RE: [Asterisk-Users] TDM 400P in Malaysia
Hello,Just follow the instruction on this link and it should workhttp://www.digium.com/index.php?menu=configuration#TDM2XB-Original Message-From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED]]On Behalf Of Darryl WareSent: Thursday, February 23, 2006 12:23To: asterisk-users@lists.digium.comSubject: [Asterisk-Users] TDM 400P in MalaysiaHey,We are deploying an Asterisk machine in Malaysia for a client. Themachine has a TDM400P with 2 x FXO 2 x FXS. I'm wondering if there areany Malaysian users out there who might be able to help me out with arunning zap config for their phone system.Cheers,Darryl.___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] Integrating with Toshiba Strata DK40i KSU
For your first problem, try using callprogres=yes in
zapata.conf. may or may not work.Its easier to integrate with Toshiba
Strata a TE110.CCF-Original
Message-From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED]]On
Behalf Of PhilipEdelbrockSent: Friday, January 06, 2006 08:01To:
asterisk-users@lists.digium.comSubject: [Asterisk-Users] Integrating with
Toshiba Strata DK40i KSUWe've done a direct swap of an old
Amanda voicemail system with a shineynew Asterisk system (Asterisk
1.0.9). The system consists of 4 FXOports on the * box (TDM400P), and
three old Wildcards we aren't using(too buggy we found).CO
lines- Toshiba - FXO ports on *We want to branch out a little
more and use it as an auto-attendant.The first problem seems to be an
asterisk problem. When ringingextensions, it thinks the first ringback
is an answer: == CDR updated on
Zap/7-1 -- Executing Macro("Zap/7-1",
"dialexten|35") in new stack -- Executing
Dial("Zap/7-1", "Zap/6/351|5|m") in new stack --
Called 6/351 -- Started music on hold, class
'default', on Zap/7-1 -- Zap/6-1 answered
Zap/7-1 -- Stopped music on hold on
Zap/7-1 -- Attempting native bridge of Zap/7-1 and
Zap/6-1To the caller, they hear on-hold music for just a brief second,
and thenringing. When they hang up, the lines remained bridged and
theextension continues to ring until I log in and do some 'soft
hangup'commands.The second problem is more of a Toshiba problem (or
rather my lack ofknowledge of). I hope that perhaps somebody might be able
to help me? Iwant to have a way to ring multiple extensions if
sombody, say, hitszero. The Toshiba can ring mutliple extensions for
fresh new incomingcalls, but once answered I can't seem to 'unanswer' the
call to get itringing at multiple stations (we have no designated reception
phone thatis staffed 100% of the
time).Thanks!Phil___--Bandwidth
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RE: [Asterisk-Users] ericsson pabx and digium card TE110P
Did you verify with the pbx engineer on how many digitsthe pbx aresending? Usually this should be the setting in the pbx. CCF -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of vador loupeSent: Sunday, October 30, 2005 10:23To: Asterisk-Users@lists.digium.comSubject: [Asterisk-Users] ericsson pabx and digium card TE110P Hi; Could some one help me: I have a problème to make call from my pabx ericsson i receive juste 4 digit from ericsson to my asterisk any idea??? thanks zaptel.conf: span=1,1,0,ccs,hdb3,crc4bchan=1-15,17-31dchan=16loadzone=frdefaultzone=fr zapata.conf: [channels]language=frswitchtype=euroisdn pridialplan=unknownprilocaldialplan=unknown hidecallerid=nothreewaycalling=yescancallforward=yescallreturn=yesechocancel=yesechocancelwhenbridged=yesrxgain=0.0txgain=0.0immediate=no context=entrant group = 0signalling=pri_netchannel = 1-15channel = 17-31 ___ --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] Zaptel TDM questions
Yes, we have an applications that needs to detect the actual answer of the call not when it is ringing. CCF -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Angus ComberSent: Friday, September 30, 2005 19:18To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Zaptel TDM questions I think the Asterisk must answer the call to be able to then dial out on the second port. This is what happens on any other PBX I have worked with in this sort of scenario. Is this a problem for you? Angus - Original Message - From: Chee Foong To: asterisk-users@lists.digium.com Sent: Friday, September 30, 2005 10:20 AM Subject: [Asterisk-Users] Zaptel TDM questions Hello, I have a TDM04B. I make a call into the first port of the card. Once asterisk receive the call, it will make another call out using the second port. From what i have observerd as soon as the called party on the second port starts ringing asterisk show the following : -- Zap/2-1 answered Zap/1-1 Any idea why asterisk thinks the call has been answered while actually the phone is still ringing? Anybody know how to avoid asterisk to answer the call while ringing? Also, I have no Answer or any Playbackcommand in the dial plan before making a call out of second port. I have also try setting callprogress to yes/no but the results are the same. Thanks CCF ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo 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] Zaptel TDM questions
Hello, I have a TDM04B. I make a call into the first port of the card. Once asterisk receive the call, it will make another call out using the second port. From what i have observerd as soon as the called party on the second port starts ringing asterisk show the following : -- Zap/2-1 answered Zap/1-1 Any idea why asterisk thinks the call has been answered while actually the phone is still ringing? Anybody know how to avoid asterisk to answer the call while ringing? Also, I have no Answer or any Playbackcommand in the dial plan before making a call out of second port. I have also try setting callprogress to yes/no but the results are the same. Thanks CCF ___ --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] TE410 stop responding
Does anyone know why my TE410 stop respoding? usually the light on the card will show red when the cable is unplug from the card. But now it shows green even if the cable is not plugged in. The other TE410 card on the same machine works fine though. The last message generated from the card is something saying the D channel link down. But i have this message before and the D channels will usually be up again automatically. Is this a hardware issue or software one? Anyone experience this? CCF ___ --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] auto restart
Hello, I have see a post in the list saying that the 'daemon' command should be remove from the asterisk startup script in /etc/rc.d/init.d/ for FC2 in order for asterisk to auto restart when crash. I wonder if this should be done on FC3 as well, because my asterisk did not restart when crash. Please help Thanks CCF ___ --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] Re: SIP/2.0 487 Request Terminated problem onCisco 7960
This may also cause a hanging SIP channel. You can check it by issuing 'sip show channels' in CLI. CCF -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Olle E. Johansson Sent: Friday, September 09, 2005 16:52 To: Chris Stenton Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Re: SIP/2.0 487 Request Terminated problem onCisco 7960 Chris Stenton wrote: With todays CVS head I am getting the following being sent after a call has been terminated on my Cisco 7960. It eventually gives up with a critical error. chan_sip.c:1132 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 102 (Critical Response) Any ideas I am sure it was working ok with cvs head a month ago. Chris Chris, one error message out of context won't say anything to me more than the phone is having a problem with it's mental state. Propably a cousin to Marwin, the depressed robot. Please give me a full SIP debug with verbose set to 4 and debug set to 4 so I can figure out what's going on!! /O ;-) ___ --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
RE: [Asterisk-Users] Hosted PBX (vPBX) and Call/PickUP Groups
i guess may be it's a 64bit variable. so you can only use 0-63. CCF -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of René Mayorga Sent: Wednesday, September 07, 2005 15:56 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Hosted PBX (vPBX) and Call/PickUP Groups Hi, I'm working with this issue for a while, Now I already solve the dialplan issues, but I still have a question about the Callgroups, I read at www.voip-info.org that , there is a 63 limit of callgroups. And I'm wondering why?? and if the 1.2.0beta version supported more than 63 Groups?? (I did'nt find any Changelog for 1.2) or If not There is any unoficial patch for that ? Thanks in advance. -- René Mayorga [EMAIL PROTECTED] El Salvador Telecom S.A. de C.V. ___ --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] SIP channels not cleared
Hello all, When I do 'sip show channels' I have seen a lot of entries where these calls has already been terminated. Some of these channels are bolong to calls being made 2 days ago but still showing from the CLI. They look like 10.223.51.1730022676583 130b36625fc 00102/00103 unknow(d) Rx: BYE 10.223.51.1730022676583 5533069e578 00102/00103 unknow(d) Rx: BYE 10.223.51.1730016513973 234f7bba140 00102/00103 unknow(d) Rx: BYE 10.223.51.1730027226765 487b770b231 00102/00103 unknow(d) Rx: BYE 10.223.51.1730016513973 69b59aa2084 00102/00103 unknow(d) Rx: BYE 10.223.51.1730199820127 60ef984904a 00102/00103 unknow(d) Rx: BYE 10.223.51.1730081805135 45bf3e8c287 00102/00103 unknow(d) Rx: BYE I have thousands of them in 'sip show channels' and is increasing but it only shows 50 calls in 'show channels'. I believe this eats up memory. Sooner or later my system will run out of memory or get the 'Too many file opened' error. I have made a sip trace on asterisk and seems like they all share a same SIP message flow. When asterisk send an INVITE to other sip server say B. B will reply with Trying. When B found out that the actual destination can not be reached, it sends a BYE to asterisk. Asterisk then reply with a 200 OK. Call is hangup succesfully but 'sip show channels' still list the call record and never go away untill asterisk is restart. See below: Aug 15 18:35:32 VERBOSE[12402] logger.c: Reliably Transmitting (no NAT) to 10.223.51.173:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0^M Via: SIP/2.0/UDP 10.21.99.221:5060;branch=z9hG4bK6caf7db4^M From: DADAS sip:[EMAIL PROTECTED];tag=as64c4813c^M To: sip:[EMAIL PROTECTED]^M Contact: sip:[EMAIL PROTECTED]^M Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE^M User-Agent: Asterisk PBX^M Date: Mon, 15 Aug 2005 10:35:32 GMT^M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY^M Content-Type: application/sdp^M Content-Length: 160^M ^M v=0^M o=root 12402 12402 IN IP4 10.21.99.221^M s=session^M c=IN IP4 10.21.99.221^M t=0 0^M m=audio 10986 RTP/AVP 8^M a=rtpmap:8 PCMA/8000^M a=silenceSupp:off - - - -^M Aug 15 18:35:32 VERBOSE[15229] logger.c: -- SIP read from 10.223.51.173:5060: SIP/2.0 100 Trying Call-Id: [EMAIL PROTECTED] CSeq: 102 INVITE From: DADAS sip:[EMAIL PROTECTED];tag=as64c4813c To: sip:[EMAIL PROTECTED] Via: SIP/2.0/UDP 10.21.99.221:5060;branch=z9hG4bK6caf7db4 Aug 15 18:35:39 VERBOSE[15229] logger.c: -- SIP read from 10.223.51.173:5060: BYE sip:[EMAIL PROTECTED] SIP/2.0 Call-Id: [EMAIL PROTECTED] Content-Length: 0 CSeq: 103 BYE From: sip:[EMAIL PROTECTED];tag=a10111834662596 To: DADAS sip:[EMAIL PROTECTED];tag=as64c4813c Via: SIP/2.0/UDP 10.223.51.173;branch=z9hG4bK05f6ab33 Via: SIP/2.0/UDP 10.21.99.221:5060;branch=z9hG4bK6caf7db4 Aug 15 18:35:39 VERBOSE[15229] logger.c: Transmitting (no NAT) to 10.223.51.173:5060: SIP/2.0 200 OK^M Via: SIP/2.0/UDP 10.223.51.173;branch=z9hG4bK05f6ab33^M Via: SIP/2.0/UDP 10.21.99.221:5060;branch=z9hG4bK6caf7db4^M From: sip:[EMAIL PROTECTED];tag=a10111834662596^M To: DADAS sip:[EMAIL PROTECTED];tag=as64c4813c^M Call-ID: [EMAIL PROTECTED] CSeq: 103 BYE^M User-Agent: Asterisk PBX^M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY^M Contact: sip:[EMAIL PROTECTED]^M Content-Length: 0^M The SIP message exchange seems to be comply to the standard. Is this a bug in asterisk? I have a system where there is always call going on and I cant schedule asterisk to be restarted at any time to clear the channels. Any idea? I have CVS HEAD runnung on fedora 3. Thanks CCF ___ How much free photo storage do you get? Store your holiday snaps for FREE with Yahoo! Photos http://uk.photos.yahoo.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] Dell Hardware
hello, What version of the linux are you using? Do you disable hyper-threading, APIC, etc?? Thanks CCF -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Michael Miller Sent: Saturday, July 23, 2005 03:33 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Dell Hardware We have 6 dual proc Dell 1850s with a TE410P in each and they have worked without fault. I know that Digium has a compatibility note on the web site regarding the NIC but I have not seen any issues. Our largest conference with a mixture of Zap, SIP, and IAX clients was close to 200 participants on a single server had no issues. Michael -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall Sent: Friday, July 22, 2005 9:48 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Dell Hardware Guys. What do you think about Dell hardware and Asterisk? Whos using it, comments, any special specs recommended or models? ___ 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 ___ 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] LED went off after loading wct4xxp
Hello, I have already configure the zaptel.conf and ztcfg -vv shows all 124 channels are configured. Its just the light was turn off when wct4xxp is loaded (with no error). CCF -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Sean Lowry Sent: Friday, July 15, 2005 18:07 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] LED went off after loading wct4xxp The lights are supposed to go off They will only come on if you have configured the span on in the /etc/zaptel.conf Sean -Original Message- From: Chee Foong [mailto:[EMAIL PROTECTED] Sent: 15 July 2005 02:28 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] LED went off after loading wct4xxp Hello, I have a Digium TE410P card. I get the knight rider lights before the module (wct4xxp) loads, but after the modules are loaded I don't get any lights. I have found the following 2 posts but still could not solve the problem http://lists.digium.com/pipermail/asterisk-users/2004-November/075277.html http://www.voip-info.org/tiki-index.php?page=Asterisk+TE410p+No+Interrupts I even disable all unused onboard modules in the bios like USB, serial, parallel port, etc. I have tried CVS HEAD and also stable version My system are: Fedora 3 TYAN S5350 (Tiger i7320) motherboard TE410P Any one has any idea or facing the same issue? CCF ___ 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 ___ 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] LED went off after loading wct4xxp
