[asterisk-users] Progress passing problem.
Hi, i have Asterisk 1.2.7.1 and outgoing trunk connected via SIP (this is Cisco AS5350)and user is connected via sip too. When user calling out (via AS5350) he receives progress tone generated by voip-phone not that passing from telco line. I turned on debug and see that the AS send: 183 Session Progreess but to user is sent Ringing, not progress. I have progressinband=never in sip.conf so shouold be transferred. Where can be a problem? Regards, Adam Rybak ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Quicknet PhoneJack questions.
Hi,
i have already bought this card and successfully configured IXJ driver in
kernel but i have few problems:
1. I have no dialtone, somtimes it appears for a very small time and dissapears.
I have in phone.conf configured mode=dialtone
2. Second progress inband, when number is placed i hear ringing tone even if
called party is busy, my voip operator passess progress inband properly - with
softphone connected to my asterisk im able to hear ringing/busy/invalid number
etc.
3. The last thing is problem with dialplan when i set:
exten = _00.,1,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN:2}
after press only 00 the system tries to callout, not waiting for other digits.
Please help if you have experience in PhoneJACK PCI.
Regards,
Adam Rybak
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[Asterisk-Users] PRI indications.
Hello, i have succesfullu setup asterisk with Sangoma E1 card, evrything works well but i don't know how to pass indications from telco switch to the user - when users call bad number telco switch shuld talk unallocated number but its only send PRI_CAUSE 1. How to pass voice indications thru asterisk to clients? My /etc/zaptel.conf: span=1,0,0,CCS,HDB3,CRC4 dchan=16 bchan=1-10 alaw=1-10 loadzone=pl defaultzone=pl My /etc/asterisk/zapata.conf: [channels] language=en context=from-pstn switchtype=euroisdn signalling=pri_cpe pridialplan=local usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=no cancallforward=no callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 immediate=no priindication=outofband group = 1 channel = 1-10 Regards, Adam Rybak ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OH323 channel in asterisk 1.2 ...Ouch ... error while writing audio data!
You should have more info in full log messages, look to this file and send output. Adam Cytowanie Rafael R. GV [EMAIL PROTECTED]: Hello I am trying to compile oh323, oh323-0.6.5 wont compile with asterisk 1.2libraries, must be oh323-0.7.3, now I have compiled this version but when reload asterisk i have this error: [chan_oh323.so]Ouch ... error while writing audio data: : Broken pipe Any idea??? -- rrgv ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OH323 channel in asterisk 1.2 ...Ouch ... error while writing audio data!
It looks like compiling oh323 with wrong version of headers or wrong version of open323/pwlib. Are you completly sure that you deleted old headers and libraries when upgraded asterisk to new version? Adam Rybak Cytowanie Rafael R. GV [EMAIL PROTECTED]: /var/log/asterisk/full.1 output: Nov 26 21:25:39 VERBOSE[14215] logger.c: [chan_oh323.so]Nov 26 21:25:39 WARNING[14215] loader.c: /usr/lib/asterisk/modules/chan_oh3 23.so: undefined symbol: _ZNK8PChannel7IsClassEPKc Nov 26 21:25:39 WARNING[14215] loader.c: Loading module chan_oh323.so failed! thanks rafael On 11/27/05, Adam Rybak [EMAIL PROTECTED] wrote: You should have more info in full log messages, look to this file and send output. Adam Cytowanie Rafael R. GV [EMAIL PROTECTED]: Hello I am trying to compile oh323, oh323-0.6.5 wont compile with asterisk 1.2libraries, must be oh323-0.7.3, now I have compiled this version but when reload asterisk i have this error: [chan_oh323.so]Ouch ... error while writing audio data: : Broken pipe Any idea??? -- rrgv ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- rrgv Pozdrawiam, Adam Rybak ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk 1.2 - IAX2 strange behavior.
