[asterisk-users] sox gsm
sox needs for gsm an optional library. I was not able to locate this one. Can anybody point me to this place? bye Ronald ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Asterisk Jobs Update
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... For more information or to start looking for Open source Asterisk VOIP employment head over to http://www.asterisk-jobs.com Is it that I don't know how to make search or there is no jobs available in any country? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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] Challenging problem regarding CallerID in TDM 04B (Trying to solve since 8 days)
Dear Leo,As you said, I have tried using dtmf and in different values. But, no reuslt. Finally, I knew that Basic Asterisk setup doesn't recognize callerid in India. To get callerid in India, we have to do some modifications in chan_zap.c source file. But, I dont know what modifications I have to do? Do you have any Idea about these modifications in source code? Can you please tell me. Looking forward to your response. Thank you.Regards,Chandra.Leo Ann Boon [EMAIL PROTECTED] wrote: Crazy Boy wrote: Hi Leo, Thank you for your quick response. In Internet, I came to know that 1) In India, we have to give dtmf and ring for cidsignallling and cidstart respectively.Have you tried settingcidsignalling=dtmf 2) Default Asterisk setup doesn't
recognise callerid in India. To recognize callerid in India, we have to do or change some modifications in chan_zap.c source file. Is it right?If cidsignalling=dtmf won't work then you might have to consider invasive surgery on chan_zap. :) 3) Please open the below link and see the values for India. http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf The zaptel driver has tone definition for india. In /etc/zaptel.conf:loadzone=indefaultzone=in Here I am giving the error messages on Asterisk console. *CLI -- Starting simple switch on 'Zap/1-1' Aug 18 14:53:13 ERROR[15499]: callerid.c:276 callerid_feed: fsk_serie made mylen 0 (-16) Aug 18 14:53:13 WARNING[15499]: chan_zap.c:6087 ss_thread: CallerID feed failed: Success Aug 18 14:53:13 WARNING[15499]: chan_zap.c:6131 ss_thread: CallerID returned with
error on channel 'Zap/1-1' -- Executing Wait("Zap/1-1", "10") in new stack -- Executing Answer("Zap/1-1", "") in new stack -- Executing NoOp("Zap/1-1", " 18082006-14:53:24") in new stack -- Executing NoOp("Zap/1-1", "CallerID is ") in new stack -- Executing NoOp("Zap/1-1", "CallerID Name is ") in new stack -- Executing NoOp("Zap/1-1", "CallerID Number is ") in new stack -- Executing SetMusicOnHold("Zap/1-1", "default") in new stack -- Executing Set("Zap/1-1", "TIMEOUT(digit)=5") in new stack -- Digit timeout set to 5 -- Executing Set("Zap/1-1", "TIMEOUT(response)=10") in new stack -- Response timeout set to 10 -- Executing BackGround("Zap/1-1", "/tmp/virg2") in new stack -- Playing '/tmp/virg2' (language 'en') == CDR updated on Zap/1-1 -- Executing Dial("Zap/1-1", "SIP/105|15|t|12") in new
stack -- Called 105 -- SIP/105-00798410 is ringing -- Nobody picked up in 15000 ms -- Executing VoiceMail("Zap/1-1", "u105") in new stack -- Playing 'vm-theperson' (language 'en') -- Playing 'digits/1' (language 'en') -- Playing 'digits/0' (language 'en') -- Playing 'digits/5' (language 'en') -- Playing 'vm-isunavail' (language 'en') -- Playing 'vm-intro' (language 'en') == Spawn extension (incoming, 105, 2) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' Please tell me. Looking forward to your response. Thank you. Regards, Chandra. *//*___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:
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Re: [asterisk-users] sox gsm
On Mon, Aug 21, 2006 at 02:02:31PM +0800, Ronald Wiplinger wrote: sox needs for gsm an optional library. I was not able to locate this one. Can anybody point me to this place? As there is a Debian package you can grab the orig tarball from: http://packages.debian.org/unstable/libs/libgsm1 And the original location is listed in the copyright file: http://packages.debian.org/changelogs/pool/main/libg/libgsm/libgsm_1.0.10-13/libgsm1.copyright -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] no audio issue
Hi im experiencing no audio problems. ive installed the latest asterisk 1.2.10 zaptel, libpri asterisk. the caller's side reception is fine but i hear nothing on my sip account. Please help Regards, John ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] queuememberstatus overwhelms manager socket connection to asterisk
I have a test application, what it does is just connect to the asterisk manager,and listen for events. I also set the connection to receive on user, call and agent events.I Noticed that everytime the queue is empty, asterisk tends to throw too many queuememberstatus events, overwhelming the connection and therefore closesit abruptly.I also got the same result when I used telnet to connect to asterisk, and then make a call which is then forwarded to an empty queue. I'm using version 1.2.7.1 and even if the queue has its eventwhencalled set to no, the problem still persists. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: no audio issue
there's actually no audio b/w sip to sip calls. I just tried 2 sip extensions and there was no audio in any of them. what could be wrong? nat = 1 i used ulaw and then gsm and one of them worked. im not using qualify. i have tried everything i could think of, even applied the patch mydiff.txt in the src directory and did the make clean; make install. John Quoting [EMAIL PROTECTED]: Hi im experiencing no audio problems. ive installed the latest asterisk 1.2.10 zaptel, libpri asterisk. the caller's side reception is fine but i hear nothing on my sip account. Please help Regards, John ___ --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] Asterisk installations in Germany
asterisk-robert wrote: I need to send some information to our German HQ regarding my experiences with VoIP. Asterisk is very prominent in those experiences. I would like to include information about installations of Asterisk at German companies/universities. We have installed Asterisk in a multi-site small business environment in Hamburg. We started out with HT-286 ATA, went to different ATAs, and ended up with SNOM 360s. Usage is light with a single queue. Connection between sites is via IAX. No central dial plan, just plain extensions, ie. Site 1 has extension 2x, Site 2 has extension 4x, and Site 3 has extensions 5x. Connection to the PSTN via chan_capi-cm (AVM C4) and bristuffed asterisk. Each site has its own PSTN connection, which can be used from the other sites via prefix. Make sure to get quality PCs, speed is not important, but I have found out some old Compaqs had weird problems, which went away by going to a new PC. Old Dell is working fine. HTH -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Is there an [EMAIL PROTECTED] specific list?
Thanks in advance Paul ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX2 Auto fallthrough
Hi all,Could anyone help me, why my calls of some clients disconnecting with the following error message: i have more than 500 IAX users but it is happening with very few customers.I will be appricate for u kind of help.Error::Auto fallthrough, channel 'IAX2/2001@2001/1' status is 'UNKNOWN' Stay in the know. Pulse on the new Yahoo.com. Check it out. ___ --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] Polycom 601 Issues
On 20/08/2006, at 8:38 PM, Paul Hales wrote: Does anything pop up on the Asterisk screen? Does music on hold work fine? PaulH On Fri, 2006-08-18 at 13:13 +0800, Nathan Alberti wrote: Nothing strange on the asterisk console... just stopped and started hold on channel. If I repeatedly take a call on and off hold sometimes it will work, other times it they will hear distorted hold music, other times they will hear silence, the same thing happens with voice. Driving me nuts :) I have tried; New firmware on the Polycom New IOS on the router 2 x New Switches Next is Asterisk version. Nathan. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP ActiveX?
Hi, I'm looking for a SIP ActiveX component to use in Visual Basic/Delphi. Thanks Kind Regards, Lennie De Villiers ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: no audio issue ([EMAIL PROTECTED])
U need 2 give more info on your setup i.e wherether u have sipclientasterisknatsipclient or whatever the situation is . Anyway in the mean time just rtp dubug on and see wherether there r rtp packets sent back and forth there's actually no audio b/w sip to sip calls. I just tried 2 sip extensions and there was no audio in any of them. what could be wrong? nat = 1 i used ulaw and then gsm and one of them worked. im not using qualify. i have tried everything i could think of, even applied the patch mydiff.txt in the src directory and did the make clean; make install. John Quoting [EMAIL PROTECTED]: Hi im experiencing no audio problems. ive installed the latest asterisk 1.2.10 zaptel, libpri asterisk. the caller's side reception is fine but i hear nothing on my sip account. Please help Regards, John ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Asterisk 'Hosting'
JM == Jeremy McNamara [EMAIL PROTECTED] writes: JM Why do you need multiple instances? Just setup your Asterisk JM configuration to separate the various 'customers' or 'tenants'. The configuration files balloon to unmanageable sizes, and changing them means that you risk breaking telephony for all customers -- not just the customer you were trying to please with the change. There is also the lovely little callgroup/pickupgroup limit of 64. We run our customer PBX's in linux-vserver. It works quite well. /Benny ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] zap channel media volume
Hi all, we do have the following configuration (non-Asterisk PBX) - T1 - ZAP (Asterisk PBX) - ZAP - T1 - (GSM Gateway) - GSM Enduser The call is originated on the (non-Asterisk PBX) - gets send over a T1 connection to the asterisk server (which does least cost routing) - the asterisk server then does send the call over a GSM Gateway into the world... The Problem we do have is - that the Users behind the non-Asterisk PBX are complaining about low volume media if the the calling through the gateway (if the are calling mobiles...). So i have started to raise the rxgain value for the connection between the asterisk box and the GSM Gateway, this does work quite well - but not really perfect. The ringback (not locally generated - does come from the GSM Provider) does get terrible loud - as soon as the callee is connected - the speech is nearly not hearable because it has such a low volume. The ringback is EARLY MEDIA - if i am right - and the speech is normal MEDIA. So, is it possible to set different gains for EARLY MEDIA and normal MEDIA ? Does anyone else have had this problem ? regards, Wolfgang ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Asterisk 'Hosting'
MR == Matt Riddell (NZ) [EMAIL PROTECTED] writes: MR And so you're thinking it would be better to run several hundred MR Asterisk instances?! Why not? As long as you stay away from the things that need zap timing, asterisk is really not much of a load. /Benny ___ --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] Re: PRI gurus - Does any one know the meaningn of span 1 received AOC-E charging 149502040 units
Tomislav Parčina wrote: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Hi Marco, as good? Well, you are use libpri-1.2.3? Believe that this is a bug of this version. Look at link´s below, contains patchs for this problem. I wait to have helped. Best Regards Josué http://bugs.digium.com/file_download.php?file_id=7499type=bug http://bugs.digium.com/file_download.php?file_id=11047type=bug Hi Josue! Does this patch enables AOC support? I'm looking for solution how to use AOC information's that my provider is sending to me. I would appreciate any info on this one. Hi! This patch does passthrough the AOC information from on ZAP channel to another ZAP channel. There is no support yet for storing the AOC value as CDR, but I think this may be easily added. regards klaus ___ --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] Ignoring PRI call?
