Re: [asterisk-users] Asterisk Redundancy
Atis Lezdins wrote: This seems nice way of sharing settings, however it wouldn't take over calls in progress. For us, currently the greatest problem is that whenever Asterisk crashes, calls are lost, and that means - lost money. Are there any ideas? Perhaps investigate/diagnose the craches? Software instability is not solved with a high-availability solution. IMHO. /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Backuping VoIP provider with PRI
Thx Steve! Steve Totaro wrote: Qualify=yes? Thanks, Steve Marc Patino Gómez wrote: Hi Adam, thanks for your quick answer, I try your tip but the problem persist, so... It seems not to be a dns problem Asterisk executes the Dial command and it tries to reach the VoIP provider until timeout, in * console appears: Called [EMAIL PROTECTED] Anybody knows howto make dial command don't wait until timeout when the provider host is unrechable? Cheers, Marc Adam KOSA wrote: Marc Patino Gómez wrote: in most cases it works well but, if my internet connection is down Asterisk tries to Dial voipprovider, but it can't resolve the dns name, so it waits 60 seconds to jump to the following priority... Any ideas to solve this problem? I can't use the IP of the provider (it has a pool of servers), I try to use dnsmgr without solving the isue Why don't you fill the ip addresses to your /etc/hosts file? In that way lookups won't need any dns resolving and still could keep the load balancing by having multiple ip addresses to the same SIP hostname. regards Adam ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4.12 Release?
Bruce McAlister wrote: I read rumors of a potential Asterisk 1.4.12 release for last week. I would like to start testing Asterisk 1.4 on Solaris, but, the fix for the segfault in res_features is only in the current development trunk. I would much rather test a release version, and as such, need to wait for the release of 1.4.12, my question is, do we have a guestimate on when it will be released, 1 week, 2 weeks, a month? I am pretty busy this week with Astricon, so if it doesn't happen this week, I would say definitely sometime next week. -- Russell Bryant Software Engineer Digium, Inc. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Music On Hold
Hi All, I need to have the same file played from MoH every time someone gets to MoH from a Dial. I want to play marketing messages from it and I want it to start from file 1 every time. Anyone know if/how this can be done? Cheers, Joel. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Do I need to run #modprobe zaptel for Digium
Hi Cohen; And do I need to run #modprobe wcfxs / #modprobe wcfxs or I have to run #modprobe wctdm? What is the difference? Regards Bilal Hi List; If I am configuring Diguim Analoge card, then I need to run #modprobe wctdm, but the question why I need to run #modprobe zaptel also? No. What #modprobe zaptel does a things that #modprobe wctdm does not do? modprobe will load all the modules on which your module depends first. wctdm depends on zaptel, and hence it would first load zaptel and later load wctdm. -- Building a website is a piece of cake. Yahoo! Small Business gives you all the tools to get online. http://smallbusiness.yahoo.com/webhosting ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple Home system with SIP
I answered because I was hoping for a repost without the licence, perhaps through gmail. Would you have been happier not knowing that you were missing out on something? /Benny ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Do I need to run #modprobe zaptel for Digium
On Wed, Sep 26, 2007 at 12:13:33AM -0700, bilal ghayyad wrote: Hi Cohen; And do I need to run #modprobe wcfxs / #modprobe wcfxs or I have to run #modprobe wctdm? What is the difference? Just use wctdm (modprobe wcfxs will likely have the same effect. wcfxs was the driver for the same device that was used up to (including) zaptel 1.0. It was later rewritten and renamed to wctdm. Currently wcfxs is usually an alias to wctdm -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DTMF signalling, SIP, and Background()
Hi, I am currently setting up a voice mail/IVR machine with asterisk (1.4.10 at the moment). During testing and evaluation, all was fine; in the - slightly different - production environment, the IVR contexts do not react sensibly. The environment is: POTS -- (ISDN) -- PBX -- (SIP) -- Asterisk with the Asterisk registering with our local PBX. When a user reaches the Asterisk machine via this path, key presses are ignored during the Background() function. My debugging possibilities have been a little restricted, unfortunately (I'm working on that), but as a wild guess, I suppose we might have the following problem: When a call is processed as a SIP call, in-band DTMF signalling does not trigger an event in Asterisk; our PBX possibly/probably does not create a SIP event for DTMF signalling. Would you think that this may be the reason for our experienced problems? Do you have any hints/solutions? I'll have to check whether we can connect PBX and Asterisk via ISDN, in which case the DTMF signals should be handled fine - we'd prefer to stick with the current setup, though. Thx, Regards, Bastian -- Collax GmbH . Burkheimer Straße 3 . 79111 Freiburg . Germany p: +49 (0) 761-45684-24 f: +49 (0) 761-45684-10www.collax.com Geschäftsführer: William K. Hite / Boris Nalbach AG München HRB 158898 . Ust.-IdNr: DE 814464942 \ Press any key... no, no, no, NOT THAT ONE! ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF signalling, SIP, and Background()
Am Mittwoch, den 26.09.2007, 11:08 +0200 schrieb Bastian Friedrich: Hi, I am currently setting up a voice mail/IVR machine with asterisk (1.4.10 at the moment). During testing and evaluation, all was fine; in the - slightly different - production environment, the IVR contexts do not react sensibly. The environment is: POTS -- (ISDN) -- PBX -- (SIP) -- Asterisk with the Asterisk registering with our local PBX. When a user reaches the Asterisk machine via this path, key presses are ignored during the Background() function. My debugging possibilities have been a little restricted, unfortunately (I'm working on that), but as a wild guess, I suppose we might have the following problem: When a call is processed as a SIP call, in-band DTMF signalling does not trigger an event in Asterisk; our PBX possibly/probably does not create a SIP event for DTMF signalling. Would you think that this may be the reason for our experienced problems? Asterisk knows of three different ways for DTMF signalling, in-band being only one of those. There are also rfc2833 and info (SIP INFO) signalling. You could try and set the dtmfmode= parameter in sip.conf to one of those. voip-info.org has some info about it. On the other hand it might be the case that your SIP PBX does _not_ generate SIP INFO or RFC messages but the DTMF signal is poor, not allowing reasonable operation. I had that one with a SIP provider once, effectively meaning I could not remote-control the voicebox. Viel Erfolg, Anselm ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel-1.4.5.1 Compile Error
Thank you, Tilghman. Your suggestion did it. I ran into similar compile problem later: - /usr/src/zaptel-1.4.5.1/xpp/xbus-sysfs.c:135: error: unknown field âhotplugâ specified in initializer make[4]: *** [/usr/src/zaptel-1.4.5.1/xpp/xbus-sysfs.o] Error 1 - and I went in and disabled the xpp in menuselect. It worked and the compile finished successfully. My question to the gurus here is this: what impact will un-selecting wcusb and xpp have later on when I go to run Asterisk? Thanks for your answers. Jeng --- Tilghman Lesher [EMAIL PROTECTED] wrote: On Tuesday 25 September 2007 09:22:01 Jeng Yu wrote: /usr/src/zaptel-1.4.5.1/wcusb.c:1451: error: unknown field âownerâ specified in initializer Type 'make menuselect', deselect wcusb, then left-arrow out to the top, hit 's' for save, then 'make' again. -- Tilghman ___ Want ideas for reducing your carbon footprint? Visit Yahoo! For Good http://uk.promotions.yahoo.com/forgood/environment.html ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Busy problem
Hi, I've a huge problem with the following: Setup: Asterisk 1.4.11 I've got two Thomson ST2030s in an queue. After a while Asterisk logs the following if somebody calls the queues number: - Got SIP response 486 Busy Here back from 172.10.3.31 -- SIP/office1-0823d190 is busy -- Nobody picked up in 0 ms The phones are NOT busy (show channels show nothing). Also show queues says not in use. Then obviously the phones gets stucked (the Thomson phone's display gets freezing) . I have the same problem also now with an Linksys SPA942. Is the Queue implementation buggy? Any idea? Kind Regards, Erik ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Busy problem
