[Asterisk-Users] Crc4 issues
Hi All, This is our 2nd E1 client that we try to use crc4 either with the e100p or with the e405p without luck. After some trials, we ask the telco to switch off crc4 on their side and everything works flawlessly. Is there anything in the crc4 calculation that may be broken? We took a look at wct1xxx.c and wct4xx.c but there doesn't seem to be much there to be fixed (apparently the crc4 calculation is done within the chip itself). We also took a look at http://lists.digium.com/pipermail/asterisk-cvs/2003-September/000126.htm l but couldn't figure out what bits should we try to set to test other card options. Is there any documentation on the card that could help us? Our zaptel looks like ... span=1,0,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 We already tried ... span=1,1,0,ccs,hdb3,crc4 span=1,1,0,ccs,hdb3,crc4,yellow span=1,0,0,ccs,hdb3,crc4,yellow ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] X100P answer in first Ring
Title: Message usecallerid=no in zapata.conf -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Senad JordanovicSent: terça-feira, 18 de maio de 2004 13:24To: [EMAIL PROTECTED]Subject: RE: [Asterisk-Users] X100P answer in first Ring I would imagine that it is I will test it , and post the result back! -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of BoaterSent: 18 May 2004 16:45To: [EMAIL PROTECTED]Subject: RE: [Asterisk-Users] X100P answer in first Ring Is this the same thing as: immediate=yes -Original Message-From: Senad Jordanovic [mailto:[EMAIL PROTECTED]Sent: Tuesday, May 18, 2004 10:21 AMTo: [EMAIL PROTECTED]Subject: RE: [Asterisk-Users] X100P answer in first Ring put: mode=immediate in your zapata.conf file -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alberto SatoSent: 18 May 2004 16:14To: [EMAIL PROTECTED]Subject: [Asterisk-Users] X100P answer in first Ring How I can do to X100P (FXO port)answer in the first Ring? Do you Yahoo!?SBC Yahoo! - Internet access at a great low price.
[Asterisk-Users] Bug in chan_iax2.c
I may have downloaded an old CVS snapshot, but the following line seems to be missing at channels/chan_iax2.c/load_module ast_mutex_init(waresl.lock); PauloHM ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Computing power for GSM codec
Hi Folks, Can someone tell me how much computing power I need on a machine running 60 channels with GSM compression? The machine will not be doing anything else but compressing 60 channels and sending them over an IAX2 trunk. Best, PauloHM ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Codec translation problems?
Hi, I'm having some problems using an IAX2 connection (using GSM) with an ALAW endpoint. Seems that the translation path GSM-SLIN-ALAW is working fine (I can hear the IAX2 party on my ALAW side perfectly), but the path ALAW-SLIN-GSM yields an distorted voice. Any clue of what can be going on? Best regards, PauloHM ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Conference server
Hi, we are setting a 120-channel conference server and would like to learn if someone already did this (hardware, problems, etc...) Best regards, PauloHM ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk compatibility list
Hi All, We are compiling an Asterisk interoperability list. If you have connected Asterisk to either a PBX or another voice/Voip device (gateway, gatekeeper, etc ...) please drop me an email. I will compile it and make it available to the list and on the wiki. Please make sure to send equipment manufacturer, signaling, protocol, and whatever else you think is relevant. Best, PauloHM ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] R2 support
Hi All, We have successfully finished implementing R2 support for *. Drop me an email off-list if you want to test it. Best, PauloHM ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] t1xxp Unable to request IRQ
Hi All, I have a e100p that is not receiving any interrupts. My /proc/interrupts look like CPU0 0: 87288 XT-PIC timer 1:104 XT-PIC keyboard 2: 0 XT-PIC cascade 8: 1 XT-PIC rtc 10: 814092 XT-PIC eth0, wcfxo 11: 0 XT-PIC t1xxp 12: 32 XT-PIC PS/2 Mouse 14: 4553 XT-PIC ide0 15: 0 XT-PIC ide1 NMI: 0 ERR: 0 My dmesg gives the following output t1xxp: Unable to request IRQ 0 Any hint? TIA, PHM ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] CAS Idle definition bits ?
Hi Daniel, AFAIK, As R2 idle bits change between countries, you may put in zaptel.conf what is the default for your locale. Something like ... cas=1-31:1001 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Daniel Bichara Sent: segunda-feira, 12 de janeiro de 2004 12:17 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] CAS Idle definition bits ? Hi, Could some one explain what are the 4 bits we should define after cas setup (zapata.com) (CAS Signalling requires idle definition in the form ':' ? Thanks in advance, Daniel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] New to asterisk? RUN... don't walk.
What about you drop your beer, stand up from your couch (if your fat belly allows you to), turn off the damn TV and try to learn some basic C programming. Then maybe you can help us in solving those frequent segmentation faults (if any). -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Me Sent: quarta-feira, 31 de dezembro de 2003 17:37 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] New to asterisk? RUN... don't walk. As a newcomer to Asterisk, you will not be welcomed with open arms. First, you will find almost no documentation on it's features. Second, if you try to ask questions, you will be flamed and pointed to worthless how-tos and 'the wiki'. These worthless documents can only be useful for explaining how things work to those already in-the-know. Lastly, Asterisk is so bug ridden, expect frequent segmentation faults. With a community so 'anti-n00b', don't expect your problems to be fixed anytime soon. RUN!!! Don't walk... away from Aterisk. __ Do you Yahoo!? Find out what made the Top Yahoo! Searches of 2003 http://search.yahoo.com/top2003 ___ 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] AGI and broken pipe
Hi All, I was able to track down what I believe is a bug when using AGI services. This bug may crash your system if your extensions.conf script is intensive in using AGI services. Depending on your system's ulimit, * keeps opening files until it reaches the system limit and then stops responding. Function app_agi/launch_script seems to leave an open and unused file. Can someone confirm this? Below is a patch that solves the problem. Index: asterisk/apps/app_agi.c === RCS file: /usr/cvsroot/asterisk/apps/app_agi.c,v retrieving revision 1.22 diff -u -r1.22 app_agi.c --- asterisk/apps/app_agi.c 5 Nov 2003 23:43:31 - 1.22 +++ asterisk/apps/app_agi.c 18 Dec 2003 13:48:38 - @@ -167,6 +167,10 @@ /* close what we're not using in the parent */ close(toast[1]); close(fromast[0]); + + // [PHM 12/18/03] + close(audio[0]) + *opid = pid; return 0; ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] AGI and broken pipe
Great ;-) Can someone else confirm this doesn't have any side effects besides solving the problem? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Angel Carpintero Sent: quinta-feira, 18 de dezembro de 2003 12:24 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] AGI and broken pipe On Thu, 18 Dec 2003 11:48:59 -0300 Paulo Mannheimer [EMAIL PROTECTED] wrote: Hi All, I was able to track down what I believe is a bug when using AGI services. This bug may crash your system if your extensions.conf script is intensive in using AGI services. Depending on your system's ulimit, * keeps opening files until it reaches the system limit and then stops responding. Function app_agi/launch_script seems to leave an open and unused file. Can someone confirm this? Below is a patch that solves the problem. Thanks Paulo, I've patched the app_agi.c and now asterisk with EAGI applications is not leaking pipes anymore :-) Angel Index: asterisk/apps/app_agi.c === RCS file: /usr/cvsroot/asterisk/apps/app_agi.c,v retrieving revision 1.22 diff -u -r1.22 app_agi.c --- asterisk/apps/app_agi.c 5 Nov 2003 23:43:31 - 1.22 +++ asterisk/apps/app_agi.c 18 Dec 2003 13:48:38 - @@ -167,6 +167,10 @@ /* close what we're not using in the parent */ close(toast[1]); close(fromast[0]); + + // [PHM 12/18/03] + close(audio[0]) + *opid = pid; return 0; ___ 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] Mysql CDR
Title: Message Hi Miklos, try starting * with -vvvc and see if there is any warning also, try connecting to your mysql server by issuing mysql asteriskcdrdb then show tables; select * from cdr; best, PHM -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of listas iPfoneSent: sexta-feira, 12 de dezembro de 2003 16:47To: [EMAIL PROTECTED]Subject: [Asterisk-Users] Mysql CDR Hi all I just installed the mysql cdr support and my database is not registering the calls :( using show modules i see that the cdr_csv.so and the cdr_addon_mysql.so are loaded It is necessary to unload the cdr_csv.so? how to do it? in crd_mysql.conf i have: [global]hostname=localhostdbname=asteriskcdrdbpassword=new_password user=asteriskcdruser;port=3306;sock=/tmp/mysql.sock i copied the crd_mysql.conf to the /etc/asterisk directory..it is to be there ..or not? and in modules.conf i have: load = cdr_addon_mysql.so It is correct? something more is needed? ( i created the database and table from wiki instructions) How can i know if asterisk is or not trying to register the calls to the database? Thanks! miklos iPFONE Telefonia IPRua Caio Graco 735 São Paulo SP iPBX +55 11 3801-3702UK +44 870 - 3403539FWD 64662sip:[EMAIL PROTECTED] www.ipfone.com.br[EMAIL PROTECTED]