So, a few questions: - If the call received by asterisk from the PRI is sent to a number not in the dialplan, what will asterisk do? Will the call be cancelled, or will asterisk signal something back to the switch to indicate dunno about this, try another? Asterisk will do whatever you tell it to do. Here is a hint of what you can do before you reject a number: http://www.voip-info.org/wiki/index.php?page=Asterisk+variable +PRI_CAUSE Sure, but I can't find a cause code that means 'try next link'. do you know if there is one, and what this is? What do you mean by try next link? Are you telling me that you have 2 PRIs connected to 2 different asterisk servers, and as far as the provider is concerned they both serve the same DID block, but you have them split up as far as what DID should go to what box? If the answer is yes, then you are doing it wrong. This is what I am trying to do, yes, as to do all DID administration myself without contacting the switch monkey. It's quite possible, it seems, by sending a cause 34, lying about no bchans being available to handle the call. roy -- Roy Sigurd Karlsbakk [EMAIL PROTECTED] (+47) 98013356 --- In space, loud sounds, like explosions, are even louder because there is no air to get in the way. ___ --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] SIP ActiveX?
Lennie De Villiers wrote: Hi, I'm looking for a SIP ActiveX component to use in Visual Basic/Delphi. Thanks You can find a proof of concept at http://www.pernau.at/kd/voip/bookmarks-sip-phones.html It's called ActXPhone regards klaus ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Zaptel install - Fedora Core 5
I'm trying to install Zaptel 1.2.7 on Fedora Core 5 with 2.6.17-1.2174_FC5 kernel from source code. When I untar Zaptel and execute this is error that I get. cc -o ztmonitor ztmonitor.o cc -o ztspeed.o -c ztspeed.c cc -o ztspeed ztspeed.o cc -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTE L_CONFIG=\/etc/zaptel.conf\zttest.c -o zttest cc -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTE L_CONFIG=\/etc/zaptel.conf\ -c -o fxotune.o fxotune.c cc -o fxotune fxotune.o -lm /lib/modules/2.6.17-1.2174_FC5/build You do not appear to have the sources for the 2.6.17-1.2174_FC5 kernel installed . make: *** [linux26] Error 1 What could be the problem? How to solve it? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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] Zaptel install - Fedora Core 5
I managed to get zaptel to compile reasonably easily on 2.6.17-1.2157_FC5smp. However, the yum repo sites do not provide devel packages for 2.6.17-2174 for some reason last time I checked, hence couldn't get it to build on that kernel. You could probably create the devel package without too much trouble from the srpm, but it's a lot easier to stick to 2157. If anyone else have managed to get FC5 to install the correct devel packages for the latest kernel, please let me know! Simon On 21 Aug 2006, at 11:52, Tomislav Parčina wrote: I'm trying to install Zaptel 1.2.7 on Fedora Core 5 with 2.6.17-1.2174_FC5 kernel from source code. When I untar Zaptel and execute this is error that I get. cc -o ztmonitor ztmonitor.o cc -o ztspeed.o -c ztspeed.c cc -o ztspeed ztspeed.o cc -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE- DSTANDALONE_ZAPATA -DZAPTE L_CONFIG=\/etc/zaptel.conf\zttest.c -o zttest cc -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE- DSTANDALONE_ZAPATA -DZAPTE L_CONFIG=\/etc/zaptel.conf\ -c -o fxotune.o fxotune.c cc -o fxotune fxotune.o -lm /lib/modules/2.6.17-1.2174_FC5/build You do not appear to have the sources for the 2.6.17-1.2174_FC5 kernel installed . make: *** [linux26] Error 1 What could be the problem? How to solve it? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Re: PRI gurus - Does any one know the meaningn of span 1 received AOC-E charging 149502040 units
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Hi! This patch does passthrough the AOC information from on ZAP channel to another ZAP channel. There is no support yet for storing the AOC value as CDR, but I think this may be easily added. Hi Klaus! I'm not programmer so I don't know how easy is to add ability to store AOC in database, but I'm sure that more than few asterisk users will benefit greatly from it. Do you know has anybody tried to do something like that? I'm willing to help, but as I have mentioned I'm not programmer. I could do testing and write documentation... -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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] Is there an [EMAIL PROTECTED] specific list?
Yeh but as [EMAIL PROTECTED] is now called Trixbox so go to www.trixbox.com Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Paul A Brown Sent: Monday, 21 August 2006 3:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Is there an [EMAIL PROTECTED] specific list? Thanks in advance Paul ___ --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 ___ --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] Announce caller-id
I did something along these lines, but I was playing the caller ID back to the caller, not after a transfer. In a perl AGI script. I split the caller ID number into an array, seperated by '//' so each number was an element. Then I played digits/$array[0]... digits/$array[1]...etc. coolbreeze wrote: I would like to transfer an incoming call and, when the call is answered, have the caller id of the call spoken when the call is answered on my cell phone. Any tips greatly appreciated AAH 2.7 using sip trunks exclusively. Regards ___ --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 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] running agi application in the background
I would like to run a fast-agi application in the background.(cmd agi()) This is because I would like to implement a disconnect after so many seconds feature or at least a log of the duration of the call. When the call is answered, the application checks to see the number of seconds (talk time)remaining then disconnects the call if the time is exceeded. How can I achieve this. Allan. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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] Zaptel install - Fedora Core 5
I couldn't find 2.6.17-1 for download but this is what I used to install the kernel source http://download.fedora.redhat.com/pub/fedora/linux/core/5/source/SRPMS/GFS-kernel-2.6.15.1-5.FC5.17.src.rpm Dennis Clark DENPRO WRK 207.618.1998 CEL 443.415.0527 FAX 1.888.811.8809 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of simon elliston ball Sent: Monday, August 21, 2006 7:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Zaptel install - Fedora Core 5 I managed to get zaptel to compile reasonably easily on 2.6.17-1.2157_FC5smp. However, the yum repo sites do not provide devel packages for 2.6.17-2174 for some reason last time I checked, hence couldn't get it to build on that kernel. You could probably create the devel package without too much trouble from the srpm, but it's a lot easier to stick to 2157. If anyone else have managed to get FC5 to install the correct devel packages for the latest kernel, please let me know! Simon On 21 Aug 2006, at 11:52, Tomislav Parčina wrote: I'm trying to install Zaptel 1.2.7 on Fedora Core 5 with 2.6.17-1.2174_FC5 kernel from source code. When I untar Zaptel and execute this is error that I get. cc -o ztmonitor ztmonitor.o cc -o ztspeed.o -c ztspeed.c cc -o ztspeed ztspeed.o cc -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE- DSTANDALONE_ZAPATA -DZAPTE L_CONFIG=\/etc/zaptel.conf\zttest.c -o zttest cc -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE- DSTANDALONE_ZAPATA -DZAPTE L_CONFIG=\/etc/zaptel.conf\ -c -o fxotune.o fxotune.c cc -o fxotune fxotune.o -lm /lib/modules/2.6.17-1.2174_FC5/build You do not appear to have the sources for the 2.6.17-1.2174_FC5 kernel installed . make: *** [linux26] Error 1 What could be the problem? How to solve it? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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 ___ --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 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Zaptel install - Fedora Core 5
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... I managed to get zaptel to compile reasonably easily on 2.6.17-1.2157_FC5smp. However, the yum repo sites do not provide devel packages for 2.6.17-2174 for some reason last time I checked, hence couldn't get it to build on that kernel. You could probably create the devel package without too much trouble from the srpm, but it's a lot easier to stick to 2157. Hi Simon! I have to use 2.6.17-1.2157 because I have precompiled vt1211 chip (sensors for VIA motherboards) driver for that kernel. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] how to set 'transfercapability'
hi, we've got a Diva Server BRI-2M PCI, SN:3485 card in our asterisk server. we use the latest divas4linux-melware-3.0.g-106.628.1-1 driver for it. the card is connected to a Bosch Integral33 PBX. the two system connected with an S0 line in order the two pbx be able to call each other. when we call from the bosch to asterisk everything is working properly. but when we call from the a x-ten soft phone client through asterisk to the bosch the it's not working. which means the asterisk pass the call to the bosch, bosch receive but don't ring the given number. after we debug the capi layer with bosch experts from bosch we found the while the bosch call asterisk it request SPEECH line bearer, but when asterisk call bosch it set bearer to MULTIUSE. i found it in divactrl/common/dbg_tapi.c LINE_BEARER_MODE__SPEECH, LINE_BEARER_MODE__MULTIUSE. so probably the problem is thet we (x-ten, asterisk, divas4linux) do not set the bearer (transfercapability) to proper value. is this the real reason? how can i set the bearer/transfercapability to speech in divas4linux or in capi or in asterisk's capi or ...? why the system do not recognize the problem? why x-ten soft phone do not ask for speech mode or why asterisk do not set the transfercapability to speech when it get a call from a soft phone? or what else can we do;-)? thank you for your help in advance. yours. -- Levente Si vis pacem para bellum! ___ --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] Re: Zaptel install - Fedora Core 5
In which case your best bet is probably to install with an rpm -- rebuilt on the source rpm. simon On 21 Aug 2006, at 12:36, Tomislav Parčina wrote: In article 344F8B3D-6591-4001-9DE6- [EMAIL PROTECTED], [EMAIL PROTECTED] says... I managed to get zaptel to compile reasonably easily on 2.6.17-1.2157_FC5smp. However, the yum repo sites do not provide devel packages for 2.6.17-2174 for some reason last time I checked, hence couldn't get it to build on that kernel. You could probably create the devel package without too much trouble from the srpm, but it's a lot easier to stick to 2157. Hi Simon! I have to use 2.6.17-1.2157 because I have precompiled vt1211 chip (sensors for VIA motherboards) driver for that kernel. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] polycom_acd_functions branch and outboundproxy
Hi, I'm using the polycom branch and have been trying to get the outboundproxy=xxx to work. Is this something that should work in the version of software? Thanks, Dean. ___ --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] Ignoring PRI call?
