incompatible pointer type chan_oh323.c:5192: error: too many arguments to function `ast_channel_register' make[1]: *** [chan_oh323.o] Error 1 make[1]: Leaving directory `/root/asterisk-oh323-0.7.1/asterisk-driver' make: *** [subdirs_build] Error 1 Looks like a compatibility problem with the asterisk functions. Had they changed? I followed the instructions at http://www.oinko.net/astrecipes/index.php?action=artikelcat=270174id=10artlang=en. And I had oh323 working before with a previous version of asterisk... Anyone else had the same problem??? Thanks for help, JO ___ 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/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Problems trying to compile H323 from CVS-STABLE
Tony Mountifield wrote: Yesterday I wrote: I'm trying to compile channels/h323 and chan_h323 from CVS-STABLE, on Fedora Core 3. [... snip ...] Well I gave up with chan_h323, which is a pity, because it should be the solution that is better integrated with Asterisk. I would still like to hear from anybody that has any ideas (please see my original post). Instead, I downloaded asterisk-oh323-0.6.5 from InAccessNetworks, along with Janus-patch4 of PWlib (1.6.6.3) and OpenH323 (1.13.5.3). Following the instructions exactly, installation went smoothly, and worked first time. When testing the ability of dual 3GHz Xeons to handle many simultaneous OH323 calls (G.711 so no heavy transcoding), I discovered that chan_oh323 is EXTREMELY profligate with file descriptors! Each open oh323 channel uses 21 fds, yes TWENTY-ONE! In order to handle upwards of 120 simultaneous calls I needed to increase the per-process file descriptor limit from the default of 1024, using the technique described at: http://www.xenoclast.org/doc/benchmark/HTTP-benchmarking-HOWTO/node7.html I then added ulimit -n 8192 to /usr/sbin/safe_asterisk. It seems to be working ok now, but I'd still like to get chan_h323 working sometime, as I have a feeling it will be much less hungry for file descriptors! Comments, anyone? We are working on pushing this number down. Be patient! Michael. ___ 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/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Problems trying to compile H323 from CVS-STABLE
Tony Mountifield wrote: In article [EMAIL PROTECTED], Michael Manousos [EMAIL PROTECTED] wrote: Tony Mountifield wrote: When testing the ability of dual 3GHz Xeons to handle many simultaneous OH323 calls (G.711 so no heavy transcoding), I discovered that chan_oh323 is EXTREMELY profligate with file descriptors! Each open oh323 channel uses 21 fds, yes TWENTY-ONE! We are working on pushing this number down. Be patient! I'll look forward to it - thanks! It would be nice if any such improvements are made available in a version compatible with Stable as well as with Head. We maintain versions compatible with both the stable and HEAD branches of Asterisk. Michael. ___ 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/listinfo/asterisk-users
Re: [Asterisk-Users] oh323 compilation
Gabriel Millerd wrote: I have been struggling with oh323 compilation for some time now. I am trying to use the voip-info suggested walk through that points to here ... http://www.oinko.net/astrecipes/index.php?action=artikelcat=270174id=10artlang=en ... which asks for versions OpenH323 (v1.13.5) PWlib (v1.6.6). Anyone know how to get these? The website http://www.inaccessnetworks.com/projects/asterisk-oh323/Libraries Actually the 1.13.x/1.6.x series is named Janus, so the Janus libraries that we have on the site are the right ones. Michael. ___ 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/listinfo/asterisk-users
Re: [Asterisk-Users] *** Asterisk 2.0 Stable release out now
Olle E. Johansson wrote: 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. AAFJ as in Asterisk April Fool's Joke? Nice :) ___ 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/listinfo/asterisk-users
Re: [Asterisk-Users] H.323 call '...' cleared, reason 15 (Call ended due to security checks)
Cenk Yabas wrote: Thanks to Yves's commitment I was able to configure oh323 channel, cleared the codec issue, registered to Gatekeeper, placed a call, but receive this message on the console. What might be the problem? Asterisk Ready. *CLI -- Registered with gatekeeper '[EMAIL PROTECTED]'. -- Executing Dial(SIP/2000-5a52, OH323/193.192.100.92/0212441) in new stack -- H.323 call to 193.192.100.92/0212441 with codec(s) g729 -- Called 193.192.100.92/0212441 -- H.323 call 'ip$localhost/5502' cleared, reason 15 (Call ended due to security checks) The gatekeeper has cleared the call. I guess because a password is required or the one provided is not correct. What version of the channel driver do you use? Michael. ___ 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/listinfo/asterisk-users
[Asterisk-Users] asterisk-oh323 pre-releases
I have prepared two new, not-final yet, releases of asterisk-oh323: - 0.6.6-pre1 for Asterisk stable - 0.7.2-pre1 for Asterisk CVS HEAD They can be found at: http://www.inaccessnetworks.com/projects/asterisk-oh323/download Please try them and report problems at the bugtracker of the channel driver at: https://skylab.inaccessnetworks.com/mantis Regards, Michael. ___ 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/listinfo/asterisk-users
Re: [Asterisk-Users] Re: chan_oh323.c ast_oh323_new Internal channel initialization failed
Hi Kamran, Kamran Ahmad wrote: hello i was searching for solution to problem (sip-h.323). any one from this list asterisk mailing have any idea how to fix it. i am getting error when i try to call from sip to h.323 user i am successfully registering my asterisk box with gnugk. but when i try to call to h.323 openphone on working on GnuGatekeeper, asterisk is not routing it to GnuGk. i am getting the following error. do you have any idea. please help i am stuck here for a week. i am unable to find anything on google on this topic. Two things: -- Executing Dial(SIP/2000-ae3f, OH323/[EMAIL PROTECTED]:1720) in new stack Since Asterisk has registered in gnugk you must not dial user@host, just user. It will find the user at the gatekeeper. Mar 16 16:14:46 ERROR[16176]: chan_oh323.c:2501 ast_oh323_new: Internal channel initialization failed. Bad binary? This is bad! Usually this happens when you uncomment flags in asterisk-oh323 Makefile while Asterisk compiled without these flags (or vice versa). So make sure that you didn't do something like that. Michael. ___ 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/listinfo/asterisk-users
[Asterisk-Users] asterisk-oh323 bugtracker
Hello all, In an attempt to make easier and more effective the management of the various issues/features/bugs of asterisk-oh323, I have setup a bugtracker at: https://skylab.inaccessnetworks.com/mantis Please direct all the bug reports and contributed patches there. Thanks, Michael. ___ 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/listinfo/asterisk-users
Re: [Asterisk-Users] problem : undefined symbol.
Kim Daeyong wrote: I downloaded asterisk to use cvs to checkout the release version. After installing, I would like to load module chan_h323.so but there is some error : *CLI load chan_h323.so Feb 17 15:22:38 WARNING[2865]: loader.c:258 ast_load_resource: /usr/lib/asterisk/m odules/chan_h323.so: undefined symbol: __use_ast_pthread_create_instead__ Unable to load module chan_h323.so *CLI How can I solve that problem? Did you try asterisk-oh323? http://www.inaccessnetworks.com/projects/asterisk-oh323 Michael. ___ 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/listinfo/asterisk-users
Re: [Asterisk-Users] stable combination of versions for asterisk and chan_oh323?
Roger Schreiter wrote: Michael Manousos schrieb: ... Use Asterisk-1.0.3 with asterisk-oh323-0.6.5. Hi, may I ask, whether that combination runs really stable at your machine? I have now those versions installed. I have asterisk crashes at least once every hour, when several simultanious calls take place. I run Asterisk-stable/asterisk-oh323-0.6.5 without a single problem. I do a lot of brutal tests on this installation and there are no crashes. Can you send a backtrace of the core file dumped? Roger. Michael. ___ 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/listinfo/asterisk-users
Re: [Asterisk-Users] stable combination of versions for asterisk and chan_oh323?
asterisk-oh323-0.7.0 is for Asterisk CVS. How did you manage to compile it with Asterisk-1.0.3? Use Asterisk-1.0.3 with asterisk-oh323-0.6.5. Michael. Roger Schreiter wrote: Hi, which is currently a stable combination of asterisk and asterisk-oh? The combination of asterisk-1.0.3 and asterisk-oh-0.7.0 is not stable at all and crashes approx once the hour when having approx 3 simultanious calls. Thanks for telling me your experience! Roger. ___ 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/listinfo/asterisk-users
Re: [Asterisk-Users] Error on compiling oh323 0.6.5 on cvs stable asterisk
Robert Rozman wrote: Hi, I have downloaded files and also local versions of pwlib oh323 (both Janus patched). Both libraries compile fine, but I get following errors on asterisk-oh323-0.6.5. Readme is a bit confusing since it doesn't mention which local libraries should be downloaded from inaccess to get everything working OK. I've also tried with/without patching oh323 with supplied patch. Any hint, advice ? Thanks in advance, regards, Rob. centrala:~/Asterisk/h323/asterisk-oh323-0.6.5 # make for x in wrapper asterisk-driver; do make -C $x build || exit 1 ; done make: *** No rule to make target `ccflags'. Stop. make: *** No rule to make target `ccflags'. Stop. make[1]: Entering directory Get the Janus_patch4 libraries. Also, from the attached log, it seems that you didn't apply the OpenH323 patch. Michael. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] reason 24 (Call ended with Q.931 cause)
Hi, Enable the driver tracing (see wrapTrace* and libTrace* in oh323.conf), re-run and send me the output file. Michael. Tola Ogunsan wrote: Hi Michael and Everyone I'm trying to connect Asterisk to a CISCO AS5350 using oh323 and I'm getting this error reason 24 (Call ended with Q.931 cause) I've checked the Asterisk wiki and several other resources. Please can anyone give me a hint on what the problem is I reach my wits end. Thanks Tola my config and debug Configuration of OpenH323 channel driver -- Version: 0.7.1 Listening on address: 0.0.0.0:1720 Gatekeeper used: No gatekeeper FastStart/H245Tunnelling/H245inSetup: OFF/OFF/OFF Supported formats in pref. order: g7290 Jitter buffer limits (min/max): 20-500 ms TCP port range: 1 - 2 UDP (RAS) port range: 1 - 2 UDP (RTP) port range: 1 - 2 IP Type-of-Service value: 0 User input mode: 3 Max number of inbound H.323 calls: 10 Max number of outbound H.323 calls: 10 Max number of simultaneous H.323 calls: 10 Max call rate (ingress direction): 1.00/30 Starting simple switch on 'Zap/3-1' -- Executing Wait(Zap/3-1, 1) in new stack -- Executing Dial(Zap/3-1, OH323/[EMAIL PROTECTED]|10) in new stack -- H.323 call to [EMAIL PROTECTED] with codec(s) g729 Outbound H.323 call 'ip$localhost/263'. -- Called [EMAIL PROTECTED] Call 'ip$localhost/263' cleared. -- H.323 call 'ip$localhost/263' cleared, reason 24 (Call ended with Q.931 cause) Call 'ip$localhost/263' cleared in INIT state. -- OH323/L263 is busy -- Hungup 'OH323/L263' == Everyone is busy/congested at this time (1:1/0/0) -- Executing Hangup(Zap/3-1, ) in new stack == Spawn extension (incoming, s, 3) exited non-zero on 'Zap/3-1' -- Hungup 'Zap/3-1' Call 'ip$localhost/263' without owner has already been cleared (2). -- Starting simple switch on 'Zap/3-1' ___ 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/listinfo/asterisk-users ___ 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/listinfo/asterisk-users
Re: [Asterisk-Users] Compiling H323 channels with FC3 and RedhatSE3
Hi Nicolas, Andrew McRory has done some some job towards packaging asterisk-oh323. The packages are available at: ftp://ftp.linuxsys.com/pub/releases/ You could start with his packages and then move on. Michael. Nicolas FOURNIL wrote: Hello I'm trying for a while to compile and install OH323 channels on my two distribs... I have downloaded the src pwlib and h323 files versions given in the documentation. Make some RPMS with googled SPECs (and seems to give good results) Tried to compile the channels failed each time... (I have also tried with at-rpms oh323 and pwlib versions). Did someone who have already done the job could help me ? -I'm looking for working specs to compile pwlib and oh323- Thanks Nicolas. ___ 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/listinfo/asterisk-users
Re: [Asterisk-Users] H323 Softphone for iPAQ
Also the following has worked great for me: http://www.wifive.net/introduction.asp Michael Radovan Mihalik wrote: http://www.sjlabs.com/sjp.html SJphoneR is a VOIP softphone that allows you to speak with any PC, PDA, stand-alone IP-phone and with any legacy wired or mobile phone (using your VOIP gateway or purchasing service from Internet Telephony Service Provider). It supports both SIP and H.323 standards and is fully interoperable with most major IP-telephony vendors and ITSP. I'm just about to try it my self ;) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Walid Azab Sent: Sunday, January 16, 2005 8:25 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] H323 Softphone for iPAQ Hi list, I was just wondering, is there any H.323 soft-phone that can be installed on a pocket PC (iPAQ). Walid ___ 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/listinfo/asterisk-users ___ 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/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk-oh323 and outgoing call
Alexander Averyanov wrote: Hello. I'm try to set up asterisk for making outgoing calls with oh323 channel driver version 0.7.1 with Asterisk CVS-1-01/09/05-01:41:37. Our provider uses Mera MVTS softswitch and supports only H.323. We don't use gatekeeper for connection but provider requires SOURCE PHONE NUMBER for route out calls and I don't know how I can specify this number. Call with this string exten = _XXX,1,Dial,OH323/[EMAIL PROTECTED] returns -- H.323 call 'ip$localhost/12715' cleared, reason 11 (Gatekeeper could not find user) Please help! How can I supply source phone number for oh323? Use the SetCallerID() app in the dialplan. Michael. ___ 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/listinfo/asterisk-users
Re: [Asterisk-Users] oh323 context for peers
Adi Linden wrote: I am experimenting with oh323 channels and h.323 gateways and a Cisco CallManager. I am not using a gatekeeper at this time. Is it possible to place calls coming into Asterisk from specific peers into specific contexts? Yes. You can associate called numbers/prefixes with contexts in the [register] section of oh323.conf. If you want to send all calls to number AAA coming from H.323 endpoints into Asterisk, then you add the following block in the [register] section: context=test alias=AAA Per H.323 endpoint configuration options in oh323.conf is something under development. In iax.conf eaxh peer has a context in which I can specify the context an inbound call will be placed in. I don't see anything like this in the oh323.conf file or the oh323 documentation. Thanks, Adi Michael. ___ 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/listinfo/asterisk-users
Re: [Asterisk-Users] oh323 context for peers
The new configuration style of OH323 will simplify the sections of
the dialplan that handle H.323 calls.