[Asterisk-Users] Iax, Iax2 and Iaxcomm
Hi, I'm trying to use iaxcomm. I can place a call from the softphone, but when I place a call to it, when I answer I get ... NOTICE[16401]: File channel.c, Line 1094 (ast_read): Dropping incompatible voice frame on IAX2[paulohm]/3 of format GSM since our native format has changed to ALAW My iax.conf looks like this .. [paulohm] type=friend host=dynamic username=... secret=... context=interno ;auth=plaintext disallow=all allow=gsm allow=ulaw allow=alaw Any hint? I'm using a cvs from 4 days ago. PauloHM ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] pridump
Sorry to bother again, but what is the syntax of a dchannel? I'm trying 1, zap/1, ... without success -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Spencer Sent: quarta-feira, 10 de dezembro de 2003 19:10 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] pridump the two dchannels. mark On Wed, 10 Dec 2003, Paulo Mannheimer wrote: Hi All, Can anyone tell me what are the dev1 dev2 parameters that I should use to run pridump? I took a look at the source code but couldn't figure this one out. Best, PauloHM ___ 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
FW: [Asterisk-Users] Iax, Iax2 and Iaxcomm
Talking to myself ... ;-) Solved this by ... disallow=all allow=gsm ;allow=ulaw ;allow=alaw -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paulo Mannheimer Sent: quinta-feira, 11 de dezembro de 2003 09:02 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Iax, Iax2 and Iaxcomm Hi, I'm trying to use iaxcomm. I can place a call from the softphone, but when I place a call to it, when I answer I get ... NOTICE[16401]: File channel.c, Line 1094 (ast_read): Dropping incompatible voice frame on IAX2[paulohm]/3 of format GSM since our native format has changed to ALAW My iax.conf looks like this .. [paulohm] type=friend host=dynamic username=... secret=... context=interno ;auth=plaintext disallow=all allow=gsm allow=ulaw allow=alaw Any hint? I'm using a cvs from 4 days ago. PauloHM ___ 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] pridump
Hi All, Can anyone tell me what are the dev1 dev2 parameters that I should use to run pridump? I took a look at the source code but couldn't figure this one out. Best, PauloHM ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Errors after re-plugging T1
Hi, not sure if this is your case, but a got rid of my error 500 messages today by changing the machine's motherboard. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Markus Mayer Sent: quarta-feira, 10 de dezembro de 2003 15:18 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Errors after re-plugging T1 Hi, After temporarily pulling the T1 cable out of our Asterisk box, we ended up getting a strange error messages even after the cable was plugged back in. [...] Dec 10 09:01:11 WARNING[1192437440]: File chan_zap.c, Line 5683 (zt_pri_error): PRI: Read on 63 failed: Unknown error 500 Dec 10 09:01:21 WARNING[1192437440]: File chan_zap.c, Line 5683 (zt_pri_error): PRI: Read on 63 failed: Unknown error 500 Dec 10 09:03:42 WARNING[1192437440]: File chan_zap.c, Line 5683 (zt_pri_error): PRI: Read on 63 failed: Unknown error 500 Dec 10 09:03:52 WARNING[1192437440]: File chan_zap.c, Line 5683 (zt_pri_error): PRI: Read on 63 failed: Unknown error 500 [...] So I stopped asterisk, unloaded the kernel modules and restarted everything, but still: Dec 10 09:06:16 WARNING[1184048960]: File chan_zap.c, Line 5683 (zt_pri_error): PRI: !! No channel map, no channel, and no ds1? What am I supposed to identify? Dec 10 09:06:16 WARNING[1184048960]: File chan_zap.c, Line 5683 (zt_pri_error): PRI: !! Unable to add IE 'Channel Identification' Dec 10 09:06:20 WARNING[1184048960]: File chan_zap.c, Line 5683 (zt_pri_error): PRI: Read on 62 failed: Unknown error 500 Dec 10 09:06:20 NOTICE[1200825920]: File chan_zap.c, Line 4634 (handle_init_event): Alarm cleared on channel 1 Dec 10 09:06:20 NOTICE[1200825920]: File chan_zap.c, Line 4634 (handle_init_event): Alarm cleared on channel 2 Dec 10 09:06:20 NOTICE[1200825920]: File chan_zap.c, Line 4634 (handle_init_event): Alarm cleared on channel 3 Dec 10 09:06:20 NOTICE[1200825920]: File chan_zap.c, Line 4634 (handle_init_event): Alarm cleared on channel 4 [...] Dec 10 09:18:07 WARNING[1184048960]: File chan_zap.c, Line 5683 (zt_pri_error): PRI: Read on 62 failed: Unknown error 500 Dec 10 09:18:07 NOTICE[1200825920]: File chan_zap.c, Line 4634 (handle_init_event): Alarm cleared on channel 1 Dec 10 09:18:07 NOTICE[1200825920]: File chan_zap.c, Line 4634 (handle_init_event): Alarm cleared on channel 2 I tried this several times, to no avail. Only rebooting the box helped. The question now is: is there a way to avoid rebooting in a situation like this and still get everything to work again? Rebooting can be a huge pain. Thanks. Regards, Markus ___ 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 termination in the Netherlands
Please drop me an email off-list if you can provide IAX termination in the Netherlands. Best regards, PauloHM ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Strage bip on ISDN/PRI
Hi All, We are just starting to deploy a new PRI IVR system, and the incoming calls sometimes get random short 'bips' while navigating our IVR menu. Any hint on what this can be? Best regards, PauloHM ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Strage bip on ISDN/PRI
Sorry for the short post - I haven't included additional info because it seemed irrelevant to the issue, mainly because we have already gone through extensive trial and error. We are using RH 7.2, testing with a cvs of 2 month ago and a fresh one downloaded yesterday. The noise doesn't seem to follow any pattern, it shows up once in a while. I was wondering if this can be a spill-off from music on hold (mpg123). Best regards, PHM -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Critchfield Sent: terça-feira, 9 de dezembro de 2003 13:40 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Strage bip on ISDN/PRI On Tue, 2003-12-09 at 11:20, Paulo Mannheimer wrote: Hi All, We are just starting to deploy a new PRI IVR system, and the incoming calls sometimes get random short 'bips' while navigating our IVR menu. Any hint on what this can be? Do they occur during changes in prompts or during single recorded prompts? What revision of software are you using? What distro and version are you on? Please provide lots of data when asking questions as it helps those who would answer your question. It is easier to weed through information overload than to send several messages asking for more information. -- 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] Erratic DTMF on E1/PRI (continuation of Strage bip on ISDN/PRI)
At the same site, DTMF recognition is functioning badly, sometimes duplicating digits and sometimes totally missing others. We have checked already /proc/interrups, there is no interrupt being shared. Our zaptel has .. span=1,1,0,ccs,hdb3 On zapata we have ... switchtype=euroisdn signalling=pri_cpe relaxdtmf=no (yes doesn't seem to help) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paulo Mannheimer Sent: terça-feira, 9 de dezembro de 2003 16:33 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Strage bip on ISDN/PRI Sorry for the short post - I haven't included additional info because it seemed irrelevant to the issue, mainly because we have already gone through extensive trial and error. We are using RH 7.2, testing with a cvs of 2 month ago and a fresh one downloaded yesterday. The noise doesn't seem to follow any pattern, it shows up once in a while. I was wondering if this can be a spill-off from music on hold (mpg123). Best regards, PHM -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Critchfield Sent: terça-feira, 9 de dezembro de 2003 13:40 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Strage bip on ISDN/PRI On Tue, 2003-12-09 at 11:20, Paulo Mannheimer wrote: Hi All, We are just starting to deploy a new PRI IVR system, and the incoming calls sometimes get random short 'bips' while navigating our IVR menu. Any hint on what this can be? Do they occur during changes in prompts or during single recorded prompts? What revision of software are you using? What distro and version are you on? Please provide lots of data when asking questions as it helps those who would answer your question. It is easier to weed through information overload than to send several messages asking for more information. -- 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 mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Iax termination in India
Hi All, Please drop me an email if you can provide Iax termination in India. PauloHM ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Pbx / channel bank install
Hi all, We are about to make our first channel bank install. This will be a one PRI outside connection and up to 70 extensions. As the schedule (and the budget) is pretty tight, I would like to learn a little bit more about general experiences with channel banks, like echo cancellation problems, Caller ID usage, etc. TIA, Paulohm ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Iax2 channel usage