Hello, I have a Digium TE410P card. I get the knight rider lights before the module (wct4xxp) loads, but after the modules are loaded I don't get any lights. I have found the following 2 posts but still could not solve the problem http://lists.digium.com/pipermail/asterisk-users/2004-November/075277.html http://www.voip-info.org/tiki-index.php?page=Asterisk+TE410p+No+Interrupts I even disable all unused onboard modules in the bios like USB, serial, parallel port, etc. I have tried CVS HEAD and also stable version My system are: Fedora 3 TYAN S5350 (Tiger i7320) motherboard TE410P Any one has any idea or facing the same issue? CCF ___ 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] Setting Caller ID after Dial
Hello, I have the following situation: I have a PRI with 200 DID numbers and I have set up 200 sip extensions that matches the last 4 digit of the corresponding DID numbers so that when any of the 200 DID number is called, asterisk can pass the call to the respective sip extension. Incomming has been fine. But when making out going calls I want the called party to always see the same number (which is one of the number selected from the 200 DID numbers). This I can be achieved in asterisk by calling SetCallerID before Dial command. However in the CDR, the caller id number of the number that i set using SetCallerID is always logged and there is no trace of which sip extension is making the call since the caller is always the same. This has become a serious trouble for billing. I have been searching around and could not seems to get a solution. I have tried DIAL_STATUS variable (only work if call is not answered), using 'g' option in Dial command (does not work if calling party hangup first), etc. Is there a solution or work around for this? Thanks in advance CCF ___ How much free photo storage do you get? Store your holiday snaps for FREE with Yahoo! Photos http://uk.photos.yahoo.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] Setting Caller ID after Dial
Hey Mark, Have you tested on doing transfer (blind and attended)? Are the extensions in the CDR still correct? CCF --- Mark Johnson [EMAIL PROTECTED] wrote: Chee Foong Chiew wrote: Hello, I have the following situation: I have a PRI with 200 DID numbers and I have set up 200 sip extensions that matches the last 4 digit of the corresponding DID numbers so that when any of the 200 DID number is called, asterisk can pass the call to the respective sip extension. Incomming has been fine. But when making out going calls I want the called party to always see the same number (which is one of the number selected from the 200 DID numbers). This I can be achieved in asterisk by calling SetCallerID before Dial command. However in the CDR, the caller id number of the number that i set using SetCallerID is always logged and there is no trace of which sip extension is making the call since the caller is always the same. This has become a serious trouble for billing. I have been searching around and could not seems to get a solution. I have tried DIAL_STATUS variable (only work if call is not answered), using 'g' option in Dial command (does not work if calling party hangup first), etc. Is there a solution or work around for this? Thanks in advance CCF I forgot in my last post to mention that I use Postgres for my CDR, and the SIP extension can be pulled from the channel column. That way, the callerid is still the way it appeared when the calls were placed. I just strip everything from the '-' to the right and it's worked great for me! Mark ___ 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 ___ How much free photo storage do you get? Store your holiday snaps for FREE with Yahoo! Photos http://uk.photos.yahoo.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] Setting Caller ID after Dial
Actually I already using account code for billing, so billing is fine. I have a 3rd party reporting software that tie extension numbers to departments. At the end of a month the person in charge will generate report on the call statistic for each department. My problem is now the report showing only one department are making calls because every outgoing call is from the same caller number. --- Chris A. Icide [EMAIL PROTECTED] wrote: What about setting and using Accountcode for each sip client? It tracks separately than callerid in the cdr. so in your sip.conf, add an accountcode= statement for each sip entry, and in the AccountCode field in the CDR, you'll have the correct entry needed to determine who made the call. -Chris Chee Foong Chiew wrote: Hello, I have the following situation: I have a PRI with 200 DID numbers and I have set up 200 sip extensions that matches the last 4 digit of the corresponding DID numbers so that when any of the 200 DID number is called, asterisk can pass the call to the respective sip extension. Incomming has been fine. But when making out going calls I want the called party to always see the same number (which is one of the number selected from the 200 DID numbers). This I can be achieved in asterisk by calling SetCallerID before Dial command. However in the CDR, the caller id number of the number that i set using SetCallerID is always logged and there is no trace of which sip extension is making the call since the caller is always the same. This has become a serious trouble for billing. I have been searching around and could not seems to get a solution. I have tried DIAL_STATUS variable (only work if call is not answered), using 'g' option in Dial command (does not work if calling party hangup first), etc. Is there a solution or work around for this? snip ___ 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 ___ Yahoo! Messenger - NEW crystal clear PC to PC calling worldwide with voicemail http://uk.messenger.yahoo.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] Core Dump
Hello, My asterisk crash from time to time, at least twice a day. Reomendation from a site is to do a gdb on core dump Below is what i get, but i have no idea what is going on Does anybody have any idea? (gdb)bt #0 0x00181aed in _int_malloc () from /lib/tls/libc.so.6 #1 0x00180dfd in malloc () from /lib/tls/libc.so.6 #2 0x00177b03 in vasprintf () from /lib/tls/libc.so.6 #3 0x0808b663 in ast_cli (fd=1, fmt=0x1 Address 0x1 out of bounds) at cli.c:54 #4 0x080a0725 in manager_event (category=2, event=0x80e812e Newexten, fmt=0x80e6da0 Channel: %s\r\nContext: %s\r\nExtension: %s\r\nPriority: %d\r\nApplication: %s\r\nAppData: %s\r\nUniqueid: %s\r\n) at manager.c:1420 #5 0x08089345 in pbx_extension_helper (c=0x9815758, con=0x1, context=0x98158a8 macro-hangupcall, exten=0x981599c s, priority=3, label=0x0, callerid=0x96bc008 Wait, action=159472384) at pbx.c:1609 #6 0x08087767 in ast_spawn_extension (c=0x1, context=0x1 Address 0x1 out of bounds, exten=0x1 Address 0x1 out of bounds, priority=1, callerid=0x1 Address 0x1 out of bounds) at pbx.c:2206 #7 0x008c96a4 in macro_exec (chan=0x9815758, data=0x200) at app_macro.c:173 #8 0x0808938b in pbx_extension_helper (c=0x9815758, con=0x1, context=0x98158a8 macro-hangupcall, exten=0x981599c s, priority=1, label=0x0, callerid=0x1cef1d0 hangupcall, action=0) at pbx.c:528 #9 0x08080c27 in ast_pbx_run (c=0x9815758) at pbx.c:2206 #10 0x00683d41 in ss_thread (data=0x9815758) at chan_zap.c:4975 #11 0x00669dac in start_thread () from /lib/tls/libpthread.so.0 #12 0x001eb9ea in clone () from /lib/tls/libc.so.6 thanks Regard CCF ___ 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] Manager Port
Hello all, I am using flash operator panel, when i stop iptables everthing is fine, but once iptables is started, the operator panel doesn't work anymore. Anyone know how to set up the iptable in order for to op panel to work? I am using tcp port 5038 for asterisk manager, and I have try open both tcp and udp port 5038 in iptables but without success. thanks CCF ___ 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] Modprobe wctdm hang at command prompt
Hello all, I have a digium TDM40B install in my Dell PowerEdge 1800. when I run modprobe wctdm nothing happen and it does not go to the next linux prompt util I press control c. It just like hanging at the prompt. Is anyone having the same problem? I have try asterisk stable (wcfxs) and CVS HEAD (wctdm) but result are the same. I have been searching and troubleshooting for 3 days but still do not know what is happening. Any help is apreciated. Thanks in advance. CCF ___ 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] Voicemail quota
Hello, Is there a way to put a voicemail quota to a SIP user? I mean a quota on the user's mailbox instead of a particular message of the user like the 'maxmessage' does currently. Quata can be total file size of message or total minutes of messages of a mailbox. Thanks Foong ___ 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 did not play music when pressing hold button on SJPhone
Hello, On my setup, I can't seem to get asterisk to play music on hold when i press the hold button on sjphone (does not work on x-lite as well). I have already set the musicclass=default in sip.conf and default = mp3:/var/lib/asterisk/mohmp3 in musiconhold.conf. the music play fine when pressing # to transfer a call, so i conclude that I installed mpg123 correctly. However, music only play if # is press but not the 'transfer' button is press on sjphone. Does anyone has the same issue? Is there anyway to solve this? Thanks in advance Foong ___ 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] SetCallerID({$NEWCALLERID})
Title: SetCallerID({$NEWCALLERID})
should
use ${NEWCALLERID}
NOTE:
$ come before {
Foong
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Steven
FrazierSent: Friday, March 11, 2005 11:23 AMTo:
asterisk-users@lists.digium.comSubject: [Asterisk-Users]
SetCallerID({$NEWCALLERID})
I am trying to SetCallerID to a variable I have
defined. This obviously is wrong. It actually sets the caller ID
to $NEWCALLERID. I have search through the examples on wiki but wasn't
able to find something similar to see what I was doing wrong. Could
someone tell me the correct way to SetCallerID to a defined
variable?
exten =
2125551212,5,SetCallerID({$NEWCALLERID})
exten =
2125551212,6,Noop(${CALLERID})
Actually shows $NEWCALLERID instead of the contents
of $NEWCALLERID.
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[Asterisk-Users] SIP URI
Hello, I try to append a URI to the SIP dial syntax, however the URI were not shown in the sip debug message. I have read one of the post in the list which actualy show the URI string in the debug message (at the To: field). Is there any setting I need to set or turn on during compilation of asterisk? I have the head version of asterisk and my extension.conf setting is proveded below: exten = 777,1,Answer exten = 777,2,SetVar(VXML_URL=play=http%3A%2F%2Flocalhost%2Fvxml%2Findex.vxml) exten = 777,3,Dial(SIP/[EMAIL PROTECTED],10,t) exten = 777,4,Hangup SIP Debug message: *CLI dial 777 -- Executing Answer(OSS/dsp, ) in new stack Console call has been answered -- Executing SetVar(OSS/dsp, VXML_URL=play=http%3A%2F%2Flocalhost%2Fvxml%2Findex.vxml) in new stack -- Executing Dial(OSS/dsp, SIP/[EMAIL PROTECTED]|10|t) in new stack We're at 192.168.1.74 port 18952 Answering with capability 0x2 (gsm) Answering with capability 0x4 (ulaw) Answering with capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) 12 headers, 12 lines Reliably Transmitting: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.74:5060;branch=z9hG4bK280927bb From: asterisk sip:[EMAIL PROTECTED];tag=as2e2564e0 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Mon, 07 Mar 2005 16:21:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 263 Thanks CFC ___ 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 supports VXML?
I have tried running 2 open source VXML browsers with * but without success: 1. sipXvxml - when started, it acts as a SIP endpoint. However I was unable to make * to pass it a URI (which is posible i guess from a post i read in this list). Also it seems to use the dsp as *, therefore if sipXvxml is installed in the same box as *, you can't issue dial command from the asterisk console. 2. publicVoiceXml - supports only CAPI. I am still looking at sipXvxml, trying to make it work. If you manage to get them work please let me know :). Foong -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Alistair Cunningham Sent: Monday, March 07, 2005 10:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] asterisk supports VXML? Marco, /me goes back and reads the rest of the email as he should have in the first place. What they're talking about is an external VoiceXML browser which they connect to over SIP, just as I've mentioned with Cisco. I don't know which browser though. Time for me to get stronger glasses, I think. Alistair Cunningham, Integrics Ltd, Telephony, Database, Unix consulting worldwide +44 (0)7870 699 479 http://integrics.com/ Alistair Cunningham wrote: Marco, There isn't. When asked about VoiceXML by my customers, I recommend using a Cisco router for VXML interpretation, and SIP to integrate it with Asterisk. There are a wide variety of PC based proprietary VXML browsers that you can use instead of Cisco. Alistair Cunningham, Integrics Ltd, Telephony, Database, Unix consulting worldwide +44 (0)7870 699 479 http://integrics.com/ Marco Parisotto wrote: Hi all where can I find infos about this VXML intepreter for asterisk? Thanks Marco Hi Foong, That's a good question you've put out there. Yes, Asterisk supports VXML and here's how it's done; Firstly in the SIP.conf, you need to have your VXML application/browser defined; sip.conf: [vxmlapp] type=friend insecure=yes username=777 reinvite=no host=123.45.67.8 Then in the EXTENSIONS.conf it will look like this; extensions.conf: exten = 777,1,Setvar,VXML_URL=voicexml=http%3A%2F%2F123.45.67.20%3A6969%2Fhellovxml% 2Fhellovxml exten = 777,2,Dial,sip/vxmlapp|10 exten = 777,3,HangUp Hope this'll clear your thoughts. Cheers! Lilantha Karunaratne MSCS Tel: (65) 90403497 _ From: asterisk-users-bounces at lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users [mailto:asterisk-users-bounces at lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users] On Behalf Of Chee Foong Sent: Friday, February 25, 2005 10:17 AM To: asterisk-users at lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users Subject: [Asterisk-Users] asterisk supports VXML? Hello, Does asterisk supports VXML? Couldn't find much resource on that on google and wiki. Thanks Foong ___ 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 ___ 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
[Asterisk-Users] asterisk supports VXML?