Michael.
Roger Schreiter wrote:
Adi Linden schrieb:
...
In iax.conf eaxh peer has a context in which I can specify the
context an
inbound call will be placed in. I don't see anything like this in the
oh323.conf file or the oh323 documentation.
Hi,
there is a workaround what is doing this job in most cases:
Use as general context in oh323.conf something like e.g.
context=oh323
Then in extensions.conf or better in a file like oh323peers.conf
included in extensions.conf switch to contexts per peer via gotoifs:
e.g.
[oh323]
;; below here are all peers with fixed ip addresses
exten = _.,1,gotoif,$[${OH323_RADDR} = \
1.2.3.4]?peer01|BYEXTENSION|1
; below only traffic from our gatekeeper
exten = _.,2,gotoif,$[${OH323_RADDR} != \
5.6.7.8]?4
;; below here are all peers which prefer authenticating
; by H.323 username and passoword via our gatekeeper
exten = _.,3,gotoif,$[${CALLERIDNAME} = \
peer02]?peer02|BYEXTENSION|1
;;; everyone who arrived here is not authorized
exten = _.,4,noOp(no such peer)
exten = _.,5,hangup
Wheras 1.2.3.4 is the fixed remote ip address from
your peer1.
5.6.7.8 is the ip address from your gatekeeper (e.g. gnugk)
which verifies the H.323 usernames via passwords.
You can expand this example with further fixed ip address
and further users, which authenticate via your gatekeeper.
Roger.
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Re: [Asterisk-Users] One-way audio in incoming calls with Asterisk + OpenGK + Innovaphone IP3000
Silviu Herchi wrote: Hello everybody, Ive been pulling my hair for a week now over a problem, and I really dont know where to look anymore. Heres my setup: There is an Innovaphone IP3000 VoIP gateway on the LAN (10.253.30.254). I can use it to send and receive calls from physical phones attached to it. I have setup Asterisk 1.0.3, with H323 and openH323, and on the same server I also setup GnuGK (10.253.30.1). I use SIP soft phones connected to the Asterisk (SJphone). Both the Innovaphone and Asterisk are configured to register on OpenGK as gateways. They do it correctly, and phone signalling is OK (phones ring in both directions, and can be picked up OK). When I call from the softphones to the hard phones, everything is OK. When I call the other way round (hard to softphones), I only have audio incoming (from hard to softphones)! All the machines are on the same LAN, attached to a hub. There is no firewall running on them. I tried switching from the supplied H323 Asterisk channel to OH323. I tried GnuGK in routed mode, in proxy mode, etc. to no avail. I also tried sniffing the communication with Ethereal. With what version of OH323 did you try? Michael. ___ 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/listinfo/asterisk-users
Re: [Asterisk-Users] RE: One-way audio in incoming calls with Asterisk + OpenGK + Innovaphone IP3000
Silviu Herchi wrote: Sorry, I mistakenly sent my mail before it was complete... Here it is again. -- Subject: One-way audio in incoming calls with Asterisk + OpenGK + Innovaphone IP3000 Hello everybody, Ive been pulling my hair for a week now over a problem, and I really dont know where to look anymore. Heres my setup: There is an Innovaphone IP3000 VoIP gateway on the LAN (10.253.30.254). I can use it to send and receive calls from physical phones attached to it. I have setup Asterisk 1.0.3, with H323, and on the same server I also setup GnuGK (10.253.30.1). I use SIP soft phones connected to the Asterisk (SJphone on 10.253.30.10) Both the Innovaphone and Asterisk are configured to register on OpenGK as gateways. They do it correctly, and phone signalling is OK (phones ring in both directions, and can be picked up OK). When I call from the softphones to the hard phones, everything is OK. When I call the other way round (hard to softphones), I only have audio incoming (from hard to softphones)! All the machines are on the same LAN, attached to a hub. There is no firewall running on them. I tried switching from the supplied H323 Asterisk channel to OH323. I tried GnuGK in routed mode, in proxy mode, etc. to no avail. I also tried sniffing the communication with Ethereal but it beats me. Here is a H.323 debug from Asterisk (with the H323 module shipped with Asterisk). The calls originated at 10.253.30.102, a hardphone - called party is 377 which rings a softphone user (silviu.herchi on 10.253.30.10). The one strange thing I noticed is the 127.0.0.1 when the second logical link is established, but I'm not sure it really is a problem. (reminder: I have the gatekeeper on the same server as the Asterisk...) Software versions: Asterisk 1.0.2 and 1.0.3 Pwlib 1.5.2 OpenH323 1.12.2 GnuGK 2.2.0 H323 module for Asterisk shipped with version 1.0.2 and 1.0.3 Asterisk-OH323 0.6.4 asterisk-oh323 0.6.4 has a known bug that may cause problems in call answering and one-way audio. Please retry with the 0.6.5 version. Thank you for your help. Silviu deleted... Michael. ___ 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/listinfo/asterisk-users
[Asterisk-Users] asterisk-oh323: New versions available
Hello all, The new versions 0.7.1 (for Asterisk CVS HEAD) and 0.6.5 (for Asterisk STABLE) fix a deadlock in outgoing H.323 calls and a bug that caused chan_oh323 to update incorretly the DIALSTATUS variable. Download from the usual location: http://www.inaccessnetworks.com/projects/asterisk-oh323 Regards, Michael. ___ 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/listinfo/asterisk-users
Re: [Asterisk-Users] OH323 channel compile error
Hi, Rafael J. Risco G.V. wrote: Hello I am trying to compile asterisk-oh323-0.7.0 with pwlib-Janus_patch4 and openh323-Janus_patch4 downloaded from inaccessnetworks so I did this: tar -zxvf openh323-Janus_patch4-src-tar.gz cd openh323 patch -p1 /root/asterisk-oh323-0.7.0/openh323_1.13.5-make.patch ./configure make opt cd asterisk-oh323-0.7.0 vi Makefile (to set the paths and options according to my system...) NOW I HAVE THIS ERROR: [EMAIL PROTECTED] asterisk-oh323-0.7.0]# make for x in wrapper asterisk-driver; do make -C $x build || exit 1 ; done make[1]: Entering directory `/root/asterisk-oh323-0.7.0/wrapper' ./check_ver /root/pwlib pwlib ./check_ver /root/openh323 openh323 g++ -DP_USE_PRAGMA -ffunction-sections -fdata-sections -D_REENTRANT -DOPENSSL_NO_KRB5 -Wall -fPIC -I/root/pwlib/include -DPTRACING -I/root/openh323/include -DHAS_OSS -Wall -x c++ -Os -DWRAPTRACING -DWRAPTRACING_LEVEL=5 -DPWLIBVERSION=\1.6.6\ -DOPENH323VERSION=\1.13.5\ -I/root/pwlib/include/ptlib/unix - I/root/pwlib/include -I/root/openh323/include -I/root/openh323/include/openh323 -I../asterisk-driver -c wrapper_misc.cxx -o wrapper_misc.o touch ../asterisk-driver/chan_oh323.c g++ -DP_USE_PRAGMA -ffunction-sections -fdata-sections -D_REENTRANT -DOPENSSL_NO_KRB5 -Wall -fPIC -I/root/pwlib/include -DPTRACING -I/root/openh323/include -DHAS_OSS -Wall -x c++ -Os -DWRAPTRACING -DWRAPTRACING_LEVEL=5 -DPWLIBVERSION=\1.6.6\ -DOPENH323VERSION=\1.13.5\ -I/root/pwlib/include/ptlib/unix - I/root/pwlib/include -I/root/openh323/include -I/root/openh323/include/openh323 -I../asterisk-driver -c asteriskaudio.cxx -o asteriskaudio.o touch ../asterisk-driver/chan_oh323.c g++ -DP_USE_PRAGMA -ffunction-sections -fdata-sections -D_REENTRANT -DOPENSSL_NO_KRB5 -Wall -fPIC -I/root/pwlib/include -DPTRACING -I/root/openh323/include -DHAS_OSS -Wall -x c++ -Os -DWRAPTRACING -DWRAPTRACING_LEVEL=5 -DPWLIBVERSION=\1.6.6\ -DOPENH323VERSION=\1.13.5\ -I/root/pwlib/include/ptlib/unix - I/root/pwlib/include -I/root/openh323/include -I/root/openh323/include/openh323 -I../asterisk-driver -c wrapconnection.cxx -o wrapconnection.o touch ../asterisk-driver/chan_oh323.c g++ -DP_USE_PRAGMA -ffunction-sections -fdata-sections -D_REENTRANT -DOPENSSL_NO_KRB5 -Wall -fPIC -I/root/pwlib/include -DPTRACING -I/root/openh323/include -DHAS_OSS -Wall -x c++ -Os -DWRAPTRACING -DWRAPTRACING_LEVEL=5 -DPWLIBVERSION=\1.6.6\ -DOPENH323VERSION=\1.13.5\ -I/root/pwlib/include/ptlib/unix - I/root/pwlib/include -I/root/openh323/include -I/root/openh323/include/openh323 -I../asterisk-driver -c wrapendpoint.cxx -o wrapendpoint.o touch ../asterisk-driver/chan_oh323.c g++ -DP_USE_PRAGMA -ffunction-sections -fdata-sections -D_REENTRANT -DOPENSSL_NO_KRB5 -Wall -fPIC -I/root/pwlib/include -DPTRACING -I/root/openh323/include -DHAS_OSS -Wall -x c++ -Os -DWRAPTRACING -DWRAPTRACING_LEVEL=5 -DPWLIBVERSION=\1.6.6\ -DOPENH323VERSION=\1.13.5\ -I/root/pwlib/include/ptlib/unix - I/root/pwlib/include -I/root/openh323/include -I/root/openh323/include/openh323 -I../asterisk-driver -c wrapper.cxx -o wrapper.o wrapper.cxx: In function `call_ret_val_t h323_clear_call(const char*)': wrapper.cxx:1230: warning: unused variable `ClearCallThread*clearCallThread' touch ../asterisk-driver/chan_oh323.c g++ -DP_USE_PRAGMA -ffunction-sections -fdata-sections -D_REENTRANT -DOPENSSL_NO_KRB5 -Wall -fPIC -I/root/pwlib/include -DPTRACING -I/root/openh323/include -DHAS_OSS -Wall -x c++ -Os -DWRAPTRACING -DWRAPTRACING_LEVEL=5 -DPWLIBVERSION=\1.6.6\ -DOPENH323VERSION=\1.13.5\ -I/root/pwlib/include/ptlib/unix - I/root/pwlib/include -I/root/openh323/include -I/root/openh323/include/openh323 -I../asterisk-driver -c wrapcaps.cxx -o wrapcaps.o touch ../asterisk-driver/chan_oh323.c g++ -DP_USE_PRAGMA -ffunction-sections -fdata-sections -D_REENTRANT -DOPENSSL_NO_KRB5 -Wall -fPIC -I/root/pwlib/include -DPTRACING -I/root/openh323/include -DHAS_OSS -Wall -x c++ -Os -DWRAPTRACING -DWRAPTRACING_LEVEL=5 -DPWLIBVERSION=\1.6.6\ -DOPENH323VERSION=\1.13.5\ -I/root/pwlib/include/ptlib/unix - I/root/pwlib/include -I/root/openh323/include -I/root/openh323/include/openh323 -I../asterisk-driver -c wrapgkserver.cxx -o wrapgkserver.o touch ../asterisk-driver/chan_oh323.c ar rc liboh323wrap_s.a wrapper_misc.o asteriskaudio.o wrapconnection.o wrapendpoint.o wrapper.o wrapcaps.o wrapgkserver.o make[1]: Leaving directory `/root/asterisk-oh323-0.7.0/wrapper' make[1]: Entering directory `/root/asterisk-oh323-0.7.0/asterisk-driver' gcc -Wall -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -I/root/asterisk/include -I../wrapper -g -c -o chan_oh323.o chan_oh323.c chan_oh323.c: In function `oh323_call': chan_oh323.c:1421: structure has no member named `cid' chan_oh323.c:1421: structure has no member named `cid' chan_oh323.c:1423: structure has no member named `cid' chan_oh323.c:1435: structure has no member named `cid' chan_oh323.c:1437: structure has no member named `cid' chan_oh323.c:1437: structure