Correct me if I'm wrong. If I have the following setup (a local user dialing through a remote gateway using IAX2 )... User - * server - IAX connection - * server - PSTN The IAX connection between the servers is completed before the call to the PSTN is succesfully completed, thus I will be using bandwidth even for calls that aren't completed yet. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Critchfield Sent: terça-feira, 11 de novembro de 2003 03:13 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Iax2 channel usage On Mon, 2003-11-10 at 05:54, Paulo Mannheimer wrote: Thanks Steven. I'll have to find a way to use bandwidth only when the call to the PSTN is completed on the other side. Why does that matter? are you on a metered connection for bytes? [mailto:[EMAIL PROTECTED] On Behalf Of Steven Critchfield On Sun, 2003-11-09 at 14:01, Paulo Mannheimer wrote: Hi all, In a forthcommming project, I'll have one * server tentatively calling 10 PSTN numbers through IAX2 and an * gateway. Can someone tell me if bandwidth is being used for each of these calls/channels even while my gateway tries to call and connect the destination numbers? Not sure I understand the question, but I'll try and answer it anyways. IAX and IAX2 is just like any other VoIP protocol and it only uses bandwidth for active calls. If there isn't any active calls the bandwidth used is very low and only what is necessary to confirm each side is there and up. When one gateway machine takes the call and decides it needs to go to the other side, the other side will answer the call immediately and yes there is bandwidth then being used. Ideally either you are now playing prompts from the newly connected to asterisk machine, or you are connecting to a phone and the call is completed. Hope this helps a bit. -- 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
RE: [Asterisk-Users] Problem in MySql-3.23.49
Try safe_mysqld --skip-grant-tables and configure your password and your allowed hosts -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of DIPAK PAUL Sent: segunda-feira, 10 de novembro de 2003 04:45 To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED]; [EMAIL PROTECTED]; [EMAIL PROTECTED]; [EMAIL PROTECTED]; [EMAIL PROTECTED] Subject: [Asterisk-Users] Problem in MySql-3.23.49 Hi I am a user of Asterisk-0.5.0. I am a final year student of MCA in IGNOU.All the system are running in Red Hat-7.3 OS. I am able to transfered call in the following procedures: PSTN(INDIA)Mediatrix 1204Asterisk server VOCAL serverMediatrix 1204PSTN(USA) Now I want to save the cdr data in my Asterisk box.I am using RedHat-7.3 OS. I am using the command mysql -u user to access mysql promt in the Konsole, When I am trying to create the root's password by using the command SET PASSWORD FOR root=PASSWORD(password); The Konsole give the error message. ERROR 1044: Access denied for user: '@localhost' to database 'mysql' And also when I am trying to create a user by using the command GRANT ALL PRIVILEGES ON *.* TO [EMAIL PROTECTED] IDENTIFIED BY password WITH GRANT OPTION; The Konsole give the error message ERROR 1045: Access denied for user: '@localhost' (Using password: NO) Please help me to create user with password, database and grant to access all works. Thanks. Dipak; _ Are you an Elvis fan? Want to visit Heartbreak Hotel? http://server1.msn.co.in/sp03/elvis/ Here's how you can win a trip! ___ 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] Iax2 channel usage
Thanks Steven. I'll have to find a way to use bandwidth only when the call to the PSTN is completed on the other side. Best, PauloHM -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Critchfield Sent: domingo, 9 de novembro de 2003 17:02 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Iax2 channel usage On Sun, 2003-11-09 at 14:01, Paulo Mannheimer wrote: Hi all, In a forthcommming project, I'll have one * server tentatively calling 10 PSTN numbers through IAX2 and an * gateway. Can someone tell me if bandwidth is being used for each of these calls/channels even while my gateway tries to call and connect the destination numbers? Not sure I understand the question, but I'll try and answer it anyways. IAX and IAX2 is just like any other VoIP protocol and it only uses bandwidth for active calls. If there isn't any active calls the bandwidth used is very low and only what is necessary to confirm each side is there and up. When one gateway machine takes the call and decides it needs to go to the other side, the other side will answer the call immediately and yes there is bandwidth then being used. Ideally either you are now playing prompts from the newly connected to asterisk machine, or you are connecting to a phone and the call is completed. Hope this helps a bit. -- 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] Iax2 channel usage
Hi all, In a forthcommming project, I'll have one * server tentatively calling 10 PSTN numbers through IAX2 and an * gateway. Can someone tell me if bandwidth is being used for each of these calls/channels even while my gateway tries to call and connect the destination numbers? Best, PauloHM ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sip bandwidth usage
This is exactly what I did. I used Xten's GSM driver to call a Zap extension. Readings where 100 Kbits/s. Using uLAW returned 80 Kbits/s !!! I also downloaded Xten pro to test their g729 codec, readings were even worse. That's why I'm so intrigued. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of WipeOut Sent: quinta-feira, 30 de outubro de 2003 10:24 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Sip bandwidth usage Paulo Mannheimer wrote: That's weird. I've done some testing both with GS and Xten products, and my iptraf readings show much more than your numbers. It depends on how you did your tests.. If you ran iptraf an PC1 and made a call from PC2(X-Lite) to GS and your sip.conf entry for either have canreinvite=no then you will get double the traffic.. Best bet is to run iptraf on the Asterisk box and then make a call from the phone to Asterisk (eg to voicemail, echo test, the pstn or a Zap channel) so that the IP traffic is only one client making a call to Asterisk using the selected codec.. That should give you the best reading.. Later.. Paulo Mannheimer wrote: Hi All- I'm working on a project that will have remote (internet)access to an * server through SIP phones, either soft or hard ones. Does anyone have any experience to share about which SIP product they are using under similar conditions, as well as which codec is being used and bandwidth usage? TIA! PauloHM Depends on the phone.. If you are using a Grand Stream then the best you will get is G.711 (+- 85Kb/s including overheads).. If you are using Snom's or X-Lite/X-Pro you have the option to use the GSM (+- 34Kb/s including overheads) codec.. X-Lite/X-Pro also support iLBC (+- 28Kb/s including overheads) although it does not currently work with Asterisk, and GrandStream have said they are going to support it as well soon.. All the phones have support for G.729 (+- 22Kb/s) either as standard or by buying a sepertate licence.. Including Asterisk.. Hope that helps.. ___ 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] Sip bandwidth usage
Hi All- I'm working on a project that will have remote (internet)access to an * server through SIP phones, either soft or hard ones. Does anyone have any experience to share about which SIP product they are using under similar conditions, as well as which codec is being used and bandwidth usage? TIA! PauloHM ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Beta testers for visual configuration tool for asterisk
Hi, thanks for you reply. I'll send you till the end of the week more info on how to download and use it. Best regards, Paulo Mannheimer -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lal, Deepak (Contractor) Sent: sexta-feira, 17 de outubro de 2003 14:52 To: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] Beta testers for visual configuration tool for asterisk Count me in too. -Original Message- From: sip [mailto:[EMAIL PROTECTED] Sent: Friday, October 17, 2003 1:56 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Beta testers for visual configuration tool for asterisk count me in - Original Message - From: Paulo Mannheimer [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, October 17, 2003 12:23 PM Subject: [Asterisk-Users] Beta testers for visual configuration tool for asterisk Hi All, We've been developing for a while an IDE for Asterisk, and the time has come to open it for beta testers. You can check at www.instant.com.br/viv.html for a snapshot of the application. Current modules are Dialplan and VoiceMail configuration. As you may see, it is all-visual, with drag and drop support and integrated sound recording, saving and cross-checking, so you dialpland doesn't crash because of a missing sound file. Beta users will have to download and install either a 16 Mb or a 4Mb Windows program, depending if you already have or not JRE 1.4.2 installed. This client works together with a tomcat-based application, which will be running on our servers during the trial. If you wish to participate, please let me know off-list. I'll get in touch with the first 5 answers to arrange how the test will be performed. Best, PauloHM ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users This message was checked by MailScan for WorkgroupMail. www.workgroupmail.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 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Beta testers for visual configuration tool f or asterisk