Erik Wartusch wrote: - Got SIP response 486 Busy Here back from 172.10.3.31 I see that response when someone presses the DND button on our Polycom phones. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [on-asterisk] Configure one call per line on Cisco 7941/7961
Ahh. Differences with the 7961 software from that of the 7960's. Sorry, need to research more. - dbc. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gary T. Giesen Sent: September-26-07 12:29 AM To: David Cook Cc: [EMAIL PROTECTED]; asterisk-users@lists.digium.com Subject: Re: [on-asterisk] Configure one call per line on Cisco 7941/7961 David, Yes, I'm aware of that, but unfortunately it does two calls on each line appearance (button), so the first two calls go on line 1, and the third will appear on line 2. I'd like to limit it to 1 call per line. Any ideas? Gary On 9/25/07, David Cook [EMAIL PROTECTED] wrote: Gary, if you register multiple lines with the same SIP credentials the phone will do rollover and take care of it. (2nd call comes in on L2, etc.) - dbc. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gary T. Giesen Sent: September-25-07 6:37 PM To: [EMAIL PROTECTED]; asterisk-users@lists.digium.com Subject: [on-asterisk] Configure one call per line on Cisco 7941/7961 Anyone aware of how to configure one call per line on a Cisco 7941/7961? The default behaviour is to have two calls per line button, and this is confusing for some of my users so I'd like to be able to have the 2nd call ring the second line button, rather than being shared with the first. I'm hoping this is something that is configurable in the XML or on the phone UI. Thanks Gary - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED] ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Redundancy
Per Jessen wrote: Atis Lezdins wrote: This seems nice way of sharing settings, however it wouldn't take over calls in progress. For us, currently the greatest problem is that whenever Asterisk crashes, calls are lost, and that means - lost money. Are there any ideas? Perhaps investigate/diagnose the craches? Software instability is not solved with a high-availability solution. IMHO. /Per Jessen, Zürich No. It's not. But there still exists the possibility even in a relatively stable situation that the software could crash or that hardware could fail. It's best, when planning a highly-available solution, to plan for the unforeseen and not assume you can avoid all mishaps. Let's assume, for the sake of argument, that the software will NEVER fail. Hardware still might, and that would still mean a lost call unless there's a way to switch running calls over to a new server seamlessly. Are there such ways? IP calls are especially troublesome in that regard. N. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] POLYCOM - SOUND POINT IP 301 - VOLUME
Hi, Does any know adjust the volume for polycom ip soun point ? I adjust by the phone on the current call, but when hangup the volume lost the volume configuration. There are any way to set phone volume by default ? Regards, Luis Morales ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] configuration of wanpipe for asterisk.
Hi: I install A102 sangoma's card and connect E1 link it now for configuring wanpipe which one should I select for dial plan context:from pstn?or from internal? Best regards. - Be a better Globetrotter. Get better travel answers from someone who knows. Yahoo! Answers - Check it out.___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] POLYCOM - SOUND POINT IP 301 - VOLUME
Luis Morales wrote: Hi, Does any know adjust the volume for polycom ip soun point ? I adjust by the phone on the current call, but when hangup the volume lost the Look in your sip.cfg for the line: volume voice.volume.persist.handset=1 voice.volume.persist.headset=1 voice.volume.persist.handsfree=1/ Change them from 0 to 1 Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Grandstream HT 502 ATA stops receiving calls
Hi, I have an annoying problem with the Grandstream HT 502 ATA. When the ATA first powers up, and registers to the asterisk server, everything works fine, and the ATA is able to receive and send calls, however after a period of time, the ATA can only make outgoing calls, but it is unable to receive calls. When I call any HT 502 after a long while from first registration, I hear ringing from the calling station, but the destination ATA never rings. I tried to decrease the Registration timeout from the HT502 configuration to 15 minutes instead of 60 minutes, the behavior was a little better in terms of the period that the ATA is able to receive calls, however after a longer time from booting, the same occurs and the HT502 stops ringing and no calls are received. When I reboot the ATA it receives calls again, but after a period of time the same problem occurs. Most the HT502 I use have this problem, and all are behind NAT (But the asterisk server is not behind NAT). Does anyone have an idea about how to fix the problem, maybe a configuration setting in the ATA, or a setting in the asterisk SIP account definition for the HT502. Your comments and replies are most appreciated. Thank you and best regards, Antoine Megalla Need a vacation? Get great deals to amazing places on Yahoo! Travel. http://travel.yahoo.com/ ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] POLYCOM - SOUND POINT IP 301 - VOLUME
Doug, Where is located sip.cfg file ? Regards, Luis Morales On Wed, 2007-09-26 at 08:32 -0400, Doug Lytle wrote: Luis Morales wrote: Hi, Does any know adjust the volume for polycom ip soun point ? I adjust by the phone on the current call, but when hangup the volume lost the Look in your sip.cfg for the line: volume voice.volume.persist.handset=1 voice.volume.persist.headset=1 voice.volume.persist.handsfree=1/ Change them from 0 to 1 Doug ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Redundancy
On 9/26/07, SIP [EMAIL PROTECTED] wrote: No. It's not. But there still exists the possibility even in a relatively stable situation that the software could crash or that hardware could fail. It's best, when planning a highly-available solution, to plan for the unforeseen and not assume you can avoid all mishaps. Let's assume, for the sake of argument, that the software will NEVER fail. Hardware still might, and that would still mean a lost call unless there's a way to switch running calls over to a new server seamlessly. Also be sure that you have a very redundant network configuration. Too often I see people spend a great deal of time and money to get redundant servers when their switches, firewalls, routers, etc are not even capable of handling a failed network element. Thanks, Scott ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] # to transfer calls
features.conf has default settings as follows: ; ; Sample Parking configuration ; [general] parkext = 700 ; What ext. to dial to park parkpos = 701-720 ; What extensions to park calls on context = parkedcalls ; Which context parked calls are in ;parkingtime = 45 ; Number of seconds a call can be parked for ; (default is 45 seconds) ;transferdigittimeout = 3 ; Number of seconds to wait between digits when transfering a call ;courtesytone = beep; Sound file to play to the parked caller ; when someone dials a parked call ;xfersound = beep ; to indicate an attended transfer is complete ;xferfailsound = beeperr; to indicate a failed transfer ;adsipark = yes ; if you want ADSI parking announcements ;findslot = next ; Continue to the 'next' parking space. Defaults to 'first' available ;pickupexten = *8 ; Configure the pickup extension. Default is *8 ;featuredigittimeout = 500 ; Max time (ms) between digits for ; feature activation. Default is 500 [featuremap] ;blindxfer = #1; Blind transfer ;disconnect = *0 ; Disconnect ;automon = *1 ; One Touch Record ;atxfer = *2 ; Attended transfer It doesn't look like call being blind transfer. I heard the annoucement transferred when '#' was pressed. Thanks. David On 9/24/07, Atis Lezdins [EMAIL PROTECTED] wrote: On Monday 24 September 2007 10:21:44 VoIP Newbie wrote: I wonder why my call was transferred when I pressed '#' in a conversation. How can I disable this kind of call transfer? Thanks. David Take a look at features.conf - probably there is blind transfer enabled on # key. Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] # to transfer calls
VoIP Newbie wrote: features.conf has default settings as follows: ; ; Sample Parking configuration ; I believe # is the default. If you don't define it, it will use that default. Set it to something that you know won't be used. Maybe ##3 Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] POLYCOM - SOUND POINT IP 301 - VOLUME