Re: [Asterisk-Users] H.323 trunking
See below.
Nardis Dome wrote:
Hi,
Could someone help me on configuring a H.323 trunk.
I am trying to set up the following scenario:
[SIPphone(2004)]--[asterisk/oh323/asterisk-oh323]--H323Trunk--[PBX]--[H323phone/(8004)]
I am using the following versions:
Linux CentOS 3.3/2.4.21-.EL.co
asterisk 1.0.1
pwlib_1.5.2
openh323_1.12.2
asterisk-oh323-0.6.3b
Calling from Asterisk (2004) to the H.323phone
(61-8004) gives me the following error
-- Executing Dial(SIP/2004-8350,
H323/192.168.204.130) in new stack
Dec 7 13:45:19 WARNING[1032209]: channel.c:1901
ast_request: No channel type registered for 'H323'
Dec 7 13:45:19 NOTICE[1032209]: app_dial.c:742
dial_exec: Unable to create channel of type 'H323'
== Everyone is busy/congested at this time
Dec 7 13:45:29 WARNING[1032209]: pbx.c:1933
ast_pbx_run: Timeout, but no rule 't' in context
'default'
[general]
static=yes
writeprotect=no
;Trunk=Modem/g1
[default]
exten = 2004,1,NoOp( call for ${EXTEN})
exten = 2004,2,Dial(SIP/${EXTEN},10,tr)
exten = 2004,3,Congestion
exten = 2005,1,NoOp( call for ${EXTEN})
exten = 2005,2,Dial(SIP/${EXTEN},10,tr)
exten = 2005,3,Congestion
exten = _61,1,Dial,H323/192.168.204.130
Change this into:
exten = _61,1,Dial,OH323/192.168.204.130
ps: 61 is a prefix. All the extensions 61xxx should be
routed to the H.323 trunk.
thx for your feedback
Michael.
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Re: [Asterisk-Users] H323 Channel
Francisco wrote: Hi, im getting mad compiling the H323 channel (Jeremy's version inAccess version). Ive tryed many versions of openh323 lib and pwlib, and i get differets errors. Does anyone uses this channel? and which version of it, openh323 lib and pwlib? asterisk-oh323-0.6.4 compiles/works perferctly with Asterisk stable. Just make sure that you follow the README file and describe the problems you get when try to compile it. Currently im using Linux Slackware 10.0, and i ask myself if is there a I also use the same distro without problems. pkg of asterisk-oh323 or something like that, precompiled.. No, not yet (although someone has offered to build some packages of asterisk-oh323). Thanks guys Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Performance problems
Tracy R Reed wrote: Some of you may recall that I have been working on building a box to convert H323 to SIP. After a significant amount of outside help and slicing and dicing of the ohh323 code to get it to compile on AMD64 we finally got it working. Now we are working on improving the performance. Do you want to share the details of the installation? I would like to make things easier for the installation of asterisk-oh323 by fixing the stuff that made your day (days?) harder. This box takes H323 from one device and converts to SIP and spits it back out to another device. The codec is g729 but we do not have any g729 licenses on the box because we are doing pass-through so I figure the cpu usage should not be a problem and some sort of bandwidth issue would be hit first. Hardware stats: AMD Athlon(tm) 64 Processor 3000+ 512M of RAM 100Mb/s full duplex switched ethernet Linux bit64.foo.com 2.6.9-1.667 #1 Tue Nov 2 14:50:10 EST 2004 x86_64 x86_64 x86_64 GNU/Linux Software: Asterisk CVS-HEAD-11/26/04-12:38:01 built by [EMAIL PROTECTED] on a x86_64 running Linux asterisk-oh323-0.7.0 The problem: If we point 24 voice channels of traffic at the box we see 5% cpu utilization and all is well. But cpu utilization scales non-linearly until we have 96 voice channels and 50% cpu utilization. At this rate we won't scale to nearly where we had hoped to. According to the voip-info.org wiki a g729 stream is usually around 30kb/s including overhead etc so 96 channels would be 2.8Mb/s. Since we have that coming in and out total bandwidth is 5.6Mb/s. Not much at all I wouldn't think. At a 20ms sample rate 96 channels is 4800 packets per second times two for incoming and outgoing and we get 9600 packets per second. Again, not that much. About 25% of the cpu seems to go to the asterisk process and 25% to the system. Absolutely no swapping is going on. At 24 channels the load average barely ticks above zero. At 96 it hits around 8. I don't know if it matters but there is no zaptel hardware at all in this box, pure voip. Anyone have any idea where the bottleneck could be or any tuning tweaks we could make? For a start perform the same test but doing SIP to SIP and not H323 to SIP and check the cpu utilization for the same figures of calls and call rate. Don't forger to disable reinvites on these calls, in order to be the test comparable to the H323 to SIP scenario. With both results we will get a clue about the portion of the utilization of the system for each protocol (SIP/H323) and the part that needs to be optimized. For sure the greatest part will be of H323 but how much of it is it? Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Process Stop After few hours
Daniel Eboa wrote: Hello to all, I have a strange behavior of my asterisk box. I'm running asterisk with asterisk-oh323 channel driver and everything works very well. But after few hours, my asterisk stop running and I have to restart it by typing asterisk -vvvc. Most of the time I connect to my asterisk with a remote host so I don't know exactly which error causes my box to stop, but I found on the console this message: Segmentation Fault. Did any one has experience this problem?? what is the solution? What versions of Asterisk/asterisk-oh323 do you run? Please provide a backtrace of the core file dumped. I use Cisco ATA 186 Boxes with my asterisk. Thank In advance. Daniel. Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Process Stop After few hours
Daniel Eboa wrote: I use asterisk-oh323-0.6.3b, pwlib-v1_6_6 and openh323-v1_13_5. This is the complete error: H245:818c6c0 PWLIB Assertion Fail: file transports.cxx, line 1637 Go up to v0.6.4 version of asterisk-oh323 (I guess that you use Asterisk CVS stable). Thanks. Daniel Michael -Original Message- From: Michael Manousos [mailto:[EMAIL PROTECTED] Sent: mercredi 1 décembre 2004 16:47 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk Process Stop After few hours Daniel Eboa wrote: Hello to all, I have a strange behavior of my asterisk box. I'm running asterisk with asterisk-oh323 channel driver and everything works very well. But after few hours, my asterisk stop running and I have to restart it by typing asterisk -vvvc. Most of the time I connect to my asterisk with a remote host so I don't know exactly which error causes my box to stop, but I found on the console this message: Segmentation Fault. Did any one has experience this problem?? what is the solution? What versions of Asterisk/asterisk-oh323 do you run? Please provide a backtrace of the core file dumped. I use Cisco ATA 186 Boxes with my asterisk. Thank In advance. Daniel. Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Process Stop After few hours
Daniel Eboa wrote: How to get it? Download it from here: http://www.inaccessnetworks.com/projects/asterisk-oh323/download -Original Message- From: Michael Manousos [mailto:[EMAIL PROTECTED] Sent: mercredi 1 décembre 2004 17:35 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk Process Stop After few hours Daniel Eboa wrote: I use asterisk-oh323-0.6.3b, pwlib-v1_6_6 and openh323-v1_13_5. This is the complete error: H245:818c6c0 PWLIB Assertion Fail: file transports.cxx, line 1637 Go up to v0.6.4 version of asterisk-oh323 (I guess that you use Asterisk CVS stable). Thanks. Daniel Michael -Original Message- From: Michael Manousos [mailto:[EMAIL PROTECTED] Sent: mercredi 1 décembre 2004 16:47 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk Process Stop After few hours Daniel Eboa wrote: Hello to all, I have a strange behavior of my asterisk box. I'm running asterisk with asterisk-oh323 channel driver and everything works very well. But after few hours, my asterisk stop running and I have to restart it by typing asterisk -vvvc. Most of the time I connect to my asterisk with a remote host so I don't know exactly which error causes my box to stop, but I found on the console this message: Segmentation Fault. Did any one has experience this problem?? what is the solution? What versions of Asterisk/asterisk-oh323 do you run? Please provide a backtrace of the core file dumped. I use Cisco ATA 186 Boxes with my asterisk. Thank In advance. Daniel. Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] overriding DTMF and codec from dialplan?