Hello, Does asterisk supports VXML? Couldn't find much resource on that on google and wiki. Thanks Foong ___ 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] G.729 pass thru Asterisk
Hello, I have te following setup: IAX client -(iax)- Asterisk -(h323) Cisco AS5300 At the present moment GSM codec is used betwee IAX client and Asterisk. G729 is used between Asterisk and Cisco AS5300. I am thinking that switching from GSM to G729 between IAX client and Asterisk. I know I need G729 licence at the IAX client, but at the Asterisk side can I make Asterisk pass through G729 to Cisco AS5300. This way I do not have to purchace G729 licence for the Asterisk server only for the IAX client. I wonder how this can be done in Asterisk? For example what should I set in the iax.conf or any other .conf file? Reading some of the post in the mailing list, someone mention Asterisk only support passthrough for G723. is that true? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H.323 and G729: Another sad tale
Hello, Can g729 calls of type g729r8 or g729br8 from a Cisco AS5300 be terminated on Asterisk systems and sent out Zap interfaces? A while ago, I only manage to get g729 call works when terminating in Cisco AS5300 from Asterisk but was unable to terminate call in Asterisk from Cisco AS53000 using g729. If the answer is Yes, then are there any specific patches I will need? Which of the two H323 drivers works? Both? Of course, I assume that the G729 licenses from Digium are required for each active channel. not sure about patches, however if you plan to use chan_h323, it is best to get the CORRECT versions of pwlib and openh323 and follow the exact installation instructions. One important thing about these libraries with chan_h323 is DO NOT 'make install' pwlib and openh323 hth ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Trouble with 2 NIC cards
Hello, I have the quite similiar problem like yours except that both of my NIC have fix public ip from different ISP provider. Unfortunately we are unable to make it work. This is due to some routing issues of the Asterisk box. My collegue was trying hard to seting up the routing tables, but did no succeed. We finally has given up trying. The solution we have make is to have 2 different asterisk iax server and make these server peer to each other, but not yet try though. - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, October 24, 2003 7:39 AM Subject: [Asterisk-Users] Trouble with 2 NIC cards Greetings everyone. Did anyone try using 2 NIC cards on the machine? For some reason, asterisk can not identify which IP should be used. In the config files (IAX.conf, sip.conf etc), there is a way to bind the IP address but if the machine is hooked to a DHCP server (such as cable modem), then fix IP doesn't work. It should be simple to bind it to a perticular ethernet card (eth0 or eth1) instead of an IP address. Anyone tried multiple NICs with asterisk? Please write your commentes. Thanks. Ricky ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX with multiple NIC
Hello, I have been using IAX to serve clients endpoints for a while with no problem. But recently, to increase the bandwidth to the Asterisk server, I add another network interface card to my Asterisk server which connected to a different service provider that I currently have. Both of my nic is assigned different public ip. the client will actually choose one of these ip and authenticate itself. However clients seems to be only able to authenticate using one of the ip(the old ip) even if i have configure (iax.conf) iax module to listen to all interface on the asterisk server. I wonder if ther is anybody having the same problem like mine when there are 2 nic on a Asterisk server and would like to share your findings and experience. Is iax designed to handle multiple network interface in the first place? Foong ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Manager
Hello all, Can I execute linux command like(ls, mkdir) through the Manager interface? I can't seem to access the manual at digium.com. I keep getting 'Forbidden' error. Looks like they are upgrading or something. CF ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX
Hello All, Is it possible to make asterisk to do authetication of IAX client through database (mysql, etc) instead of creating all the client username in iax.conf? How hard is to implementthe feature i describe above? We plan to use IAX as part of our VOIP infrastructure mainly because it penetrate NAT/firewall with ease. Foong
[Asterisk-Users] Radius + Asterisk
Hello all, I have read a post saying that someone is implementing Radius function in Asterisk. Does it come with the current version of Asterisk? I wonder if Asterisk can be register as a NAS to an Radius server? Foong
Re: [Asterisk-Users] G729 experiences..
Can't really remember, If I am not mistaken you dont have to reregister the codec. unless you format your harddisk. If your using chan_h323, you need to modify its makefile to compile with g.729 support every time you download from cvs.(something that I always forgot to do) :). Foong - Original Message - From: WipeOut [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, September 26, 2003 3:37 PM Subject: Re: [Asterisk-Users] G729 experiences.. Chee Foong wrote: Quality are good, However doesn't seem to get the codec to work with incomming call from Cisco AS5300. Outgoing call to AS5300 is ok. safe_asterisk does work. Foong When recompling Asterisk is there anything special that you have to do if you have G.729 installed? in otherwords do you have to reinstall it or re-register it or anything else.. Later ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G729 experiences..
No, I dont think you need a zap device. I used to run meetme, where all conference participants are from IP endpoints (G.729) without any zaptel device. I just added a digium E100P recently, works without problem so far. I am not sure about the relationship, may be what they mean is IP endpoints callling PSTN lines through asterisk(with zap devices) works using digium's G.729. Foong - Original Message - From: WipeOut [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, September 26, 2003 3:43 PM Subject: Re: [Asterisk-Users] G729 experiences.. Chee Foong wrote: Quality are good, However doesn't seem to get the codec to work with incomming call from Cisco AS5300. Outgoing call to AS5300 is ok. safe_asterisk does work. Foong Another question.. Is zaptel hardware required in order to use the G.729 codec?? The reason for the, what may seem like a silly question is that on the digium website they comment The G.729 codec works with all Digium cards.. I am wondering what relationship there is between digium cards and codecs?? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX calling number
Hello, I am recentlyinspecting the IAX protocol.. I wonder if there away to associate a user name to a number say I have a client register to the IAX server with username 'John' andI want to associate a number say '12345678' tho John so otherregister users can call john by dialing 12345678. Something like the H323_id and the E164 alias in H323 protocol. Foong
Re: [Asterisk-Users] IAX calling number
Ahh...Understood. That's possible.
But my problem is I will have 500 users (and increasing). I can't have an
entry for every users in the config file. The only way to handle this so far
I found is to use number as username, therefore we can use only 1 extension:
exten = _700XX,1,Dial(IAX/${EXTEN})
But user wont like it if username is a long string of number, they prefer
meaningful name.
Thanks anyway.
Foong
- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, September 26, 2003 3:47 PM
Subject: Re: [Asterisk-Users] IAX calling number
On Fri, 26 Sep 2003, Chee Foong wrote:
I wonder if there away to associate a user name to a number say I have a
client register to the IAX server with username 'John' and I want to
associate a number say '12345678' tho John so other register users can
call john by dialing 12345678. Something like the H323_id and the E164
alias in H323 protocol.
exten = 12345768,1,Dial(IAX/john)
- wasim of the it doesn't get any simpler than this cult
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Re: [Asterisk-Users] G729 experiences..
Where did you install asterisk? foong - Original Message - From: Max Speransky [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, September 26, 2003 7:07 PM Subject: Re: [Asterisk-Users] G729 experiences.. On Fri, Sep 26, 2003 at 03:54:57PM +0800, Chee Foong wrote: And what I need to do if my asterisk box don't have a harddisk ? I plan to make it on flash or tftpbooting ... May be somebody comment this ? Can't really remember, If I am not mistaken you dont have to reregister the codec. unless you format your harddisk. If your using chan_h323, you need to modify its makefile to compile with g.729 support every time you download from cvs.(something that I always forgot to do) :). Foong - Original Message - From: WipeOut [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, September 26, 2003 3:37 PM Subject: Re: [Asterisk-Users] G729 experiences.. Chee Foong wrote: Quality are good, However doesn't seem to get the codec to work with incomming call from Cisco AS5300. Outgoing call to AS5300 is ok. safe_asterisk does work. Foong When recompling Asterisk is there anything special that you have to do if you have G.729 installed? in otherwords do you have to reinstall it or re-register it or anything else.. Later ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- ... All opinions expressed are mine and not those of my employer. Yours, Max [Msg N 2278] --- mailto: [EMAIL PROTECTED] phone: +380-44-2054455 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Meetme question
Have you got a zaptel device?? Can you post you meetne.conf? - Original Message - From: C. Johnson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, September 25, 2003 3:43 PM Subject: [Asterisk-Users] Meetme question Ok.. I got * and SIP working internally now .. still wrestling with connecting two remote * pbx's together.. I'll save that for another day though :) I setup Meetme on this new * PBX, but when I try to dial to join the conference, I hear a recording saying the conference is invalid or something to that effect. Here's a copy of my log files: == Parsing '/etc/asterisk/meetme.conf': Found WARNING[24592]: File app_meetme.c, Line 154 (build_conf): Unable to open pseudo channel It then hangs up.. Anyone seen this before?? -cj ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G729 experiences..
Quality are good, However doesn't seem to get the codec to work with incomming call from Cisco AS5300. Outgoing call to AS5300 is ok. safe_asterisk does work. Foong - Original Message - From: Matthew Hardeman [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, September 26, 2003 4:27 AM Subject: RE: [Asterisk-Users] G729 experiences.. It's ok... The voice sounds fine. It's superior to most cell phone calls, anyway. I've used it with the Cisco 7960's without any trouble. You can use asterisk in any way that uses it in console mode. Safe asterisk does so, so you can use it. This may be otherwise fixed, but I'm not sure. As safe asterisk works, I don't worry about it. Voicemail will use one license for each output stream it has to transcode. Therefore, it is preferable if you are using G729 to only write out one format of voicemail recording. I use WAV49, which is small like GSM, but easier to play on default windows installs with any kind of decent media player installed. It *does* properly release the license when done. (At least now, on my system, it does.) Matt Hardeman PaperSoft -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of WipeOut . Sent: Thursday, September 25, 2003 7:02 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] G729 experiences.. Hi, I am still toying with the idea of going ahead with using the G.729.. Can those using it tell me about some of your experiences using G.729.. Things like and problems you had running it, the voice quality and anything else you can think of... I have read in the archives that asterisk has to be run with -c.. Is this still the case? and if so does this mean that * can't be run using the safe_asterisk script? or started remotely via an SSH session?? I have also read that the voicemailmain app uses up licences.. Does this still happen and how many does it use?? Thanks.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AGI: getting the return code from an exec()'d application?