Luis Morales wrote: Doug, Where is located sip.cfg file ? Where ever you are provisioning your phones from. I do my provisioning with FTP and the files are located in the polycom home directory that I created. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] My G729 problem re-visited
Ok, I built a test system to duplicate my problem and provide myself a platform that I can mess around with to try and break any features. My problem is G729 pass-through from a gateway to a phone. I think I even have transcoding working, which makes me more confused on what's wrong with my pass-through. It must be a configuration issue. The basics... *CLI core show version Asterisk 1.4.11 built by root @ fwd-tst02 on a i686 running Linux *CLI show modules like 723 Module Description Use Count codec_g723.so G.723.1 Coder/Decoder 0 format_g723.so G.723.1 Simple Timestamp File Format 0 *CLI show modules like 729 Module Description Use Count codec_g729.so G.729 Coder/Decoder 0 format_g729.so Raw G729 data 0 *CLI show translation [truncated] g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 ulaw 5 2 - 1 2 2 1 3 7 - 11 2 - alaw 5 2 1 - 2 2 1 3 7 - 11 2 - g729 5 2 2 2 2 2 1 3 - - 11 2 - The configuration... [gateway] type=friend host=gateway context=default-inbound disallow=all allow=g729 [phone] type=friend context=sip host=dynamic username=phone secret=scott dtmfmode=RFC2833 disallow=all allow=g729 callerid=Scott qualify=yes canreinvite=no exten = 1266,1,Dial(SIP/[number],30,t) exten = 1266,2,Congestion exten = 1266,1,Dial(SIP/[number],30) exten = 1266,2,Congestion (The same results using both of the above dialplans...) The environment... PSTN - Gateway - Asterisk - Phone What I'm seeing works... With the gateway setup to send both G711 and G729, it sends an INVITE which includes both G711 and G729 codecs. Asterisk sends an INVITE to my phone with only G729. The call is made and there's a conversation in G711 with the gateway and G729 with the phone. I assume this means Asterisk is transcoding. What Im seeing fails... With the gateway setup to send only G729, it sends an INVITE to Asterisk which includes only G729. Asterisk send an INVITE to the phone using G729, too. The 200 OK from the phone to the Asterisk includes G729. The 200 OK going from Asterisk to the gateway doesn't include ANY codec. The call is dropped the moment I pickup the phone to answer the call. My question... Why does Asterisk not want to respond to my gateway in G729? Even if the gateway requests it, Asterisk seems to just ignore it. From the transcoding call, and phone to phone G729 calls, I have proof that Asterisk knows how to handle G729 calls. Where do I go from here??? Thanks, Scott ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] chan_sip falls over with undefined symbol ast_pickup_ext
I have just downloaded and built asterisk 1.4.11 on my Fedora Core 6 box. All seemed to go well but once I had configured the server for SIP and sent my first SIP call to the server then asterisk crashed with the message *CLI asterisk: symbol lookup error: /usr/lib/asterisk/modules/chan_sip.so: undefined symbol: ast_pickup_ext This looks like a library has not been installed. Does anybody know which one? Steve Hindmarch ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music On Hold
Make the file the only one in the /var/lib/asterisk/moh directory. Forrest Beck [EMAIL PROTECTED] www.shift8.biz On Sep 26, 2007, at 3:07 AM, Joel Hill wrote: Hi All, I need to have the same file played from MoH every time someone gets to MoH from a Dial. I want to play marketing messages from it and I want it to start from file 1 every time. Anyone know if/how this can be done? Cheers, Joel. ___ Sign up now for AstriCon 2007! September 25-28th. http:// www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Supermicro PDSME+ and TE110P [ ref:00D36mPe.50033qy57:ref ] NEW CASE 22828
Hello, Digium support kindly proposed to ship a TE120P card to help resolve the issue. I plugged in the card, and introduced the loopback plug. I cleared the red alarm for a while and then i started seeing alarm switching from Yel/Recovering to Blue/Rec with a lot of IRQ Misses. I call Digium that assisted us, and we noticed IRQ sharing with the VGA adapter and the Ethernet port. I changed to all the available slot but the DIGIUM card IRQ did not change. Mainboard: SUPERMICRO PDSME+ CPU: Intel Core2Duo E4300 RAM: 2 x DIMM DDR2 1GB PC667/5300 HDD: 1 x 80GB SATA VGA: onboard LAN: 2 x 10/100/1000 onboard I disabled the Ethernet card, but i still have a lot of IRQ Misses, and could not disable the VGA adapter. I plugged in the Same card in a DELL Dimension 3100, and It worked right away and like a charm. I thank the whole digium team for their kind support, for their patience and professionalism, and promise to buy only Digium gear to show my gratitude :-) Kido Digium Support a écrit : Rod, Jared, I do not believe that this customer's problems are a result of hardware incompatibility, due to the fact that his system had no problems detecting the card and ztcfg did not report any errors during configuration. The problem he is experiencing is that the span stays in red alarm after it is configured and connected to the PRI. We attempted to test the card by configuring for loopback and inserting a loopback plug into the span, but the red alarm persisted. My conclusion was and is that the customer's TE110P is faulty. As noted in the case description and by the customer himself, I recommended that he have his card RMA'd if it is still under warranty (which, if I understand correctly, would result in him receiving a TE120P due to the TE110P being past end-of-life). With that cleared up, I'll send him a followup anyhow, since he seems not to understand that the card itself is unusable--it may be that I did not explain it clearly enough to him over the phone. Thanks for bringing this to my attention! --- Original Message --- From: Rod Montgomery [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Fwd: Re: [asterisk-users] Supermicro PDSME+ and TE110P [ ref:00D36mPe.50033qy57:ref ] Sure, I'll ask Patrick to re-open this case and offer a TE120P exchange to this customer. Patrick, here's some text I wrote for a similar offer recently, feel free to re-purpose as you see fit: Digium is committed to manufacturing quality products and to providing top-flight support that exceeds your expectations. Sometimes the mailing list traffic fosters the wrong perception of Digium, but we are eager to assist every customer -- including logging in remotely to provide installation assistance and troubleshoot if necessary. We would like you to evaluate our newer model single-span PCI card, the Digium Wildcard TE120P. May we ship you one to evaluate, please? If it works in your Dell box, you're welcome to return the older TE110P, but all we really ask is that you tell the mailing list about your experience. If you're willing to give the TE120P a try, please reply with your mailing address. Thanks, rm - Forwarded Message - From: Jared Smith [EMAIL PROTECTED] To: Rod Montgomery [EMAIL PROTECTED] Sent: Wednesday, September 19, 2007 10:39:21 AM (GMT-0600) America/Chicago Subject: [Fwd: Re: [asterisk-users] Supermicro PDSME+ and TE110P] Follow-up to the message I sent you this morning... Can we get this guy a TE120P for his TE110P and see if that solves his incompatibility problems? -Jared Forwarded Message From: kido [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Supermicro PDSME+ and TE110P Date: Wed, 19 Sep 2007 15:01:23 + Interesting. I called Digium support. Very friendly guys but they were unable to tell me if it was a hardware compatibility. They only suggested an RMA, but it is an incompatibility issue, that won't help. That is why, I asked for your experience. Thanks Jared Smith a écrit : On Wed, 2007-09-19 at 13:01 +, kido wrote: Has anyone use the Supermicro PDSME+ in combination with the TE110P successfully? My experience so far is not very good. If you're having a motherboard compatibility issue with a Digium card under warranty, you should contact the Digium support department and they'll help you out. Regards, Patrick Anderson Digium Hardware Support Specialist Digium, Inc. 150 West Park Loop Suite 100 Huntsville, AL 35806 [EMAIL PROTECTED] Toll Free 1-877-LINUX-ME (1-877-546-8963) Local 1-256-428-6000 -- Kido NOAGBODJI Directeur NTIC C.A.F.E. Informatique Télécom' Cité Maman
Re: [asterisk-users] Zaptel-1.4.5.1 Compile Error