Brian West wrote: OH But it is just that simple. You also have: -= Info about application 'ImportVar' =- [Synopsis]: Set variable to value [Description]: ImportVar(#n=channel|variable): Sets variable n to variable as evaluated on the specified channel (instead of current). If prefixed with _, single inheritance assumed. If prefixed with __, infinite inheritance is assumed. I give up, my mistake. bkw It's not so simple. Check http://bugs.digium.com/bug_view_advanced_page.php?bug_id=928 for the details. Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OH323 Rocks :) --- H323 guys, use it to solve no answer at this time problem!!!
Thanks. I appreciate that. Michael. kido noagbodji wrote: i have had some problems with the H323 channel ... Other party not anwsering SIP 2 H323 bridge. the chan_oh323 solves the problem. Use it. (Even though it is quite complicated to install but READ the README file) Nahuel that should solve it!! Kido ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] overriding DTMF and codec from dialplan?
Roy Sigurd Karlsbakk wrote: is it possible, from an agi script or directly in extensions.conf, to override the DTMF and codec settings? to answer my own question SetVar(SIP_CODEC=g726) allowed me to force g726, but only on outgoing calls. when dialling in from the iax server, I do the same, setting the codec etc, but this does not work. sip show channels only shows the channel using alaw Change this into SetVar(_SIP_CODEC=g726) and it will work. roy Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] oh323 compile issue
administrator tootai wrote: Hi all, I want to give a try to oh323 (currently nufone h323 channel is setup and compiling fine) on a yesterday CVS update of asterisk. I have _pwlib 1.8.1_ and _openh323 1.15.1_ What I made: Wrong, wrong, wrong! 1) Read the README. 2) Get the right versions of OpenH323/Pwlib. 3) Follow the instructions. Michael. openh323 dir: make clean apply the oh323 patch configure make opt asterisk-oh323-0.7 dir: make [...] wrapendpoint.cxx: In method `BOOL WrapH323EndPoint::OpenAudioChannel (H323Connection , int, unsigned int, H323AudioCodec )': wrapendpoint.cxx:915: no matching function for call to `H323AudioCodec::IsDescendant (const char *)' wrapendpoint.cxx:916: no matching function for call to `H323AudioCodec::IsDescendant (const char *)' make[1]: *** [wrapendpoint.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk-oh323-0.7.0/wrapper' make: *** [subdirs_build] Error 1 [EMAIL PROTECTED] asterisk-oh323-0.7.0]# Someone know what's the problem? Regards ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] oh323 compile issue
administrator tootai wrote: Michael Manousos a crit : administrator tootai wrote: Hi all, I want to give a try to oh323 (currently nufone h323 channel is setup and compiling fine) on a yesterday CVS update of asterisk. I have _pwlib 1.8.1_ and _openh323 1.15.1_ What I made: Wrong, wrong, wrong! 1) Read the README. Done 2) Get the right versions of OpenH323/Pwlib. Can't come back to an earlier version Use the OH323STAT flag in the top-level Makefile to build a channel driver with staically linked the libraries if you can't setup multiple OpenH323/Pwlib versions on one machine. 3) Follow the instructions. Give up the test of oh323. You told me to test it, I try it ;-) If my configuration don't meet the required one, too bad. You can do it. Just try harder! Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] oh323/g729 and DTMF
Al Escasa wrote:
In my oh323.conf, i am using:
userInputMode=TONE
Is everyone trying to say that i have no hope using
oh323 when using inband DTMFs? is this problem of
asterisk? the protocol? the codec? i wish there is
still some kind of workaround.. =(
What I meant was that inband DTMFs do not work when G.729 is used.
Out-of-band DTMFs work just fine. You could send me your config
with a screen log (with -cd options) when you make H.323 calls
to check if there is something weird.
I also set inBandDTMF=yes (am not sure if that helped
but nothing happened when i tested again).
Whats the differnce between purchased licences and
passthru mode? I am able to make calls using oh323 and
the codec being used is g729 (since this is the codec
used by our VoIP provider).
But my problem is, the incoming VoIP call seems like
it could not select any keys coz there's no response
(my analysis it is not responding to the DTMF signal).
Anyways, here is part of my extensions.conf under
h323:
[voip-h323]
exten = ${DNIS_TEST},1,Ringing
exten = ${DNIS_TEST},2,Playback(record1)
exten = ${DNIS_TEST},3,Background(silence/3)
exten = 1,1,Goto,nmailbox|s|1
exten = ${DNIS_TEST},4,Dial(Zap/7,5,T)
exten = ${DNIS_TEST},5,Goto,operator1|s|1
exten = ${DNIS_TEST},6,hangup
If you will notice, step 3 will wait for the user to
input 1 if he wants to go to voicemail. This config
works when coming from a PSTN line. But when using
Voip, there is no response.
Lastly, if this is really going nowhere.. Can I use
SIP instead of oh323 in solving this problem of
capturing user's input?? If so, any ideas to go about
it?
If you guys need to view some more of my config, I'd
gladly post it.. =)
Thanks again! and more power!
-Alejandrino
Michael.
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Re: [Asterisk-Users] Problem compiling H323 channel
Roy Layson wrote: hope it can help [deleted] I. INSTALL OS OS is Fedora Core2 installed only -textbased internet (elinks) -web server (apache and etc...) -SQL (mysql and DBD/DBI) -Development tools (default) -kernnel Development (default) II. Downloaded the following: pwlib Janus patch 4 (1.6.6.3) from http://unc.dl.sourceforge.net/sourceforge/openh323/pwlib-Janus_patch4-src-tar.gz openh323 Janus Patch 4 (1.13.5.3) from http://unc.dl.sourceforge.net/sourceforge/openh323/openh323-Janus_patch4-src-tar.gz asterisk-oh323-0.7.0 from http://www.inaccessnetworks.com/projects/asterisk-oh323/download/asterisk-oh323-0.7.0.tar.gz Performed: # cd /usr/src # export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot # cvs login - the password is anoncvs. # cvs checkout zaptel asterisk *all where etracted to: /root/pwlib /root/openh323 /root/asterisk-oh323-0.7.0 /root/zaptel /root/asterisk III. INSTALLATION a.) pwlib # cd /root/pwlib # ./configure # make b.) openh323 # cd /root/openh323 # patch -p1 /root/asterisk-oh323-0.7.0/openh323_1.13.5-make.patch # ./configure # make c.) asterisk # cd /root/asterisk # make d.) asterisk-oh323-0.7.0 # cd /root/asterisk-oh323-0.7.0 # vi Makefile SETTINGS (of Makefile) DESTDIR= PWLIBDIR=/root/pwlib OPENH323DIR=/root/openh323 ASTERISKINCDIR=/root/asterisk/include ASTERISKMODDIR=/usr/lib/asterisk/modules ASTERISKETCDIR=/etc/asterisk OH323WRAPLIBDIR=/usr/local/lib # make error encountered! What is the error? Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_h323 on AMD64
Tracy R Reed wrote: On Tue, Nov 23, 2004 at 12:42:07PM +0100, administrator tootai spake thusly: with my other regular x86 box running H323. One odd thing I note is that when looking at the UDP traffic with tcpdump I see the * box receiving my Same problem here. My * box is connected to GnuGK. CVS Head 11/02/04, kernel 2.4.26-SMP Same problem as in you ran tcpdump or something and saw the odd behavior of receiving but not sending any packets? VERY interesting. Were you on an x86-64 bit box or regular x86? I was thinking this odd behavior was some odd interation with x86-64. Same versions. Before, I was running pwlib 1.5 and openh323 1.12 and had no problem. Perhaps I should switch to that version and see how it goes then. I was afraid my problem was related to the platform. I am really hoping H323 stabilizes on Asterisk. It's a shame that it is such a pain in the neck add-on when it is still really the backbone of VOIP. Have you tried asterisk-oh323? Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] oh323/g729 and DTMF
Al Escasa wrote: Hi everyone, Could somebody enlighten me on this one? I have configured my asterisk to run on oh323 using codec g729. Incoming calls are working okay. But the thing I want to work is say pressing some options, say dial 1 to go to voicemail or dial a certain number to dial a specific extension. I have a config for this and tried calling from a normal PSTN and is working. But i just can't seem to make it work using oh323/coded g729? Its like it does not respond to DTMF signals? I have dig into many mailing list and not any clear solutions. Could someone help me or even just send me any procdures on how to do this. And also could someone please verify that what I am doing is wrong in the first place.. If you try to accomplish this with inband DTMFs, then there is no hope. What 'userInputMode' in oh323.conf are you using? Thanks in advance for the support! -Alejandrino Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H323 Problems
Peter, Peter Landy wrote: New to Asterisk so I am sure this has been answered before. I can compile PWLIB and OpenH323 but when it comes to compiling asterisk-oh323 then I get all kinds of errors even though I have set the paths up in the source files. I can attach the errors if it is useful. I though however that someone must have gone through this exercise successfully. Any chance of someone giving me a quick how to so I can check I am doing it right? Did you apply the OpenH323 patch BEFORE configuring/compiling the library? Regards Peter Landy Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is H323 dying?
kido noagbodji wrote: Hello, I just downloaded and installed the latest version of asterisk under Fedora. (had it under FreeBSD but was having TOOO many problems) After my installation i noticed that the channel H323 was not included ( I remember that i did not have to install it under freeBSD) but I have seen that SIP and IAX are supported though. So i am wondering: Does asterisk consider H323 so achaic that it does not bother including it anymore? According to you specialists, are we looking at the end of H323? or maybe i just did not install asterisk properly :-). H.323 support for Asterisk based on the original code (asterisk-oh323) is far from dying. Check: http://www.inaccessnetworks.com/projects/asterisk-oh323 for the latest code. Thanks Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk-OH323 OUTCODEC
Try:
SetGlobalVar(OH323_OUTCODEC=g723.1)
Michael.
M. Ehsanul Karim wrote:
Hello,
What would be the outcodec value for g723.1 (6.3k). I have g723
support which works with SIP (not pass thru) , but when I use OH323 it
always
Unsupported ${OH323_OUTCODEC} value (G72316K3)!
I have enabled all g723 in oh323.conf
SetGlobalVar(OH323_OUTCODEC=G72316K3)
Regards,
Ehsan
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Re: [Asterisk-Users] Forward incoming SIP calls to H323 ipphone?
Since you are able to receive H.323 calls with chan_oh323, I assume that the module is loaded. You could check the incoming/outgoing/simultaneous limits or submit the oh323.conf. Additionally, what are the full messages that you get on the console? Michael. Alex van Es wrote: Hi all, For my setup I need to forward incoming SIP and ZAP calls to my IP phone using H323. I have managed to set up the OH323 and when I enter my asterisk's ip number into sjphone, it will answer and give me the welcome message. So receiving calls with H323 is not a problem.. but I want to be able to dial out. I have set up a extention that looks like; exten = 1234,1,Dial(OH323/192.168.1.20) I keep on getting the message unable to create channel of type ' OH323'. I have tried also the names h323, h.323, oh323, OH323/h323.. but none of them seem to exist. When I receive the incoming call it says channel OH323, so I assume that is the correct name. However.. I still can't forward calls out. I could do without OH323, but when I forward incoming SIP calls to my IP phone using SIP I just get silence after I answer the phone (both parties can't hear each other) so I wanted to try it this way. Anyone has any ideas? Alex -- Alex van Es - [EMAIL PROTECTED] http://photography.icepick.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Forward incoming SIP calls to H323 ipphone?