Hello Steven, I am planing to do the same thing: make dial return correct dial status and use agi to detect it. Is it possible for you to share the modified dial source. If not, can you provide some pointers on how to hack it? I am no real C programmer :( If i am not mistaken, result return by exec is like: 200 Result=number additional information, if any does the status appear in the 'additional ionformation' portion? - Original Message - From: Steven J. Sobol [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, September 26, 2003 8:48 AM Subject: [Asterisk-Users] AGI: getting the return code from an exec()'d application? So I hacked up the Dial app to return a numeric return code instead of changing contexts based on a number being busy or unanswered. The purpose for this modified dial app, which I call AGIDial, is to help me concoct a follow-me type of application. The app returns -1 for a completed call, 0 for unanswered, or 1 for busy. Well, I hooked the thing up to an AGI script that uses perl and AGI.pm, and ran some tests. The AGIDial app is definitely returning the right status codes and is able to differentiate between the three types of call termination. But the AGI script always reports a status code of 0. And I figured out why. $AGI-exec() seems to grab the return code of a Perl print() command which outputs the command to the server - but the return code of the print() is not what I want - the return code of the application is what I want. How do I exec an app through AGI and get *its* return code? -- JustThe.net Internet Multimedia Services 22674 Motnocab Road * Apple Valley, CA 92307-1950 Steve Sobol, Proprietor 888.480.4NET (4638) * 248.724.4NET * [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] G.729A + Cisco AS5300
Hello, I have 5 digium's g.729 codecs and succesfully register with asterisk, I have incomming call from my cisco AS5300 to Asterisk through IP. But Asterisk always use g711 ulaw instead of g.729. When I disable all other codecs other than g.729 in both cisco and asterisk, calls get dropped once connected. The codec list show onmy cisco AS5300for g.729 are: g729r8 g729br8 I suspect thatdigium's g.729 is not compatible with these codec found on cisco AS5300. Am I correct? Any advice will be helpful Foong
Re: [Asterisk-Users] G.729A + Cisco AS5300
IC, does that means they are not compatible?. Funny thing is, call make from asterisk to AS5300is fine using codec G.729. But call from AS5300 to asterisk result in incompatible codec. This is very strange. Foong - Original Message - From: Tjardick van der Kraan To: [EMAIL PROTECTED] Sent: Monday, September 22, 2003 3:50 PM Subject: Re: [Asterisk-Users] G.729A + Cisco AS5300 the G.729 from digium are the G.729A type. Greetings, Tj -- Tjardick van der Kraan Tel +32 4 34 40 522Fax +32 4 34 40 525GSM +32 497 45 27 36 IAXtel: 1 700 344 0522FWD: 26322IPtel: 91331 Belgium - Original Message - From: Chee Foong To: [EMAIL PROTECTED] Sent: Monday, September 22, 2003 9:10 AM Subject: [Asterisk-Users] G.729A + Cisco AS5300 Hello, I have 5 digium's g.729 codecs and succesfully register with asterisk, I have incomming call from my cisco AS5300 to Asterisk through IP. But Asterisk always use g711 ulaw instead of g.729. When I disable all other codecs other than g.729 in both cisco and asterisk, calls get dropped once connected. The codec list show onmy cisco AS5300for g.729 are: g729r8 g729br8 I suspect thatdigium's g.729 is not compatible with these codec found on cisco AS5300. Am I correct? Any advice will be helpful Foong
Re: [Asterisk-Users] G.729A + Cisco AS5300
Hello, I am using H.323 with chan_h323. Here is my config in h323.conf: allow=g729 if I set allow=ulaw, G7.11 alway get used. Therefore I disallow it. I want to use G.729. G.711 is too heavy for my network Any with AS5300 manage to get the digium's g.729 working Foong - Original Message - From: Eric Wieling [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, September 22, 2003 4:10 PM Subject: Re: [Asterisk-Users] G.729A + Cisco AS5300 Are you using SIP or H323? If SIP, what are the allow= and disallow= lines in your sip.conf? On Mon, 2003-09-22 at 03:08, Chee Foong wrote: IC, does that means they are not compatible?. Funny thing is, call make from asterisk to AS5300 is fine using codec G.729. But call from AS5300 to asterisk result in incompatible codec. This is very strange. Foong - Original Message - From: Tjardick van der Kraan To: [EMAIL PROTECTED] Sent: Monday, September 22, 2003 3:50 PM Subject: Re: [Asterisk-Users] G.729A + Cisco AS5300 the G.729 from digium are the G.729A type. Greetings, Tj -- Tjardick van der Kraan Tel +32 4 34 40 522 Fax +32 4 34 40 525 GSM +32 497 45 27 36 IAXtel: 1 700 344 0522 FWD: 26322 IPtel: 91331 Belgium - Original Message - From: Chee Foong To: [EMAIL PROTECTED] Sent: Monday, September 22, 2003 9:10 AM Subject: [Asterisk-Users] G.729A + Cisco AS5300 Hello, I have 5 digium's g.729 codecs and succesfully register with asterisk, I have incomming call from my cisco AS5300 to Asterisk through IP. But Asterisk always use g711 ulaw instead of g.729. When I disable all other codecs other than g.729 in both cisco and asterisk, calls get dropped once connected. The codec list show on my cisco AS5300 for g.729 are: g729r8 g729br8 I suspect that digium's g.729 is not compatible with these codec found on cisco AS5300. Am I correct? Any advice will be helpful Foong __ This message has been 'sanitized'. This means that potentially dangerous content has been rewritten or removed. The following log describes which actions were taken. Sanitizer (start=1064217921): Part (pos=3455): SanitizeFile (filename=unnamed.txt, mimetype=text/plain): Match (names=unnamed.txt, rule=1): ScanFile (file=/tmp/att-3f6ead42-MII-unnamed.txt): Scan succeeded, file is clean. Enforced policy: unknown Match (names=unnamed.txt, rule=3): Enforced policy: accept Added 1 bytes of scratch space. Total modifications so far: 1 Part (pos=5049): SanitizeFile (filename=unnamed.html, mimetype=text/html): Match (names=unnamed.html, rule=1): ScanFile (file=/tmp/att-3f6ead42-PQH-unnamed.html): Scan succeeded, file is clean. Enforced policy: unknown Match (names=unnamed.html, rule=3): Enforced policy: accept Added 1 bytes of scratch space. Note: Styles and layers give attackers many tools to fool the user and common browsers interpret Javascript code found within style definitions. References: - http://www.securityfocus.com/bid/630 - http://archives.indenial.com/hypermail/bugtraq/2001/January2001/0512.html Rewrote HTML tag: _style_0 _/STYLE_ as: _DANGEROUS_style_0 _/STYLE_ Rewrote HTML tag: _DIV_ as: _p__DANGEROUS_DIV_ Rewrote HTML tag: _/DIV_ as: _/p__DANGEROUS_DIV_ Rewrote HTML tag: _DIV_ as: _p__DANGEROUS_DIV_ Rewrote HTML tag: _/DIV_ as: _/p__DANGEROUS_DIV_ Rewrote HTML tag: _DIV_ as: _p__DANGEROUS_DIV_ Rewrote HTML tag: _/DIV_ as: _/p__DANGEROUS_DIV_ Rewrote HTML tag: _DIV_ as: _p__DANGEROUS_DIV_ Rewrote HTML tag: _/DIV_ as: _/p__DANGEROUS_DIV_ Rewrote HTML tag: _DIV_ as: _p__DANGEROUS_DIV_ Rewrote HTML tag: _/DIV_ as: _/p__DANGEROUS_DIV_ Rewrote HTML tag: _DIV_ as: _p__DANGEROUS_DIV_ Rewrote HTML tag: _/DIV_ as: _/p__DANGEROUS_DIV_ Rewrote HTML tag: _DIV_ as: _p__DANGEROUS_DIV_ Rewrote HTML
Re: [Asterisk-Users] G.729A + Cisco AS5300
hello, I have tried that but get disconnected once asterisk answer the call. Got the following error 1:02.899 H225 Answer:813ae50 h323.cxx(4167) H323 CreateLogicalChannel - unknown data type Guess it's the difference btw g.729 on AS5300 and g.729 on asterisk. Cisco AS5300 has G.729 and G.729 Annex-B while digium's is G.729 Annex-A. Still wondering why calling from asterisk to AS5300 works using the digium codec since they are different. Thanks Foong - Original Message - From: Eric Wieling [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, September 22, 2003 5:30 PM Subject: Re: [Asterisk-Users] G.729A + Cisco AS5300 add a disallow=all above the allow=g729 line. On Mon, 2003-09-22 at 04:28, Chee Foong wrote: Hello, I am using H.323 with chan_h323. Here is my config in h323.conf: allow=g729 if I set allow=ulaw, G7.11 alway get used. Therefore I disallow it. I want to use G.729. G.711 is too heavy for my network Any with AS5300 manage to get the digium's g.729 working Foong - Original Message - From: Eric Wieling [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, September 22, 2003 4:10 PM Subject: Re: [Asterisk-Users] G.729A + Cisco AS5300 Are you using SIP or H323? If SIP, what are the allow= and disallow= lines in your sip.conf? On Mon, 2003-09-22 at 03:08, Chee Foong wrote: IC, does that means they are not compatible?. Funny thing is, call make from asterisk to AS5300 is fine using codec G.729. But call from AS5300 to asterisk result in incompatible codec. This is very strange. Foong - Original Message - From: Tjardick van der Kraan To: [EMAIL PROTECTED] Sent: Monday, September 22, 2003 3:50 PM Subject: Re: [Asterisk-Users] G.729A + Cisco AS5300 the G.729 from digium are the G.729A type. Greetings, Tj -- Tjardick van der Kraan Tel +32 4 34 40 522 Fax +32 4 34 40 525 GSM +32 497 45 27 36 IAXtel: 1 700 344 0522 FWD: 26322 IPtel: 91331 Belgium - Original Message - From: Chee Foong To: [EMAIL PROTECTED] Sent: Monday, September 22, 2003 9:10 AM Subject: [Asterisk-Users] G.729A + Cisco AS5300 Hello, I have 5 digium's g.729 codecs and succesfully register with asterisk, I have incomming call from my cisco AS5300 to Asterisk through IP. But Asterisk always use g711 ulaw instead of g.729. When I disable all other codecs other than g.729 in both cisco and asterisk, calls get dropped once connected. The codec list show on my cisco AS5300 for g.729 are: g729r8 g729br8 I suspect that digium's g.729 is not compatible with these codec found on cisco AS5300. Am I correct? Any advice will be helpful Foong __ This message has been 'sanitized'. This means that potentially dangerous content has been rewritten or removed. The following log describes which actions were taken. Sanitizer (start=1064217921): Part (pos=3455): SanitizeFile (filename=unnamed.txt, mimetype=text/plain): Match (names=unnamed.txt, rule=1): ScanFile (file=/tmp/att-3f6ead42-MII-unnamed.txt): Scan succeeded, file is clean. Enforced policy: unknown Match (names=unnamed.txt, rule=3): Enforced policy: accept Added 1 bytes of scratch space. Total modifications so far: 1 Part (pos=5049): SanitizeFile (filename=unnamed.html, mimetype=text/html): Match (names=unnamed.html, rule=1): ScanFile (file=/tmp/att-3f6ead42-PQH-unnamed.html): Scan succeeded, file is clean. Enforced policy: unknown Match (names=unnamed.html, rule=3): Enforced policy: accept Added 1 bytes of scratch space. Note: Styles and layers give attackers many tools to fool the user and common browsers interpret Javascript code found within style definitions. References: - http://www.securityfocus.com/bid/630 - http://archives.indenial.com/hypermail
[Asterisk-Users] Chan_h323 config
Hello, Camparing chan_h323 config with chan_oh323 config, In the codec section chan_oh323 allow me to specify frame value. Is there a equivalent in chan_h323? Or if not, what is the default frame value if I use G.729(digium). Foong
[Asterisk-Users] Meetme Admin menu
Hello, Is there a asterisk developer guide/source code doc or something like that? I want to see if I can implement the admin menu function for meetme. Foong
Re: [Asterisk-Users] G.729A + Cisco AS5300
Hello, Actually call from asterisk to AS5300 works fine with G.729. But not the other way round. I have tried enable all codecs, enable only g.729 on AS5300 but did not manage to get it work May I know what's you setting on both side Jeremy? Thanks for the reply Foong - Original Message - From: Jeremy McNamara [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, September 23, 2003 12:18 AM Subject: Re: [Asterisk-Users] G.729A + Cisco AS5300 Eric Wieling wrote: I doubt that it's a codec problem. Maybe chan_h323 doesnt' support G729. JerJer would know. I babysit systems that terminate hundreds of thousands of G.729 based H.323 calls per day using chan_h323 and As5300. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G.729A + Cisco AS5300
Yes, you are right. H.323 incoming call from the As5300 doesn't succeed. outgoing call to AS5300 works fine like your system. Foong - Original Message - From: Jeremy McNamara [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, September 23, 2003 11:14 AM Subject: Re: [Asterisk-Users] G.729A + Cisco AS5300 Chee Foong wrote: Hello, Actually call from asterisk to AS5300 works fine with G.729. But not the other way round. I have tried enable all codecs, enable only g.729 on AS5300 but did not manage to get it work May I know what's you setting on both side Jeremy? My systems only do termination: Asterisk---Dial,H3235300PSTN To be clear you are talking about a H.323 incoming call from the As5300 doesn't succeed? Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] E1 config - Telekom Malaysia
Hello does anyone has experience in setting upE100P with E1 provided by telekom Malaysia? If so, is anyone happy to share theit config or provide some guidance? Foong
Re: [Asterisk-Users] Dial + disconnect
Luckily, I have a E100P. could you tell me how to get the dial status within the extension logic or in AGI script? - Original Message - From: Eric Wieling [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, September 10, 2003 1:39 PM Subject: Re: [Asterisk-Users] Dial + disconnect Yes, on ISDN PRI. On analog you can try the busytetect and progress detect but that always disconnects my calls at random times. On Wed, 2003-09-10 at 00:37, Chee Foong wrote: Yes you are right, Sorry my mistake. So, is there a way to detect busy, answer, or no answer call? Foong - Original Message - From: Richard Lyman [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, September 10, 2003 12:50 PM Subject: Re: [Asterisk-Users] Dial + disconnect based on what dial string you have a zap device '0122740900' (looks more like an exten/phone# to me) maybe you meant Zap/1/0122740900 (notice the /1/, you could also use groups /g1/ (if setup in zapata.conf)) Chee Foong wrote: Hello, When I have the following extension: exten = 900,1,dial(Zap/0122740900) can I know whether 'dial' actually gets through or the called party is busy at the moment. I want to perform different action based on whether the 'dail' success or not. Foong ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- BTEL Consulting 850-484-4535 x2111 (Office) 504-595-3916 x2111 (Experimental) 877-552-0838 (Backup Phone) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] LEDs on E100P card