On Sunday 20 August 2006 10:55, Roy Sigurd Karlsbakk wrote: - If the call received by asterisk from the PRI is sent to a number not in the dialplan, what will asterisk do? Will the call be cancelled, or will asterisk signal something back to the switch to indicate dunno about this, try another? If the call comes in to a context and the DID does not match, Asterisk rejects the call outright. I had a patch submitted a LONG time ago (I don't see it in the tracker anymore) which allowed chan_zap to jump to the context 'i' extension if no DID matched, but Mark didn't like the idea at the time. (I think he may have something against consistency, heh) - If the call is received by Asterisk from a wildchar extensions like exten = _X.,1,..., will it be possible to signal the switch dunno about this, try another later in the chain? Since Asterisk's current dialplan implementation does not scale wery well, we're doing sip user/callerid lookups in mysql from an AGI script and the MYSQL app from -addons. Anyway, this means we'll accept all incoming calls before handling them (not answering, though, just accepting them to the dialplan). _X. will match any DID, so the dialplan logic will consider it matched. This seems logical. If you want to match something else, make the dialplan do it, or revive the 'jump to i' issue I brought up so long ago. :-) -A. ___ --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] running agi application in the background
Allan Kamau schrieb: I would like to run a fast-agi application in the background.(cmd agi()) This is because I would like to implement a disconnect after so many seconds feature or at least a log of the duration of the call. What about using an Option of the Dial-App instead ?? S(n): Hangup the call n seconds AFTER called party picks up. http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Dial ___ --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] Sending signals to asterisk
On Sun, Aug 20, 2006 at 06:20:42PM -0300, Danko Miocevic wrote: Hello, is there any way to send signals to asterisk, for example, I send a sign to a parallel port and it calls an extension. I can´t modify asterisk code to make it. Any ideas? Thanks for your time, Danko BTW: try using a serial port instead. While it is technically feasible to use the parallel port, there aren't really standard ways of connecting over parallel (except plip). -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Choppy sound zap-to-sip, but not sip-to-sip?
Hi, I have lately noticed that we sometimes get choppy sound when recieving calls from the PSTN (on a TE410P-card) that get sent to an external SIP extension (over the internet) who has a somewhat bad connection. The strange thing is that it still sounds good when calling internally to the SIP-to-SIP. Is there any simple answer to why Zap-to-SIP (external) sounds bad when there is a bad connection, but SIP-to-SIP doesn't? The problem (I think) is not with the card or drivers since the problem only occurs when the connection is bad and never on our phones that are on the same internal network with the server. Thanks! Regards, Jan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Joining calls via manager.api or AGI
Is there a way to initiate 2 different calls and connect them together with Asterisk, using the manager.api or the AGI system? I want to link the calls without using DTMF, such as with an SMS or web triggered script. I thought the call files would be able to set the necessary AGI variables for the outbound leg but the AGI variables do not include the DNID equivalent. Any ideas? /Obelix ___ --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] Joining calls via manager.api or AGI
Is there a way to initiate 2 different calls and connect them together with Asterisk, using the manager.api or the AGI system? I want to link the calls without using DTMF, such as with an SMS or web triggered script. The only way right now is using meetme. There is a patch with a 'bridge' function but is marked as post 1.4 ( http://bugs.digium.com/view.php?id=5841 ). This is a very much needed feature. -- Nicolás Gudiño Buenos Aires - Argentina ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Configure mailserver to deliver voicemail
G'Day List, I am looking for documentation on how to configure sendmail to deliver asterisk voicemails to the recipient's mailbox. I Googled it but found many many references to the fact that asterisk can do that but no How-To's. I believe sendmail is running on my asterisk box as: [EMAIL PROTECTED] /] # mail returns... Mail version 8.1 6/6/93 Also, my voicemail.conf is already configured. Thanks ___ --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] Joining calls via manager.api or AGI
This feature was supposed to be in 1.2, in fact Kevin promised me that it would be since I had it in before the feature freeze for 1.2. It did not go in. Since then I have had to move on to other things and others have tried to keep it going. This is really a very basic function that should be in the Core of Asterisk, it just isn't a priority for the development team so it keeps getting pushed aside and it looks like for 1.4 it has yet again. I have been using this feature in production on many servers for well over a year and it works great. Hopefully we'll see it added to Asterisk core some day. MATT--- On 8/21/06, Nicolás Gudiño [EMAIL PROTECTED] wrote: Is there a way to initiate 2 different calls and connect them together with Asterisk, using the manager.api or the AGI system? I want to link the calls without using DTMF, such as with an SMS or web triggered script. The only way right now is using meetme. There is a patch with a 'bridge' function but is marked as post 1.4 ( http://bugs.digium.com/view.php?id=5841 ). This is a very much needed feature. -- Nicolás Gudiño Buenos Aires - Argentina ___ --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 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX2 TRUNK CPU consumption
Hi, I have a strange problem about the cpu consumption of a IAX trunk. I have two asterisk connected by a IAX trunk. The asterisk number 1 is installed on a Soekris Box Asterisk 1 Asterisk 2 IAX T | -- | I use another asterisk to generate some traffic Asterisk 0* 1* 2 SIP IAX T | - | - | I see that when on the * n. 2 the console prints this message the cpu consumption is low: -- Hungup 'IAX2/test-16' -- Accepting AUTHENTICATED call from 192.168.x.xxx: requested format = ulaw, requested prefs = (unknown), actual format = ulaw, host prefs = (ulaw), priority = mine But when the console prints this message the cpu consumption is high: requested prefs = (unknown|unknown|jpeg|lpc10|ulaw|unknown) MOST IMPORTANT: the two different messages appears with the same call generator, and I not understand why the same situation create different messages? Any idea to resolve the problem? Thanks Matteo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Is it possible to call System dialplan application via AMI?
Hi guys, Does anyone know whether is it possible to call System (Execute a system Linux shell command) dialplan application via AMI? If so, how? Thanks in advance, * ___ --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] Configure mailserver to deliver voicemail
On Mon, Aug 21, 2006 at 09:52:23AM -0400, Ferguson, Michael wrote: G'Day List, I am looking for documentation on how to configure sendmail to deliver asterisk voicemails to the recipient's mailbox. Nothing special about sendmail. Basically any standard MTA: sendmail, postfix, or whatever. Follow you distro's documentation on configuring an MTA. You should be able to send mail from the command line: echo test | mailx -s helllo there [EMAIL PROTECTED] -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] t.38 bounty
hi, bounty for t.38 is $11,750. that looks good! http://www.voip-info.org/wiki/view/Asterisk+T.38+Bounty how high must be bounty for Digium to hire programmer for this? thanks --- Marek Cervenka === ___ --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
SV: [asterisk-users] Choppy sound zap-to-sip, but not sip-to-sip?
Sorry. It sould say SIP-to-Zap not the other way around. Meaning that the Zap user is heard fine, but the external-SIP user is choppy when calling out on Zap (not when calling SIP-to-SIP though). -Ursprungligt meddelande- Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För [EMAIL PROTECTED] Skickat: den 21 augusti 2006 15:15 Till: asterisk-users@lists.digium.com Ämne: [asterisk-users] Choppy sound zap-to-sip, but not sip-to-sip? Hi, I have lately noticed that we sometimes get choppy sound when recieving calls from the PSTN (on a TE410P-card) that get sent to an external SIP extension (over the internet) who has a somewhat bad connection. The strange thing is that it still sounds good when calling internally to the SIP-to-SIP. Is there any simple answer to why Zap-to-SIP (external) sounds bad when there is a bad connection, but SIP-to-SIP doesn't? The problem (I think) is not with the card or drivers since the problem only occurs when the connection is bad and never on our phones that are on the same internal network with the server. Thanks! Regards, Jan ___ --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 ___ --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] Apache for FastAGI
On 8/18/06, Shidan [EMAIL PROTECTED] wrote: I don't know if I responded to the original poster before but if you are looking for a python fastAGI server, there already is one, its called starpy. Anders, since you know Erlang, do you know of any media processig libraries in Erlang, do the ericsson softswitches do the media processing themselves? First off, I don't work for Ericsson, but it is my impression that normally the call control is done in Erlang and the media processing is done in either hardware or C. I You really are interested I recommend You to ask on [EMAIL PROTECTED] /Anders ___ --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] Ignoring PRI call?