Alex van Es wrote: Michael, Yeah.. for sure the channel is loaded.. calling to my asterisks works fine. I have included the oh323.conf and the original message. Thanks a lot for you help. I would would like to get this baby working. Alex The log; Nov 8 18:04:01 WARNING[294930]: channel.c:1901 ast_request: No channel type registered for 'OH323' Hmm, according to this message, chan_oh323.so isn't loaded. Your config is fine. Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Forward incoming SIP calls to H323 ipphone?
format_ilbc.so Raw iLBC data 0 format_sln.so Raw Signed Linear Audio support (SLN) 0 format_jpeg.so JPEG (Joint Picture Experts Group) Image 0 cdr_csv.so Comma Separated Values CDR Backend 0 cdr_manager.so Asterisk Call Manager CDR Backend 0 chan_oh323.so OpenH323 Channel Driver 0 chan_zap.so Zapata Telephony 0 app_zapras.so Zap RAS Application 0 app_meetme.so MeetMe conference bridge 0 app_flash.so Flash zap trunk application 0 app_zapbarge.so Barge in on Zap channel application 0 app_zapscan.so Scan Zap channels application 0 On 8-nov-04, at 18:29, Michael Manousos wrote: Alex van Es wrote: Michael, Yeah.. for sure the channel is loaded.. calling to my asterisks works fine. I have included the oh323.conf and the original message. Thanks a lot for you help. I would would like to get this baby working. Alex The log; Nov 8 18:04:01 WARNING[294930]: channel.c:1901 ast_request: No channel type registered for 'OH323' Hmm, according to this message, chan_oh323.so isn't loaded. Your config is fine. Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex van Es - [EMAIL PROTECTED] http://photography.icepick.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk-oh323: New versions now available!
Hello all, The asterisk-oh323 package has been updated. From now on, there are two series of releases: - 0.6.x releases, latest is 0.6.4. These will work with Asterisk v1-0 source code. - 0.7.x and above, latest is 0.7.0. These are for CVS code of Asterisk. Also, the latest versions now use OpenH323/Pwlib Janus-patch4 libraries. 0.6.4/0.7.0 versions contain major stability fixes and some fixes for dynamic codec negotiation (although there are still same cases that do not work). Additionally, the build process supports the building of a chan_oh323.so binary that has statically linked the OpenH323/Pwlib/oh323wrap libraries (to avoid common runtime errors with conflicting versions of OpenH323/Pwlib libraries). No new features have been added. In the following versions, I'm thinking about extending the configuration of the channel driver (oh323.conf) for supporting specific configuration options per H.323 endpoint and gateway used. Something like the user/peer/friend philosophy of Asterisk config files, but closer to the needs of chan_oh323. Download from the usual location: http://www.inaccessnetworks.com/projects/asterisk-oh323 Regards, Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] More than one OH323 Gatekeeper Registration
Sergio (RED) wrote: Hi, Anybody know if I can register my Asterisk in more than one h323 Gatekeeper. I need to call to diferents providers depending on convenients destinations prices. This is purely an OpenH323 issue. The library does not permit such a usage. I guess that Craig (Southeren) is the most appropriate person to comment on the validity of this one, and also, if this is doable. Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OH323 Trunking
Huddleston, Robert wrote: The only disadvantage we found to using the OH323 channel driver is that we cannot now register netmeeting or other h323 directly to the * With the What do you mean cannot now register? asterisk-oh323 doesn't implement gatekeeper functionality. It never did. Just use it as a gateway (no registration is needed). Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems with 0penh323 Channel Driver
[EMAIL PROTECTED] wrote: Hi, I have asterisk,openh323-v1_13_5 and pwlib-v1_6_6 installed on my PC. each time i run asterisk -c, i get the following error: [chan_oh323.so] = (OpenH323 Channel Driver) == Parsing '/etc/asterisk/rtp.conf': Found == Parsing '/etc/asterisk/oh323.conf': Found [1]WrapH323EndPoint::WrapH323EndPoint: Compile-time libraries OpenH323 v1.13.5, PWlib v1.6.6 segmentation error [EMAIL PROTECTED] root]# Can you help me? What versions of Asterisk, asterissk-oh323 do you use? What is the current configuration of oh323? Can you send the backtrace of the core file dumped? Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dial Out w/ OH323
Huddleston, Robert wrote: Due to the format of the message coming from the H323 channels included w/ Asterisk we were unable to use our gatekeeper. For a quick solution we tried the OH323 channel drivers and can receive inbound calls from the parent gatekeeper. We are trying to do a dial to gatekeeper... I am trying exten = 5551212,1,Wait,2 exten = 5551212,2,Dial,OH323/5551212 But I am not sure if this is the correct protocol... Please help Check the CONFIGURATION file included in asterisk-oh323. All valid Dial strings are explained there. Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dial Out w/ OH323
Huddleston, Robert wrote:
Okay - read it... my configuration works... what I want
exten = XX,1,Wait,2
exten = XX,2,Dial(OH323/XX)
I want it to pass the 10 digits to the DIAL string... I'm not sure I
understand the macros
can I just put the ${EXTEN} in there??
Of course. The following will do the job:
exten = _XX,2,Dial(OH323/${EXTEN})
Michael.
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Re: [Asterisk-Users] Problem Loading asterisk_oh323-0.6.3b eith last *cvs...
Rafael J. Risco G.V wrote: Hello Ive just install last cvs version (Mon Sep 6) of Asterisk with asterisk-oh323-0.6.3b and pwlib-v1_6_6-src.tar.gz, openh323-v1_13_5-src.tar.gz and . this is the error loading asterisk with chan_oh323 module:: [cdr_csv.so] = (Comma Separated Values CDR Backend) [cdr_manager.so] = (Asterisk Call Manager CDR Backend) == Parsing '/etc/asterisk/cdr_manager.conf': Found [format_sln.so] = (Raw Signed Linear Audio support (SLN)) == Registered file format sln, extension(s) sln|raw * [chan_oh323.so]Sep 6 16:58:13 WARNING[1076236928]: loader.c:248 ast_load_resource: /usr/local/lib/liboh323wrap.so: undefined symbol: _ZNK26H323CapabilityRegistration8GetClassEj Sep 6 16:58:13 WARNING[1076236928]: loader.c:429 load_modules: Loading module chan_oh323.so failed!* ** any idea? Just make sure that at runtime the correct openh323/pwlib libraries are used. thank you Rafael Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H323 Control Protocol Error
[EMAIL PROTECTED] wrote: Hi there ! I searched the whole web to find some helping information about H323 Control Protocol, but there is no way to find that information. We compiled and installed asterisk_0.9.0 + pwlib 1.5.2 + openh323_1.12.2 + 'asterisk-oh323_1.5 channel driver + wrapper' and configured the Please use current versions of Asterisk/OpenH323/asterisk-oh323. Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OH323 0.6.3b compilation problem with 1.0 RC2 on RH9
Joa~o Amaro wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I Vlasis, I'm using those versions (Fedora COre 1) and it compiled without problems, but when i try to initialize asterisk i get the folowwing error: ERROR [-1084337504]: chanoh323.c:4636 load_module: H.323 listener creation failed. There is some other process listening on the TCP port used for H.323 signaling (default is 1720). This port can be specified in oh323.conf. Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OH323 0.6.3b compilation problem with 1.0 RC2 on RH9
It works fine for me on a Slack9.1 laptop. Michael. Vlasis Chatzistayrou wrote: Hello, I just tried to compile OH323 0.6.3b on a RH9 machine with Asterisk 1.0 RC2 installed but failed. I applied the patch to the required OpenH323 library according to the instructions, and set the proper directories in the Makefile. Here is what I receive after I issue make: *** g++ -DP_USE_PRAGMA -fno-rtti -ffunction-sections -fdata-sections -D_REENTRANT - DOPENSSL_NO_KRB5 -Wall -fPIC -I/Downloads/pwlib/v1.6.6/pwlib/include - DPTRACING -I/Downloads/openh323/v1.13.5/openh323/include -DHAS_OSS -Wall -x c++ -Os -DPWLIBVERSION=\1.6.6\ -DOPENH323VERSION=\1.13.5\ - I/Downloads/pwlib/v1.6.6/pwlib/include/ptlib/unix - I/Downloads/pwlib/v1.6.6/pwlib/include - I/Downloads/openh323/v1.13.5/openh323/include - I/Downloads/openh323/v1.13.5/openh323/include/openh323 -I../asterisk-driver -c wrapcaps.cxx -o wrapcaps.o touch ../asterisk-driver/chan_oh323.c gcc -shared -Wl,-soname,liboh323wrap.so -o liboh323wrap.so wrapper_misc.o asteriskaudio.o wrapendpoint.o wrapconnection.o wrapper.o wrapcaps.o make[1]: Leaving directory `/Downloads/oh323/oh323-0.6.3b/asterisk-oh323- 0.6.3b/wrapper' make[1]: Entering directory `/Downloads/oh323/oh323-0.6.3b/asterisk-oh323- 0.6.3b/asterisk-driver' gcc -Wall -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing- declarations -D_REENTRANT -D_GNU_SOURCE -I/usr/include/asterisk -I../wrapper - g -c -o chan_oh323.o chan_oh323.c In file included from /usr/include/stdio.h:34, from chan_oh323.c:34: /usr/lib/gcc-lib/i386-redhat-linux/3.2.2/include/stddef.h:213: syntax error before typedef In file included from chan_oh323.c:34: /usr/include/stdio.h:46: syntax error before typedef /usr/include/stdio.h:62: syntax error before typedef In file included from /usr/include/_G_config.h:44, from /usr/include/libio.h:32, from /usr/include/stdio.h:72, from chan_oh323.c:34: /usr/include/gconv.h:176: parse error before __flexarr In file included from /usr/include/libio.h:32, from /usr/include/stdio.h:72, from chan_oh323.c:34: /usr/include/_G_config.h:47: field `__cd' has incomplete type /usr/include/_G_config.h:50: field `__cd' has incomplete type /usr/include/_G_config.h:52: confused by earlier errors, bailing out make[1]: *** [chan_oh323.o] Error 1 make[1]: Leaving directory `/Downloads/oh323/oh323-0.6.3b/asterisk-oh323- 0.6.3b/asterisk-driver' make: *** [subdirs_all] Error 1 *** I'm not a very experienced Linux user so I can't really figure out what the problem may be in this case. Does anyone have any suggestions? Thank you in advance, Vlasis Hatzistavrou. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk codecs and packet size
Andres wrote: The quick and dirty way: In rtp.c, function ast_rtp_write, in the switch statement, AST_FORMAT_G729A case, change the smoother creation to something larger. E.g.: rtp-smoother = ast_smoother_new(40); Keep in mind that you must set this into something valid (45 obviously is not valid). Recompile and you should be fine. Michael, this little nugget made my day. Last year we offered to pay for this development. Too bad you didn't collect:) Just out of curiosity. What was the offering for this one-line patch? Thanks! Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk codecs and packet size
Luis Vazquez wrote: Does anybody knows if it's posible or if there is some develoment in course to be able to use longer transmit packet sizes (as long as I know this is fixed in 20ms now) with the compressed voip codecs in asterisk (g729, g726, gsm, etc). I need to use asterisk to connect remote sip clients with 24kb bandwidth lines and I'm using a licences g729 codec but because I can't increase the packet size to 40 or 60 ms in asterisk the connection is useless. The quick and dirty way: In rtp.c, function ast_rtp_write, in the switch statement, AST_FORMAT_G729A case, change the smoother creation to something larger. E.g.: rtp-smoother = ast_smoother_new(40); Keep in mind that you must set this into something valid (45 obviously is not valid). Recompile and you should be fine. The right (but longer) way: --- The ability to packetize variable number of frames per RTP packet for various codecs should be configurable from within the rtp.conf file. This requires some coding of course. Currently, I don't have time available to do it, but I could do it as soon as I find some free time. Thanks very much Luis Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Compile error H323
Enrico Stahn wrote: Hi! Have a look at the following entry. I solved this problem: http://enrico.todo.de/weblog/item/asterisk-oh323-compile-error That's the wrong way to do it. You use incorrect versions of the libraries. Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PLC (Packet loss cancel) questions