Hello, There are 2 leds at the back of the E100P card. I search the mailing list and digium website. There seems to beno documentation about them. On the card itself, 1 led is labeled D1 and the other is labled D2. Can someone explain or point me to the right resources about these leds. Thanks Foong
Re: [Asterisk-Users] LEDs on E100P card
Hi thanks for your reply, I have plugged in a E1 trunk to my E100P card and there is flashing red light. The E1 has no signalling, it has not been configured yet at the provider side. In this scenario, would I get a green light or flashing red? By the way what is red alarm? is it a E1 terminology? - Original Message - From: Mark Spencer [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, September 10, 2003 10:06 PM Subject: Re: [Asterisk-Users] LEDs on E100P card One is a bicolor LED which indicates: off - span not configured / driver not loaded green - OK red (flashing) - RED Alarm yellow - Yellow Alarm The second is an orange LED which indicates a loopback (local or remote) is up for testing purposes. Mark On Wed, 10 Sep 2003, Chee Foong wrote: Hello, There are 2 leds at the back of the E100P card. I search the mailing list and digium website. There seems to be no documentation about them. On the card itself, 1 led is labeled D1 and the other is labled D2. Can someone explain or point me to the right resources about these leds. Thanks Foong ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dial + disconnect
Ahhh...I see those config options. But how would I know which case(busy, no answer) happen in the extension.conf. What I am planing to do is somthing like: if(Busy detected) put a retry flag in MySQL db, for retying later else if(no answer detected) then stop trying the destination number, and log the call info in MySQL. thanks Foong - Original Message - From: Richard Lyman [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, September 11, 2003 12:18 AM Subject: Re: [Asterisk-Users] Dial + disconnect well depending on the hardware you are using and where you are using it at, in some cases there is. look in zapata.conf search 'callprogress'. Chee Foong wrote: Yes you are right, Sorry my mistake. So, is there a way to detect busy, answer, or no answer call? Foong - Original Message - From: Richard Lyman [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, September 10, 2003 12:50 PM Subject: Re: [Asterisk-Users] Dial + disconnect based on what dial string you have a zap device '0122740900' (looks more like an exten/phone# to me) maybe you meant Zap/1/0122740900 (notice the /1/, you could also use groups /g1/ (if setup in zapata.conf)) Chee Foong wrote: Hello, When I have the following extension: exten = 900,1,dial(Zap/0122740900) can I know whether 'dial' actually gets through or the called party is busy at the moment. I want to perform different action based on whether the 'dail' success or not. Foong ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dial + disconnect
Hi, Sorry, forgot about the info. I have a E100P digium card and is planing to implement an automatic dialing system using the spooling mechanism. In this service, asterisk will call Party A then Party B and finally let them talk. As you know, Party A or Party B might not be reached. Therefore I need to know, if a particular party is busy, the party does not answer the call or the call get through well. This is to determine whether to retry the call if something fail or do some other operations. I looked at the handbooks, tutorial on digium.com and also the mailing list but nothing really related. I checked the dial application through the CLI, It mention something about the priority n+101 if dial return busy, but if I am not mistaken that is only happend if no channel is available (in my case 30 channels has been occupied). It is not the destination party that is busy. Am I right? Thanks for you reply - Original Message - From: Richard Lyman [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, September 11, 2003 12:03 PM Subject: Re: [Asterisk-Users] Dial + disconnect well, you never said what hardware you had, and where you were using it... so even if everything is working... look at handbook for extension handling. you'll also have to work out the mysql updates, given that the only current sql module is app_sql-postresql (or there abouts). (there used to be some mysql (mostly for cdr) floating about the mailing list (search the archives)). www.digium.com/handbook-draft.pdf has alot of helpful info, you should take a peek. Chee Foong wrote: Ahhh...I see those config options. But how would I know which case(busy, no answer) happen in the extension.conf. What I am planing to do is somthing like: if(Busy detected) put a retry flag in MySQL db, for retying later else if(no answer detected) then stop trying the destination number, and log the call info in MySQL. thanks Foong - Original Message - From: Richard Lyman [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, September 11, 2003 12:18 AM Subject: Re: [Asterisk-Users] Dial + disconnect well depending on the hardware you are using and where you are using it at, in some cases there is. look in zapata.conf search 'callprogress'. Chee Foong wrote: Yes you are right, Sorry my mistake. So, is there a way to detect busy, answer, or no answer call? Foong - Original Message - From: Richard Lyman [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, September 10, 2003 12:50 PM Subject: Re: [Asterisk-Users] Dial + disconnect based on what dial string you have a zap device '0122740900' (looks more like an exten/phone# to me) maybe you meant Zap/1/0122740900 (notice the /1/, you could also use groups /g1/ (if setup in zapata.conf)) Chee Foong wrote: Hello, When I have the following extension: exten = 900,1,dial(Zap/0122740900) can I know whether 'dial' actually gets through or the called party is busy at the moment. I want to perform different action based on whether the 'dail' success or not. Foong ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dial + disconnect
Hello, When I have the following extension: exten = 900,1,dial(Zap/0122740900) can I know whether 'dial' actually gets through or the called party is busy at the moment. I want to perform different action based on whether the 'dail' success or not. Foong
Re: [Asterisk-Users] Dial + disconnect
Yes you are right, Sorry my mistake. So, is there a way to detect busy, answer, or no answer call? Foong - Original Message - From: Richard Lyman [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, September 10, 2003 12:50 PM Subject: Re: [Asterisk-Users] Dial + disconnect based on what dial string you have a zap device '0122740900' (looks more like an exten/phone# to me) maybe you meant Zap/1/0122740900 (notice the /1/, you could also use groups /g1/ (if setup in zapata.conf)) Chee Foong wrote: Hello, When I have the following extension: exten = 900,1,dial(Zap/0122740900) can I know whether 'dial' actually gets through or the called party is busy at the moment. I want to perform different action based on whether the 'dail' success or not. Foong ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] G729 codec
Hello People, I am a little unsure about the licensing of the digium g729 codec. Can anyone advice me on the following scenario? I have a digium card (T100) 30 channels and I want to use it to provide conference service. I wonder how many licence should I buy if: 1. I only use the PRI to connect callers to the asterisk. 2. Besides PRI, callers can also call in to asterisk from ip endpoints. I already have 5 g729 licence from digium. In case I need to add some more to make a total of 10, can I just buy another 5 licences and register them to asterisk? I am confused since 1 licence key is valid only to how many licence you purchase in one order. Separate order will have a different licence key. I wonder if I can run registration twice with different licence key on a single asterisk server. Please help ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Conference Leader
Hello, Is meetme able to do the following scenario: Say a group of caller calling in to the same conference room where one of them is a conference leader.The leadercan press a key which gives that person options like dial out to a particular person and tranfer him/her to the conference. The leader's password is different from the other. I think the tricky part in this scenario is how todetectthe key press by leader. any idea.
Re: [Asterisk-Users] conference authorization
Perhaps you should check out the AGI module. Write a perl script to compare
DTMF(pin) with any data storage(text file, Database). See this doc
http://home.cogeco.ca/~camstuff/.
The other solution is of course modify the source code to check for pin.
You can also use the Autheticate module.
I found that the first option is easier to implement and provide more
control.
Foong
- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, August 27, 2003 3:19 PM
Subject: Re: [Asterisk-Users] conference authorization
Well, going by apps/app_meetme.c, for some of it, we see
inpin = strchr(inflags, '|');
if (inpin) {
*inpin = '\0';
inpin++;
/* XXX Need to do something with pin XXX
*/
ast_log(LOG_WARNING, MEETME WITH
PIN=(%s)\n, inpin);
}
and a bit further down, we see:
/* XXX Should prompt user for pin if pin is
required XXX */
/* Run the conference */
res = conf_run(chan, cnf, confflags);
Therefore, we conclude asterisk does not do conference authentication,
yet.
On Wed, Aug 27, 2003 at 09:10:46AM +0200, radan wrote:
Hello all !
How can I make conference authorization
based on pin number ?
I have:
exten = 1,1,Meetme,1234|ps|
where is a pin number
and this doesn't works
Where do I have to add information about pin number ??
Greetings
Andrzej Radke
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Re: [Asterisk-Users] Limit Number of user in Conference
Cheers mate! After getting the latest CVS, I manage to get it work in my AGI script. Excellent patch, thanks a lot. Foong ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Conference + time limit
Hello All, Sorry about the html, I always send mail using plain text, not sure why it contains html. Yes I should patch my outlook :). My purpose is to limit the conference call for 1 hour. After that all callers involve in the conference will be disconnected. AbsoluteTimeOut hangup a particlular channel xx seconds after the caller get connected. If I want to make the conference stop at 5pm I can calculate the seconds from the time connected to 5pm then set the AbsoluteTimeOut for each caller. I will expect few seconds off between the disconnect time of callers depend on how fast my machine do the calculation. Before I start any major scripting(AGI), I want to make sure if there is a way I can set the time (not seconds) where a caller will be hangup so I dont have to worry about calculating seconds? Thanks again for the replies. Foong On Thu, 2003-08-21 at 04:12, Chee Foong wrote: Hello Conference again. Meetme can now limit number of users in a room. Can it also limit how long a conference session? Someone ask the same question (from achive) but doesn't have a solid answer. Please do not use HTML in your email. You should look at the junk that is created by it. BTW, what size is size=3D2 ? It seems to be in all HTML email from Microsoft products. Chee, You need to look into the TimeOut and AbsoluteTimeOut functions to get a user out after a timeframe. I may be wrong, but I think this will terminate the call though, and that may not be what is wanted. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Limit Number of user in Conference
Hello, Is it possible to limit the number of user in a particular conference room? Foong
Re: [Asterisk-Users] Where to find correct ver of OpenH323 PWLIB for Chan_h323
should be CVS Foong - Original Message - From: Steven Thomas [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, August 20, 2003 2:42 PM Subject: [Asterisk-Users] Where to find correct ver of OpenH323 PWLIB for Chan_h323 Hi, Can someone tell me where to find the stated correct versions of Openh323 and PWLIB for Chan_h323? The README states the versions required are: Open H.323 v1.11.7 PWLib v1.4.11 I am still trying to resolve my continuing one way audio problem by using these versions.. Thanks. Regards, Steven Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Where to find correct ver of OpenH323 PWLIB for Chan_h323
you can do cvs update -r v1_11_7 to get version 1.11.7 for openh323 Foong - Original Message - From: Steven Thomas [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, August 20, 2003 2:51 PM Subject: Re: [Asterisk-Users] Where to find correct ver of OpenH323 PWLIB for Chan_h323 I thought that the CVS would only contain the lastest code - being: OpenH323: v1.12.2 PWLib: v1.5.2 Is this not the case? Thanks Regards, Steven Thomas Chee Foong [EMAIL PROTECTED]To: [EMAIL PROTECTED] Sent by: cc: [EMAIL PROTECTED]Subject: Re: [Asterisk-Users] Where to find correct ver of OpenH323 PWLIB for .digium.comChan_h323 20-08-03 04:53 PM Please respond to asterisk-users should be CVS Foong - Original Message - From: Steven Thomas [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, August 20, 2003 2:42 PM Subject: [Asterisk-Users] Where to find correct ver of OpenH323 PWLIB for Chan_h323 Hi, Can someone tell me where to find the stated correct versions of Openh323 and PWLIB for Chan_h323? The README states the versions required are: Open H.323 v1.11.7 PWLib v1.4.11 I am still trying to resolve my continuing one way audio problem by using these versions.. Thanks. Regards, Steven Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Where to find correct ver of OpenH323 PWLIB for Chan_h323