Hi, thanks for you reply. I'll send you till the end of the week more info on how to download and use it. Best regards, Paulo Mannheimer -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Thomas Wienecke Sent: sexta-feira, 17 de outubro de 2003 17:43 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Beta testers for visual configuration tool f or asterisk Am Freitag, 17. Oktober 2003 19:51 schrieb Lal, Deepak (Contractor): i am willing to assist also. mostly on weekends, i´ m afraid, but willing. Thomas W. Count me in too. -Original Message- From: sip [mailto:[EMAIL PROTECTED] Sent: Friday, October 17, 2003 1:56 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Beta testers for visual configuration tool for asterisk count me in - Original Message - From: Paulo Mannheimer [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, October 17, 2003 12:23 PM Subject: [Asterisk-Users] Beta testers for visual configuration tool for asterisk Hi All, We've been developing for a while an IDE for Asterisk, and the time has come to open it for beta testers. You can check at www.instant.com.br/viv.html for a snapshot of the application. Current modules are Dialplan and VoiceMail configuration. As you may see, it is all-visual, with drag and drop support and integrated sound recording, saving and cross-checking, so you dialpland doesn't crash because of a missing sound file. Beta users will have to download and install either a 16 Mb or a 4Mb Windows program, depending if you already have or not JRE 1.4.2 installed. This client works together with a tomcat-based application, which will be running on our servers during the trial. If you wish to participate, please let me know off-list. I'll get in touch with the first 5 answers to arrange how the test will be performed. Best, PauloHM ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users This message was checked by MailScan for WorkgroupMail. www.workgroupmail.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 ___ 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] Beta testers for visual configuration tool for asterisk
Hi, thanks for you reply. I'll send you till the end of the week more info on how to download and use it. Best regards, Paulo Mannheimer -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Coberly Sent: sábado, 18 de outubro de 2003 14:49 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Beta testers for visual configuration tool for asterisk Hi, We would be interested in this project also. Paulo Mannheimer wrote: Hi All, We've been developing for a while an IDE for Asterisk, and the time has come to open it for beta testers. You can check at www.instant.com.br/viv.html for a snapshot of the application. Current modules are Dialplan and VoiceMail configuration. As you may see, it is all-visual, with drag and drop support and integrated sound recording, saving and cross-checking, so you dialpland doesn't crash because of a missing sound file. Beta users will have to download and install either a 16 Mb or a 4Mb Windows program, depending if you already have or not JRE 1.4.2 installed. This client works together with a tomcat-based application, which will be running on our servers during the trial. If you wish to participate, please let me know off-list. I'll get in touch with the first 5 answers to arrange how the test will be performed. Best, PauloHM ___ 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] Beta testers for visual configuration tool for asterisk
Hi, thanks for you reply. I'll send you till the end of the week more info on how to download and use it. Best regards, Paulo Mannheimer -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Josh Roberson Sent: sábado, 18 de outubro de 2003 01:21 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Beta testers for visual configuration tool for asterisk I would like to beta test this tool. :) Looks like it could be a good thing. -- Josh Roberson Indigent Networks 1.877.677.9647 x1 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paulo Mannheimer Sent: Friday, October 17, 2003 11:24 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Beta testers for visual configuration tool for asterisk Hi All, We've been developing for a while an IDE for Asterisk, and the time has come to open it for beta testers. You can check at www.instant.com.br/viv.html for a snapshot of the application. Current modules are Dialplan and VoiceMail configuration. As you may see, it is all-visual, with drag and drop support and integrated sound recording, saving and cross-checking, so you dialpland doesn't crash because of a missing sound file. Beta users will have to download and install either a 16 Mb or a 4Mb Windows program, depending if you already have or not JRE 1.4.2 installed. This client works together with a tomcat-based application, which will be running on our servers during the trial. If you wish to participate, please let me know off-list. I'll get in touch with the first 5 answers to arrange how the test will be performed. Best, PauloHM ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.522 / Virus Database: 320 - Release Date: 9/29/2003 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.522 / Virus Database: 320 - Release Date: 9/29/2003 ___ 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] Beta testers for visual configuration tool for asterisk
Hi All, We've been developing for a while an IDE for Asterisk, and the time has come to open it for beta testers. You can check at www.instant.com.br/viv.html for a snapshot of the application. Current modules are Dialplan and VoiceMail configuration. As you may see, it is all-visual, with drag and drop support and integrated sound recording, saving and cross-checking, so you dialpland doesn't crash because of a missing sound file. Beta users will have to download and install either a 16 Mb or a 4Mb Windows program, depending if you already have or not JRE 1.4.2 installed. This client works together with a tomcat-based application, which will be running on our servers during the trial. If you wish to participate, please let me know off-list. I'll get in touch with the first 5 answers to arrange how the test will be performed. Best, PauloHM ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Manager
Here is a patch that I posted to Mark a couple of days ago. Haven't tested it too much. It basically implements the system command through the manager interface. Due to security issues, you have to create a system.conf file at /etc/asterisk with the commands that you wish to allow. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chee Foong Sent: terça-feira, 14 de outubro de 2003 04:53 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk Manager Hello mate, I tried that, i get No such command 'System(ls)'. I can't even make it work on CLI. I am able to execute linux command (via CLI) by prefix command with a !. I would like to know how to do it throut the manager appllication. Thanks for you reply. CF - Original Message - From: [EMAIL PROTECTED] To: Chee Foong [EMAIL PROTECTED] Sent: Tuesday, October 14, 2003 2:08 PM Subject: Re: [Asterisk-Users] Asterisk Manager On Tue, 14 Oct 2003, Chee Foong wrote: Can I execute linux command like(ls, mkdir) through the Manager interface? nain*CLI show application system nain*CLI -= Info about application 'System' =- [Synopsis]: Execute a system command [Description]: System(command): Executes a command by using system(). Returns -1 on failure to execute the specified command. If the command itself executes but is in error, and if there exists a priority n + 101, where 'n' is the priority of the current instance, then the channel will be setup to continue at that priority level. Otherwise, System returns 0. -- Mirza Wasim Baig | Principal Consultant | Convergence Business Systems PK #48, St 32, Sector F-6/1, Islamabad, Pakistan 44000 | US: +1(800)460-1446 VOX: +92(51)282-0628 | FAX: +92(51)282-0621 | GSM: +92(300)850-8070 This mail is confidential intended solely for the use of the addressee. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users systemcli2.diff Description: Binary data
RE: [Asterisk-Users] indications.conf
Take a look at zaptel/zonedata.c, I guess you have to change it. Greetings from Rio de Janeiro ;-) PHM -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andre Lomonaco Sent: quarta-feira, 15 de outubro de 2003 16:40 To: '[EMAIL PROTECTED]' Subject: [Asterisk-Users] indications.conf Hi, I´m trying to make * work with Brazilian analog signalling.. I´m using the following in indications.conf file... [br] description = Brasil ringcadence = 1000,4000 dial = 425 busy = 425/250,0/250 ring = 425/1000,0/4000 callwaiting = 425/60,0/250,425/60,0/5000 I changed zaptel.conf to loadzone=br #loadzone=fr #loadzone=de #loadzone=uk #loadzone=fi #loadzone=jp #loadzone=sp #loadzone=no defaultzone=br Now when I try to load zaptel, wcfxs and wcfxo, I got an error: [EMAIL PROTECTED] asterisk]# modprobe zaptel [EMAIL PROTECTED] asterisk]# modprobe wcfxo Notice: Configuration file is /etc/zaptel.conf line 128: No such tone zone known: br 1 error(s) detected /lib/modules/2.4.20-8/misc/wcfxo.o: post-install wcfxo failed /lib/modules/2.4.20-8/misc/wcfxo.o: insmod wcfxo failed Any tip to solve this problem... Thanks a Lot... Andre Lomonaco ___ 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] (still) channel problems