This is what I am trying to do, yes, as to do all DID administration myself without contacting the switch monkey. It's quite possible, it seems, by sending a cause 34, lying about no bchans being available to handle the call. Thanks for reporting back, I like this idea :) thanks again. ___ --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] Analog-to-VoIP: blade?
www.zhone.com. Their MALC can handle 500 POTS lines in a 23 shelf with POTS - VoIP (SIP/MGCP). 'Telco quality' and the per port cost for high density isn't that bad. You could probably also go with a bunch of CAC AccesBanks connected to a CAC Widebank, connected to a Lucent TNT and get 672 DS0's - g. 729 VoIP On Aug 20, 2006, at 4:55 PM, Ken D'Ambrosio wrote: I've seen analog-to-VoIP gateways such as the Audiocodes one -- which, truthfully, looks very, very nice -- but I've got several hundreds of analog phones to deal with, and I was wondering if anyone has seen something with even higher concentrations than the Audiocodes 24-ports-per-rack-unit. Thanks for any suggestions! -Ken ___ --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 -- Matthew S. Crocker Vice President Crocker Communications, Inc. Internet Division PO BOX 710 Greenfield, MA 01302-0710 http://www.crocker.com ___ --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] Re: Asterisk 'Hosting'
-Original Message- From: Benny Amorsen [mailto:[EMAIL PROTECTED] Sent: Monday, August 21, 2006 3:39 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Re: Asterisk 'Hosting' JM == Jeremy McNamara [EMAIL PROTECTED] writes: JM Why do you need multiple instances? Just setup your Asterisk JM configuration to separate the various 'customers' or 'tenants'. The configuration files balloon to unmanageable sizes, and changing them means that you risk breaking telephony for all customers -- not just the customer you were trying to please with the change. There is also the lovely little callgroup/pickupgroup limit of 64. We run our customer PBX's in linux-vserver. It works quite well. Awesome. How many instances are you running on a single system? Doug. ___ --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] SIP ActiveX?
This is a commercial activex you may want to evaluate:http:/.www.vaxvoip.comIt worksLennie De Villiers [EMAIL PROTECTED] wrote: Hi,I'm looking for a SIP ActiveX component to use in Visual Basic/Delphi.ThanksKind Regards,Lennie De Villiers ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users Elpidio Ramos PresidentRM International ServicesSA CV Web: http://www.ramosoft.com Mex: +52 (55) 5116-9804 Office +52 (55) 5116-9805 Fax+52 (55)1755-6601 CellUSA: +1 (801) 494-1415 Office +1 (240) 250-8264 Fax +1 (801) 938-4740Direct ___ --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] t.38 bounty
marek cervenka wrote: hi, bounty for t.38 is $11,750. that looks good! http://www.voip-info.org/wiki/view/Asterisk+T.38+Bounty how high must be bounty for Digium to hire programmer for this? thanks Do you really think T.38 can be implemented on a contract basis for $11,750? Besides, these bounties are rarely paid. Most of those pledges are quite old, and I really wouldn't expect them to be paid. Digium has no interest in implementing T.38. They have actually been quite obstructive. Regards, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DTMF + voipjet
Hello list, Was wondering if anyone knows how to get DTMF to work on voipjet.. Tried, dtmf=rfc2833 dtmfmode=rfc2833 doesn't seem to work... Any clues? Cheers! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cancelling outbound call: is Asterisk behaving correctly
Hi, we have a setup with an Asterisk, an openser and a Cisco 5400 in place. Asterisk is the frontend to the users, providing registering and RTP proxy functionality and openser is the gate-keeper of the Cisco. I can call in and out, everything is fine so far. But there is one strange fact: when I place a outbound call (from a VoIP phone to the asterisk, from the asterisk to the openser, from the openser to the Cisco, from the Cisco into the PSTN) and cancel this call by hanging up the VoIP phone before the call was establish, I can see the correct message flow for a cancelation (Asterisk says CANCEL to openser, openser says 200 cancelling to Asterisk, openser talks to the Cisco, finally openser says 487 Request cancelled to Asterisk and Asterisk says ACK). So far so fine. But then, Asterisk says CANCEL again. With the same call-id, the same tags, just completely the same CANCEL as before. Why? I was under the impression, that it has correctly identified the messages from openser, since it has answered with ACK. I've already compared this message-flow to the one of the other direction (inbound call, cancelled from PSTN phone), which correctly ends with the ACK. I can not find any significant difference. I've attached on ngrep trace. 212.153.11.54 is asterisk, 212.153.11.19 is openser and 146.188.127.31 is the Cisco. Can anyone give me any hint? Thanks, Wolfgang interface: hme0 (212.153.111.0/255.255.255.192) filter: (ip) and ( port 5060 ) # U 212.153.111.54:5060 - 212.153.111.19:5060 INVITE sip:[EMAIL PROTECTED] SIP/2.0. Via: SIP/2.0/UDP 212.153.111.54:5060;branch=z9hG4bK170b;rport. From: 49694525 sip:[EMAIL PROTECTED];tag=as3898. To: sip:[EMAIL PROTECTED]. Contact: sip:[EMAIL PROTECTED]. Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE. User-Agent: Asterisk PBX. Max-Forwards: 70. Remote-Party-ID: 49694525 sip:[EMAIL PROTECTED];privacy=off;screen=no. Date: Mon, 21 Aug 2006 12:06:56 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Content-Type: application/sdp. Content-Length: 211. . v=0. o=root 26661 26661 IN IP4 212.153.111.54. s=session. c=IN IP4 212.153.111.54. t=0 0. m=audio 42670 RTP/AVP 0 3 8. a=rtpmap:0 PCMU/8000. a=rtpmap:3 GSM/8000. a=rtpmap:8 PCMA/8000. a=silenceSupp:off - - - -. # U 212.153.111.19:5060 - 212.153.111.54:5060 SIP/2.0 100 trying fast. Via: SIP/2.0/UDP 212.153.111.54:5060;branch=z9hG4bK170b;rport=5060. From: 49694525 sip:[EMAIL PROTECTED];tag=as3898. To: sip:[EMAIL PROTECTED]. Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE. Server: OpenSer (1.2.0-dev1-notls (sparc64/solaris)). Content-Length: 0. Warning: 392 212.153.111.19:5060 Noisy feedback tells: pid=10911 req_src_ip=212.153.111.54 req_src_port=5060 in_uri=sip:[EMAIL PROTECTED] out_uri=sip:[EMAIL PROTECTED] via_cnt==1. . # U 212.153.111.19:5060 - 212.153.111.54:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 212.153.111.54:5060;branch=z9hG4bK170b;rport=5060. From: 49694525 sip:[EMAIL PROTECTED];tag=as3898. To: sip:[EMAIL PROTECTED]. Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE. Server: OpenSer (1.2.0-dev1-notls (sparc64/solaris)). Content-Length: 0. Warning: 392 212.153.111.19:5060 Noisy feedback tells: pid=10911 req_src_ip=212.153.111.54 req_src_port=5060 in_uri=sip:[EMAIL PROTECTED] out_uri=sip:[EMAIL PROTECTED] via_cnt==1. . # U 212.153.111.19:5060 - 146.188.127.31:5060 INVITE sip:[EMAIL PROTECTED] SIP/2.0. Record-Route: sip:212.153.111.19;lr;ftag=as3898. Via: SIP/2.0/UDP 212.153.111.19;branch=z9hG4bKa4cf.d514.0. Via: SIP/2.0/UDP 212.153.111.54:5060;branch=z9hG4bK170b;rport=5060. From: 49694525 sip:[EMAIL PROTECTED];tag=as3898. To: sip:[EMAIL PROTECTED]. Contact: sip:[EMAIL PROTECTED]. Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE. User-Agent: Asterisk PBX. Max-Forwards: 20. Remote-Party-ID: 49694525 sip:[EMAIL PROTECTED];privacy=off;screen=no. Date: Mon, 21 Aug 2006 12:06:56 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Content-Type: application/sdp. Content-Length: 211. . v=0. o=root 26661 26661 IN IP4 212.153.111.54. s=session. c=IN IP4 212.153.111.54. t=0 0. m=audio 42670 RTP/AVP 0 3 8. a=rtpmap:0 PCMU/8000. a=rtpmap:3 GSM/8000. a=rtpmap:8 PCMA/8000. a=silenceSupp:off - - - -. # U 146.188.127.31:5060 - 212.153.111.19:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 212.153.111.19;branch=z9hG4bKa4cf.d514.0,SIP/2.0/UDP 212.153.111.54:5060;branch=z9hG4bK170b;rport=5060. From: 49694525 sip:[EMAIL PROTECTED];tag=as3898. To: sip:[EMAIL PROTECTED];tag=E8EB016C-295. Date: Mon, 21 Aug 2006 12:06:56 GMT. Call-ID: [EMAIL PROTECTED] Server: Cisco-SIPGateway/IOS-12.x. CSeq: 102 INVITE. Allow-Events: telephone-event. Content-Length: 0. . # U 146.188.127.31:5060 - 212.153.111.19:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 212.153.111.19;branch=z9hG4bKa4cf.d514.0,SIP/2.0/UDP 212.153.111.54:5060;branch=z9hG4bK170b;rport=5060. From: 49694525 sip:[EMAIL
[asterisk-users] failed calls
I am trying to track down a problem which is occurring on about 1% of the phone calls through a customers system. Layout looks like this: PSTN PRI Asterisk A IAX Trunk over point to point T1 Asterisk B SIP over LAN Polycom IP501 1) The user on the Polycom IP501 phone dials a number. 2) It is routed across the LAN to an Asterisk PBX 3) The call is then routed across the T1 via IAX to another Asterisk Server 4) This server drops the call on a PRI line 5) The callee will hear their phone ring 6) On the Polycom you hear 5-10 seconds of silence then a fast busy. 7) The callee answers but no one is there. I see the following in my debug log (on Asterisk B) but Im not sure if any of these messages are abnormal: Aug 21 08:39:18 DEBUG[16560] channel.c: Didn't get a frame from channel: SIP/101-40c4 Aug 21 08:39:18 DEBUG[16560] channel.c: Bridge stops bridging channels SIP/101-40c4 and IAX2/ROUTING-6 Aug 21 08:39:18 DEBUG[16560] chan_iax2.c: We're hanging up IAX2/ROUTING-6 now... Aug 21 08:39:18 DEBUG[16560] app_dial.c: Exiting with DIALSTATUS=ANSWER. Aug 21 08:39:18 DEBUG[16560] chan_sip.c: update_call_counter(101) - decrement call limit counter Anyone have any ideas on this? -Jonathan Jonathan Creasy Network Engineer BluegrassNet Development www.bgnd.com www.bluegrass.net o. 502-589-4638 c. 502-889-5567 h. 812-206-1830 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Size of realtime appdata field under MySQL
Hi all, I'm trying to use a bigger appdata column for realtime, the reason being that I'm moving to a new setup where the SIP devices are named according to the name of the user and some of my dial/page commands need to dial a goodly number of phones which then exceeds the 255 max size of the column. I've tried turning the column into a text and Asterisk copes with that but still truncates it somewhere. Is this possible / is there a constant I can change somewhere? :) Thanks, Peter. This message has been comprehensively scanned for viruses, please visit http://www.avg.power.net.uk/ for details. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk in Xen 3.0
Hello! Are there any known (bad) issues / experience running Asterisk inside Xen VM? Has anyone experienced running Asterisk inside a Xen VM with PCI access to PRI adapter? Regards, Tomer. ___ --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] Announce caller-id