Kevin Walsh wrote: [EMAIL PROTECTED] wrote: On 27 Aug 2004 at 2:33, Kevin Walsh wrote: There is no packet loss concealment in Asterisk at this time. Why doesn't asterisk clock to the 1000 interrupts per second instead of the incoming audio? Were there no interrupts available when it started? Even if you had no card you could use the ztdummy module and even though that might be off by a bit, surely it'd sound better than a connection which is experiencing packet loss? I'm note sure what you're referring to with the 1000 interrupts per second. Asterisk, as it stands, only reacts to incoming frames. If nothing is received then nothing is sent. The authors obviously didn't take packet loss into consideration. When a packet is received, the expected time of the next packet is calculated. A while ago, I proposed that some sort of empty frame frame could be scheduled for now + next ETA. The arrival of the empty frame would wake up the receiver and, with the help of the jitter buffer, it could determine whether to pass on that frame to the translator, or to drop the packet as a duplicate. Some codecs could recognise the empty frame as a trigger to run their perform packet loss concealment code, whereas others (with no PLC) could simply treat it as a silent frame. This approach also is not fully right. On a system that implements silence suppression and uses discontinuous transmission (DTX), the receiver has a very tough job. I know that the current implementation of Asterisk doesn't work well with silence suppression but this doesn't mean that the design of a solution shouldn't take into account the full scenario. Look at the RTP stack of the receiver. When a packet is received, there are two cases: a) An RTP packet carrying voice frames is received. In that case the decoder will play the voice frames. b) A CN (Comfort Noise) packet is received. In that case the decoder will generate background noise (or do nothing). Now the hard part. Nothing is received (while something was expected). These are the normal interpretations of this situation: a) The transmitter detected silence and sent nothing (Silence). The receiver knows it from the last packet received (a CN packet). b) The transmitter sent a packet but the packet was lost (Packet loss). The receiver knows it from the last packet received (an RTP packet). These conditions can be identified at the RTP stack and signalled to Asterisk through the use of a new frame type (as you propose above). But, of course these are not always correct and the following situations could also happen: a) The transmitter detected silence and sent nothing but the last CN packet was lost. According to the above interpretations, the receiver will try to conseal a packet loss, which is wrong. b) The transmitter sent an RTP packet, that packet was lost and the last packet correctly received at the receiver was a CN packet. Again, following the above interpretation, the receiver will do nothing (or more accurate, will play some background noise), while it should conseal the packet loss. These cases cannot be identified, so the receiver just can only guess about what really happened and act accordingly. This all seems possible to me, but I haven't seen a discussion relating to this proposal nor any other alternatives. I hope that the above issues will start a discussion and result to a solution, no just for PLC, but also for the DTX operation. [deleted] Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PLC (Packet loss cancel) questions
Kevin Walsh wrote: Michael Manousos [EMAIL PROTECTED] wrote: Look at the RTP stack of the receiver. When a packet is received, there are two cases: a) An RTP packet carrying voice frames is received. In that case the decoder will play the voice frames. b) A CN (Comfort Noise) packet is received. In that case the decoder will generate background noise (or do nothing). Agreed. Now the hard part. Nothing is received (while something was expected). These are the normal interpretations of this situation: a) The transmitter detected silence and sent nothing (Silence). The receiver knows it from the last packet received (a CN packet). b) The transmitter sent a packet but the packet was lost (Packet loss). The receiver knows it from the last packet received (an RTP packet). Both of the above cases are identifiable using a line state flag. Asterisk can (a) continue to generate CN or (b) generate a new frame type to get the codec to handle the concealment - where possible. These conditions can be identified at the RTP stack and signalled to Asterisk through the use of a new frame type (as you propose above). But, of course these are not always correct and the following situations could also happen: a) The transmitter detected silence and sent nothing but the last CN packet was lost. According to the above interpretations, the receiver will try to conseal a packet loss, which is wrong. I would propose that after x lost packets, Asterisk should treat all further lost packets as CN. The proceeding x packets should be interpreted as RTP packet loss and run through the concealment routine. Well, no matter what kind of concealment algorithm is used, just the first one or two packets will be concealed. The rest losses will result in no-playback. No CN interpretation, just absolute silence. b) The transmitter sent an RTP packet, that packet was lost and the last packet correctly received at the receiver was a CN packet. Again, following the above interpretation, the receiver will do nothing (or more accurate, will play some background noise), while it should conseal the packet loss. In this case, there is nothing to conceal anyway, as the last received data was a CN packet. In this case, the CN state should be continued until an RTP packet is received and the line state can be changed. Exactly. So the receiver, in case of no-receiption, should go back and see what was the last packet correctly received and act as I described above. The difficult part to handle would be late or out-of-sequence RTP Actually this is not so difficult, if there is a jitter buffer. packets. These should be ironed out by the jitter buffer. Late, lost and juggled packets are to be expected when dealing with UDP. This all seems possible to me, but I haven't seen a discussion relating to this proposal nor any other alternatives. I hope that the above issues will start a discussion and result to a solution, no just for PLC, but also for the DTX operation. I hope so too. Asterisk would benefit greatly from these improvements. Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_oh323 and cdr
Roger Schreiter wrote: Hi, there are some posts about that topic, but unfortunatelly I do not yet know what to do. I find every call in Master.csv, but those coming in via chan_oh323. In oh323.conf I have accountcode=oh323 but there is no other file in the directory cdr-csv than Master.csv. Can anyone give me any hint, what to do, in order to have calls from chan_oh323 logged in any file? In oh323.conf, set: amaFlags=billing Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RC2 and Netmeeting 3.01 ?
Zineddin Karzazi wrote: --- Robert Rozman [EMAIL PROTECTED] schrieb: Hi, I'd kindly ask for any guidance how to setup Netmeeting to work with Asterisk. I've setup Asterisk as Gateway, selected GSM codec, and I'm able to call local extensions (no calls into PBX functions) but get no sound. Any hint, advice ? Anyone using Netmeeting (maybe also windows messenger) with Asterisk sucessfully ? Thanks in advance, regards, Robert. I have same Problems with Netmeeting, just wanted to test H.323 with Astersik , it rings, but as soon as i answer it dissconnects. im getting the Following Error: oh323_exception: OH323/R27469: Invalid format of RTP addresses. Aug 13 10:19:05 ERROR[524304]: chan_oh323.c:1933 oh323_write: OH323/R27469: Failed to create smoother. There is no common codec between Asterisk and Netmeeting. Today i tried Openphone (H.323 Client) and it works. Zineddin. Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk+IVR functions trouble
alex3377 wrote: I' got a problem, using asterisk-rc2 :IVR functions (Background...Playback...etc) doesn't works : Executing Background(OH323/RX, vm-extension) in new stack channel.c:1650 ast_set_write_fornat: Unable to find path from GSM to G729A---Asterisk box supplied only with network adapter.---Asterisk box registered in Mera (soft-switch with H323 protocol) and doing SIP-endpoints (such as ATA). and G729A is preferred codec to my needs.Is this trouble associated with G729A codec? Do you have G.729 codec for Asterisk installed? Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RC2 and Netmeeting 3.01 ?
Robert Rozman wrote: - Original Message - From: Michael Manousos [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Thursday, August 26, 2004 4:52 PM Subject: Re: [Asterisk-Users] RC2 and Netmeeting 3.01 ? snip There is no common codec between Asterisk and Netmeeting. --snip Isn't GSM codec that can be run on both ? Netmeeting has a different GSM codec (MS-GSM). Try to use another codec (e.g. PCMA) Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk-oh323-0.6.3a
M. Willigs wrote: Hi there. I thy to compile asterisk-oh323-0.6.3a but it fail in the make command. I have the pwlib-v1_6_6-1 and openh323-v1_13_5-1 as saying in the README file of the packet asterisk-oh323-0.6.3a You must apply the included OpenH323 patch before trying to configure/compile OpenH323 (as mentioned in the README). Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OH323 and codec selection
Chris A. Icide wrote: I'm having a small issue with the oh323 implementation when it comes to codec selection. Version info: CVS Head 6/30/2004 OH323 0.6.3 OpenPhone for windows version 1.8.1 Asterisk is configured as a h323 endpoint which either terminates to the PSTN locally through a PRI or terminates the h323 call to an IAX provider remotely. Asterisk also has G729 licences installed. in oh323.conf we set codecs allowed in the following order: G729 GSM ULAW ALAW When dialing in with OpenPhone with all codecs besides g729 disabled in the audio codec configuration panel, oh323 in Asterisk still picks and uses GSM as the selected codec. Only if I disable all but G729 in oh323.conf will Asterisk use G729 for an incomming h323 call. Am I doing something wrong? Is the order of the codecs in the oh323.conf significant, or is some other method of codec selection being used? Yes, the order of the codecs in oh323.conf is significant. I use it with ohphone (v1.4.3) without problems. Just make sure that you declare the codec of openphone as the preferred one. I don't know what happens when you just disable the codecs in openphone (and it seems that it doesn't work, since the codec that is selected is one of the disabled codecs). -Chris Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] queue_log question: which endpoint was connected?
lenz wrote: Hello list, as I'm writing a little perl parser for queue_log analysis, I'd like to know *which* telephone answered a specific queue call. Unfortunately app_queue only logs the call id but does not log the call end point. This is okay for SIP endpoints, because their call id is something like SIP/endpointname-1234 so you can reasonably understand who was on answering, but for OH323 I get ID's like OH323/LJ5645 that are meaningless. Is there a way to extract from some other log the fact that OH323/LJ234 was a call placed to - say - OH323/[EMAIL PROTECTED] or can I extract it from some field of the peer data structure queue_log seems to extract data from? (to obtain call id, they gust print peer-name) The IP of the connected endpoint can be obtained from the OH323_RADDR variable. For incoming H.323 calls you can get the name of the channel and the IP address inside the dialplan, write them to a file and process them later. For outgoing H.323 calls [Dial(OH323/...)], you can't do it from the dialplan. In that case the OH323_RADDR variable is accessible only through the Dial() app. Anyway, it seems that the name of the OH323 channels needs to be more useful (added to my TODO list). Any help will be greatly appreciated. Thanks l. Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems Compiling Asterisk-oh323-0.6.2
Zineddin Karzazi wrote: Hi. im compiling the wrapper for oh323(under Suse 9.0) -pwlib 1.6.6 -openh323 1.13.5. (with oh323 Patch) i execute: ./samples/simple/obj_linux_x86_r/simph323 and it works fine. When i Run asterisk-oh323 0.6.2: make Download and install version 0.6.3a. Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] h323 to SIP Server Load
Steve Totaro wrote: Does anyone do any large scale SIP to H323 conversion? How many simultaneous calls can your server handle and on what hardware? I think I read on the wiki that twenty five would max out most servers. Not true for asterisk-oh323. Micheal. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] still can't load oh323 - Are we not supporting H.323 any more?
ruixun wu wrote: Hi Alexey, I followed your steps, but Asterisk still didn't work. I am a little crazy. I show my envirement and ld.so.conf here. Could somebody tell me if I am using the correct libraries? Thanks a lot ld.so.conf: /usr/kerberos/lib /usr/X11R6/lib /usr/lib/qt-3.1/lib /usr/local/lib envirement: PWLIBDIR=/usr/src/pwlib OPENH323DIR=/usr/src/openh323 LD_LIBRARY_PATH=/usr/src/pwlib/lib:/usr/src/openh323/lib ldconfig -v /lib: libiw.so.25 - libiw.so.25 [snip] /usr/local/lib: liboh323wrap.so - liboh323wrap.so Remove these two from /usr/local/lib -- libh323_linux_x86_r.so.1.13.5 - libh323_linux_x86_r.so.1.13.5 libpt_linux_x86_r.so.1.6.6 - libpt_linux_x86_r.so.1.6.6 --- You should be using the binaries in $PWLIBDIR/lib and $OPENH323DIR/lib (unless you are sure that they are the same). Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] still can't load oh323 - Are we not supporting H.323 any more?
ruixun wu wrote: Hi Michael, Thanks for your time. I deleted these two files ,libh323_linux_x86_r.so.1.13.5 and libpt_linux_x86_r.so.1.6.6. And startd asterisk, the error still exist. Then I copy these two files from $PWLIBDIR/lib and $OPENH323DIR/lib to /usr/local/lib. Startd the asterisk, the error still exist. It's very strange why most of others can install oh323 easily and I met this hard problem. The installation of asterisk-oh323 is pretty easy if you follow the README file. I can't think of anything else that could cause a problem. Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip - H323 using oh323 and G729
There is nothing wrong with asterisk-oh323, the call is rejected from
the remote endpoint. Try to turn on only G.729 and retry.