Hi folks, I'm still having the following problem, maybe someone can help me out of it. Two IDENTICAL MACHINES (same motherboard, same RH 7.2, same *) communicate through IAX2. Everything works ok on machine 1. On machine 2, if I try to use 4 fxo's from a TDM400 card, sound gets lousy. If I manually destroy one of the zap channels (e.g. zap destroy channel 4), sound gets good again. Help! PauloHM ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Incomming call management
Hi all, I'm looking for the following functionality: if my queues reach a certain threshold, I would like to disable any available zap / PRI channels, so my telco doesn't try to connect more people. After a while, I will enable them again. Any hints on how to implement this? Should I be looking to patch * on chan_zap level, or should I somehow ioctl zapata and disable these channels somehow? Best regards, PauloHM ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Interface with PBX
Hi Folks, I'm trying to interface * with a PBX, but seems that his ring cadence is somewhat different, and my T100 doesn't show any call coming in. I've tried to change zaptel to new values but still couldn't make it work. Is there any other place where I should be changing some parameter? Is there any tool to measure the cadence timing that this pbx is providing? Thanks! PauloHM ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sip call waiting
Hi folks, As none of the SIP softphones that I tested can disable more than one incoming call, I decided to implement it by software ;-) I'm attaching a patch that does it. To make it work, modify your sip.conf file and include callwaiting=[0|1] at the general section, or for each peer that you wish to control. Please note that I haven't tested it too much, and my source tree is quite old, so I'm not sure if this patch will apply to the current CVS. Let me know if you find something wrong asap, as this goes into production tomorrow ! Best regards, PauloHM sipcallwaiting.diff Description: Binary data
RE: [Asterisk-Users] Sip call waiting
Damn. Seems to implement what I was looking for ... ;-( Does anyone know if the incominglimit works if the call is being generated from a queue? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Todd Sent: September 17, 2003 2:19 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Sip call waiting Hi folks, As none of the SIP softphones that I tested can disable more than one incoming call, I decided to implement it by software ;-) I'm attaching a patch that does it. To make it work, modify your sip.conf file and include callwaiting=[0|1] at the general section, or for each peer that you wish to control. Please note that I haven't tested it too much, and my source tree is quite old, so I'm not sure if this patch will apply to the current CVS. Let me know if you find something wrong asap, as this goes into production tomorrow ! Best regards, PauloHM Paulo - Have you tried using the already-existing feature of outgoinglimit= in sip.conf? I have not tried it as a call waiting canceller, but you might be able to set it to 1 to get what you want. http://bugs.digium.com/bug_view_page.php?bug_id=098 JT ___ 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 center design question
Hi Rich, We have done this before. We basically developed a small client that sits on every machine and communicates with * to get information about an incoming call. Contact me off-list and I will be glad to tell you more about the entire solution. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: September 16, 2003 1:39 PM To: Asterisk-users-list Subject: [Asterisk-Users] call center design question Would like to deploy * in a small help desk environment (five to ten people) using call queues and some sort of CTI interface to pop Remedy screen data in front of the help desk person receiving the call. Data to be popped would be based on CallerID. Anyone doing something similar? Anyone interfacing to an external Remedy system? Any reference sites that I could read/learn more of the requirements and/or 10,000 foot implementation? Rich ___ 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 center design question
Sure, here it it goes. We first developed a small client that sits on a Windows machine taskbar (sorry guys, but customer had only windows machines ... Hehehe). Upon boot, the client is loaded and communicates with the * server telling its IP address and extension number. When a call is about to be transferred to that extension, an * AGI sends the client all information that was programmed to be transferred. We had to patch app_queue.c to do this (giving it the ability to call an AGI just before a call is being answered by a queue member). I've submitted a patch with this change but I'm not sure if it was accepted. Once the client receives the data, it makes it available through the clipboard. All your application has to do is to monitor the clipboard waiting for any data. If something shows up, it's an indication that the agent's phone is going to ring pretty soon! We are currently expanding our small client to handle much more tasks, maybe even a complete SIP/IAX softphone, so we can deploy an entire contact center based on VoIP. Anyone willing to help? Best, PHM -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of PJ Welsh Sent: September 16, 2003 4:09 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] call center design question Yes, Please share. On Tue, Sep 16, 2003 at 03:05:33PM -0400, Yifang Dai wrote: On Tue, Sep 16, 2003 at 03:27:44PM -0300, Paulo Mannheimer wrote: Hi Rich, We have done this before. We basically developed a small client that sits on every machine and communicates with * to get information about an incoming call. Contact me off-list and I will be glad to tell you more about the entire solution. Hi, I'm interested in this solution too, can you share it with the group? Thanks! -- Yifang Dai | eFax: (847)628-0255 |Debian GNU/Linux [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
RE: [Asterisk-Users] call center design question
I'm not sure I understood your question. As far as I know, listening to the manager interface wouldn't give me enough information. At the moment where the call is transferred, the client has already browsed through a couple of menus, setting some variables. The AGI sends the content of these variables to the client. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of TC Sent: September 16, 2003 7:39 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] call center design question Sure, here it it goes. When a call is about to be transferred to that extension, an * AGI sends the client all information that was programmed to be transferred. We had to patch app_queue.c to do this (giving it the ability to call an AGI just before a call is being answered by a queue member Having gone down this route now, and the benefit of hindsight ... what advantage do you find with made to * channel/app vs. listening on events from the manager interface then doing your client communication when getting the correct state msg ___ 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 center design question
Is there anyone out there with a custom client softphone and is interested in integrating both solutions? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of TC Sent: September 16, 2003 3:53 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] call center design question Hi Rich, We have done this before. We basically developed a small client that sits on every machine and communicates with * to get information about an incoming call. Contact me off-list and I will be glad to tell you more about the entire solution. Actually you might be surpised that there are others who are interested in the details of your solution :) The standard designs of have seen fall into a few categories 1) modifications to a channel driver pushing via a socket to clients 2) server listening to manager events, pushing via a socket to clients 3) clients listening to manager events selecting events of interest 4) custom client softphones receving urls/callerid 5) other unique solutions ... ___ 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] SIP busy
Thanks John and all,
Unfortunatelly this will not work for me, because the SIP phones are
agents and I'm managing incomming calls through a queue.
Anyone knows a SIP softphone that supports disabling call waiting?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Todd
Sent: September 11, 2003 8:20 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] SIP busy
[message re-ordered]
- Original Message -
From: Paulo Mannheimer [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, September 11, 2003 4:32 PM
Subject: [Asterisk-Users] SIP busy
Hi,
I would like * to treat a SIP extension as a normal extension, when
it comes to the busy functionality. In other words, if someone tries
to call the SIP phone and there is already an ongoing conversation,
the new caller should get a busy message/tone
Is there any parameter that I can set? Is this something that should
be configured at my softphone?