Hello, i found in my system logs problem with handling IAX2 calls - its looks: Connected to Asterisk CVS-Nv1-2-0-11/19/05-23:19:49 currently running on ast-serv (pid = 13312) Verbosity is at least 13 ast-serv*CLI show channels Channel Location State Application(Data) IAX2/xxx-1 [EMAIL PROTECTED]:1 Up Bridged Call(OOH323/VOICE-eb0d OOH323/VOICE-eb0d[EMAIL PROTECTED]:3 Up Dial(IAX2/[EMAIL PROTECTED]/0114920 2 active channels 1 active call Nov 26 21:49:19 WARNING[25064]: chan_iax2.c:7971 network_thread: chan_iax2: ast_sched_runq ran 1984 scheduled tasks all at once Nov 26 21:49:53 WARNING[25064]: chan_iax2.c:7971 network_thread: chan_iax2: ast_sched_runq ran 1921 scheduled tasks all at once Nov 26 21:49:55 WARNING[25064]: chan_iax2.c:7971 network_thread: chan_iax2: ast_sched_runq ran 229 scheduled tasks all at once Nov 26 21:50:02 WARNING[25064]: chan_iax2.c:7971 network_thread: chan_iax2: ast_sched_runq ran 1456 scheduled tasks all at once Nov 26 21:50:28 WARNING[25064]: chan_iax2.c:7971 network_thread: chan_iax2: ast_sched_runq ran 3493 scheduled tasks all at once Nov 26 21:50:40 WARNING[25064]: chan_iax2.c:7971 network_thread: chan_iax2: ast_sched_runq ran 1611 scheduled tasks all at once Nov 26 21:50:51 WARNING[25064]: chan_iax2.c:7971 network_thread: chan_iax2: ast_sched_runq ran 1563 scheduled tasks all at once Nov 26 21:51:25 WARNING[25064]: chan_iax2.c:7971 network_thread: chan_iax2: ast_sched_runq ran 115 scheduled tasks all at once [... many the same WARNINGS snipped ...] Nov 26 22:04:16 WARNING[25064]: chan_iax2.c:7971 network_thread: chan_iax2: ast_sched_runq ran 45 scheduled tasks all at once ast-serv*CLI show channels Channel Location State Application(Data) IAX2/xxx-1 [EMAIL PROTECTED]:1 Up Bridged Call(OOH323/VOICE-eb0d OOH323/VOICE-eb0d[EMAIL PROTECTED]:3 Up Dial(IAX2/[EMAIL PROTECTED]/0114920 2 active channels 1 active call Nov 26 22:07:37 WARNING[25064]: chan_iax2.c:7971 network_thread: chan_iax2: ast_sched_runq ran 76 scheduled tasks all at once Nov 26 22:07:45 WARNING[25064]: chan_iax2.c:7971 network_thread: chan_iax2: ast_sched_runq ran 3045 scheduled tasks all at once ast-serv*CLI What it can be? System dont have much load - there is one call only. There is free memory, cpu usage is low. $ w 22:09:21 up 17 days, 4:22, 2 users, load average: 0.02, 0.25, 0.20 Regards, Adam ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk 1.2 stability problem.
Hello, i have succesfully ipgraded my system to asterisk 1.2 with OOH323C channel driver, today i got hangup of my asterisk after this messages: Nov 25 21:03:22 WARNING[24395] channel.c: Avoided initial deadlock for '0x8198118', 10 retries! Nov 25 21:03:25 WARNING[24395] channel.c: Avoided initial deadlock for '0x8198118', 10 retries! Nov 25 21:03:28 WARNING[24395] channel.c: Avoided initial deadlock for '0x8198118', 10 retries! Nov 25 21:03:29 WARNING[24395] channel.c: Avoided initial deadlock for '0x8198118', 10 retries! Nov 25 21:03:30 WARNING[24395] channel.c: Avoided initial deadlock for '0x8198118', 10 retries! Nov 25 21:03:30 WARNING[24395] channel.c: Avoided initial deadlock for '0x8198118', 10 retries! Nov 25 21:03:37 WARNING[24395] channel.c: Avoided initial deadlock for '0x8198118', 10 retries! Nov 25 21:03:38 WARNING[24395] channel.c: Avoided initial deadlock for '0x8198118', 10 retries! I was able to connect to asterisk using asterisk -r but stop now command does nothing, and asfter some seconds freezes, i tried to killall asterisk but doesnt work, i was able kill -9 asterisk only. At this time were two calls active - 4 channels: 2 IAX, 1 SIP, 1 OOH323C What it can be a problem? Regards, Adam ___ --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] Passing parametrs to php agi scripts.