And yes, you don't need g729 licenses to do g729 passthrough.
Michael.
David Allen wrote:
Hi All,
I have set up a box that will be used as follows:
SIP Phone Asterisk Cisco H323 VoIP Server
192.168.1.5 192.168.1.50 192.168.1.80
Asterisk is running the latest CVS and oh323 driver.
The SIP phone is a Grandstream Budgetone 100.
I have everything setup and running with G.711 ALAW and ULAW and i'm able
to make calls through Asterisk between the SIP phone and the Cisco VOIP
server, however if I want to use G.729 between the SIP Phone and the Cisco,
the Call is unable to be completed and responds back with:
-- Executing Dial(SIP/200-ebdf, OH323/[EMAIL PROTECTED]) in new stack
-- H.323 call to [EMAIL PROTECTED] with codec G729A
-- Called [EMAIL PROTECTED]
-- H.323 call 'ip$localhost/1573' cleared, reason 24 (Call ended with
Q.931 cause)
-- Hungup 'OH323/L1573'
== No one is available to answer at this time
-- Executing Hangup(SIP/200-ebdf, ) in new stack
== Spawn extension (internal, 10290071717, 2) exited non-zero on
'SIP/200-ebdf'
and the oh323 outputs the following:
2:21.275 ThreadID=0x49399b30 H323Making call to:
[EMAIL PROTECTED]
2:21.276 ThreadID=0x49399b30 H323Added capability:
G.711-ALaw-64k{hw} 1
2:21.276 ThreadID=0x49399b30 H323Added capability:
G.711-uLaw-64k{hw} 2
2:21.276 ThreadID=0x49399b30 H323Added capability: G.729{hw}
3
2:21.276 ThreadID=0x49399b30 H323Added capability:
UserInput/hookflash 4
2:21.276 ThreadID=0x49399b30 H323Added capability:
UserInput/basicString 5
2:21.276 ThreadID=0x49399b30 H323Added capability:
UserInput/dtmf 6
2:21.276 ThreadID=0x49399b30 H323Added capability:
UserInput/RFC2833 7
2:21.277 ThreadID=0x49399b30 H323Found capability:
G.711-ALaw-64k{hw} 1
2:21.277 ThreadID=0x49399b30 H323Found capability:
UserInput/hookflash 4
2:21.277 ThreadID=0x49399b30 H323Found capability:
UserInput/basicString 5
2:21.277 ThreadID=0x49399b30 H323Found capability:
UserInput/dtmf 6
2:21.277 ThreadID=0x49399b30 H323Found capability:
UserInput/RFC2833 7
2:21.277 ThreadID=0x49399b30 H323Found capability:
G.711-uLaw-64k{hw} 2
2:21.277 ThreadID=0x49399b30 H323Found capability: G.729{hw}
3
2:21.278 ThreadID=0x49399b30 RFC2833 Handler created
2:21.278 ThreadID=0x49399b30 H323Added capability: G.729{hw}
1
2:21.278 ThreadID=0x49399b30 H323Created new connection:
ip$localhost/23866
2:21.279 H225 Caller:80f1490 H225Started call thread
2:21.328 H225 Caller:80f1490 H323TCP Started connection:
host=192.168.1.80:1720, if=192.168.1.50:10001, handl$
2:21.328 H225 Caller:80f1490 H225Sending Setup PDU
2:21.329 H225 Caller:80f1490 H225Check for Fast start by
local endpoint
2:21.329 H225 Caller:80f1490 H245Default
OnSelectLogicalChannels, FastStartInitiate
2:21.330 H225 Caller:80f1490 RTP_UDP Session 1 created:
192.168.1.50:10002-10003 ssrc=204629209
2:21.330 H225 Caller:80f1490 RTP Adding session RTP_UDP
2:21.330 H225 Caller:80f1490 H323RTP Receiver created using
session 1
2:21.331 H225 Caller:80f1490 RTP Found existing session 1
2:21.331 H225 Caller:80f1490 H323RTP Transmitter created using
session 1
2:21.331 H225 Caller:80f1490 H225Fast start begun by local
endpoint
2:21.332 H225 Caller:80f1490 H323RTP OnSendingPDU
2:21.333 H225 Caller:80f1490 RTP OnSendingPDU
2:21.333 H225 Caller:80f1490 LID Created codec: pt=G729,
bytes=10, samples=80
2:21.334 H225 Caller:80f1490 H225Built fastStart for
G.729{hw} 1
2:21.335 H225 Caller:80f1490 H323RTP OnSendingPDU
2:21.336 H225 Caller:80f1490 RTP OnSendingPDU
2:21.336 H225 Caller:80f1490 LID Created codec: pt=G729,
bytes=10, samples=80
2:21.336 H225 Caller:80f1490 H225Built fastStart for
G.729{hw} 1
2:21.338 H225 Caller:80f1490 H225Sending PDU: setup
2:21.338 H225 Caller:80f1490 H225Reading PDUs: callRef=23866
2:21.398 H225 Caller:80f1490 H225Receiving PDU:
releaseComplete
2:21.398 H225 Caller:80f1490 H225Handling PDU:
ReleaseComplete callRef=23866
2:21.399 H225 Caller:80f1490 H225Set protocol version to 4
and implying H.245 version 7
2:21.399 H225 Caller:80f1490 H323Clearing connection
ip$localhost/23866 reason=EndedByQ931Cause
2:21.399 H225 Caller:80f1490 H323Call end reason for
ip$localhost/23866 set to EndedByQ931Cause
2:21.399 H225
Re: [Asterisk-Users] error while compiling asterisk-oh323
Try to describe your problem. A first guess is that you didn't apply the patch for the OpenH323. Michael. Mandar Pise wrote: Hi Folks, I am breaking my head for compiling asterisk-oh323 properly on my asterisk box from past 1 week. But still after my all efforts, I unable to make it compile properly, My box is Fedora core 2 with asterisk-0.9.0. I was trying for following configuration with openh323 and pwlib. Openh323 and pwlib are installed properly. But problem is asterisk-oh323. asterisk-oh323-0.6.2a.tar.gz openh323-v1_13_5-src.tar.gz pwlib-v1_6_6-src.tar.gz I don't know how to make it work. I went through mailing list but it couldn't help me to solve my problem. Has anybody faced similar problems? Thanks Regards, Mandar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk-oh323 on fedora Core 2 - Anyone has a working install?
Kanuri, Seshu wrote: I am wondering if anyone has a working install of oh323 on fedora Core2. I'll try this when I find some time (I have to setup FC2 on a box). You could help me by describing where it fails to install. Michael. An replies would be appreciated as we need this urgently. Seshu Kanuri -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] Sent: Thursday, July 22, 2004 6:12 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] error while compiling asterisk-oh323 -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Thu, 22 Jul 2004, Mandar Pise wrote: My box is Fedora core 2 with asterisk-0.9.0. I was trying for following configuration with openh323 and pwlib. Openh323 and pwlib are installed properly. But problem is asterisk-oh323. I've had the same problem - seems to compile with not much trouble on RedHat 9, but spectacularly fails on Fedora 2. After spending a long time banging my head against it I eventually gave up. I wanted H323 support so I could use GnomeMeeting, but since I couldn't get it working I've switched to IAXComm. On that note, I was having a problem with the current version of IAXComm occasionally missing calls. I believe I've now fixed that problem and the patch is available on my website if anyone else is struggling with that problem: http://www.nexusuk.org/projects/VoIP/iaxcomm/missedcalls.php (I have also sent a copy to the IAXComm author) - -- - Steve Jabber: [EMAIL PROTECTED] Web: http://www.nexusuk.org/ Servatis a periculum, servatis a maleficum - Whisper, Evanescence -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) Comment: Public key available at http://www.nexusuk.org/pubkey.txt iD8DBQFA/5Le5zUOsIV3bqERAkSwAJ9t0cQnHW9agZFBKgKp/tcwjglFLwCfUdGR /ZQUFSrY895W4cnGa3JzsJs= =H+Wc -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] STILL NO AUDIO
Why don't you use asterisk-oh323? Michael. Sebastian Nocetti wrote: I WANT TO USE G729, I HAVE TO USE IT... -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Eric Wieling Enviado el: Lunes, 19 de Julio de 2004 11:46 a.m. Para: [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] STILL NO AUDIO I suspect it will be solved when you put disallow=all and allow=ulaw in sip.conf and h323.conf (and NO OTHER ALLOW= LINES) On Mon, 2004-07-19 at 09:25, Sebastian Nocetti wrote: I cant do SIP - CHAN_H323 transmit audio!!! I can hear rings, but when connected, NOTHING It happened in both: SIP - CHAN_H323 and CHAN_H323 - SIP... when it will be solved? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] oh323 dial structure and oh323 debug?
Hi Chris,
Chris A. Icide wrote:
According to the wiki at voip-info.org, the dial structure for using
oh323 without a gatekeeper is:
OH323/exten@host:port
or
OH323/exten
The second option is valid only in the case where a gatekeeper is used.
NOTE: OpenH323 library v1.12.0 has a bug in the parsing of the destination
host. When this version is used then the above syntax should be:
OH323/h323:exten@host:port
That's right (I wrote it). For a detailed description check also
the CONFIGURATION file in asterisk-oh323 package.
Now, I've got a '*' box that has oh323 running, and it accepts inbound
h323 calls and processes them perfectly (well, not perfectly, but thats
because they are coming in with g729 and going out as gsm, and we don't
have our g729 licenses from digium yet), and now that this is working as
expected, I've been asked to pass any calls with prefix 572 back out to
another h323 gateway. Simple enough, in the dial plan I just matched
_572. and tried sending it out to the other h323 gateway (not an
asterisk platform). This is where the problem is.
I can't seem to get the system to send the extension along no matter
what form I try. And to worsen this the oh323 debug toggle CLI command
does nothing (I haven't checked if I need to go back and compile a debug
It seems that you haven't enabled the logging of debug info
(check logger.conf and add the debug option in the line
console = ...). Anyway, the debug of the OH323 channel driver
is basically for debugging and not tracing a call.
flag into oh323 yet). I've tried the following in the way of dial commands:
First, Dial(OH323/gw-ip-addy) works in that I actually contact the
remote gw and it gives me a bad user message '-- H.323 call
'ip$localhost/6740' cleared, reason 24 (Call ended with Q.931 cause)
', so I know I'm getting to the right gw.