export CVSROOT=:pserver:[EMAIL PROTECTED]:/cvsroot/openh323 cvs login CVS password: press enter cd /root cvs checkout openh323 cd openh323 cvs update -r v1_11_7 I usually get the latest version then down grade to older version, If you know how to get the older version directly, let me know. Foong - Original Message - From: Steven Thomas [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, August 20, 2003 3:05 PM Subject: Re: [Asterisk-Users] Where to find correct ver of OpenH323 PWLIB for Chan_h323 Thanks - because of my ignorance using the CVS archive - could you please give me the full command - thanks. Regards, Steven Thomas Chee Foong [EMAIL PROTECTED]To: [EMAIL PROTECTED] Sent by: cc: [EMAIL PROTECTED]Subject: Re: [Asterisk-Users] Where to find correct ver of OpenH323 PWLIB for .digium.comChan_h323 20-08-03 05:03 PM Please respond to asterisk-users you can do cvs update -r v1_11_7 to get version 1.11.7 for openh323 Foong - Original Message - From: Steven Thomas [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, August 20, 2003 2:51 PM Subject: Re: [Asterisk-Users] Where to find correct ver of OpenH323 PWLIB for Chan_h323 I thought that the CVS would only contain the lastest code - being: OpenH323: v1.12.2 PWLib: v1.5.2 Is this not the case? Thanks Regards, Steven Thomas Chee Foong [EMAIL PROTECTED]To: [EMAIL PROTECTED] Sent by: cc: [EMAIL PROTECTED]Subject: Re: [Asterisk-Users] Where to find correct ver of OpenH323 PWLIB for .digium.comChan_h323 20-08-03 04:53 PM Please respond to asterisk-users should be CVS Foong - Original Message - From: Steven Thomas [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, August 20, 2003 2:42 PM Subject: [Asterisk-Users] Where to find correct ver of OpenH323 PWLIB for Chan_h323 Hi, Can someone tell me where to find the stated correct versions of Openh323 and PWLIB for Chan_h323? The README states the versions required are: Open H.323 v1.11.7 PWLib v1.4.11 I am still trying to resolve my continuing one way audio problem by using these versions.. Thanks. Regards, Steven Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Conference + E100P + H323
Hello Martin, Yes, I have span configure in zaptel.conf: span=1,0,0,esf,b8zs I dont have a PRI plugged in to the card. Would it be an issue? Reason is I am current only testing the call originating from H323 endpoints. Firewall shouldn't be a issue since the call works fine with ztdummy loaded. I debug the chan_h323 and it uses the right codec G729 from digium. Only cant hear the Meetme prompt. Foong - Original Message - From: Martin Pycko [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, August 13, 2003 12:28 PM Subject: Re: [Asterisk-Users] Conference + E100P + H323 On Wed, 13 Aug 2003, Chee Foong wrote: Hello, I have a E100P card from digium and I try to implement a conference bridge in asterisk. I wonder since I got the E100P card do I still need to load ztdummy for caller from h323 endpoints to work with Meetme? It's not necessary. I load the E100P driver but i did not load the ztdummy driver. My h323 Do you have the span configured in /etc/zaptel.conf ? caller does not hear any voice play by Meetme. Looks like ztdummy is required as long as h323 is concern and not depend on whether there is a zaptel device. Check the firewall and codecs. regards Martin Foong ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_OH323
Hello, I downloaded the chan_oh323. I experience few problems: When I dial from console I get all the object creation and deletion message, and when a call get connected it gives me the following output. Wrong Pitch 1st subfr. ! ! Wrong Pitch 1st subfr. ! !Wrong Pitch 1st subfr. ! ! Wrong Pitch 1st subfr. ! this message keep outputed to the console untill I end the call. When I dial in to asterisk, I get WARNING[524312]: File chan_oh323.c, Line 948 (oh323_read): H323:7160: Invalid size for G.729 (2 bytes). then i got disconnected. I am using Digium g.729 codec. In oh323.conf i set codec=G729 Any idea? Foong
Re: [Asterisk-Users] chan_oh323 + dtmf
Hello Michael My extensio.conf are as follows: I have try it with H323 phone, it works ok all digits detected. Only when call is coming from pstn cause the problem Also, the console output when digit is press is: Invalid extension '1 ' in context...' There is a space after the 1, I believe its a # key. It could possible be the problem? Any idea to fix it? [conference] ; ; conference: Conference Call ; exten = s,1,Ringing exten = s,2,DigitTimeout,10; Set Digit Timeout to 5 seconds exten = s,3,ResponseTimeout,10; Set Response Timeout to 10 seconds exten = s,4,Answer exten = s,5,Background(conf-getconfno) exten = t,6,Goto(s,5) exten = 1234,1,Meetme,1234|ps|9888 - Original Message - From: Michael Manousos [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, August 06, 2003 4:40 PM Subject: Re: [Asterisk-Users] chan_oh323 + dtmf Chee Foong wrote: Hello all, I have a cisco AS5300 which is register with a gatekeeper and a Asterisk server also register with the gatekeeper. PSTN AS5300 Gatekeeper Asterisk I set up a conference room on the Asterisk sever (Room No 1234). I try to call from PSTN to AS5300, The AS5300 will call the Asterisk server through the gatekeeper. I manage to get to the start of the conference where the 'Please key in conference number' is played. But when I press the room no (1234), Asterisk only get the first digit which is 1 and play 'Invalid conference number' right a way. What are the contents of your extensions.conf at the point that you are trying to enter the conference number? After the H.323 channel has been answered, the DTMFs are handled by the application connected to the channel (conferencing here). I am using chan_oh323, I am close to get this thing to work (having sorted the correct codec), just the dtmf issue now. I am using digium's g729. By the way how many variation of g729 are there. I know g729a, g729b, but there seem to be others. Please help. Thanks Foong Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] H323 + DTMF detection
Hello all, Does anybody has a problem where asterisk only grap the first digit of a string of multiple DTMF digits? My setup is: PSTN --- AS5300 ---Gatekeeper --- Asterisk When call coming from PSTN all the way toAsterisk to access a conference room,I press the conference room number which is 1234, but asterisk only grap the first digit of the digits pressed and I end up getting Invalid Conference Number. Any idea? Foong
[Asterisk-Users] reload
Hello All, I wonder is there a way where I reload asterisk on CLI without disconnect any call that is currently taken place. Foong
Re: [Asterisk-Users] Conference + E100P + H323
Hi, I manage to solve the problem. I just change the span configuration in zaptel.conf to E1 configuration. Unload zaptel driver and load it again. It seems to work fine. I would like to know if RFC2833 is equavalent to out of band DTMF? Foong - Original Message - From: Jeremy McNamara [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, August 13, 2003 4:00 PM Subject: Re: [Asterisk-Users] Conference + E100P + H323 Chee Foong wrote: Firewall shouldn't be a issue since the call works fine with ztdummy loaded. I debug the chan_h323 and it uses the right codec G729 from digium. H.323 does NOT deal with NAT or Firewalls without a smart edge device. chan_h323 does not use ztdummy whatsoever, so that has no bearing. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Conference + E100P + H323
Hello, I have a E100P card from digium and I try to implement a conference bridge in asterisk. I wonder since I got the E100P card do I still need to load ztdummyfor caller from h323 endpoints to work with Meetme? I load the E100P driver but i did not load theztdummy driver. My h323 caller does not hear any voice play byMeetme. Looks like ztdummy is required as long as h323 is concern and not depend on whether there is a zaptel device. Foong
Re: [Asterisk-Users] SendDtmf + chan_h323
Hello,
I have tried you suggestion. I found out that dtmf does not even send out if
even if i press manually.
I have the following structure
IVR --- Cisco Gateway -- Gatekeeper (GNUGK)-Asterisk
^
|
H323
phone
When I dial into asterisk from a H323 endpoint, I can call an extension
which GK will route me to Cisco gateway then to IVR. When IVR prompt me for
input, no matter what I press, the IVR seems to get nothing. I am using
chan_h323.
The Cisco gateway serve as a bridge between PSTN and IP network. The IVR is
on PSTN network.
Someone could have experience the same problem?
cheers
Foong
- Original Message -
From: Michael Manousos [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, August 05, 2003 5:37 PM
Subject: Re: [Asterisk-Users] SendDtmf
WipeOut . wrote:
Hello all,
I am trying to use asterisk to call a local access gateway by dialing a
fix number, after getting connected, the is a IVR prompt for pin number and
finally the real destination number. I manage to use asterisk to dial to the
gateway but have no idea how to send the pin number and destination number.
This is due to asterisk only process the next ext only if dial app has
terminated. My extension.conf are as follows:
[test]
exten = _0,1,Dial(H323/${EXTEN:0})
exten = _0,2,SendDTMF(PIN_NUMBER_HERE)
I saw someone post the similiar question but with no reply. Does anybody
has any idea?
cheers
Foong
I have a similar problem when trying to use a long distance access
number but was never able to find a solution.. The reason that the method
you are trying does not work is becasue the call is connected on priority 1
and then does not move on to the next priority so the SendDTMF is never
processed..
Why don't you just Dial() and then press the DTMFs, when the
channel has been answered?
Michael.
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[Asterisk-Users] chan_oh323 + dtmf
Hello all, I have a cisco AS5300 which is register with a gatekeeper and a Asterisk server also register with the gatekeeper. PSTN AS5300 Gatekeeper Asterisk I set up a conference room on the Asterisk sever (Room No 1234). I try to call from PSTN to AS5300, The AS5300 will call the Asterisk server through the gatekeeper. I manage to get tothe start of the conference where the 'Please key inconference number' is played. But when I press the room no (1234), Asterisk only get the first digit which is 1 and play 'Invalid conference number' right a way. I am using chan_oh323, I am close to get this thing to work (having sorted the correct codec), just the dtmf issue now. I am using digium's g729. By the way how many variation of g729 are there. I know g729a, g729b, but there seem to be others. Please help. Thanks Foong
[Asterisk-Users] WipeOut - gateway access with pin solution
Helo WipeOut,
I have found a solution for sending dtmf after
dial.
I use spooling. Take a look at the sample.call file
inside asterisk dir. You need to edit this file and dump it in
/var/spool/asterisk/outgoing. Asterisk will precess this file
automaticlly
Icreate the sample.call do something like
this:
Channel: OH323/4324324324 #dial the access
way
MaxRetries: 3RetryTime: 60WaitTime:
30
Context: test-context #after connected to access
gateway, proceed to context 'test-contet' in extension.confExtension:
1Priority: 1
# set var to be used in extension.conf
SetVar: PINNO=1234
SerVar: NUMTOCALL=123123213123 # actual dest
number
My extension(test-extension) is: (in
extension.conf)
exten =
1,1,SendDTMF(${PINNO})
exten = 1,2, Wait, 3exten =
1,3,SendDTMF(${NUMTOCALL})
However, this might not suitable for you, if your
user dial in manually. My situation works fine cause everyting is automated
where calling number and called number is inserted into db in
advanced.
also, chan_h323 has proplem sending DTMF,
chan_oh323 works but sound quality is bad.
Foong
Re: [Asterisk-Users] chan_oh323 + dtmf
Hello, Yes that's the only extension. I tried add an extension 1 to the config. end up the extension getting execute. Foong - Original Message - From: Michael Manousos [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, August 06, 2003 5:36 PM Subject: Re: [Asterisk-Users] chan_oh323 + dtmf Chee Foong wrote: Hello Michael My extensio.conf are as follows: I have try it with H323 phone, it works ok all digits detected. Only when call is coming from pstn cause the problem Also, the console output when digit is press is: Invalid extension '1 ' in context...' There is a space after the 1, I believe its a # key. It could possible be the problem? Any idea to fix it? [conference] ; ; conference: Conference Call ; exten = s,1,Ringing exten = s,2,DigitTimeout,10; Set Digit Timeout to 5 seconds exten = s,3,ResponseTimeout,10; Set Response Timeout to 10 seconds exten = s,4,Answer exten = s,5,Background(conf-getconfno) exten = t,6,Goto(s,5) exten = 1234,1,Meetme,1234|ps|9888 Do you have any other extension in this context that could match a single digit? Michael. - Original Message - From: Michael Manousos [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, August 06, 2003 4:40 PM Subject: Re: [Asterisk-Users] chan_oh323 + dtmf Chee Foong wrote: Hello all, I have a cisco AS5300 which is register with a gatekeeper and a Asterisk server also register with the gatekeeper. PSTN AS5300 Gatekeeper Asterisk I set up a conference room on the Asterisk sever (Room No 1234). I try to call from PSTN to AS5300, The AS5300 will call the Asterisk server through the gatekeeper. I manage to get to the start of the conference where the 'Please key in conference number' is played. But when I press the room no (1234), Asterisk only get the first digit which is 1 and play 'Invalid conference number' right a way. What are the contents of your extensions.conf at the point that you are trying to enter the conference number? After the H.323 channel has been answered, the DTMFs are handled by the application connected to the channel (conferencing here). I am using chan_oh323, I am close to get this thing to work (having sorted the correct codec), just the dtmf issue now. I am using digium's g729. By the way how many variation of g729 are there. I know g729a, g729b, but there seem to be others. Please help. Thanks Foong Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SendDtmf
Hello all,
I am trying to use asterisk to call a local access
gateway by dialing a fix number, after getting connected, the is a IVR prompt
for pin number and finally the real destination number. I manage to use asterisk
to dial to the gateway but have no idea how to send the pin number and
destination number. This is due to asterisk only process the next ext only if
dial app has terminated. My extension.conf are as follows:
[test]
exten =
_0,1,Dial(H323/${EXTEN:0})
exten =_0,2,SendDTMF(PIN_NUMBER_HERE)
I saw someone post the similiar question but with
no reply. Does anybody has any idea?