On Wed, Sep 26, 2007 at 11:13:31AM +0100, Jeng Yu wrote: Thank you, Tilghman. Your suggestion did it. I ran into similar compile problem later: - /usr/src/zaptel-1.4.5.1/xpp/xbus-sysfs.c:135: error: unknown field âhotplugâ specified in initializer make[4]: *** [/usr/src/zaptel-1.4.5.1/xpp/xbus-sysfs.o] Error 1 - and I went in and disabled the xpp in menuselect. It worked and the compile finished successfully. My question to the gurus here is this: what impact will un-selecting wcusb and xpp have later on when I go to run Asterisk? wcusb is a driver for the Digium s100U single FXS USB device. xpp/ includes the driver for the Xorcom Astribank. If you just need ztdummy (or any other Zaptel device): you should have no problems. The problem is caused by a backport of the Fedora f Kernel that is not included in the original 2.6.15 kernel . Later versions of Fedora kernels (from updates) should have no issues building Zaptel. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] chan_sip falls over with undefined symbolast_pickup_ext
Silly me. I solved it myself. I was not loading res_features.so Steve Hindmarch From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: 26 September 2007 14:31 To: asterisk-users@lists.digium.com Subject: [asterisk-users] chan_sip falls over with undefined symbolast_pickup_ext I have just downloaded and built asterisk 1.4.11 on my Fedora Core 6 box. All seemed to go well but once I had configured the server for SIP and sent my first SIP call to the server then asterisk crashed with the message *CLI asterisk: symbol lookup error: /usr/lib/asterisk/modules/chan_sip.so: undefined symbol: ast_pickup_ext This looks like a library has not been installed. Does anybody know which one? Steve Hindmarch ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] POLYCOM - SOUND POINT IP 301 - VOLUME
That's an good tips. Where i find information or help to provisioning the phones with ftp ? In my case the setup was made on each phone using polycom web interface. Regards, Luis Morales On Wed, 2007-09-26 at 09:23 -0400, Doug Lytle wrote: Luis Morales wrote: Doug, Where is located sip.cfg file ? Where ever you are provisioning your phones from. I do my provisioning with FTP and the files are located in the polycom home directory that I created. Doug ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] # to transfer calls
From the asterisk CLI do show features you'll find # is default for Blind transfer your entry below is commented out, ie has a ;in_front_of_it hope this helps Bails VoIP Newbie wrote: features.conf has default settings as follows: ; ; Sample Parking configuration ; [general] parkext = 700 ; What ext. to dial to park parkpos = 701-720 ; What extensions to park calls on context = parkedcalls ; Which context parked calls are in ;parkingtime = 45 ; Number of seconds a call can be parked for ; (default is 45 seconds) ;transferdigittimeout = 3 ; Number of seconds to wait between digits when transfering a call ;courtesytone = beep; Sound file to play to the parked caller ; when someone dials a parked call ;xfersound = beep ; to indicate an attended transfer is complete ;xferfailsound = beeperr; to indicate a failed transfer ;adsipark = yes ; if you want ADSI parking announcements ;findslot = next ; Continue to the 'next' parking space. Defaults to 'first' available ;pickupexten = *8 ; Configure the pickup extension. Default is *8 ;featuredigittimeout = 500 ; Max time (ms) between digits for ; feature activation. Default is 500 [featuremap] ;blindxfer = #1; Blind transfer ;disconnect = *0 ; Disconnect ;automon = *1 ; One Touch Record ;atxfer = *2 ; Attended transfer It doesn't look like call being blind transfer. I heard the annoucement transferred when '#' was pressed. Thanks. David On 9/24/07, Atis Lezdins [EMAIL PROTECTED] wrote: On Monday 24 September 2007 10:21:44 VoIP Newbie wrote: I wonder why my call was transferred when I pressed '#' in a conversation. How can I disable this kind of call transfer? Thanks. David Take a look at features.conf - probably there is blind transfer enabled on # key. Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel-1.4.5.1 Compile Error
On Wednesday 26 September 2007 05:13:31 Jeng Yu wrote: Thank you, Tilghman. Your suggestion did it. I ran into similar compile problem later: - /usr/src/zaptel-1.4.5.1/xpp/xbus-sysfs.c:135: error: unknown field âhotplugâ specified in initializer make[4]: *** [/usr/src/zaptel-1.4.5.1/xpp/xbus-sysfs.o] Error 1 - and I went in and disabled the xpp in menuselect. It worked and the compile finished successfully. My question to the gurus here is this: what impact will un-selecting wcusb and xpp have later on when I go to run Asterisk? xpp is the driver for the Xorcom channel bank, and wcusb is the driver for a device which is long since discontinued by Digium. If you have neither device, the drivers are unnecessary. For that matter, you can disable pretty much any driver that isn't directly related to the hardware you're using (other than modules like zaptel and zttranscode, which are dependencies). -- Tilghman ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Supermicro PDSME+ and TE110P [ ref:00D36mPe.50033qy57:ref ] NEW CASE 22828
On Wednesday 26 September 2007 08:36:01 kido wrote: Mainboard: SUPERMICRO PDSME+ snip For whatever reason, I've seen a lot of issues with SuperMicro boards, which is why the reseller I've worked for tends to use Abit motherboards, not SuperMicro, as the Abit boards do not exhibit these problems. -- Tilghman ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Grandstream GXP2020 / 2000
Erik Wartusch wrote: Hi, Has somebody experiences with the Grandstream GXP2020 / 2000 phones in a business graded installation (with really traffic on not 3 calls a day ;-) ) Of course with Asterisk PBX (1.4.1 or 1.4.11 or 1.4 in generall) Hi Erik, we have about 75 Grandstream GXP2000 phones running with Asterisk 1.2. We also have about 25 Aastra 480i phones in our call centre and 10 Cisco phones in meeting rooms. The Grandstreams work fine on the whole, they are probably good value for money in smaller installations. We use some simple bash scripts to manage the configurations so the need to convert the config files from text to a binary format is not an issue. Sound quality is OK, the speaker phone is not great. Most users are happy with the Grandstreams but we give Aastras to anyone who spends a lot of time on the phone. The reason I cannot recommend these devices for larger installations is the mediocre response from Grandstream's technical support. Tech support will acknowledge your initial problem report but then ignore you if they don't have an immediate fix. This is a pattern repeated over several reported incidents. Our latest issue is with the GXP2000's running f/w = 1.1.1.14. The phone does not send a keep-alive packet when the mute function is used, despite this bug being documented as fixed in a much earlier release. This results in a disconnect after 5 minutes of being on mute. Very annoying when on a conference call or on hold to tech support. This is fixed in 1.1.4.18 but this release introduces an issue with very loud (for our environment) ring tones rendering the GXP2000 unusable in our office. We no longer purchase Grandstream phones but will consider them again in the future should the support issues be resolved. regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] POLYCOM - SOUND POINT IP 301 - VOLUME
Luis Morales wrote: That's an good tips. Where i find information or help to provisioning http://www.voip-info.org/wiki-Polycom+Phones Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Grandstream GXP2020 / 2000
We've a site with about 200 Grandstream GXP2000 phones, and they work quite well. We made some CGI Perl scripts to mass-deploy and manage their configurations from a MySQL DB into a TFTP server, where the phones go to download their binaries. With some initial work, now it has become easy to manage the site. All phones have firmware version 1.1.1.14; we are testing new stable version 1.1.4.18 but by now we found that some phones freeze sometimes - version 1.1.1.14 seems more stable. One thing they lack is the ability to dial alphanumeric contacts (URI dials), we hope future firmware corrects this issue. Older ones hadn't so much good hands-free speaker, but recent ones have a better DSP from Texas Instruments. Althow they're not the best choice in the market (like Cisco or Polycom), they represent a good price/quality ratio. Regards, Ricardo Carvalho. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Manager Originate Action and Cancel
I'm using the Originate Action on the Asterisk Manager to place calls between two extensions in async mode. Is there any way to cancel the Originate Action before I get the OriginateResponse action? I'm unable to perform a Hangup because I can't know the channel name before I get the response... thanks in advance! -- santiago aguiar *netlabs* / Palmar 2548 Montevideo, Uruguay Tel. +(598 2) 707 7687 Fax. +(598 2) 709 4866 / http://www.netlabs.com.uy ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Panel?