Hello, i have problem with pass parameters into php agi script from extensions.conf, how to get this parameter from php variables? Im passing paramterer: s,1,DaeadAGI,test.php,parameter1 How get value of parameter1 in php script? Regards, Adam Rybak ___ --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] Call transfer.
Hello, how i can tranfer call to another user? Im using X-Lite, i have configured in features.conf: [featuremap] blindxfer = #1 disconnect = *0 automon = *1 atxfer = *2 But when im dial *2 in conversation nothig happens. What can br problem? Im using asterisk CVS-HEAD from 02/09/05. Regards, Adam Rybak ___ --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 and Caller ID missing
I have the same problem with asterisk CVS HEAD and asterisk-ooh323 module from obj-sys. This driver is included into asterisk-addon package. Adam. Cytowanie Asterisk guy [EMAIL PROTECTED]: I get the same problem. ( asterisk1.2.0beta1+oh323 0.73), any suggestion for this ? On 6/13/05, Federico Alves [EMAIL PROTECTED] wrote: I am sending calls using Oh323 to a Cisco Gateway (AS5300), and although I set the caller id correctly in my perl AGI script $AGI-set_callerid($ani); , the gateway does not see any caller id coming from my Asterisk box. I use the very latest version of Oh323 as published in the Inaccess web site, and Asterisk HEAD from two days ago. The caller ID important for this client, because he will further authenticate the call based on the ANI. I am only doing a codec conversion. Any help is appreciated from Jeremy McNamara. ___ 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 Pozdrawiam, Adam Rybak ___ --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 and Caller ID missing
I have the same problem with asterisk CVS HEAD and asterisk-ooh323 module from obj-sys. This driver is included into asterisk-addon package. Adam. Cytowanie Asterisk guy [EMAIL PROTECTED]: I get the same problem. ( asterisk1.2.0beta1+oh323 0.73), any suggestion for this ? On 6/13/05, Federico Alves [EMAIL PROTECTED] wrote: I am sending calls using Oh323 to a Cisco Gateway (AS5300), and although I set the caller id correctly in my perl AGI script $AGI-set_callerid($ani); , the gateway does not see any caller id coming from my Asterisk box. I use the very latest version of Oh323 as published in the Inaccess web site, and Asterisk HEAD from two days ago. The caller ID important for this client, because he will further authenticate the call based on the ANI. I am only doing a codec conversion. Any help is appreciated from Jeremy McNamara. ___ 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 Pozdrawiam, Adam Rybak - Koniec przekazywanej wiadomoĊci - Pozdrawiam, Adam Rybak ___ --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] H323 with asterisk-ooh323c
Hello, i have succesfully compiled and installed newest channel driver ooh323c with asterisk CVS-HEAD. I have small problem - when the asterisk logins to the GnuGK its shchown as unknown type: RCF|195.214.XXX.XXX:1720|ASTERIX2:h323_ID|unknown|9681_endp Sun, 11 Sep 2005 22:34:34 +0200 C(0/0/0) 1 and when im seting prefixes for routing in gnugk.ini this not working. If i use oh323 channel (0.7.1pre) this logins as gateway: RCF|195.214.XXX.XXX:1720|ASTERIX:h323_ID|gateway|7452_endp Sun, 11 Sep 2005 22:29:51 +0200 C(0/0/6) 2 Prefixes: 881,871 how to change that ooh323c login as gateway type? I need send traffic from H.323 network to the asterisk. How to configuree h323.conf? My h323 conf is: [general] port=1720 bindaddr=195.214.XXX.XXX faststart=yes h245tunneling=no h323id=ASTERIX2 gatekeeper = 195.214.XXX.XXX logfile=/var/log/asterisk/h323_log context=in disallow=all allow=gsm allow=ilbc dtmfmode=rfc2833 [incoming] type=user context=in allow=all prefix=* Pozdrawiam, Adam Rybak ___ --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] Connecting Asterisk with Microsoft LCS (Live Communication Server)
Hello, im trying to connect LCS to asterisk which will act as pstn gateway for LCS. Microsoft system supports only SIP TCP connections but asterisk UDP. im was searching about conversion beetwen TCP and UDP and i found that SER can do that but i don't know SER and my trying to configure SER fails. is there any other possibility to connect this together? Maybe someone has correct config files for SER? Thanks in advance, Adam ___ 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] How to pass Asterisk -SIP- Cisco AS -H323- world ?