cheers
Foong
[Asterisk-Users] Mysql CDR
hello all, I am using the msql cdr module to store cdr in db, I realised thatit does't capture the start and end time af a particular call record. Therefore I dive into the source code toadd the start and end timeinto the query (add something like cdr-start, cdr-end), but end up getting segfault. the original version of cdr_mysql.so works fine but Ineed the start time and end time of calling as well. I wonder what would Iget with cdr-start?the start time field in my dbis of type date or should i use varchar? thanks Foong
Re: [Asterisk-Users] Call Transfer
Excellent idea mate, Now I am able to do what I wanted with Great help from Jeremy McNamara. Thanks alot Foong - Original Message - From: Andy Powell [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 30, 2003 6:56 PM Subject: Re: [Asterisk-Users] Call Transfer Foong Take a look at the sample.call file, modifying the settings in there and copying the file to /var/spool/asterisk/outgoing will cause asterisk to dial the call.. an example config is below Channel: SIP/[EMAIL PROTECTED] MaxRetries: 2 RetryTime: 60 WaitTime: 30 Context: mysipcontext2 Extension: 2000 Priority: 1 This will make asterisk dial exten 1000 in the context mysipcontext when it's answered it will then call exten 2000 in mysipcontext2.. All you need is a script to lookup in the database and generate the script file for you and it's done. HTH Andy *** REPLY SEPARATOR *** On 30/07/2003 at 16:30 Chee Foong wrote: Hello Dan, Thanks for you reply. Base on you recomendation using the 'T' argument. I manage to do call transfer an it works really well. My problem comes when my boss comes out with a superb idea where the transfering process is automated without involving a human :( Say asterisk get 2 numbers (from database, text file, etc), one belongs party A and the other belongs to party B. Asterisk will calls both parties and do the tranfer automatically. In another words, asterisk is resposible to 'press' the '#' to do the transfer. I don't this can be achieve in the extension.conf not matter how you structure you dial plan. Perhaps, the only way is to write a apps and plug it into asterisk like all the asterisk modules such as Meetme. Any ideas? Foong - Original Message - From: Dan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 30, 2003 3:42 PM Subject: Re: [Asterisk-Users] Call Transfer Hi, It works if you put the 'T' switch in the dial line. You can then transfer the call from the caller. I have tested it in the folllowing configuration and it works: Call from a Cisco 7960 to an ATA 186. Select 'Transfer on 7960 Call another extension (X-Lite) Select again transfer on 7960. The call remain between ATA and X-Lite. This is what you need? BR, Dan - Original Message - From: Chee Foong [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 30, 2003 7:08 AM Subject: [Asterisk-Users] Call Transfer Hello all, I am in a situation where I need to use asterisk to call someone say Party A. After the call to Party A got through, asterisk will put Party A on hold, then asterisk will call Party B. If call to Party B got through, asterisk will transfer Party A to Party B. I wonder if this features is implemented into asterisk. I have found a post in asterisk mailing list: http://lists.digium.com/pipermail/asterisk-users/2003-June/013253.html but that doesn't help much. If this features is not implemented, can anyone give me some point on how to implement this in asterisk? Do I need to write an app like the Dial apps for asterisk to load at start up? thanks Foong ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Transfer
Hello Dan, Thanks for you reply. Base on you recomendation using the 'T' argument. I manage to do call transfer an it works really well. My problem comes when my boss comes out with a superb idea where the transfering process is automated without involving a human :( Say asterisk get 2 numbers (from database, text file, etc), one belongs party A and the other belongs to party B. Asterisk will calls both parties and do the tranfer automatically. In another words, asterisk is resposible to 'press' the '#' to do the transfer. I don't this can be achieve in the extension.conf not matter how you structure you dial plan. Perhaps, the only way is to write a apps and plug it into asterisk like all the asterisk modules such as Meetme. Any ideas? Foong - Original Message - From: Dan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 30, 2003 3:42 PM Subject: Re: [Asterisk-Users] Call Transfer Hi, It works if you put the 'T' switch in the dial line. You can then transfer the call from the caller. I have tested it in the folllowing configuration and it works: Call from a Cisco 7960 to an ATA 186. Select 'Transfer on 7960 Call another extension (X-Lite) Select again transfer on 7960. The call remain between ATA and X-Lite. This is what you need? BR, Dan - Original Message - From: Chee Foong [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 30, 2003 7:08 AM Subject: [Asterisk-Users] Call Transfer Hello all, I am in a situation where I need to use asterisk to call someone say Party A. After the call to Party A got through, asterisk will put Party A on hold, then asterisk will call Party B. If call to Party B got through, asterisk will transfer Party A to Party B. I wonder if this features is implemented into asterisk. I have found a post in asterisk mailing list: http://lists.digium.com/pipermail/asterisk-users/2003-June/013253.html but that doesn't help much. If this features is not implemented, can anyone give me some point on how to implement this in asterisk? Do I need to write an app like the Dial apps for asterisk to load at start up? thanks Foong ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Transfer
Dan, Asterisk is suppose to trigger the transfer when it successfully call both extensions Do you mean I have to create conference room for every call? that would not be practicle. Or do you have a example dialplan to to illustrate you suggestion? Actually, we have a client that is too lazy to do all the dialing, he want a system that will call him and also the person he wanted to call, just like some receptionists do theese days. The different is that asterisk is taking over the receptionist's job thanks Foong Hi Foong, But then... who and when will trigger the transfer between the two remote extensions? I think to something like that. One of the extension calls a special number, entering a password (or check after the Caller ID). Asterisk close the call, wait for answer Call the second extension, wait for answer Then, in some way (eventually through a conference mode using local CONSOLE as master) bridge the two calls. What do you think about that? Dan - Original Message - From: Chee Foong [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 30, 2003 11:30 AM Subject: Re: [Asterisk-Users] Call Transfer Hello Dan, Thanks for you reply. Base on you recomendation using the 'T' argument. I manage to do call transfer an it works really well. My problem comes when my boss comes out with a superb idea where the transfering process is automated without involving a human :( Say asterisk get 2 numbers (from database, text file, etc), one belongs party A and the other belongs to party B. Asterisk will calls both parties and do the tranfer automatically. In another words, asterisk is resposible to 'press' the '#' to do the transfer. I don't this can be achieve in the extension.conf not matter how you structure you dial plan. Perhaps, the only way is to write a apps and plug it into asterisk like all the asterisk modules such as Meetme. Any ideas? Foong - Original Message - From: Dan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 30, 2003 3:42 PM Subject: Re: [Asterisk-Users] Call Transfer Hi, It works if you put the 'T' switch in the dial line. You can then transfer the call from the caller. I have tested it in the folllowing configuration and it works: Call from a Cisco 7960 to an ATA 186. Select 'Transfer on 7960 Call another extension (X-Lite) Select again transfer on 7960. The call remain between ATA and X-Lite. This is what you need? BR, Dan - Original Message - From: Chee Foong [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 30, 2003 7:08 AM Subject: [Asterisk-Users] Call Transfer Hello all, I am in a situation where I need to use asterisk to call someone say Party A. After the call to Party A got through, asterisk will put Party A on hold, then asterisk will call Party B. If call to Party B got through, asterisk will transfer Party A to Party B. I wonder if this features is implemented into asterisk. I have found a post in asterisk mailing list: http://lists.digium.com/pipermail/asterisk-users/2003-June/013253.html but that doesn't help much. If this features is not implemented, can anyone give me some point on how to implement this in asterisk? Do I need to write an app like the Dial apps for asterisk to load at start up? thanks Foong ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Transfer
Hello But If i do that I have to create lots of conference room if I have lots of caller. Foong - Original Message - From: Sip Rtp [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 30, 2003 5:44 PM Subject: Re: [Asterisk-Users] Call Transfer Yes, I second to that idea. I think thats only available option to put them in a local conference. Rgds Manoj K Gupta - Original Message - From: Dan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 30, 2003 2:04 PM Subject: Re: [Asterisk-Users] Call Transfer Hi Foong, But then... who and when will trigger the transfer between the two remote extensions? I think to something like that. One of the extension calls a special number, entering a password (or check after the Caller ID). Asterisk close the call, wait for answer Call the second extension, wait for answer Then, in some way (eventually through a conference mode using local CONSOLE as master) bridge the two calls. What do you think about that? Dan - Original Message - From: Chee Foong [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 30, 2003 11:30 AM Subject: Re: [Asterisk-Users] Call Transfer Hello Dan, Thanks for you reply. Base on you recomendation using the 'T' argument. I manage to do call transfer an it works really well. My problem comes when my boss comes out with a superb idea where the transfering process is automated without involving a human :( Say asterisk get 2 numbers (from database, text file, etc), one belongs party A and the other belongs to party B. Asterisk will calls both parties and do the tranfer automatically. In another words, asterisk is resposible to 'press' the '#' to do the transfer. I don't this can be achieve in the extension.conf not matter how you structure you dial plan. Perhaps, the only way is to write a apps and plug it into asterisk like all the asterisk modules such as Meetme. Any ideas? Foong - Original Message - From: Dan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 30, 2003 3:42 PM Subject: Re: [Asterisk-Users] Call Transfer Hi, It works if you put the 'T' switch in the dial line. You can then transfer the call from the caller. I have tested it in the folllowing configuration and it works: Call from a Cisco 7960 to an ATA 186. Select 'Transfer on 7960 Call another extension (X-Lite) Select again transfer on 7960. The call remain between ATA and X-Lite. This is what you need? BR, Dan - Original Message - From: Chee Foong [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 30, 2003 7:08 AM Subject: [Asterisk-Users] Call Transfer Hello all, I am in a situation where I need to use asterisk to call someone say Party A. After the call to Party A got through, asterisk will put Party A on hold, then asterisk will call Party B. If call to Party B got through, asterisk will transfer Party A to Party B. I wonder if this features is implemented into asterisk. I have found a post in asterisk mailing list: http://lists.digium.com/pipermail/asterisk-users/2003-June/013253.html but that doesn't help much. If this features is not implemented, can anyone give me some point on how to implement this in asterisk? Do I need to write an app like the Dial apps for asterisk to load at start up? thanks Foong ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software http://sitebuilder.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Transfer
Hi Sip,
I achieve that by adding the following extension into extension.conf:
exten = _9,1,Dial(H323/{EXTEN:1})
foong
- Original Message -
From: Sip Rtp [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, July 30, 2003 5:45 PM
Subject: Re: [Asterisk-Users] Call Transfer
Hi
I would like to further ask if it is possible to
transfer a call from
openphone to pstn. i.e. i use openphone and asterisk
-oh323 channel driver
to make a call to a PSTN number through zap channel
connected on that
end.Then i wanna transfer that PSTN number to some
other openphone
extension/alias
May i have a look at your extension to conf, as i am
not clear with how to
implement this.
Rgds
Manoj k Gupta
- Original Message -
From: Chee Foong [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, July 30, 2003 2:00 PM
Subject: Re: [Asterisk-Users] Call Transfer
Hello Dan,
Thanks for you reply.
Base on you recomendation using the 'T' argument. I
manage to do call
transfer an it works really well.
My problem comes when my boss comes out with a
superb idea where the
transfering process is automated without involving a
human :(
Say asterisk get 2 numbers (from database, text
file, etc), one belongs
party A and the other belongs to party B. Asterisk
will calls both parties
and do the tranfer automatically. In another words,
asterisk is resposible
to 'press' the '#' to do the transfer. I don't this
can be achieve in the
extension.conf not matter how you structure you dial
plan.
Perhaps, the only way is to write a apps and plug it
into asterisk like
all
the asterisk modules such as Meetme.
Any ideas?
Foong
- Original Message -
From: Dan [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, July 30, 2003 3:42 PM
Subject: Re: [Asterisk-Users] Call Transfer
Hi,
It works if you put the 'T' switch in the dial
line.
You can then transfer the call from the caller.
I have tested it in the folllowing configuration
and it works:
Call from a Cisco 7960 to an ATA 186.
Select 'Transfer on 7960
Call another extension (X-Lite)
Select again transfer on 7960.
The call remain between ATA and X-Lite.
This is what you need?
BR,
Dan
- Original Message -
From: Chee Foong [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, July 30, 2003 7:08 AM
Subject: [Asterisk-Users] Call Transfer
Hello all,
I am in a situation where I need to use asterisk
to call someone say
Party
A. After the call to Party A got through, asterisk
will put Party A on
hold,
then asterisk will call Party B. If call to Party
B got through,
asterisk
will transfer Party A to Party B.
I wonder if this features is implemented into
asterisk. I have found a
post
in asterisk mailing list:
http://lists.digium.com/pipermail/asterisk-users/2003-June/013253.html
but that doesn't help much.