Yes, I have. It is not difficult. I use the Asterisk Manager interface. Is there a particular question? - Walt Terry Giufre-Sweetser wrote: Dear List, Has anyone found or written a status panel application, windows or linux, that uses SIP notifies and subscriptions, to gather the status of SIP extensions from Asterisk? And displsy nicely on a GUI? ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] POLYCOM - SOUND POINT IP 301 - VOLUME
Thxs!! On Wed, 2007-09-26 at 10:26 -0400, Doug Lytle wrote: Luis Morales wrote: That's an good tips. Where i find information or help to provisioning http://www.voip-info.org/wiki-Polycom+Phones Doug ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Grandstream GXP2020 / 2000
On Sep 26, 2007, at 10:58 AM, Ricardo Carvalho wrote: All phones have firmware version 1.1.1.14; we are testing new stable version 1.1.4.18 but by now we found that some phones freeze sometimes - version 1.1.1.14 seems more stable. I'm not sure which firmware I'm running on my GXP2000 (I only have one at current), but I did find that it does not like my DHCP server. If I set the phone to use DHCP it would freeze periodically when it tried to renew the DHCP lease. I'd have to yank the power to get it to reset. Changing to static IP fixed the problem and I haven't had any freezing with it since (6 months+ now). I can't say if this is specifically a Grandstream issue with the DHCP as I also know that Windows 98 doesn't like my DHCP server either and fails to renew leases properly as well. So I could just have a crappy DHCP server in place at that location and it may be the true source of the Grandstream's DHCP problems. Figured I'd let you know in case you are using DHCP you may want to try static and see if the freezing stops for you as well. -chris www.mythtech.net ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] faster timeout in ENUMLOOKUP() function
Hi all, In my server dialplan, it first tries to dial possible SIP URI contacts returned by DNS lookups using the ENUMLOOKUP function; it only sends calls to PSTN when there aren't any NAPTR records of the dialed number. Problem arises when my Internet connection is down to some locations, which inhibits my Asterisk server to reach the DNS servers to do those lookups. In those cases, calls only get sent to the PSTN after ENUMLOOKUP function times out (which takes very long)! Is it possible to configure a shorter timeout for the ENUMLOOKUP function, so the next priority in my dialplan comes faster? Or any ideas to avoid this problem? Regards, Ricardo Carvalho. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk - Spandsp Fax not working?
Hi all, I'm trying to setup an asterisk based fax receiving machine. i'm using asterisk 1.2.18 and app_rxfax with spandsp 0.0.4pre9 I have no problems with a modem-fax, but with the fax machines i have tried almost every fax fails, both in sending and receive. the machines are sending a receiving a lot of faxes every day and working well, so i think the problem is on the spandsp side. i have tried almost every spandsp version from 0.0.2 to the current one, both with and without ECM, but without luck. has anybody succeeded in receiving faxes with asterisk app_rxfax and spandsp? I'm noticicing a lot of different behaviours: sending w ECM gave me an OK, and the second half of the page was missing, other faxes fail with Sep 26 17:26:18 DEBUG[4741] app_rxfax.c: == Sep 26 17:26:18 DEBUG[4741] app_rxfax.c: Fax receive not successful - result (11) Unexpected message received. Sep 26 17:26:18 DEBUG[4741] app_rxfax.c: == can anybody help me? thank you in advance, marco ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Routing issue
Hi list I'm kinda new to asterisk and I'm woriking for a company that sells Asterisk solutions and appliances. I installed TrixBox on a litle PC @ home and a x100p card which is recognized as a Zaptel card, I made some in/outbound routes and they seem to work but I have a problem with SIP softphones. I created 2 estensions 1000 and 1001 they're both in different cities, when I 1000 (on the same network as TrixBox) dial 1001 (the other city) they answer and can hear me, but I don't hear them, and when they call *43 for echo test it plays the You're entering echo test... but when it tries to start echo it just hangs up. and the log says -- Executing [EMAIL PROTECTED]:1] Answer(SIP/1000-08939150, ) in new stack -- Executing [EMAIL PROTECTED]:2] Wait(SIP/1000-08939150, 1) in new stack -- Executing [EMAIL PROTECTED]:3] Playback(SIP/1000-08939150, demo-echotest) in new stack -- SIP/1000-08939150 Playing 'demo-echotest' (language 'en') == Spawn extension (from-internal, *43, 3) exited non-zero on 'SIP/1000-08939150' -- Executing [EMAIL PROTECTED]:1] Macro(SIP/1000-08939150, hangupcall) in new stack -- Executing [EMAIL PROTECTED]:1] ResetCDR(SIP/1000-08939150, w) in new stack -- Executing [EMAIL PROTECTED]:2] NoCDR(SIP/1000-08939150, ) in new stack -- Executing [EMAIL PROTECTED]:3] GotoIf(SIP/1000-08939150, 1?skiprg) in new stack -- Goto (macro-hangupcall,s,6) -- Executing [EMAIL PROTECTED]:6] GotoIf(SIP/1000-08939150, 1?skipblkvm) in new stack -- Goto (macro-hangupcall,s,9) -- Executing [EMAIL PROTECTED]:9] GotoIf(SIP/1000-08939150, 1?theend) in new stack -- Goto (macro-hangupcall,s,11) -- Executing [EMAIL PROTECTED]:11] Hangup(SIP/1000-08939150, ) in new stack == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/1000-08939150' in macro 'hangupcall' == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/1000-08939150' I'd appreciate your help a lot, I'm not if this is a forewall issue or something wrong with my asterisk config. Thanks a lot. -- DAVID GONZALEZ H. GNU/Linux Debian+SuSE+RedHat+LFS TECNICO EN REDES NETWORK ADMIN http://www.computrabajo.com.co/cvs/dgonzalezh ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fwd: Routing issue
Hi list I'm kinda new to asterisk and I'm woriking for a company that sells Asterisk solutions and appliances. I installed TrixBox on a litle PC @ home and a x100p card which is recognized as a Zaptel card, I made some in/outbound routes and they seem to work but I have a problem with SIP softphones. I created 2 estensions 1000 and 1001 they're both in different cities, when I 1000 (on the same network as TrixBox) dial 1001 (the other city) they answer and can hear me, but I don't hear them, and when they call *43 for echo test it plays the You're entering echo test... but when it tries to start echo it just hangs up. and the log says -- Executing [EMAIL PROTECTED]:1] Answer(SIP/1000-08939150, ) in new stack -- Executing [EMAIL PROTECTED]:2] Wait(SIP/1000-08939150, 1) in new stack -- Executing [EMAIL PROTECTED]:3] Playback(SIP/1000-08939150, demo-echotest) in new stack -- SIP/1000-08939150 Playing 'demo-echotest' (language 'en') == Spawn extension (from-internal, *43, 3) exited non-zero on 'SIP/1000-08939150' -- Executing [EMAIL PROTECTED]:1] Macro(SIP/1000-08939150, hangupcall) in new stack -- Executing [EMAIL PROTECTED]:1] ResetCDR(SIP/1000-08939150, w) in new stack -- Executing [EMAIL PROTECTED]:2] NoCDR(SIP/1000-08939150, ) in new stack -- Executing [EMAIL PROTECTED]:3] GotoIf(SIP/1000-08939150, 1?skiprg) in new stack -- Goto (macro-hangupcall,s,6) -- Executing [EMAIL PROTECTED]:6] GotoIf(SIP/1000-08939150, 1?skipblkvm) in new stack -- Goto (macro-hangupcall,s,9) -- Executing [EMAIL PROTECTED] :9] GotoIf(SIP/1000-08939150, 1?theend) in new stack -- Goto (macro-hangupcall,s,11) -- Executing [EMAIL PROTECTED]:11] Hangup(SIP/1000-08939150, ) in new stack == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/1000-08939150' in macro 'hangupcall' == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/1000-08939150' I'd appreciate your help a lot, I'm not if this is a forewall issue or something wrong with my asterisk config. Thanks a lot. -- DAVID GONZALEZ H. GNU/Linux Debian+SuSE+RedHat+LFS TECNICO EN REDES NETWORK ADMIN http://www.computrabajo.com.co/cvs/dgonzalezh ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX gsm bandwith calls
Hi everybody I have 2 asterisk server connected by iax trunk using gsm over a 64Kbps Frame relay circuit, my questions are:whats is bandwith of each call?, and how to limit this on asterisk? Thanks.. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Yikes! Polycom 501 chokes on BootRom 4.0.0?