Dial(OH323/${EXTEN:3}@gw-ip-addy) doesn't work, we don't seem to parse
the exten@ip-addy and just try to reach the whole argument as an
address, error is '-- H.323 call 'ip$localhost/6738' cleared, reason 11
(Gatekeeper could not find user)'
Dial(OH323/h323:${EXTEN:3}@gw-ip-addy) (yes, pulling at straws here
because I'm running 1.13.5 lib), also doesn't work... ' -- H.323 call
'ip$localhost/6737' cleared, reason 11 (Gatekeeper could not find user)'
I've also tried mixing and matching the exten and ip-addy all around to
no avail. Can someone point out the right format of the Dial command
for oh323 when routing a call with a dialed extension to gateway with a
known ip-address. No gatekeepers involved at all.
From the messages above it is obvious that: 1) there is a gatekeeper
somewhere in the path, 2) you try to contact a non-existent user.
Your Dial string is just fine.
-Chris
Michael.
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Re: [Asterisk-Users] Re: Cann't load oh323 0.6.3a
It seems that you use wrong versions of the libraries at run-time (probably the distribution's libraries?). Do a ldd /usr/lib/asterisk/modules/chan_oh323.so Michael. Fathallah Soumaya wrote: when I put ldd /usr/local/lib/liboh323wrap.so, it tells me: libc.so.6 = /lib/tls/libc.so.6 (0x4200) /lib/ld-linux.so.2 = /lib/ld-linux.so.2 (0x8000) thank you very much for your answer best regards, -- Lars Degenhardt [EMAIL PROTECTED] a crit : Fathallah Soumaya wrote: I tried and I still have the same error: Jul 14 12:31:29 WARNING[1076298368]: loader.c:242 ast_load_resource: /usr/local/lib/liboh323wrap.so: undefined symbol: _ZTI14PAbstractArray Jul 14 12:31:29 WARNING[1076298368]: loader.c:423 load_modules: Loading module chan_oh323.so failed! what does ldd /usr/local/lib/liboh323wrap.so tell you? --- Emmanuel OYOUA [EMAIL PROTECTED] a crit : try : ldconfig ; asterisk -cvvv -- Emmanuel OYOUA ABIDJAN, Systmes et rseaux -- AFRIPA TELECOM, Africa switch on (http://www.afripatelecom.net) -- Original Message --- From: Fathallah Soumaya [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wed, 14 Jul 2004 12:12:30 +0200 (CEST) Subject: Re: [Asterisk-Users] Re: Cann't load oh323 0.6.3a I have already autoload = yes and the channel fails to load... --- Lars Degenhardt [EMAIL PROTECTED] a crit : Fathallah Soumaya wrote: yes, but if I say noload this channel it does not load and I cannot use this channel to make h323 calls... I didn't say put noload = chan_oh323.so into that file, I said leave it alone, the module will be loaded automatically if the param autoload is set to yes --- Lars Degenhardt [EMAIL PROTECTED] a crit : ---snip [chan_oh323.so]Jul 13 12:56:47 WARNING[1074464512]: loader.c:240 ast_load_resource: /usr/local/lib/liboh323wrap.so: undefined symbol: _ZTI14PAbstractArray Jul 13 12:56:47 WARNING[1074464512]: loader.c:421 load_modules: Loading module chan_oh323.so failed! ---snap look in your /etc/asterisk/modules.conf, set autoload to yes and don't load chan_oh323.so specifically. cheers ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Crez gratuitement votre Yahoo! Mail avec 100 Mo de stockage ! Crez votre Yahoo! Mail sur http://fr.benefits.yahoo.com/ Dialoguez en direct avec vos amis grce Yahoo! Messenger !Tlchargez Yahoo! Messenger sur http://fr.messenger.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Lars Degenhardt phon: +49 76814749263| mobile: +49 1736936968| box: +49 891488262647 BOFH excuse #83: Support staff hung over, send aspirin and come back LATER. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users === message truncated === Crez gratuitement votre Yahoo! Mail avec 100 Mo de stockage ! Crez votre Yahoo! Mail sur http://fr.benefits.yahoo.com/ Dialoguez en direct avec vos amis grce Yahoo! Messenger !Tlchargez Yahoo! Messenger sur http://fr.messenger.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OH323 and G729
If you want to be absolutely right then you have to use the following patch that I have submitted on the bugtracker: http://bugs.digium.com/bug_view_page.php?bug_id=928 SetVar doesn't work as expected with the Dial() app. The patch above provides a new app (SetInheritVar) for that purpose. Michael. Serge wrote: Yes, it's work, Thanks, But possible don't use Global Var?, due in this situation all other destinations use this codec, after 1 time global setup. And g729 - limited:( Regards, Serge. - Original Message - From: Michael Manousos [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, July 13, 2004 1:46 PM Subject: Re: [Asterisk-Users] OH323 and G729 Try with 'SetGlobalVar' instead of 'SetVar'. Michael. Serge wrote: Dear All, I have problem with new oh323 0.6.3a , I try use var OH323_OUTCODEC, but it don't work. oh323 driver don't want connect to gateway with g729, it's work if I only use in oh323.conf one codec ( g729 ). If I enable 2 or more codecs - always in use other codec: -- Executing SetVar([EMAIL PROTECTED]/1, OH323_OUTCODEC=g729a) in new stack -- Executing Dial([EMAIL PROTECTED]/1, OH323/##|70) in new stack -- H.323 call to # with codec GSM Due Gateway don't support GSM and ulaw, always return: No one is available to answer at this time Many thanks for your help, Regards, Serge. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OH323 and G729
Try with 'SetGlobalVar' instead of 'SetVar'. Michael. Serge wrote: Dear All, I have problem with new oh323 0.6.3a , I try use var OH323_OUTCODEC, but it don't work. oh323 driver don't want connect to gateway with g729, it's work if I only use in oh323.conf one codec ( g729 ). If I enable 2 or more codecs - always in use other codec: -- Executing SetVar([EMAIL PROTECTED]/1, OH323_OUTCODEC=g729a) in new stack -- Executing Dial([EMAIL PROTECTED]/1, OH323/##|70) in new stack -- H.323 call to # with codec GSM Due Gateway don't support GSM and ulaw, always return: No one is available to answer at this time Many thanks for your help, Regards, Serge. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems with cdr_csv
In oh323.conf set: amaFlags=billing Michael. Oleg A. Arkhangelsky wrote: Hello All, It seems that this question is very stupid, but anyway. Do I need any additional configuration for cdr_csv.so? This module is loaded by default at Asterisk's startup (asterisk -fvvv): [cdr_csv.so] = (Comma Separated Values CDR Backend) But when I place call I didn't see anything in /var/log/asterisk/cdr-csv. There is also no errors or warnings regarding this module on console. P.S.: Asterisk CVS-HEAD-07/07/04-11:48:42 P.P.S.: I'm using chan_oh323.so channel driver (by InAccess Networks). ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sound quality IAX client GSM to ALAW with oh323
Hi, Do IAX(GSM) - IAX(ALAW) calls sound ok? What is the configuration of OH323 channel (oh323.conf)? Also, run asterisk with '-vvvcd', make a call and send the output. Don't forget to enable the logging of debug messages (logger.conf). Michael. Arne Scheffer wrote: Hello veryone, I have a strange problem. I have an asterisk (latest from CVS) with latest oh323 channel driver. I place calls with DIAX. The H323 gateways only support G711A De DIAX only supports GSM When I perform an inbound call: H323 - asterisk - DIAX :: sound is ok. When I perform an outbound call: DIAX - Asterisk - h323 :: sound is terrible and CPU load is 80% When I perform an asteisk internal call with DIAX: DIAX - asterisk IVR :: sound is good and cpu OK. Does anyone else have this problem ? Know how to solve it ? regards, Arne. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] strange problem with oh323 loaded!
What exactly is the problem with v0.6.3(a)? Michael. Anthony Law wrote: I too tried 0.6.3 and it is behaving the same. I have now downloaded oh323 to 0.6.2a and it seems fine. Regards, Anthony ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- *** * * *G R E E C E * * * * EUROPEAN CHAMPION EURO 2004 * * * *** ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OH323-COMPILE
You are trying to compile an ancient asterisk-oh323 with fresh Asterisk code. It won't work. Download and install asterisk-oh323-0.6.3a. Also, download and compile the recommended versions of OpenH323/Pwlib (OpenH323/Pwlib 1.12.2/1.5.2 are too old). Michael. mohammad mirzaee wrote: HI ALL HI MICHAEL; My name is mohammad and I am iranian.I have been trying to install oh323 channel but I come up with dead end. In fact it makes me crazy. plz help me michael. I saw mailing list and I trid serevel CVS headers such as , 2004-06-07( seven of june) 0r 2004-07-02( second of july) besides I use: 1-openh323 v1.12.2 2-pwlib v1.5.2 3- asterisk CVS (2004-06-07, 2004-07-02, .) 4- oh323 v.5-10 / oh323 v.5.9 5- my linux box is redhat 8.0 the error looks like the following: make[1]: *** [chan_oh323.o] Eroor 1 make[1]: Leaving directory '/root/asterisk/asterisk-oh323-0.5.9/asterisk-driver' make:*** [subdirs_all] Error 1 I think there is a mismatch between my oh323 and asterisk. But I donot know the excat asterisk CVS I will be waiting for your help warmest regards mohammad -- *** * * *G R E E C E * * * * EUROPEAN CHAMPION EURO 2004 * * * *** ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] strange problem with oh323 loaded!
OK, I'll look at it. Michael. T. Chan wrote: Dear All, I don't know but I tried all 0.6.x version of OH323 and normally I use safe_asterisk to start asterisk, and everytime when I use 'stop now' to terminate asterisk, it does not do anything, and you are rite, I have to use kill -9 to kill the PIDs and threads. However, if I use asterisk -vvvgc to start, 'stop now' works. Thanks all TC -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Scott Stingel Sent: Friday, July 02, 2004 4:37 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] strange problem with oh323 loaded! Same problem here - with latest 0.6.3a oh323. Locks up on exit. Had to kill -9 This didn't happen with 0.6.2a, but that's on a different machine. Maybe you could try this older version which worked fine (same PwLib and OpenH323) Regards Scott Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anthony Law Sent: Friday, July 02, 2004 1:15 PM To: Mailing List Asterisk Subject: [Asterisk-Users] strange problem with oh323 loaded! Hi, Here it is when I start it with /etc/rc.d/init.d/asteriskd found in asterisk source contrib/init.d/rc.redhat.asterisk It started without problem and when i issue stop now It freezes, please see below, tai*CLI add debug dontdumpextensions helpiax2 include init loadlocal logger mgcpno oh323 reload remove save set showsip skinny softunload == Registered channel type 'OH323' (OpenH323 Channel Driver) == OpenH323 Channel Ready (v0.6.3) == Parsing '/etc/asterisk/enum.conf': == Parsing '/etc/asterisk/enum.conf': Found == Parsing '/etc/asterisk/logger.conf': == Parsing '/etc/asterisk/logger.conf': Found Asterisk Event Logger restarted Asterisk Ready. tai*CLI stop now tai*CLI It freezes right here and does nothing else - If I do it with safe_asterisk , it died and loops [EMAIL PROTECTED] init.d]# /usr/sbin/safe_asterisk -vvc [EMAIL PROTECTED] init.d]# Asterisk ended with exit status 127 Asterisk died with code 127. Automatically restarting Asterisk. Asterisk ended with exit status 127 Asterisk died with code 127. Automatically restarting Asterisk. Asterisk ended with exit status 127 As I have mentioned, if I noload oh323 this won't happen *CLI stop now Beginning asterisk shutdown Executing last minute cleanups == Destroying any remaining musiconhold processes Asterisk cleanly ending (0). Any ideas? Regards, Anthony ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.713 / Virus Database: 469 - Release Date: 6/30/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.713 / Virus Database: 469 - Release Date: 6/30/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- *** * * *G R E E C E * * * * EUROPEAN CHAMPION EURO 2004 * * * *** ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