Best,
PHM
Basically you need to disable call waiting on your SIP device (if it
supports call waiting to begin with). When the second call comes into
the SIP device with call waiting disabled, it should send a 486 SIP
message (mine says 486 Busy Here) back to the Asterisk. You can see
this in sip debug mode on the console.
Then setup your extensions.conf to take the appropriate action on Busy
like any other extension.
Sean
___
Sean Robertson
NETXUSA
p. 800-289-6389
f. 864-233-4344 Ask me about Voice over IP.
http://www.netxusa.com/
Another method would simply be to keep a call counter for existing
calls, and increment it/decrement it when calls are made and then
hung up. Put a short GotoIf before your Dial statement to check if
the line is occupied and then reject the call if that is the case.
[test]
exten = 1234,1,DBGet(STATUS=${EXTEN}/OFFHOOK)
exten = 1234,2,SetVar(CALLEDNUMBER=${EXTEN})
exten = 1234,3,GotoIf($[${STATUS} = 1}]?106:3)
exten = 1234,4,DBPut(${EXTEN}/OFFHOOK=1)
exten = 1234,5,Dial(SIP/1234,20)
exten = 1234,6,DBPut(${EXTEN}/OFFHOOK=0)
exten = 1234,7,Voicemail2(u1234)
exten = 1234,106,DBPut(${EXTEN}/OFFHOOK=0)
exten = 1234,107,Voicemail2(b1234)
exten = h,1,DBPut(${CALLEDNUNMBER}/OFFHOOK=0)
exten = h,2,Hangup
JT
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RE: [Asterisk-Users] Is there any MFC-R2 implementation for asterisk?
Me too. I sent Steve an email about this, but didn't get a reply. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of LQ (Asterisk) Sent: September 11, 2003 10:19 AM To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Subject: [Asterisk-Users] Is there any MFC-R2 implementation for asterisk? The last thing that I read about it was: Steve Underwood [EMAIL PROTECTED] wrote on Sep 3: Is EM designed to work with the E1 driver code? I think probably not. I had to fix some things to get proper access to the CAS signaling bits when I implemented MFC/R2... So, apparently he implemented it. I was trying to contact Steve, but he isn't answering me. Does anybody have any news about it? Regards, Pablo. -Original Message- From: Herry Sitepu [mailto:[EMAIL PROTECTED] Posted At: Thursday, September 11, 2003 5:07 Posted To: Asterisk Conversation: [Asterisk-Users] Is there any MFC-R2 implementation for asterisk? Subject: [Asterisk-Users] Is there any MFC-R2 implementation for asterisk? Hi guys, Is there anyone has implemented MFC-R2 for astrisk? Regards Herry Sitepu ___ 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] SIP busy
Hi, I would like * to treat a SIP extension as a normal extension, when it comes to the busy functionality. In other words, if someone tries to call the SIP phone and there is already an ongoing conversation, the new caller should get a busy message/tone Is there any parameter that I can set? Is this something that should be configured at my softphone? Best, PHM ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Noise over iax2 and FXO
Hi, I have an installation connecting two machines through IAX2. Each machine has 3 FXS and 4 FXO ports. Everything seems to work fine, except on one FXO port, where I constantly get a strange locomotive noise when I use it to terminate an IAX2 incomming call. Usually after a while the strange noise goes away, but it is very annoying. Any hints on what can be causing this? Thanks! PHM ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Urgent help - File size limit exceeded error
Hi, My installation that was working flawlessly for 2 weeks stopped working when I installed a g729 codec license. Now, if I try to start * I get a File size limit exceeded error and the program aborts. Any clue of what's going on? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Urgent help - File size limit exceeded error
Found what was going on ... My debug file at /var/log/asterisk was greater than 2 gigs (don't ask me why ...) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paulo Mannheimer Sent: September 08, 2003 8:47 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Urgent help - File size limit exceeded error Hi, My installation that was working flawlessly for 2 weeks stopped working when I installed a g729 codec license. Now, if I try to start * I get a File size limit exceeded error and the program aborts. Any clue of what's going on? ___ 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] Arraycom voip phone
Hi All, Does anyone have any experience with the ArrayCom VoIP phone? I bought one a couple of weeks ago, it used to work quite well with * until I misconfigured one option. I now cannot make it work anymore, because the phone boots up, doesn't find a valid SIP gateway, resets itself and keeps rebooting indefinetely ;-( Their technical support refuses to answer my questions. Any hint on a master reset? PauloHM ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] E1 problems
Hi, I'm testing an E1 with EM signaling. Some of the problems I'm running into are the following: 1) if I try to configure any channel above channel 15, I start getting a multiframe alignment error on my telco test equipment. So I have my zaptel file only configured for 15 channels, like this span=1,1,0,cas,hdb3 em=1-15 2) When the test equipment tries to send me a DTMF string, I only get the first one. Any thoughts? Best, PauloHM ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Why doesnt anyone reply me ?
Am I crazy or do you have a Goto just before your Record command?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of kaku ustaad
Sent: August 25, 2003 8:33 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Why doesnt anyone reply me ?
I have posted soo many times in the past but never recieved even a
single
reply . seem like you people are ignoring me or either way too busy ..
never mind this is my last try .
How can record a conversation with asterisk ?
I tried to use Record() but dint work for me .. here is what i tried .
exten = s,1,Wait,1 ; Wait a second, just for fun
exten = s,2,Answer ; Answer the line
exten = s,3,DigitTimeout,5 ; Set Digit Timeout to 5 seconds
exten = s,4,ResponseTimeout,10 ; Set Response Timeout to 10
seconds
exten = s,5,BackGround(seattle); Play a congratulatory message
exten = i,1,Goto,sip|${EXTEN}|1
exten = i,2,Record(input:wav)
include = sip
_
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[Asterisk-Users] * and IAX as a gateway to video conferencing
Has anyone used * and IAX in a gateway to a videoconferencing application? Best, PauloHM
RE: [Asterisk-Users] new on E100P
Answering myself, It seems that my zaptel service script wasnt loading the wct1xxp module. Should I load something else? Torisa? Tor2? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paulo Mannheimer Sent: August 12, 2003 11:23 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] new on E100P Hi, Im installing my first E100P. My zaptel reads the following: Span=1,0,0,ccs,hdb3,crc4 Em=1-31 My Zapata.conf reads the following: Signaling = em_w Channel =1-15 Channel =16-31 After starting the zapter service I get: ZT_SPANCONFIG failed on span 1: No such device or address (6) ??? PauloHM
[Asterisk-Users] new on E100P
Hi, Im installing my first E100P. My zaptel reads the following: Span=1,0,0,ccs,hdb3,crc4 Em=1-31 My Zapata.conf reads the following: Signaling = em_w Channel =1-15 Channel =16-31 After starting the zapter service I get: ZT_SPANCONFIG failed on span 1: No such device or address (6) ??? PauloHM
[Asterisk-Users] R2 support
Hi folks, where can I find the R2 beta code for Asterisk? Best, PauloHM
[Asterisk-Users] voicemail file access problems
Hi folks, Im having problems accessing my voicemail files through the web interface. I remember that this was discussed on the list, and it seems to be a permission problem, but I couldnt find any answer by searching the archives. Any hint? PauloHM
RE: [Asterisk-Users] voicemail file access problems
Thanks! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman Lesher Sent: July 30, 2003 4:06 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] voicemail file access problems On Wednesday 30 July 2003 01:41 pm, Paulo Mannheimer wrote: Hi folks, I'm having problems accessing my voicemail files through the web interface. I remember that this was discussed on the list, and it seems to be a permission problem, but I couldn't find any answer by searching the archives. Any hint? chown root vmail.cgi chmod u+s vmail.cgi -Tilghman ___ 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] voicemail file access problems