I use a 650, so YMMV, but it's working with mine. Mike _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Al lists Sent: Wednesday, September 26, 2007 01:19 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Yikes! Polycom 501 chokes on BootRom 4.0.0? One more thing i noticed today, with SIP 2.2 and Polycom 601 i wasnt able to enable buddy watch to use with hints. I'll spend more time on it later to see what is up with that. On 9/25/07, Mike [EMAIL PROTECTED] wrote: I am having a similar issue with 4.0.0. Mine is that it doesn't get any DHCP address (gets stuck waiting for an address). I fixed it by going back one to the previous bootrom version, worked like a charm. Mike -Original Message- From: [EMAIL PROTECTED] [mailto: mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Dave Fullerton Sent: Tuesday, September 25, 2007 08:49 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Yikes! Polycom 501 chokes on BootRom 4.0.0? Doug wrote: I was progressively upgrading this phone from 3.1.2 to 3.2.3, then to 4.0.0. v3.2.3 worked fine, but when I went to 4.0.0 (Even adding the more specific 2345-11500-040.bootrom.ld ), it won't run, and just keeps rebooting. Now I've got a really nice doorstop unless someone knows how to get out of this predicament. Help! 0925003705|cfg |3|00|Beginning to provision phone dns |3|00|DNS 0925003705|lookup for 'somedomain.com'(66.16.26.106) TTL=83485 copy |3|00|' ftp://someuser:[EMAIL PROTECTED]/2345-11500-040.bootrom.ld ftp://someuser:[EMAIL PROTECTED]/2345-11500-040.bootrom.ld' from 'somedomain.com(66.16.26.106)' 0925003706|cfg |3|00|Image 2345-11500-040.bootrom.ld has not changed 0925003706|copy |3|00|Download of '2345-11500-040.bootrom.ld' succeeded on attempt 1 (addr 1 of 1) 0925003706|cfg |3|00|Downloaded bootROM is identical to current 0925003706|version 4.0.0 copy |3|00|'ftp://someuser:[EMAIL PROTECTED]/0004f210.cfg' from 'somedomain.com(66.16.26.106 http://66.16.26.106 )' 0925003707|copy |3|00|Download of '0004f210.cfg' succeeded on attempt 1 (addr 1 of 1) 0925003708|cfg |5|00|Could not get the list of CONFIG_FILES cfg 0925003708||5|00|Could not get the list of MISC_FILES 0925003709|cfg |5|00|Couldn't get parameter APP_FILE_PATH cfg 0925003709||3|00|Unspecified error occured with downloaded application image 0925003709|app1 |6|00|Error in saving application. 0925003709|app1 |6|00|Uploading boot log, time is TUE SEP 25 00:37:10 0925003709|2007 I've upgraded my 501 to bootrom 4.0.0. It did reboot and reformat the filesystem about three times in a row before it finally finished but it did work for me. I'm still using SIP 2.1.2 though. Don't know if that information helps any. -Dave ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] faster timeout in ENUMLOOKUP() function
I need to change my email address for this list but the website is having issues doing that. Can anyone give me another method to accomplish this? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ricardo Carvalho Sent: Wednesday, September 26, 2007 10:32 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] faster timeout in ENUMLOOKUP() function -- This message has been scanned for viruses and dangerous content by Athens Hyperion Scanner, and is believed to be clean. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Yikes! Polycom 501 chokes on BootRom 4.0.0?
At 00:18 9/26/2007, Al lists wrote: One more thing i noticed today, with SIP 2.2 and Polycom 601 i wasnt able to enable buddy watch to use with hints. I'll spend more time on it later to see what is up with that. I guess they still haven't fixed that. The 601 that we have is using: 1.6.7.0098 On 9/25/07, Mike mailto:[EMAIL PROTECTED][EMAIL PROTECTED] wrote: I am having a similar issue with 4.0.0. Mine is that it doesn't get any DHCP address (gets stuck waiting for an address). I fixed it by going back one to the previous bootrom version, worked like a charm. Mike -Original Message- From: mailto:[EMAIL PROTECTED][EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] On Behalf Of Dave Fullerton Sent: Tuesday, September 25, 2007 08:49 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Yikes! Polycom 501 chokes on BootRom 4.0.0? Doug wrote: I was progressively upgrading this phone from 3.1.2 to 3.2.3, then to 4.0.0. v3.2.3 worked fine, but when I went to 4.0.0 (Even adding the more specific 2345-11500-040.bootrom.ld ), it won't run, and just keeps rebooting. Now I've got a really nice doorstop unless someone knows how to get out of this predicament. Help! 0925003705|cfg |3|00|Beginning to provision phone dns |3|00|DNS 0925003705|lookup for 'http://somedomain.comsomedomain.com'(http://66.16.26.10666.16.26.106) TTL=83485 copy |3|00|' ftp://someuser:[EMAIL PROTECTED]/2345-11500-040.bootrom.ld' from 'somedomain.com(http://66.16.26.10666.16.26.106)' 0925003706|cfg |3|00|Image 2345-11500-040.bootrom.ld has not changed 0925003706|copy |3|00|Download of '2345-11500-040.bootrom.ld' succeeded on attempt 1 (addr 1 of 1) 0925003706|cfg |3|00|Downloaded bootROM is identical to current 0925003706|version 4.0.0 copy |3|00|'ftp://someuser:[EMAIL PROTECTED]/0004f210.cfgftp://someuser:[EMAIL PROTECTED]/0004f210.cfg' from 'somedomain.com(http://66.16.26.10666.16.26.106 )' 0925003707|copy |3|00|Download of '0004f210.cfg' succeeded on attempt 1 (addr 1 of 1) 0925003708|cfg |5|00|Could not get the list of CONFIG_FILES cfg 0925003708||5|00|Could not get the list of MISC_FILES 0925003709|cfg |5|00|Couldn't get parameter APP_FILE_PATH cfg 0925003709||3|00|Unspecified error occured with downloaded application image 0925003709|app1 |6|00|Error in saving application. 0925003709|app1 |6|00|Uploading boot log, time is TUE SEP 25 00:37:10 0925003709|2007 I've upgraded my 501 to bootrom 4.0.0. It did reboot and reformat the filesystem about three times in a row before it finally finished but it did work for me. I'm still using SIP 2.1.2 though. Don't know if that information helps any. -Dave ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com--http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-usershttp://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com--http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-usershttp://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Yikes! Polycom 501 chokes on BootRom 4.0.0?