If this features is not implemented, can anyone
give me some point on
how
to
implement this in asterisk? Do I need to write an
app like the Dial apps
for
asterisk to load at start up?
thanks
Foong
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Re: [Asterisk-Users] Call Transfer
Thanks Andy Will try that Thanks again. Foong - Original Message - From: Andy Powell [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 30, 2003 6:56 PM Subject: Re: [Asterisk-Users] Call Transfer Foong Take a look at the sample.call file, modifying the settings in there and copying the file to /var/spool/asterisk/outgoing will cause asterisk to dial the call.. an example config is below Channel: SIP/[EMAIL PROTECTED] MaxRetries: 2 RetryTime: 60 WaitTime: 30 Context: mysipcontext2 Extension: 2000 Priority: 1 This will make asterisk dial exten 1000 in the context mysipcontext when it's answered it will then call exten 2000 in mysipcontext2.. All you need is a script to lookup in the database and generate the script file for you and it's done. HTH Andy *** REPLY SEPARATOR *** On 30/07/2003 at 16:30 Chee Foong wrote: Hello Dan, Thanks for you reply. Base on you recomendation using the 'T' argument. I manage to do call transfer an it works really well. My problem comes when my boss comes out with a superb idea where the transfering process is automated without involving a human :( Say asterisk get 2 numbers (from database, text file, etc), one belongs party A and the other belongs to party B. Asterisk will calls both parties and do the tranfer automatically. In another words, asterisk is resposible to 'press' the '#' to do the transfer. I don't this can be achieve in the extension.conf not matter how you structure you dial plan. Perhaps, the only way is to write a apps and plug it into asterisk like all the asterisk modules such as Meetme. Any ideas? Foong - Original Message - From: Dan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 30, 2003 3:42 PM Subject: Re: [Asterisk-Users] Call Transfer Hi, It works if you put the 'T' switch in the dial line. You can then transfer the call from the caller. I have tested it in the folllowing configuration and it works: Call from a Cisco 7960 to an ATA 186. Select 'Transfer on 7960 Call another extension (X-Lite) Select again transfer on 7960. The call remain between ATA and X-Lite. This is what you need? BR, Dan - Original Message - From: Chee Foong [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 30, 2003 7:08 AM Subject: [Asterisk-Users] Call Transfer Hello all, I am in a situation where I need to use asterisk to call someone say Party A. After the call to Party A got through, asterisk will put Party A on hold, then asterisk will call Party B. If call to Party B got through, asterisk will transfer Party A to Party B. I wonder if this features is implemented into asterisk. I have found a post in asterisk mailing list: http://lists.digium.com/pipermail/asterisk-users/2003-June/013253.html but that doesn't help much. If this features is not implemented, can anyone give me some point on how to implement this in asterisk? Do I need to write an app like the Dial apps for asterisk to load at start up? thanks Foong ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Transfer
Dan, The time to call could be stored into database with party A and party B phone number. Asterisk or perhaps a script (mentions by Andy Powel in another reply) just keep checking the database and make calls if time is current time and the call has not been processed yet. In this manner, the caller can even schedule a call for tomorrow mornnig, all he do is just insert a record in database and wait :). Foong - Original Message - From: Dan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 30, 2003 7:15 PM Subject: Re: [Asterisk-Users] Call Transfer Foong, Actually, we have a client that is too lazy to do all the dialing, he want a system that will call him and also the person he wanted to call, just like some receptionists do theese days. The different is that asterisk is taking over the receptionist's job ... then... who decide when the call must be initiated and how? Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sample.call + chan_h323 gives seg fault
I dumped the following test.call file into /var/spool/asterisk/outgoing gives me segmentation fault :( Channel: H323/0143126544 MaxRetries: 2 RetryTime: 60 WaitTime: 30 Context: voip-test Extension: 90324324433 Priority: 1 same thing happend if I execute dial command on console. I figure out that this happen only if I dial through a H323 channel. I am using chan_h323. Any one experience the same thing? Foong - Original Message - From: Andy Powell [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 30, 2003 6:56 PM Subject: Re: [Asterisk-Users] Call Transfer Foong Take a look at the sample.call file, modifying the settings in there and copying the file to /var/spool/asterisk/outgoing will cause asterisk to dial the call.. an example config is below Channel: SIP/[EMAIL PROTECTED] MaxRetries: 2 RetryTime: 60 WaitTime: 30 Context: mysipcontext2 Extension: 2000 Priority: 1 This will make asterisk dial exten 1000 in the context mysipcontext when it's answered it will then call exten 2000 in mysipcontext2.. All you need is a script to lookup in the database and generate the script file for you and it's done. HTH Andy *** REPLY SEPARATOR *** On 30/07/2003 at 16:30 Chee Foong wrote: Hello Dan, Thanks for you reply. Base on you recomendation using the 'T' argument. I manage to do call transfer an it works really well. My problem comes when my boss comes out with a superb idea where the transfering process is automated without involving a human :( Say asterisk get 2 numbers (from database, text file, etc), one belongs party A and the other belongs to party B. Asterisk will calls both parties and do the tranfer automatically. In another words, asterisk is resposible to 'press' the '#' to do the transfer. I don't this can be achieve in the extension.conf not matter how you structure you dial plan. Perhaps, the only way is to write a apps and plug it into asterisk like all the asterisk modules such as Meetme. Any ideas? Foong - Original Message - From: Dan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 30, 2003 3:42 PM Subject: Re: [Asterisk-Users] Call Transfer Hi, It works if you put the 'T' switch in the dial line. You can then transfer the call from the caller. I have tested it in the folllowing configuration and it works: Call from a Cisco 7960 to an ATA 186. Select 'Transfer on 7960 Call another extension (X-Lite) Select again transfer on 7960. The call remain between ATA and X-Lite. This is what you need? BR, Dan - Original Message - From: Chee Foong [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 30, 2003 7:08 AM Subject: [Asterisk-Users] Call Transfer Hello all, I am in a situation where I need to use asterisk to call someone say Party A. After the call to Party A got through, asterisk will put Party A on hold, then asterisk will call Party B. If call to Party B got through, asterisk will transfer Party A to Party B. I wonder if this features is implemented into asterisk. I have found a post in asterisk mailing list: http://lists.digium.com/pipermail/asterisk-users/2003-June/013253.html but that doesn't help much. If this features is not implemented, can anyone give me some point on how to implement this in asterisk? Do I need to write an app like the Dial apps for asterisk to load at start up? thanks Foong ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] File chan_h323.c, Line 875
Hello, Anybody experience this error: ERROR[237594]: File chan_h323.c, Line 875 (create_connection): Unable to allocate private structure, this is very bad. the call still get through, but both party cannot hear each other Pls Help. Foong - Original Message - From: Steven Critchfield [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, July 25, 2003 1:47 PM Subject: Re: [Asterisk-Users] Voicemail() problems - Long pause afterincoming message recording ended. On Thu, 2003-07-24 at 22:38, [EMAIL PROTECTED] wrote: I'm having the following problem: I call into my Asterisk box (RedHat Linux 9.0, 1 Digium X100P card) to access voicemail. After dialing the appropriate extension I get voicemail, am presented with the user's unavailable message, and can leave a message normally. The problem comes when I press # to end the recording, at which point I am told Your message has been saved. Then there is a long lag of about 20 seconds of silence, during which Asterisk does not respond to DTMF at all, before I am finally dropped back into the priority list for the extension, which in this case is a simple Goodbye hangup. Any idea why this long lag after message-recording termination is happening? I'd like Asterisk to hang up immediately after the incoming caller terminates their VM recording. Check your mail settings, also whether your DNS is working fast. This is all probably due to the time to get the vm out the app and into the mail. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] help on chan_h323
Hello, I have a voip endpoint calling the asterisk, when this endpoint press extension 2, asterisk will dial to another voip endpoint. However when asterisk try to dial the second endpoint I go the following error: Can somebody help me? -- Executing Ringing(H323/ip$202.75.145.111:2131/15497, ) in new stack -- Executing Wait(H323/ip$202.75.145.111:2131/15497, 2) in new stack -- Executing BackGround(H323/ip$202.75.145.111:2131/15497, demo-instruct) in new stack -- Playing 'demo-instruct' -- Executing Ringing(H323/ip$202.75.145.111:2131/15497, ) in new stack -- Executing Dial(H323/ip$202.75.145.111:2131/15497, H323/0122736111) in new stack dest: 0122746011 ERROR[204826]: File chan_h323.c, Line 875 (create_connection): Unable to allocate private structure, this is very bad. ERROR[204826]: File chan_h323.c, Line 875 (create_connection): Unable to allocate private structure, this is very bad. ERROR[204826]: File chan_h323.c, Line 875 (create_connection): Unable to allocate private structure, this is very bad. ERROR[204826]: File chan_h323.c, Line 875 (create_connection): Unable to allocate private structure, this is very bad. -- Called 0122736111 WARNING[196633]: File chan_h323.c, Line 528 (oh323_write): Asked to transmit frame type 64, while native formats is 256 (read/write = 256/256) == Spawn extension (Inovas-PBX, 2, 2) exited non-zero on 'H323/ip$202.75.145.111:2131/15497' ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] help on chan_h323
Hello, After further testing. I have to manually issue the reload command after every call to avoid the 'Unable to allocate private structure' error. Pretty bad :( Foong - Original Message - From: Chee Foong [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, July 25, 2003 10:37 AM Subject: [Asterisk-Users] help on chan_h323 Hello, I have a voip endpoint calling the asterisk, when this endpoint press extension 2, asterisk will dial to another voip endpoint. However when asterisk try to dial the second endpoint I go the following error: Can somebody help me? -- Executing Ringing(H323/ip$202.75.145.111:2131/15497, ) in new stack -- Executing Wait(H323/ip$202.75.145.111:2131/15497, 2) in new stack -- Executing BackGround(H323/ip$202.75.145.111:2131/15497, demo-instruct) in new stack -- Playing 'demo-instruct' -- Executing Ringing(H323/ip$202.75.145.111:2131/15497, ) in new stack -- Executing Dial(H323/ip$202.75.145.111:2131/15497, H323/0122736111) in new stack dest: 0122746011 ERROR[204826]: File chan_h323.c, Line 875 (create_connection): Unable to allocate private structure, this is very bad. ERROR[204826]: File chan_h323.c, Line 875 (create_connection): Unable to allocate private structure, this is very bad. ERROR[204826]: File chan_h323.c, Line 875 (create_connection): Unable to allocate private structure, this is very bad. ERROR[204826]: File chan_h323.c, Line 875 (create_connection): Unable to allocate private structure, this is very bad. -- Called 0122736111 WARNING[196633]: File chan_h323.c, Line 528 (oh323_write): Asked to transmit frame type 64, while native formats is 256 (read/write = 256/256) == Spawn extension (Inovas-PBX, 2, 2) exited non-zero on 'H323/ip$202.75.145.111:2131/15497' ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] g729 + oh323
Thanks for the info mate. Looking forward to the bug fix release. :) cheers Foong - Original Message - From: Michael Manousos [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, July 22, 2003 7:02 PM Subject: Re: [Asterisk-Users] g729 + oh323 Chee Foong wrote: Hello, Is Oh323 supports g729 codec from digium? I saw an g729 option in the oh323.conf but I have also read some post in the mailing list saying that oh323 doesn't support g729 codec from digium. asterisk-oh323 had some problems with G.729 formats. I have fixed them and soon I 'll make a new bug-fix release. But I have't tested it with digium's G.729 codec, just with some Cisco boxes. Foong Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] g729 + oh323
Hello, From my personal experience, Latest version of OpenH323 wont work. Suggest you use OpenH323 version 1.11.7. Version 1.12.0 will lead you to compilation error regarding H323Capability undeclared or something like that. Good Luck! Cheers Foong - Original Message - From: Jeremy McNamara [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 23, 2003 1:21 AM Subject: Re: [Asterisk-Users] g729 + oh323 You need the latest cvs version of Open H.323 to use iLBC. Use at your own risk. I will see if i can find time to pull the H323Capability out of 1.12.0 and put it in chan_h323 directly. Jeremy McNamara Michael Bielicki wrote: hmmm we tried today to run it with ilbc and there was no sound. something seems to be funny in that codec in regards to chan_h323 just my 2c Michael Bielicki On Tuesday 22 July 2003 4:36 am, Jeremy McNamara wrote: You should run chan_h323. It is distributed with Asterisk and works with G.729 and any other codec asterisk supports TODAY. There is no need to run a 3rd party driver. Jeremy McNamara Chee Foong wrote: Thanks for the info mate. Looking forward to the bug fix release. :) cheers Foong - Original Message - From: Michael Manousos [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, July 22, 2003 7:02 PM Subject: Re: [Asterisk-Users] g729 + oh323 Chee Foong wrote: Hello, Is Oh323 supports g729 codec from digium? I saw an g729 option in the oh323.conf but I have also read some post in the mailing list saying that oh323 doesn't support g729 codec from digium. asterisk-oh323 had some problems with G.729 formats. I have fixed them and soon I 'll make a new bug-fix release. But I have't tested it with digium's G.729 codec, just with some Cisco boxes. Foong Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] g729 + oh323
Hello, Is Oh323 supports g729 codec from digium? I saw an g729 option in the oh323.conf but I have also read some post in the mailing list saying that oh323 doesn't support g729 codec from digium. Foong ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Connect 2 party with asterisk
Hello all, I wonder if the following possible with Asterisk: 1. Use Asterisk to call party A, put party A on hold. 2. Use Asterisk to call party B 3. Finally, connect party A to party B so they can talk to each other. Note: Asterisk is suppose to do all the dialing. Thanks in advance. Foong ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