How where you able to interact with the callee after they had answered the call? You lose control of the dial plan after someone answers, until they hang up. -Original Message- From: Roy Kidder [mailto:[EMAIL PROTECTED] Sent: Monday, August 21, 2006 5:05 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Announce caller-id I did something along these lines, but I was playing the caller ID back to the caller, not after a transfer. In a perl AGI script. I split the caller ID number into an array, seperated by '//' so each number was an element. Then I played digits/$array[0]... digits/$array[1]...etc. coolbreeze wrote: I would like to transfer an incoming call and, when the call is answered, have the caller id of the call spoken when the call is answered on my cell phone. Any tips greatly appreciated AAH 2.7 using sip trunks exclusively. Regards ___ --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 ___ --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 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Status of Monitor
Is there a way to find out if a channel is currently being recorded/monitored via the Asterisk Manager API. Currently, if I issue a Action: Status, it lists all channels as unmonitored, regardless if they're being recorded or not. (In my setup, I'm not doing automatic monitoring, I have a web interface to send Action: Monitor to allow recording to start at a certain place in the conversation. I would like to make this interface a bit more robust, and present the correct actions (start or stop) depending if the channel is recording or not) Any help would be appreciated, Thanks ___ --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] Asterisk in Xen 3.0
Hi, Tomer Horn wrote: Are there any known (bad) issues / experience running Asterisk inside Xen VM? Has anyone experienced running Asterisk inside a Xen VM with PCI access to PRI adapter? We do this a lot, although I believe our engineers are still using Xen2 for systems with BRI/PRI adapters. Xen3 is fine if there is only software involved. Florian ___ --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] Polycom IP430 won't finish boot
Hi Doug - Let me start by saying when I first plugged it in, I didn't have the files set up on my ftp server yet, and the phone used it's default settings and it completed bootup. Now... I started with sip v1.6.6b and bootrom 3.1.3 on the ftp server. Phone boots, d/l's files, reaches Welcome screen and stops. After several minutes, it will reboot on it's own. So, I copied the v1.6.7 sip firmware using default files (sip.cfg, phone1.cfg ) and still doing same thing. One thing you could try that may or may not help: a different FTP server. I've been using ProFTPd (particularly because you can configure it to use the Polycom default username and password, so you don't have to manually type in those settings on each phone). If that doesn't do it, go for Polycom support. It sounds like you're doing everything according to their book, and it isn't working. If so, they should be the ones to fix it. - Noah ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Portuguese sound files available?
Hi, I've been searching for sound files in Portuguese language to use in Asterisk for example for voicemail, but I couldn't find anything... Does anyone know where I could find them for download, if there is such thing already? Regards, Ricardo. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Variable to show caller id for a current call?
Is there a variable that can be gotten with GetVar to show the callerid of the current incoming call in progress at a sip extension? For instance, a caller from 516-922-9463 calls extension 234. I would like to be able to be able to get back the 516-922-9463 if I pass 234. Also, can this be done while the extension is ringing? TIA, Warren ___ --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] t.38 bounty
Steve Underwood wrote: marek cervenka wrote: hi, bounty for t.38 is $11,750. that looks good! http://www.voip-info.org/wiki/view/Asterisk+T.38+Bounty how high must be bounty for Digium to hire programmer for this? thanks Do you really think T.38 can be implemented on a contract basis for $11,750? Besides, these bounties are rarely paid. Most of those pledges are quite old, and I really wouldn't expect them to be paid. Digium has no interest in implementing T.38. They have actually been quite obstructive. Regards, Steve No point making pledges anyway, nowadays you may as well just paypal the money to steve. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] t.38 bounty
What is the status of it anyway? I followed the bug for it and it appears that the bug was closed and maybe it was incorporated into Trunk. Is this true? And should it be (fully) functional now? PA ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Zapand SendDTMF??
I have tried it with exten = 5481,2,DIAL(IAX2/5480,,D(w1)) and it also does
not work.
I have since moved it to an analog extension on a legacy PBX.
I have tried:
exten = 5481,3,DIAL(Zap/g2/5110,,D(1))
and a macro with SendDTMF.
It works fine if I dial 5110, then enter the number of the zone I wish to page.
If I dial 5481 with is intended to dial zone 1 automatically, I get a 3-4
second delay before I can speak or it gets cut off.
Note: I am referring to 3-4 seconds after the DTMF digit 1 is sent. I
understand that it should be muted during the D option.
--
--
Steven
http://www.glimasoutheast.org
Alexander Lopez [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]
Try This
exten = 5481,1,NoOp(${TIMESTAMP} paging Group 1 Office Page)
exten = 5481,2,DIAL(IAX2/5480/w1||)
SNIP
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RE: [asterisk-users] Variable to show caller id for a current call?
${CALLERID(number)}
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Warren (mailing lists)
Sent: Monday, August 21, 2006 1:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Variable to show caller id for a
current call?
Is there a variable that can be gotten with GetVar to show the
callerid of the current incoming call in progress at a sip extension?
For instance, a caller from 516-922-9463 calls extension 234.
I would like to be able to be able to get back the
516-922-9463 if I pass 234.
Also, can this be done while the extension is ringing?
TIA,
Warren
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[asterisk-users] Realtime and labels
Does anyone know if realtime extensions support the use of labels?
ie:
exten = acdpause,1,Answer
exten = acdpause,n,Wait,1
exten = acdpause,n,PauseQueueMember(|Agent/${CALLERIDNUM})
exten = acdpause,n,GotoIf($[${PQMSTATUS} =
PAUSED]?paused:error)
exten = acdpause,n(paused),Playback(unavailable)
exten = acdpause,n,Hangup
exten = acdpause,n(error), Playback(an-error-has-occured)
exten = acdpause,n,Hangup
Do I just put the label in the extension column?
Doug.
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Re: [asterisk-users] Metermaid - Parking Slot
David Gagnon wrote: Finally, in the trunk all the states of my device are broken. If I downgrade to 1.2.10, everything is fine. The device get busy and ringing. But in the current trunk Asterisk SVN-trunk-r40632M none of my hints works. Anyone could confim this bugs ? David, I haven't heard of anyone using the metermaid function in the svn trunk. I haven't even seen any documentation for it - I guess its buried in the source code :-( According to bug 5779, oej extensively rewrote everything for svn trunk... better open a bug report. - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Realtime and hints
Can realtime be used with hints? How would you get the following into the database given that the priority column is numeric, and that there is no application for the first entry? exten = 2944006,hint,SIP/2944006 exten = 2944006,1,Dial(SIP/2944006) Every time I touch realtime I hit obstacles. How are others getting around this limitation? Doug. ___ --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] Variable to show caller id for a current call?
But how do you get that with GetVar? I am trying to do this through the
API. I tried:
Action: GetVar
Variable CALLERID(227)
and I tries:
Action: GetVar
Variable ${CALLERID(227)}
Neither returned anything.
How can I do this? Alternately... Is there a way to have a program
fired off when an extension rings that will have the caller id passed to
it as part of the call?
W
Rushowr wrote:
${CALLERID(number)}
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Warren (mailing lists)
Sent: Monday, August 21, 2006 1:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Variable to show caller id for a
current call?
Is there a variable that can be gotten with GetVar to show the
callerid of the current incoming call in progress at a sip extension?
For instance, a caller from 516-922-9463 calls extension 234.
I would like to be able to be able to get back the
516-922-9463 if I pass 234.
Also, can this be done while the extension is ringing?
TIA,
Warren
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Re: [asterisk-users] Portuguese sound files available?
Ricardo Carvalho wrote: I've been searching for sound files in Portuguese language to use in Asterisk for example for voicemail, but I couldn't find anything... Does anyone know where I could find them for download, if there is such thing already? Brazilian Portuguese only... http://www.google.com/search?q=asterisk+sound+files+site%3Avoip-info.org ___ --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] Variable to show caller id for a current call?
Well, for one, you could set something like CID = ${CALLERID(number)} in the
dialplan, and then GetVar CID
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Warren (mailing lists)
Sent: Monday, August 21, 2006 3:54 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Variable to show caller id for a
current call?
But how do you get that with GetVar? I am trying to do this
through the API. I tried:
Action: GetVar
Variable CALLERID(227)
and I tries:
Action: GetVar
Variable ${CALLERID(227)}
Neither returned anything.
How can I do this? Alternately... Is there a way to have a
program fired off when an extension rings that will have the
caller id passed to it as part of the call?
W
Rushowr wrote:
${CALLERID(number)}
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Warren (mailing lists)
Sent: Monday, August 21, 2006 1:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Variable to show caller id for a
current call?
Is there a variable that can be gotten with GetVar to show the
callerid of the current incoming call in progress at a sip
extension?