yea thats what i did i put SIP 1.6 and its working like a champ, there should be a way to get it working with 2.2, i'll wait for my next 601 and play with it. On 9/26/07, Doug [EMAIL PROTECTED] wrote: At 00:18 9/26/2007, Al lists wrote: One more thing i noticed today, with SIP 2.2 and Polycom 601 i wasnt able to enable buddy watch to use with hints. I'll spend more time on it later to see what is up with that. I guess they still haven't fixed that. The 601 that we have is using: 1.6.7.0098 On 9/25/07, Mike mailto:[EMAIL PROTECTED][EMAIL PROTECTED] wrote: I am having a similar issue with 4.0.0. Mine is that it doesn't get any DHCP address (gets stuck waiting for an address). I fixed it by going back one to the previous bootrom version, worked like a charm. Mike -Original Message- From: mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] On Behalf Of Dave Fullerton Sent: Tuesday, September 25, 2007 08:49 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Yikes! Polycom 501 chokes on BootRom 4.0.0? Doug wrote: I was progressively upgrading this phone from 3.1.2 to 3.2.3, then to 4.0.0. v3.2.3 worked fine, but when I went to 4.0.0 (Even adding the more specific 2345-11500-040.bootrom.ld ), it won't run, and just keeps rebooting. Now I've got a really nice doorstop unless someone knows how to get out of this predicament. Help! 0925003705|cfg |3|00|Beginning to provision phone dns |3|00|DNS 0925003705|lookup for 'http://somedomain.comsomedomain.com'(http://66.16.26.106 66.16.26.106) TTL=83485 copy |3|00|' ftp://someuser:[EMAIL PROTECTED]/2345-11500-040.bootrom.ld' from 'somedomain.com(http://66.16.26.10666.16.26.106)' 0925003706|cfg |3|00|Image 2345-11500-040.bootrom.ld has not changed 0925003706|copy |3|00|Download of '2345-11500-040.bootrom.ld' succeeded on attempt 1 (addr 1 of 1) 0925003706|cfg |3|00|Downloaded bootROM is identical to current 0925003706|version 4.0.0 copy |3|00|'ftp://someuser:[EMAIL PROTECTED]/0004f210.cfg ftp://someuser:[EMAIL PROTECTED]/0004f210.cfg' from 'somedomain.com(http://66.16.26.10666.16.26.106 )' 0925003707|copy |3|00|Download of '0004f210.cfg' succeeded on attempt 1 (addr 1 of 1) 0925003708|cfg |5|00|Could not get the list of CONFIG_FILES cfg 0925003708||5|00|Could not get the list of MISC_FILES 0925003709|cfg |5|00|Couldn't get parameter APP_FILE_PATH cfg 0925003709||3|00|Unspecified error occured with downloaded application image 0925003709|app1 |6|00|Error in saving application. 0925003709|app1 |6|00|Uploading boot log, time is TUE SEP 25 00:37:10 0925003709|2007 I've upgraded my 501 to bootrom 4.0.0. It did reboot and reformat the filesystem about three times in a row before it finally finished but it did work for me. I'm still using SIP 2.1.2 though. Don't know if that information helps any. -Dave ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com--http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com--http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Slightly OT: Help choosing a free software license?
I'm a little boggled by all the license models one can choose to release a piece of software under. GPL, AFPL, etc? I'm hoping someone can point me to a CLEAR resource that talks about the pros and cons of choosing one over another. All I've found seems to go right over my head. (it's basically a contract, so maybe that's the point) Although I've already made the source to my AstSee Asterisk monitor available, I would like to do it formally, protecting myself in all the ways I am too ignorant to itemize alone. Thanks! Moj Mojo with Horan Company, LLC wrote: Dinesh Nair wrote: On Mon, 10 Sep 2007 13:43:46 -0800, Mojo with Horan Company, LLC wrote: Though still in the proof-of-concept stage, my project AstSee from http://www.astsee.com/ might be fun to play with if you're using linux/XWindows. There are screenshots there. that may be so, but without source, there's no way we can test it on freebsd. i'll stick with fop for the timebeing, thank you. Ok, so source is available now. Do your worst: Innovate! ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music On Hold
So concatenate all the files you've got into one to follow Forrest's suggestion :) Forrest Beck wrote: Make the file the only one in the /var/lib/asterisk/moh directory. Forrest Beck [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] www.shift8.biz On Sep 26, 2007, at 3:07 AM, Joel Hill wrote: Hi All, I need to have the same file played from MoH every time someone gets to MoH from a Dial. I want to play marketing messages from it and I want it to start from file 1 every time. Anyone know if/how this can be done? Cheers, Joel. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Ast_log
Hi all, Anyone know where the asterisk log file is stored? I have some failed calls into my Asterisk box, and I just want to find out why those calls failed. Thnx. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to busy out zap channels
I know this topic came up many months back and some discussions were being had on how to do this within the Zaptel drivers. However, I'm looking for even a crude hack that someone has put together to get this done. We have PRI's and LD T1's that are load balanced on two boxes. The hunt order goes from box to box as far as the spans are concerned. There are times that I would like to busy one out so that calls gradually role to the new box and I can eventually take one out of service. What I was thinking is to create a script that I could tell the specific channels and it would go through and initiate zap calls to an empty meetme. Basically bridging all of the available zap channels on a given span together. Then the trick is monitoring the hangups so that it can initiate a subsequent call immediately following. Once all of the channels in a span have been bridged, I can then bring the box down. Nasty huh? Anyone have a better idea? Or do they have anything like this so I'm not putting it together? Thanks, -Brian ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to busy out zap channels
Very nasty indeed. Through my experience with PRI, the TelCo switchs are not that present to deal with. Your method will work, kind of. However, if the TelCo decides to send you a call during that split second of idle, how are you going to handle it. The best way is still to call your TelCo to take the span down. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian Roy Sent: Wednesday, September 26, 2007 3:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] How to busy out zap channels I know this topic came up many months back and some discussions were being had on how to do this within the Zaptel drivers. However, I'm looking for even a crude hack that someone has put together to get this done. We have PRI's and LD T1's that are load balanced on two boxes. The hunt order goes from box to box as far as the spans are concerned. There are times that I would like to busy one out so that calls gradually role to the new box and I can eventually take one out of service. What I was thinking is to create a script that I could tell the specific channels and it would go through and initiate zap calls to an empty meetme. Basically bridging all of the available zap channels on a given span together. Then the trick is monitoring the hangups so that it can initiate a subsequent call immediately following. Once all of the channels in a span have been bridged, I can then bring the box down. Nasty huh? Anyone have a better idea? Or do they have anything like this so I'm not putting it together? Thanks, -Brian ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Networking Question
I have an Asterisk server running REL 4 with two NICs. One NIC has a 192.168.1.x IP address and is connected to a POE switch with Polycom phones that have 192.168.1.x IP addresses. The other NIC has a 172.17.x.x IP address connected to a router. The router is connected to the Internet. If the Internet goes down or the cable between the 172.17.x.x. NIC is disconnected, the phones start to lose their connection to asterisk. Why? What have I screwed up? Thanks, Brian___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ast_log
Wai Wu wrote: Hi all, Anyone know where the asterisk log file is stored? I have some failed /var/log/asterisk Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Networking Question