Hello, i know thats is * mailing list but maybe here are cisco guru's which can help. My network schema is: Softphones -SIP- Asterisk -SIP-\/-H323- WORLD2 Cisco AS5350 - E1 - WORLD (Phones - Traditional PBX -E1---/) - will be developed soon. Communications beetween WORLD and softphones works well but i have an H323 link to other site and i want to allow calling from softthones (in future from Phones too) calling to the WORLD2. I tried to add dialpeers but this doesnt work - all calls are routed via E1 to WORLD. This is part of my config: ! GK Config: interface FastEthernet0/0 ip address 192.168.X.X 255.255.255.0 duplex auto speed auto h323-gateway voip interface h323-gateway voip id TGK1 ipaddr 194.X.X.X 1719 h323-gateway voip h323-id MYID ! gateway ! For World - Softphones communication dial-peer voice 14 pots incoming called-number 2323. direct-inward-dial ! dial-peer voice 15 voip destination-pattern 2323. session protocol sipv2 session target ipv4:192.168.X.X codec g711alaw ! For outgoing Softphones - World dial-peer voice 1000 pots application session destination-pattern .T direct-inward-dial port 3/1:D forward-digits all ! i tried to add ! dial-peer voice 999 voip application session destination-pattern 0. target session ras ! but all calls are still routed via dial peer 1000 - why ? I want to pass all calls thru cisco becouse i need one point for billing for asterisk and PBX calls and in future i need to make calls from PBX to the WORLD2 destinantion. PLEASE HELP! Thanks, Adam ___ 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] OH323 and outgoing calls problem.
Hello,
i have just installed OH323 and configured all outgoing calls from sip
softphones, sip context in extensions files is:
[sip]
exten = _.,1,Dial(OH323/${EXTEN})
this is only one in this context, all softphones uses this context.
After call system trying to cal h :O
It looks:
-- Registered SIP '111' at 195.XXX.XXX.XXX port 5060 expires 1800
-- Saved useragent X-Lite release 1103m for peer 111
-- Executing Dial(SIP/111-3d65, OH323/4812XXX) in new stack
-- H.323 call to 4812XXX with codec(s) XX
-- Called 4812XXX
-- OH323/48122863865-70bc is ringing
-- Hungup 'OH323/4812XXX-70bc'
== Spawn extension (sip, 4812XXX, 1) exited non-zero on 'SIP/111-3d65'
-- Executing Dial(SIP/111-3d65, OH323/h) in new stack
-- H.323 call to h with codec(s) XX
-- Called h
-- Hungup 'OH323/h-1357'
== Spawn extension (sip, h, 1) exited non-zero on 'SIP/111-3d65'
-- H.323 call 'ip$localhost/27188' cleared, reason 1 (Cleared by local user)
-- H.323 call 'ip$localhost/27189' cleared, reason 1 (Cleared by local user)
And what is:
-- Called h
-- Hungup 'OH323/h-1357'
== Spawn extension (sip, h, 1) exited non-zero on 'SIP/111-3d65'
??
On GK displays:
ACF|195.XXX.XXX.XXX:1720|3429_endp|27190|4812XXX:dialedDigits|X:dialedDigits=111:dialedDigits|false;
ARJ|195.XXX.XXX.XXX:1720|h:h323_ID|X:dialedDigits=111:dialedDigits|false|calledPartytRegistered;
DCF|195.XXX.XXX.XXX|3429_endp|27190|normalDrop;
What is the ARJ packet?
The same problem I see in this mail:
http://lists.digium.com/pipermail/asterisk-users/2005-April/098884.html
im using
asterisk-oh323-0.7.2-pre1
openh323-v1_13_5-1
pwlib-v1_6_6-1
Maybe this is configuration problem but there is no other extensions inx sip
context.