For instance, a caller from 516-922-9463 calls extension 234.
I would like to be able to be able to get back the
516-922-9463 if I pass 234.
Also, can this be done while the extension is ringing?
TIA,
Warren
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RE: [asterisk-users] Apache for FastAGI
-Original Message- From: Tzafrir Cohen [mailto:[EMAIL PROTECTED] Sent: Sunday, August 20, 2006 5:08 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Apache for FastAGI On Sun, Aug 20, 2006 at 04:18:11PM -0600, Douglas Garstang wrote: I'm not sure there's much point in developing it in Erlang anyways. I'll usually do a quick look and see how popular a language or technology is, in the job market before I spend time and effort on learning it. A search on dice for Erlang gets about 3 results. Thanks troll. This post of yours actually made me look into erlang, and it looks interesting. It's called time management. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Text to Speech
Can someone recommend a good text to speech engine that is usable by Asterisk? I have tried the Festival one and it just doesnt cut it for commercial applications. We are willing to pay for a good one that works. Anyone tried the ATT speech engine? The IBM ViaVoice sounds no better then Festival. Thanks for your input. _ Kevin Savoy Business Unit Telecom Analyst 2218 4th Ave W Williston, ND 58801 Ph: 701-774-4023 Fax: 701-774-2901 http://www.novo1.com Novo 1 is a service mark of Novo 1, Inc ___ --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] Text to Speech
N.B.: Please use plain text when sending to this list Can someone recommend a good text to speech engine that is usable by Asterisk? I have tried the Festival one and it just doesn't cut it for commercial applications. We are willing to pay for a good one that works. Anyone tried the ATT speech engine? The IBM ViaVoice sounds no better then Festival. You have flite that is free and, IMHO better than festival (http://nerdvittles.com/index.php?p=134). I also tried Prophecy (http://www.voxeo.com/) but I'm still waiting for the Linux version as I don't have time to babysit a Windows server :) hth ___ --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] Text to Speech
Cepstral seems to sound descent...But if you have more than one voice installed (Example: different languages) I can't say it in realtime in the dialplan...I have to do a little trick like: exten = 1,1,System(/opt/swift/bin/swift -n Diane-8kHz "Hello World" -o /var/lib/asterisk/sounds/swift.wav)exten = 1,n,System(sox /var/lib/asterisk/sounds/swift.wav /var/lib/asterisk/sounds/swift.gsm)exten = 1,n,Playback(swift) When you show app cepstral in theCLI it says you can do it like this: exten= 1,1,Cepstral(voice name="William"hello world/voice) But it doesn't work for me trying to select the voice in the Dialplan just tells me my voice is missing or corrupt...but if you only have one in the voice directory it works fine... But someone else made the app_cepstral.so If I could find a correct syntax that worked I would change the source and recompile the module. - Original Message - From: Kevin Savoy To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Monday, August 21, 2006 4:12 PM Subject: [asterisk-users] Text to Speech Can someone recommend a good text to speech engine that is usable by Asterisk? I have tried the Festival one and it just doesnt cut it for commercial applications. We are willing to pay for a good one that works. Anyone tried the ATT speech engine? The IBM ViaVoice sounds no better then Festival. Thanks for your input. _ Kevin Savoy Business Unit Telecom Analyst 2218 4th Ave W Williston, ND 58801 Ph: 701-774-4023 Fax: 701-774-2901 http://www.novo1.com Novo 1 is a service mark of Novo 1, Inc ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message.Checked by AVG Free Edition.Version: 7.1.405 / Virus Database: 268.11.3/423 - Release Date: 8/18/2006 ___ --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] Zaptel install - Fedora Core 5
Hi Simon, I did yum update last week and here is my current kernel: # uname -vr 2.6.17-1.2174_FC5smp #1 SMP Tue Aug 8 16:00:39 EDT 2006 # # ls -l /usr/src/kernels total 12 drwxr-xr-x 18 root root 4096 Jul 8 19:43 2.6.17-1.2145_FC5-smp-i686 lrwxrwxrwx 1 root root 26 Jul 8 19:43 2.6.17-1.2145_FC5smp-i686 - 2.6.17-1.2145_FC5-smp-i686 drwxr-xr-x 18 root root 4096 Jul 20 07:27 2.6.17-1.2157_FC5-smp-i686 lrwxrwxrwx 1 root root 26 Jul 20 07:27 2.6.17-1.2157_FC5smp-i686 - 2.6.17-1.2157_FC5-smp-i686 drwxr-xr-x 18 root root 4096 Aug 14 05:29 2.6.17-1.2174_FC5-smp-i686 lrwxrwxrwx 1 root root 26 Aug 14 05:29 2.6.17-1.2174_FC5smp-i686 - 2.6.17-1.2174_FC5-smp-i686 # I had no problem at all with zaptel. I am only using TDM400P though, in case that matters. Cheers, Anto - Original Message - From: simon elliston ball [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, August 21, 2006 1:02 PM Subject: Re: [asterisk-users] Zaptel install - Fedora Core 5 I managed to get zaptel to compile reasonably easily on 2.6.17-1.2157_FC5smp. However, the yum repo sites do not provide devel packages for 2.6.17-2174 for some reason last time I checked, hence couldn't get it to build on that kernel. You could probably create the devel package without too much trouble from the srpm, but it's a lot easier to stick to 2157. If anyone else have managed to get FC5 to install the correct devel packages for the latest kernel, please let me know! Simon On 21 Aug 2006, at 11:52, Tomislav Parcina wrote: I'm trying to install Zaptel 1.2.7 on Fedora Core 5 with 2.6.17-1.2174_FC5 kernel from source code. When I untar Zaptel and execute this is error that I get. cc -o ztmonitor ztmonitor.o cc -o ztspeed.o -c ztspeed.c cc -o ztspeed ztspeed.o cc -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE- DSTANDALONE_ZAPATA -DZAPTE L_CONFIG=\/etc/zaptel.conf\zttest.c -o zttest cc -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE- DSTANDALONE_ZAPATA -DZAPTE L_CONFIG=\/etc/zaptel.conf\ -c -o fxotune.o fxotune.c cc -o fxotune fxotune.o -lm /lib/modules/2.6.17-1.2174_FC5/build You do not appear to have the sources for the 2.6.17-1.2174_FC5 kernel installed . make: *** [linux26] Error 1 What could be the problem? How to solve it? -- Tomislav Parcina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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 ___ --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 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Double dial dtmf sounds
Hi, I have site using only softphones (SJPhone under Windows). Once in a while the users complain that they hear double and triple dial dtmf when they dial out. What could be causing that on the asterisk side? Andre Courchesne ___ --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] Text to Speech
All I can find for Flite is for AAH, does it work as well with plain Asterisk? Is the setup the same? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Time Bandit Sent: Monday, August 21, 2006 3:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Text to Speech N.B.: Please use plain text when sending to this list Can someone recommend a good text to speech engine that is usable by Asterisk? I have tried the Festival one and it just doesn't cut it for commercial applications. We are willing to pay for a good one that works. Anyone tried the ATT speech engine? The IBM ViaVoice sounds no better then Festival. You have flite that is free and, IMHO better than festival (http://nerdvittles.com/index.php?p=134). I also tried Prophecy (http://www.voxeo.com/) but I'm still waiting for the Linux version as I don't have time to babysit a Windows server :) hth ___ --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 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voicemail and languages other than english doesn't seem to work well
I want to hear french messages. I put language=fr in the [globals] section of extensions.conf and in the [general] section of sip.conf. If I call an unavailable number, the digits are read in english even if the trace says french : -- Executing VoiceMail(SIP/103-6441, [EMAIL PROTECTED]) in new stack -- Playing 'vm-theperson' (language 'fr') ; -- ok, french -- Playing 'digits/1' (language 'fr') ; non ok, english -- Playing 'digits/0' (language 'fr') ; non ok, english -- Playing 'digits/4' (language 'fr') ; non ok, english -- Playing 'vm-isunavail' (language 'fr') ; -- ok, french -- Playing 'vm-intro' (language 'fr') ; -- ok, french The french messages are at the right place : /var/lib/asterisk/sounds/fr/digits/*.gsm By the way, the days and months are also in english. They are in the same dir (digits). Somebody has an idea? Thank you. --- Dominique Dartois ___ --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] Voicemail and languages other than english doesn't seem to work well
On Mon, 2006-08-21 at 23:03 +0200, Dominique Dartois wrote: I want to hear french messages. I put language=fr in the [globals] section of extensions.conf and in the [general] section of sip.conf. The french messages are at the right place : /var/lib/asterisk/sounds/fr/digits/*.gsm The correct location for sounds is: /var/lib/asterisk/sounds/fr/*.gsm - for general sounds /var/lib/asterisk/sounds/digits/fr/*.gsm /var/lib/asterisk/sounds/letters/fr/*.gsm /var/lib/asterisk/sounds/phonetic/fr/*.gsm -- Carlos Chavez Prats Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 signature.asc Description: This is a digitally signed message part ___ --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] Text to Speech
Quoting Kevin Savoy [EMAIL PROTECTED]: Can someone recommend a good text to speech engine that is usable by Asterisk? I have tried the Festival one and it just doesn't cut it for commercial applications. I like Cepstral. Using the information here: http://www.oldskoolphreak.com/tfiles/voip/installing_app_cepstral.txt You can install app_cepstral after you have installed the package and libs from Cepstral. Then you can do something like this: [macro-cepstral-demo] exten = s,1,Cepstral(voice name=DuchessHello and welcome to the world of text to speech using Cepstral. My name is Duchess./voice) exten = s,n,Cepstral(voice name=WalterHello and welcome to the world of text to speech using Cepstral. My name is Walter./voice) exten = s,n,Cepstral(voice name=ShoutyHello and welcome to the world of text to speech using Cepstral. My name is Shouty./voice) exten = s,n,Cepstral(voice name=WilliamHello and welcome to the world of text to speech using Cepstral. My name is William./voice) exten = s,n,Cepstral(voice name=WhisperyHello and welcome to the world of text to speech using Cepstral. My name is Whispery./voice) exten = s,n,Cepstral(voice name=RobinHello and welcome to the world of text to speech using Cepstral. My name is Robin./voice) exten = s,n,Cepstral(voice name=LindaHello and welcome to the world of text to speech using Cepstral. My name is Linda./voice) exten = s,n,Cepstral(voice name=EmilyHello and welcome to the world of text to speech using Cepstral. My name is Emily./voice) exten = s,n,Cepstral(voice name=DianeHello and welcome to the world of text to speech using Cepstral. My name is Diane./voice) exten = s,n,Cepstral(voice name=DavidHello and welcome to the world of text to speech using Cepstral. My name is David./voice) exten = s,n,Cepstral(voice name=DuncanHello and welcome to the world of text to speech using Cepstral. My name is Duncan./voice) exten = s,n,Cepstral(voice name=DamienHello and welcome to the world of text to speech using Cepstral. My name is Damien./voice) exten = s,n,Cepstral(voice name=CallieHello and welcome to the world of text to speech using Cepstral. My name is Callie./voice) exten = s,n,Cepstral(voice name=DogHello and welcome to the world of text to speech using Cepstral. My name is Dog./voice) exten = s,n,Cepstral(voice name=AmyHello and welcome to the world of text to speech using Cepstral. My name is Amy./voice) --Shane This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Manager API: matching an Originate to the Newchannel event