This is getting too confusing for me ;-( Could someone summarize what are the steps necessary to make vmail.cgi work on a system? Something like this: 1) copy vmail.cgi to your cgi-bin directory 2) copy images/*.gif to your img directory 3) grant 4) grant -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Patrick Sent: July 30, 2003 5:33 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] voicemail file access problems Did it work after you left a new voice mail message? I was looking into the source code to fix it so that the euid was set to nobody, create the file and then change it back to uid 0, but that didn't work. Or, maybe change the file mode was 770 with the group set so that the webserver could modify the file so I wouldn't have to run a suid .cgi script. Patrick On Wed, 30 Jul 2003, Todd Lieberman wrote: I fixed my own problem. I had just did chmod 755 vmail.cgi and it worked. you still need to make sure nobody has read/write permission on /var/spool/asterisk/vm/$MBOX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Todd Lieberman Sent: Wednesday, July 30, 2003 3:50 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] voicemail file access problems I did the chown and now I get [Wed Jul 30 15:51:11 2003] [error] [client 216.183.124.45] Setuid/gid script is writable by world., referer: http://asterisk.weichertrents.com/cgi-bin/vmail.cgi -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Paulo Mannheimer Sent: Wednesday, July 30, 2003 3:23 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] voicemail file access problems Thanks! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman Lesher Sent: July 30, 2003 4:06 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] voicemail file access problems On Wednesday 30 July 2003 01:41 pm, Paulo Mannheimer wrote: Hi folks, I'm having problems accessing my voicemail files through the web interface. I remember that this was discussed on the list, and it seems to be a permission problem, but I couldn't find any answer by searching the archives. Any hint? chown root vmail.cgi chmod u+s vmail.cgi -Tilghman ___ 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] Call Dropping
Try increasing busycount (a hidden parameter) at Zapata.conf Mine works like a charm with busydetect=yes busycount=6 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerk Face Sent: July 29, 2003 9:03 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Call Dropping Some of my end users have reported to me that occasionally they'll be in the middle of a conversation and the call will be dropped. I have yet to catch anything unusual when debugging the channels. Has anybody had this problem before, if so, how did you solve it? My hardware: 2 X100P 1 TDM40B Thanks for your time _ STOP MORE SPAM with the new MSN 8 and get 2 months FREE* http://join.msn.com/?page=features/junkmail ___ 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] busydetect and random hangups
This was it! Thanks! BTW, is busycount a hidden feature, or should it be listed in Zapata.conf.sample ? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brancaleoni Matteo Sent: July 22, 2003 5:38 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] busydetect and random hangups increase busycount in zapata.conf busycount=6 is ok for me. the default is 3 , I think, and sometimes it hangsup on speaking (or some other moh ;) ) Matteo. Il mar, 2003-07-22 alle 22:11, Paulo Mannheimer ha scritto: Hi, I'm having random hangup problems with zap channels. If I turn busydetect off in Zapata.conf, * fails completely to detect a user hangup in the middle of a script. On the other hand, if I turn it on, everything works much better, but long calls tend to be hung up without a motive. Any other parameter that I can try? Any #define that I can tweak and recompile? Will callprogress be of any help, as I'm outside the US? Thanks! PauloHM -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ 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] busydetect and random hangups
Hi, Im having random hangup problems with zap channels. If I turn busydetect off in Zapata.conf, * fails completely to detect a user hangup in the middle of a script. On the other hand, if I turn it on, everything works much better, but long calls tend to be hung up without a motive. Any other parameter that I can try? Any #define that I can tweak and recompile? Will callprogress be of any help, as Im outside the US? Thanks! PauloHM
[Asterisk-Users] new voicemail messages
Hi,
Im localizing the voicemail messages to Portuguese. To
make it possible for another person to translate it, Ive set up a couple
of extensions that call the following macro for each message on the system. After
recording, I can perfectly hear each message using Playback.
When I try to play the new recorded message using VoiceMailMain, I cant hear the new message (line
goes silent), and the new file grows in size. The previous recorded content is erased
Any hint? Yes, Im setting setlanguage
before calling VoiceMailMain.
[macro-record]
exten =
s,1,setlanguage(us)
exten =
s,2,Playback(${ARG1})
exten =
s,3,setlanguage(br)
exten =
s,4,Playback(br/${ARG1})
exten =
s,5,Wait(2)
exten =
s,6,Record(br/${ARG1}:gsm)
exten =
s,7,Goto(s,1)
Paulo H. Mannheimer
Instant Solutions
+55 21 2512.7999
+55 21 8818.7999
[Asterisk-Users] gotoiftime error
Hi folks, There was a bug with the GotoIfTime built-in command, under certain circumstances a variable contained garbage, screwing up correct time identification. Im submitting now a patch to Mark so this can be fixed. PauloHM
RE: [Asterisk-Users] gotoiftime error
Sure, here it goes. As you may notice, a local instance of the variable ast_include is used in function pbx_builtin_gotoiftime As the local variable is not initialized to zero, its minmask bitfields contain garbage, thus sometimes yielding true for unallowed times. BTW, nice work on the bitfield logic. PauloHM -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman Lesher Sent: July 01, 2003 1:24 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] gotoiftime error On Tuesday 01 July 2003 09:08 am, Paulo Mannheimer wrote: Hi folks, There was a bug with the GotoIfTime built-in command, under certain circumstances a variable contained garbage, screwing up correct time identification. I'm submitting now a patch to Mark so this can be fixed. What exactly was the error? Could you post the patch here? Since I wrote the GotoIfTime application, I'm curious to know what the bug was. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users gotoiftime.diff Description: Binary data
[Asterisk-Users] stuck channel
Im getting this intermittent problem, sometimes a zap channel gets stuck after a call. Below is a snapshot of the channel. Any ideas what can be happening? Name: Zap/1-1 Type: Zap UniqueID: 1056988772.10 Caller ID: (N/A) DNID Digits: (N/A) State: Up (6) Rings: 0 NativeFormat: 68 WriteFormat: 4 ReadFormat: 4 1st File Descriptor: 19 Frames in: 3524 Frames out: 0 Time to Hangup: 0 -- PBX -- Context: interno Extension: 24242646 Priority: 2 Call Group: 2 Pickup Group: 2 Application: Congestion Data: (Empty) Stack: 0 Blocking in: ast_waitfor_nandfds Paulo H. Mannheimer Instant Solutions +55 21 2512.7999 +55 21 8818.7999
[Asterisk-Users] app_queue ringing all available channels
I just noticed that app_queue here rings together all available extensions, which may not be the best for a call center. Is this the correct functionality or something specific from my installation? PauloHM
RE: [Asterisk-Users] dynamic queue channels
Title: Message
Just posted a patch do Mark implementing
this.
There are two new commands:
-
AddQueueMember(queuename[|interface])
-
RemoveQueueMember(queuename[|interface])
An example would be
AddQueueMember(techsupport|Zap/3-1)
Hope you find it useful
PauloHM
-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Benjamin Miller
Sent: June 24, 2003 11:41 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users]
dynamic queue channels
There may be some
trickiness that can be done with chan_local asagents of the
call queue. However, a much more elegant way to do this would be to
create an app_addagent and app_removeagent that allows the dynamic addition and
removal of extensions from the agent pool for a given queue.
addagent(${CHANNEL}, techsupport) or something like that.
Ben
-Original Message-
From: Paulo Mannheimer
[mailto:[EMAIL PROTECTED]
Sent: Monday, June 23, 2003 6:36 PM
To:
[EMAIL PROTECTED]
Subject: [Asterisk-Users] dynamic
queue channels
Hi, Im trying to build a call
center application that allows attendants to come in the morning and dial a
certain extension to make their extension available.
I wouldnt like to use the
AgentLogin app because their line would need to stay off-hook (is this
correct?)
Is there any SET channel status command
that would allow me to do something like this?
PauloHM
RE: [Asterisk-Users] dynamic queue channels
Title: Message
Sure, here it goes. PLEASE READ THE
DISCLAIMER BELOW ;-)
This is my first true patch to asterisk,
no money back guarantee. Please backup all your hard disk before applying it !!! (just kidding )
PauloHM
-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of TC
Sent: June 26, 2003 5:02 PM
To:
[EMAIL PROTECTED]
Subject: Re: [Asterisk-Users]
dynamic queue channels
Could
you post to the list,so we could take look just in case mark has a
plate load of items to merge ??