Thanks,
Adam
My oh323.conf:
My Oh323.conf:
[general]
listenAddress=ALL
listenPort=1720
tcpStart=1
tcpEnd=2
udpStart=1
udpEnd=2
fastStart=yes
h245Tunnelling=yes
h245inSetup=yes
inBandDTMF=yes
jitterMin=20
jitterMax=100
outboundMax=100
inboundMax=100
simultaneousMax=200
wrapLibTraceLevel=9
libTraceLevel=9
libTraceFile=/tmp/oh323_debug.log
gatekeeper=195.XXX.XXX.XXX
gatekeeperTTL=300
userInputMode=TONE
amaFlags=default
accountCode=H323
musionhold=default
context=voip-h323
[register]
alias=ASTERIX
prefix=*
[codecs]
odec=GSM0610
frames=4
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[Asterisk-Users] OH323 and outgoing calls.
Hello, i have configured OH323 and i have to pass outgoing calls via H.323. How to write extensions.conf rules that all numbers send to H323. Now i have only one number for tests: [sip] exten = 2929,2,Dial(OH323/48122345678) how write that any number accessed should be send to OH323? Regards, Adam Rybak ___ 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] Problems with meetme.
You need to compile zaptel with ztdummy module. Uncomment ztdummy in Makefile. You need have to loaded this module as kernel module before executing meetme. Here you will find more detailed info: http://www.voip-info.org/tiki-index.php?page=Asterisk%20timer%20ztdummy Adam Cytowanie Xisco (Personal) [EMAIL PROTECTED]: Hi everybody, I'm new in *, i have installed over fedora core 3, with kernel version 2.6 and ztdummy. I have created one conference in meetme.conf and I have modified properly extension.conf. But when I try to do a call to this extension I get the following errors: -- Executing Answer(SIP/xmg-cba9, ) in new stack -- Executing Wait(SIP/xmg-cba9, 3) in new stack Apr 9 15:53:19 NOTICE[12916]: rtp.c:453 ast_rtp_read: RTP: Received packet with bad UDP checksum -- Executing MeetMe(SIP/xmg-cba9, 1234) in new stack == Parsing '/etc/asterisk/meetme.conf': Found Apr 9 15:53:20 WARNING[12916]: chan_zap.c:841 zt_open: Unable to open '/dev/zap/pseudo': No such file or directory Apr 9 15:53:20 ERROR[12916]: chan_zap.c:6959 chandup: Unable to dup channel: No such file or directory Apr 9 15:53:20 WARNING[12916]: app_meetme.c:304 build_conf: Unable to open pseudo channel - trying device Apr 9 15:53:20 WARNING[12916]: app_meetme.c:307 build_conf: Unable to open pseudo device -- Playing 'conf-invalid' (language 'en') == Spawn extension (sip-incoming, 1000, 3) exited non-zero on 'SIP/xmg-cba9' Can anybody help me? O tell me where I have to look for... tnxs in advance. Pozdrawiam, Adam Rybak ___ 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 as protocol conventer beetwen SIP and H.323
Hello, have successfully installed Asterisk 1.o with H.323 driver and made configuration: GW (Hardware)- GnuGK - Asterisk and i call into asterisk from the PSTN network and it's work fine, but i need to make conversion from SIP small gateways to H.323. I need to make configuration like that: (Normal Phones - SIP Gateways -) x many - Asterisk - GnuGK (H.323) - Gateway (H.323) SIP Gateway and H.323 Gateway supports g.729 - i need the g729 codec into Asterisk? Can i mark sip gateways that i will can see on h.323 gateway witch from SIP gateway it comes? Can you write sample configs for me? Im Asterisk newbie :) Regards, Adam Rybak ___ 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 as protocol conventer beetwen SIP and H.323
Cytowanie Sahil Gupta [EMAIL PROTECTED]: [...] Hi, Try the OH323 implementation, we found it works better. Everyone has different experiences oviously.. Thanks, just compiled oh323 0.6.5. But still don't know how force asterisk to act as protocol converter. Regards, Adam ___ 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