Hi, I'm having a bit of trouble matching up Newchannel (and Newexten, etc. etc.) events with the Originate that created them. Basically, what I want to do is have software automatically initiate a call, and then track the status of that call through to completion. I can match to some degree with the Channel name in later events, but I can't see a way to do this that isn't inherently racey - ie. the person dials out, or someone calls in, at the same time as I'm doing my Originate, I'm not going to be able to match the events with any degree of certainty. Am I missing the obvious somewhere? Cheers, Nic. -- Nic Bellamy, Head Of Engineering, Vadacom Ltd - http://www.vadacom.co.nz/ ___ --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] Voicemail and languages other than englishdoesn't seem to work well
Thank you very much Carlos, you are absolutely right. Now it works! Thanks again. --- Dominique Dartois -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Carlos Chavez Envoyé : lundi 21 août 2006 23:09 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [asterisk-users] Voicemail and languages other than englishdoesn't seem to work well On Mon, 2006-08-21 at 23:03 +0200, Dominique Dartois wrote: I want to hear french messages. I put language=fr in the [globals] section of extensions.conf and in the [general] section of sip.conf. The french messages are at the right place : /var/lib/asterisk/sounds/fr/digits/*.gsm The correct location for sounds is: /var/lib/asterisk/sounds/fr/*.gsm - for general sounds /var/lib/asterisk/sounds/digits/fr/*.gsm /var/lib/asterisk/sounds/letters/fr/*.gsm /var/lib/asterisk/sounds/phonetic/fr/*.gsm -- Carlos Chavez Prats Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 ___ --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] Text to Speech
You might try runtime Dectalk for Linux available from http://www.fonix.com -- its not free, but it sounds quite nice. on Monday 08/21/2006 Time Bandit([EMAIL PROTECTED]) wrote N.B.: Please use plain text when sending to this list Can someone recommend a good text to speech engine that is usable by Asterisk? I have tried the Festival one and it just doesn't cut it for commercial applications. We are willing to pay for a good one that works. Anyone tried the ATT speech engine? The IBM ViaVoice sounds no better then Festival. You have flite that is free and, IMHO better than festival (http://nerdvittles.com/index.php?p=134). I also tried Prophecy (http://www.voxeo.com/) but I'm still waiting for the Linux version as I don't have time to babysit a Windows server :) hth ___ --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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [EMAIL PROTECTED] ___ --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] Realtime and labels
Douglas Garstang wrote: Does anyone know if realtime extensions support the use of labels? I don't believe so. As I understand it, the dialplan parser internally converts n-type and labeled priorities to a straight numeric format, which is then used internally. Becuase the Realtime engine bypasses that parser, it has to have extensions in strict, old-style numeric priority order. If this isn't correct I'm sure someone will point it out. B. ___ --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] Text to Speech
Anytime I try and specify a voice when there is more than 1 voice in my voices directory...it has an error with the syntax you show here... Like I was saying in a previous post... - Original Message - From: Shane Young [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com; Kevin Savoy [EMAIL PROTECTED] Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Monday, August 21, 2006 5:13 PM Subject: Re: [asterisk-users] Text to Speech Quoting Kevin Savoy [EMAIL PROTECTED]: Can someone recommend a good text to speech engine that is usable by Asterisk? I have tried the Festival one and it just doesn't cut it for commercial applications. I like Cepstral. Using the information here: http://www.oldskoolphreak.com/tfiles/voip/installing_app_cepstral.txt You can install app_cepstral after you have installed the package and libs from Cepstral. Then you can do something like this: [macro-cepstral-demo] exten = s,1,Cepstral(voice name=DuchessHello and welcome to the world of text to speech using Cepstral. My name is Duchess./voice) exten = s,n,Cepstral(voice name=WalterHello and welcome to the world of text to speech using Cepstral. My name is Walter./voice) exten = s,n,Cepstral(voice name=ShoutyHello and welcome to the world of text to speech using Cepstral. My name is Shouty./voice) exten = s,n,Cepstral(voice name=WilliamHello and welcome to the world of text to speech using Cepstral. My name is William./voice) exten = s,n,Cepstral(voice name=WhisperyHello and welcome to the world of text to speech using Cepstral. My name is Whispery./voice) exten = s,n,Cepstral(voice name=RobinHello and welcome to the world of text to speech using Cepstral. My name is Robin./voice) exten = s,n,Cepstral(voice name=LindaHello and welcome to the world of text to speech using Cepstral. My name is Linda./voice) exten = s,n,Cepstral(voice name=EmilyHello and welcome to the world of text to speech using Cepstral. My name is Emily./voice) exten = s,n,Cepstral(voice name=DianeHello and welcome to the world of text to speech using Cepstral. My name is Diane./voice) exten = s,n,Cepstral(voice name=DavidHello and welcome to the world of text to speech using Cepstral. My name is David./voice) exten = s,n,Cepstral(voice name=DuncanHello and welcome to the world of text to speech using Cepstral. My name is Duncan./voice) exten = s,n,Cepstral(voice name=DamienHello and welcome to the world of text to speech using Cepstral. My name is Damien./voice) exten = s,n,Cepstral(voice name=CallieHello and welcome to the world of text to speech using Cepstral. My name is Callie./voice) exten = s,n,Cepstral(voice name=DogHello and welcome to the world of text to speech using Cepstral. My name is Dog./voice) exten = s,n,Cepstral(voice name=AmyHello and welcome to the world of text to speech using Cepstral. My name is Amy./voice) --Shane This message was sent using IMP, the Internet Messaging Program. ___ --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 -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.405 / Virus Database: 268.11.3/423 - Release Date: 8/18/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Manager API: matching an Originate to the Newchannel event
In article [EMAIL PROTECTED], Nic Bellamy [EMAIL PROTECTED] wrote: Hi, I'm having a bit of trouble matching up Newchannel (and Newexten, etc. etc.) events with the Originate that created them. Basically, what I want to do is have software automatically initiate a call, and then track the status of that call through to completion. I can match to some degree with the Channel name in later events, but I can't see a way to do this that isn't inherently racey - ie. the person dials out, or someone calls in, at the same time as I'm doing my Originate, I'm not going to be able to match the events with any degree of certainty. Am I missing the obvious somewhere? No, there isn't a clean way to do it. This was discussed here a week ago; see my suggestions at http://lists.digium.com/pipermail/asterisk-users/2006-August/162581.html and at http://lists.digium.com/pipermail/asterisk-users/2006-August/162582.html Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call file do 2 outbound call
Hello, I am not so really familar with asterisk at the moment, but i am working hard on it. Please could anybody advise me how to write a call file for the queue to do 2 outbounds call and connect both via my SIP interface. Thanks in advance. Daniel ___ --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] Re: Manager API: matching an Originate to the Newchannel event
Yes there is but only in Bristuff asterisk.In bristuff when you enter Originate command you receive feedback with uniqueid of created call.So than you can trace uniqueidgreetingsmk 2006/8/21, Tony Mountifield [EMAIL PROTECTED]: In article [EMAIL PROTECTED],Nic Bellamy [EMAIL PROTECTED] wrote: Hi, I'm having a bit of trouble matching up Newchannel (and Newexten, etc. etc.) events with the Originate that created them. Basically, what I want to do is have software automatically initiate a call, and then track the status of that call through to completion. I can match to some degree with the Channel name in later events, but I can't see a way to do this that isn't inherently racey - ie. the person dials out, or someone calls in, at the same time as I'm doing my Originate, I'm not going to be able to match the events with any degree of certainty. Am I missing the obvious somewhere?No, there isn't a clean way to do it. This was discussed here a week ago; see my suggestions at http://lists.digium.com/pipermail/asterisk-users/2006-August/162581.htmland at http://lists.digium.com/pipermail/asterisk-users/2006-August/162582.htmlCheersTony--Tony MountifieldWork: [EMAIL PROTECTED] - http://www.softins.co.ukPlay: [EMAIL PROTECTED] - http://tony.mountifield.org___ --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
[asterisk-users] Call file do 2 outbound call
Hello, I am not so really familar with asterisk at the moment, but i am working hard on it. Please could anybody advise me how to write a call file for the queue to do 2 outbounds call and connect both via my SIP interface. Thanks in advance. Daniel ___ --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] Re: Manager API: matching an Originate to the Newchannel event