-Original Message-
From: Benjamin Miller [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
[EMAIL PROTECTED]
Date: June 26, 2003 11:45 AM
Subject: RE: [Asterisk-Users]
dynamic queue channels
Nice work! :-)
Thanks
Cant wait to see it in
cvs.
-Original Message-
From: Paulo Mannheimer [mailto:[EMAIL PROTECTED]]
Sent: Thursday, June 26, 2003
11:23 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users]
dynamic queue channels
Just posted a patch do
Mark implementing this.
There are two new
commands:
- AddQueueMember(queuename[|interface])
- RemoveQueueMember(queuename[|interface])
An example would be
AddQueueMember(techsupport|Zap/3-1)
Hope you find it useful
PauloHM
-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Benjamin Miller
Sent: June
24, 2003 11:41 AM
To:
[EMAIL PROTECTED]
Subject: RE: [Asterisk-Users]
dynamic queue channels
There
may be some trickiness that can be done with chan_local
asagents of the call queue. However, a much more elegant way to do
this would be to create an app_addagent and app_removeagent that allows the
dynamic addition and removal of extensions from the agent pool for a given
queue. addagent(${CHANNEL}, techsupport) or something like that.
Ben
-Original Message-
From: Paulo Mannheimer [mailto:[EMAIL PROTECTED]
Sent: Monday,
June 23, 2003 6:36 PM
To:
[EMAIL PROTECTED]
Subject: [Asterisk-Users] dynamic
queue channels
Hi, Im trying to build a call
center application that allows attendants to come in the morning and dial a
certain extension to make their extension available.
I wouldnt like to use the
AgentLogin app because their line would need to stay off-hook (is this correct?)
Is there any SET channel status
command that would allow me to do something like this?
PauloHM
newappqueue.diff
Description: Binary data
RE: [Asterisk-Users] dynamic queue channels
Title: Message
Correct.
This should be part of my disclaimer.
Sorry about that.
-Original Message-
From: TC [mailto:[EMAIL PROTECTED]]
Sent: June 26, 2003 6:31 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users]
dynamic queue channels
FYI
Not a biggie but that pacth is not
against current cvs
-Original Message-
From: Paulo Mannheimer [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
[EMAIL PROTECTED]
Date: June 26, 2003 2:13 PM
Subject: RE: [Asterisk-Users]
dynamic queue channels
Sure, here it goes. PLEASE
READ THE DISCLAIMER BELOW ;-)
This is my first true
patch to asterisk, no money back guarantee. Please backup all your hard disk
before applying it !!! (just kidding )
PauloHM
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of TC
Sent: June
26, 2003 5:02 PM
To:
[EMAIL PROTECTED]
Subject: Re: [Asterisk-Users]
dynamic queue channels
Could
you post to the list,so we could take look just in case mark has a
plate load of items to merge ??
-Original Message-
From: Benjamin Miller [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
[EMAIL PROTECTED]
Date: June 26, 2003 11:45 AM
Subject: RE: [Asterisk-Users]
dynamic queue channels
Nice
work! :-) Thanks
Cant
wait to see it in cvs.
-Original Message-
From: Paulo Mannheimer [mailto:[EMAIL PROTECTED]]
Sent: Thursday, June 26, 2003
11:23 AM
To:
[EMAIL PROTECTED]
Subject: RE: [Asterisk-Users]
dynamic queue channels
Just
posted a patch do Mark implementing this.
There
are two new commands:
- AddQueueMember(queuename[|interface])
- RemoveQueueMember(queuename[|interface])
An
example would be
AddQueueMember(techsupport|Zap/3-1)
Hope you
find it useful
PauloHM
-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Benjamin Miller
Sent:
June 24, 2003 11:41 AM
To:
[EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] dynamic
queue channels
There
may be some trickiness that can be done with chan_local
asagents of the call queue. However, a much more elegant way to do
this would be to create an app_addagent and app_removeagent that allows the
dynamic addition and removal of extensions from the agent pool for a given
queue. addagent(${CHANNEL}, techsupport) or something like that.
Ben
-Original Message-
From: Paulo Mannheimer
[mailto:[EMAIL PROTECTED]
Sent:
Monday, June 23, 2003 6:36 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] dynamic
queue channels
Hi, Im trying to build a call
center application that allows attendants to come in the morning and dial a
certain extension to make their extension available.
I wouldnt like to use the
AgentLogin app because their line would need to stay off-hook (is this correct?)
Is there any SET channel status
command that would allow me to do something like this?
PauloHM
RE: [Asterisk-Users] Web interface for Asterisk
I think it's a good start, and would be willing to work on expanding the concept. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dylan VanHerpen Sent: June 26, 2003 6:29 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Web interface for Asterisk Hi everybody, I've been tinkering with a web based interface for Asterisk. I tried to stick as closely to the current configuration format as possible. The web interface should help to do things a little easier (sort by extension, context, do bulk changes). www.packetbell.com/asterisk Feedback appreciated! Dylan. ___ 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 termination in the US
Hi, Can someone provide information about IAX termination in the US and other countries? I tried Google but nothing showed up ;-( PauloHM
[Asterisk-Users] dynamic queue channels
Hi, Im trying to build a call center application that allows attendants to come in the morning and dial a certain extension to make their extension available. I wouldnt like to use the AgentLogin app because their line would need to stay off-hook (is this correct?) Is there any SET channel status command that would allow me to do something like this? PauloHM
[Asterisk-Users] queue application
Hi, Im working on a call center application where callers input some information and get transferred to an attendant, or waits in a queue until one is available. The operator is using a PC-based system that needs to have access to the information previously input by the caller. I was thinking about making * write some control info somewhere and then make the application get it through samba/file sharing. Any other insights? Also, how to make this work if the call is queued? Best regards, PHM
RE: [Asterisk-Users] lost variables
Sorry, my mistake. The point was that I had a message playing with Background() and a couple of Setvar() after it. As I started to dial an extension before the message had finished, the setvar() calls didn't get invoked. PHM -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Martin Pycko Sent: June 11, 2003 3:51 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] lost variables Why do you think so? Local variables get lost only when the call gets hanged up. Martin On Wed, 11 Jun 2003, Paulo Mannheimer wrote: Hi, Seems that my local variable content get lost when I call an AGI program. Is this the correct functionality? Thanks, Paulo H. Mannheimer ___ 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] lost variables
Hi, Seems that my local variable content get lost when I call an AGI program. Is this the correct functionality? Thanks, Paulo H. Mannheimer
[Asterisk-Users] answering calls with SIP phones
Hi, I have an incoming call that I would like answered every time by a different SIP phone (out of 50). Also, some of the phone may not be available (may be turned off and thus unregistered with Asterisk). Any way of doing this? Paulo H. Mannheimer
RE: [Asterisk-Users] answering calls with SIP phones
Thanks, very good insights. The proposed method has a single flaw - it's very difficult to detect that all SIP channels are busy, and thus queue the call. It's a petty that SIP does not support call groups, it would make it automatic. Best, PHM -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: June 05, 2003 2:25 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] answering calls with SIP phones On Thu, 5 Jun 2003, Paulo Mannheimer wrote: I have an incoming call that I would like answered every time by a different SIP phone (out of 50). hmm... pass the call through an AGI first, that picks a random number and then pass the call to that SIP phone number Also, some of the phone may not be available (may be turned off and thus unregistered with Asterisk). set the unavailable priority to go back to the AGI for another try Any way of doing this? i'm sure there are more, and intelligent ways of doing this but if you want a kludge, this'll work - wasim ___ 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] Asterisk localization
Hi All, Ive been working with asterisk for about two months, and I would like to contribute to the project on the localization side, mostly making it easier to translate text output and pre-recorded messages. My goal is to discuss with you guys a framework for localization/translation, and then slowly start to implement it. I believe its important to discuss this before any code is programmed because it can become a pain in the neck to program new things and support old ones if its not implemented in a flexible and easy way. Any thoughts? Paulo H. Mannheimer