If you are originating a call with a Local/ channel you cannot use the uniqueID alone to track it. The only field that will follow all legs of a Local/ channel originated call is the CallerID, and that is only if you add the o flag to your Dial string. It's a very messy prospect to track calls through the Manager API. In fact two years ago I drafted this Whitepaper that would help tremendously, nothing ever came of it: http://www.freedomphones.net/Manager_API_modification_whitepaper.txt MATT--- On 8/21/06, Miloš Kocbek [EMAIL PROTECTED] wrote: Yes there is but only in Bristuff asterisk. In bristuff when you enter Originate command you receive feedback with uniqueid of created call. So than you can trace uniqueid greetings mk 2006/8/21, Tony Mountifield [EMAIL PROTECTED]: In article [EMAIL PROTECTED], Nic Bellamy [EMAIL PROTECTED] wrote: Hi, I'm having a bit of trouble matching up Newchannel (and Newexten, etc. etc.) events with the Originate that created them. Basically, what I want to do is have software automatically initiate a call, and then track the status of that call through to completion. I can match to some degree with the Channel name in later events, but I can't see a way to do this that isn't inherently racey - ie. the person dials out, or someone calls in, at the same time as I'm doing my Originate, I'm not going to be able to match the events with any degree of certainty. Am I missing the obvious somewhere? No, there isn't a clean way to do it. This was discussed here a week ago; see my suggestions at http://lists.digium.com/pipermail/asterisk-users/2006-August/162581.html and at http://lists.digium.com/pipermail/asterisk-users/2006-August/162582.html Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --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 ___ --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 ___ --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] Text to Speech
All I can find for Flite is for AAH, does it work as well with plain Asterisk? Is the setup the same? Never tried it, but it should be the same. Have a look here : http://dialogpalette.sourceforge.net/extras.html hth ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SLA.conf
I found this indication that Shared Line Appearance is possibly in SVN. Is it or is this just an indication that it is up and coming? http://bugs.digium.com/view.php?id=7701 ___ --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] Apache for FastAGI
First off, I don't work for Ericsson, but it is my impression that normally the call control is done in Erlang and the media processing is done ineither hardware orC.I never said you work in Ericsson ;) I You really are interested I recommend You to ask on[EMAIL PROTECTED]I did but they stopped responding as soon as I said asterisk isn't pretending to be a pbx, it is a pbx Definitely I think the lightweight processes in Erlang are mind blowing, but unless it can handle media it won't be as interesting or motivational to learn for whats currently on my plate.. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP Encryption in China
Hi all, Anyone has information on how Chinese equipment makers are encrypting the SIP signaling + media packets to avoid ISP firewalls? Recently, I was sent a sample FXS/O gateway with support for 3 flavors (Seawolf, etc) of such encryption. I don't believe they're using SIP/TLS and SRTP. At first glance, it looks like a simple shared key scheme, certainly without all the key management of SRTP. Cheers. Leo. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Indonesian MFC-R2
Hi, What's wrong with T3 timed out? I use asterisk-1.2.10 package from ScopServ http://www.scopserv.com/v2/home.php?section=news (ScopServ Telephony Server 1.2.20). Here there are four pages of the scanned report from our telco http://www.flickr.com/photos/[EMAIL PROTECTED]/ = These lines appears in asterisk console: [EMAIL PROTECTED] ~]# asterisk -rvv == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Found Asterisk 1.2.10, Copyright (C) 1999 - 2006 Digium, Inc. and others. Created by Mark Spencer [EMAIL PROTECTED] Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'show license' for details. = Connected to Asterisk 1.2.10 currently running on localhost (pid = 2173) Verbosity was 0 and is now 14 Aug 21 13:34:04 WARNING[2185]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/11 - 1001 [2/ 2/Group A /ANI request ] Aug 21 13:34:04 WARNING[2185]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/11 Far end disconnected(cause=Normal, unspecified cause [31]) - state 0x2 Aug 21 13:34:04 WARNING[2185]: chan_unicall.c:2644 handle_uc_event: Unicall/11 event Far end disconnected Aug 21 13:34:04 WARNING[2185]: chan_unicall.c:2930 handle_uc_event: CRN 32769 - far disconnected cause=Normal, unspecified cause [31] Aug 21 13:34:04 WARNING[2185]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/11 Call control(6) Aug 21 13:34:04 WARNING[2185]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/11 Drop call(cause=Normal Clearing [16]) Aug 21 13:34:04 WARNING[2185]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/11 Call disconnected(cause=Normal, unspecified cause [31]) - state 0x800 Aug 21 13:34:04 WARNING[2185]: chan_unicall.c:2644 handle_uc_event: Unicall/11 event Drop call Aug 21 13:34:04 WARNING[2185]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/11 Call control(7) Aug 21 13:34:04 WARNING[2185]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/11 Release call Aug 21 13:34:04 WARNING[2185]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/11 1001 - [1/1000/Clear fwd /ANI request ] Aug 21 13:34:04 WARNING[2185]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/11 Release guard expired Aug 21 13:34:04 WARNING[2185]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/11 Destroying call with CRN 32769 Aug 21 13:34:04 WARNING[2185]: chan_unicall.c:2644 handle_uc_event: Unicall/11 event Release call -- Unicall/11 released Aug 21 13:34:04 WARNING[2185]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/11 Channel echo cancel Aug 21 13:34:08 WARNING[2185]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/12 - 0001 [1/ 1/Idle /Idle ] Aug 21 13:34:08 WARNING[2185]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/12 Detected Aug 21 13:34:08 WARNING[2185]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/12 Making a new call with CRN 32769 Aug 21 13:34:08 WARNING[2185]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/12 1101 - [2/ 2/Idle /Idle ] Aug 21 13:34:08 WARNING[2185]: chan_unicall.c:2644 handle_uc_event: Unicall/12 event Detected Aug 21 13:34:08 WARNING[2185]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/12 - 2 on [2/ 2/Seize ack /Seize ack] Aug 21 13:34:08 WARNING[2185]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/12 6 on - [2/ 2/Seize ack /Seize ack] Aug 21 13:34:08 WARNING[2185]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/12 - 2 off [2/ 2/Group A /Category req ] Aug 21 13:34:08 WARNING[2185]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/12 6 off - [2/ 2/Group A /Category req ] Aug 21 13:34:08 WARNING[2185]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/12 - 2 on [2/ 2/Group A /Category req ] Aug 21 13:34:08 WARNING[2185]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/12 5 on - [2/ 2/Group A /Category req ] Aug 21 13:34:08 WARNING[2185]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/12 - 2 off [2/ 2/Group A /ANI request ] Aug 21 13:34:08 WARNING[2185]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/12 5 off - [2/ 2/Group A /ANI request ] Aug 21 13:34:23 WARNING[2185]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/12 R2 prot. err. [2/ 2/Group A /ANI request ] cause 32771 - T3 timed out Aug 21 13:34:23 WARNING[2185]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/12 1001 - [1/ 1/Idle /Idle ] Aug 21 13:34:23 WARNING[2185]: chan_unicall.c:2644 handle_uc_event:
[asterisk-users] Re: SIP Debug to file - Is it possible?
Hello List -I'm a big fan of call traces to diagnose a problem; I often use pri set debug file X to write PRI traces out to a file, anyone know of a similar method of saving IP traces (SIP,IAX) to a file? Anyone have any ngrep scripts that do the trick?Thanks!-- --Christopher T Aloi-- -- --Christopher T Aloi-- ___ --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] Re: SIP Debug to file - Is it possible?
Christopher Aloi wrote: Hello List - I'm a big fan of call traces to diagnose a problem; I often use pri set debug file X to write PRI traces out to a file, anyone know of a similar method of saving IP traces (SIP,IAX) to a file? Anyone have any ngrep scripts that do the trick? tcpdump :) ? ___ --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] Re: SIP Debug to file - Is it possible?
Try Ethereal (I think it's called WireShark now). Does nice decoding of the packet stream to show you what's going on. Supports SIP for sure, not so sure about IAX though.-brandon On 8/21/06, Leo Ann Boon [EMAIL PROTECTED] wrote: Christopher Aloi wrote: Hello List - I'm a big fan of call traces to diagnose a problem; I often use pri set debug file X to write PRI traces out to a file, anyone know of a similar method of saving IP traces (SIP,IAX) to a file? Anyone have any ngrep scripts that do the trick?tcpdump :) ?___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Brandon GalbraithEmail: [EMAIL PROTECTED] AIM: brandong00Voice: 630.400.6992A true pirate starts drinking before the sun hits the yard-arm. Ya. --thelost ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Quick, hopefully easy, question
Hey all, I've done some peeking around and can't find a GOOD listing of what the currently supported SIP headers are that Asterisk supports. My main reason is to get the CallerID/RPID settings for whether or not to display, but there's others as well. Anyone have a link? SKM ___ --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] Re: SIP Debug to file - Is it possible?
ngrep is also good if you only want to see SIP traffic and filter all the lower level stuff. -Original Message- From: Brandon Galbraith [mailto:[EMAIL PROTECTED] Sent: Mon 8/21/2006 8:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [asterisk-users] Re: SIP Debug to file - Is it possible? Try Ethereal (I think it's called WireShark now). Does nice decoding of the packet stream to show you what's going on. Supports SIP for sure, not so sure about IAX though. -brandon On 8/21/06, Leo Ann Boon [EMAIL PROTECTED] wrote: Christopher Aloi wrote: Hello List - I'm a big fan of call traces to diagnose a problem; I often use pri set debug file X to write PRI traces out to a file, anyone know of a similar method of saving IP traces (SIP,IAX) to a file? Anyone have any ngrep scripts that do the trick? tcpdump :) ? ___ --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 -- Brandon Galbraith Email: [EMAIL PROTECTED] AIM: brandong00 Voice: 630.400.6992 A true pirate starts drinking before the sun hits the yard-arm. Ya. --thelost ___ --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