Do your phones have the 172.17.x.x as the proxy address? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian M. Arlinghaus Sent: Wednesday, September 26, 2007 4:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Networking Question I have an Asterisk server running REL 4 with two NICs. One NIC has a 192.168.1.x IP address and is connected to a POE switch with Polycom phones that have 192.168.1.x IP addresses. The other NIC has a 172.17.x.x IP address connected to a router. The router is connected to the Internet. If the Internet goes down or the cable between the 172.17.x.x. NIC is disconnected, the phones start to lose their connection to asterisk. Why? What have I screwed up? Thanks, Brian ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ast_log
The Asterisk log file is normally located in /var/log/asterisk But you may want to read your asterisk.conf file to make sure the path in which your system store it. You will see something like this astlogdir = /var/log/asterisk -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wai Wu Sent: Wednesday, September 26, 2007 3:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Ast_log Hi all, Anyone know where the asterisk log file is stored? I have some failed calls into my Asterisk box, and I just want to find out why those calls failed. Thnx. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ChanSpy issue
Hello list I am having an issue with Chanspy/SIP that Im hoping someone has come across and resolved in the past. I am sending calls that come in TDM through T1 ZAP channels and go out to a SIP trunk. If I spy on the SIP channel, I can hear the person on the SIP side of the call just fine, but the person on the ZAP channel fades in and out. If I spy on the ZAP channel, and can hear both sides just fine, but I dont know who I am spying on since I have other calls coming in on the same T1. If I spy on a SIP extension instead of a SIP trunk, I hear both sides just fine. I am using a recent version of Asterisk 1.2 and I am using g729 licenses. This is the command I am using to spy. exten = 8011,1,ChanSpy(Sip/SIPTRUNK|bqv(4)) image001.png___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ast_log
Thanks to all who replied. -Original Message- From: [EMAIL PROTECTED] on behalf of Ed Nuñez Sent: Wed 9/26/2007 4:28 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Ast_log The Asterisk log file is normally located in /var/log/asterisk But you may want to read your asterisk.conf file to make sure the path in which your system store it. You will see something like this astlogdir = /var/log/asterisk -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wai Wu Sent: Wednesday, September 26, 2007 3:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Ast_log Hi all, Anyone know where the asterisk log file is stored? I have some failed calls into my Asterisk box, and I just want to find out why those calls failed. Thnx. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users attachment: winmail.dat___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ChanSpy issue
The parameter to Chanspy should be the whole or part of the channel name. I do not understand what you mean by sip trunk. It make perfect sense that you can hear both streams of voice when you use the phone's extension as Asterisk usually uses SIP/extension+xxx as the channel name of the call. -Original Message- From: [EMAIL PROTECTED] on behalf of Ed Nuñez Sent: Wed 9/26/2007 4:48 PM To: [EMAIL PROTECTED] Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] ChanSpy issue Hello list I am having an issue with Chanspy/SIP that I'm hoping someone has come across and resolved in the past. I am sending calls that come in TDM through T1 ZAP channels and go out to a SIP trunk. If I spy on the SIP channel, I can hear the person on the SIP side of the call just fine, but the person on the ZAP channel fades in and out. If I spy on the ZAP channel, and can hear both sides just fine, but I don't know who I am spying on since I have other calls coming in on the same T1. If I spy on a SIP extension instead of a SIP trunk, I hear both sides just fine. I am using a recent version of Asterisk 1.2 and I am using g729 licenses. This is the command I am using to spy. exten = 8011,1,ChanSpy(Sip/SIPTRUNK|bqv(4)) attachment: image001.png___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music On Hold
Thanks for the suggestion, but I need it to play multiple messages. Always starting with the same one. Cheers, Joel. On Wed, 2007-09-26 at 09:36 -0400, Forrest Beck wrote: Make the file the only one in the /var/lib/asterisk/moh directory. Forrest Beck [EMAIL PROTECTED] www.shift8.biz On Sep 26, 2007, at 3:07 AM, Joel Hill wrote: Hi All, I need to have the same file played from MoH every time someone gets to MoH from a Dial. I want to play marketing messages from it and I want it to start from file 1 every time. Anyone know if/how this can be done? Cheers, Joel. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HOWTO/FAQ question (Location: Sweden)
Yes you can :) that's what asterisk can do. Im running all sip in my asterisk in my 2 call centers. that all SIP On 9/26/07, Turbo Fredriksson [EMAIL PROTECTED] wrote: zoachien == zoachien [EMAIL PROTECTED] writes: zoachien Turbo Fredriksson wrote: How do I connect to a 'normal' (i.e. analog) telephone? zoachien - you can take a voip provider and not buy any hardware. I was thinking in this way to, but I was unsure if I can still use Asterisk in all it's glory (i.e. with all the cool modules like MP3 player, call center stuff etc), or will this be in the hands of the telco? ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Regards, Mark Quitoriano, CCNA Fan the flame... http://www.spreadfirefox.com/?q=user/registerr=19441 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music On Hold
Hi All, I need to have the same file played from MoH every time someone gets to MoH from a Dial. I want to play marketing messages from it and I want it to start from file 1 every time. Anyone know if/how this can be done? On Wed, 2007-09-26 at 09:36 -0400, Forrest Beck wrote: Make the file the only one in the /var/lib/asterisk/moh directory. Forrest Beck [EMAIL PROTECTED] www.shift8.biz Thanks for the suggestion, but I need it to play multiple messages. Always starting with the same one. Cheers, Joel. Create a new MOH class with one large file consisting of every message you want heard, in the order you want them heard. Since there will be only one file, you know which will be first ;) We actually do this with some of our queues, so I know it works. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Networking Question
A few questions for you: Where is your DNS Server for your LAN located by using the 172.17.x.x address I suppose there is more to your network than two segments, (Asterisk may drop connections if it has a problem with DNS) How are your Polycom phones configured? Are they using a ftp/tftp server to get the configs, or are they configured one by one via the web interface? (Once again loosing DNS service if you are using hostnames on the phone configs would cause the phone not to be able to reach the * server) From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian M. Arlinghaus Sent: Wednesday, September 26, 2007 4:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Networking Question I have an Asterisk server running REL 4 with two NICs. One NIC has a 192.168.1.x IP address and is connected to a POE switch with Polycom phones that have 192.168.1.x IP addresses. The other NIC has a 172.17.x.x IP address connected to a router. The router is connected to the Internet. If the Internet goes down or the cable between the 172.17.x.x. NIC is disconnected, the phones start to lose their connection to asterisk. Why? What have I screwed up? Thanks, Brian ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music On Hold
Concatenate the files into one larger file, in the order you want them to play -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Joel Hill Sent: Wednesday, September 26, 2007 7:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Music On Hold Thanks for the suggestion, but I need it to play multiple messages. Always starting with the same one. Cheers, Joel. On Wed, 2007-09-26 at 09:36 -0400, Forrest Beck wrote: Make the file the only one in the /var/lib/asterisk/moh directory. Forrest Beck [EMAIL PROTECTED] www.shift8.biz On Sep 26, 2007, at 3:07 AM, Joel Hill wrote: Hi All, I need to have the same file played from MoH every time someone gets to MoH from a Dial. I want to play marketing messages from it and I want it to start from file 1 every time. Anyone know if/how this can be done? Cheers, Joel. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk realtime error
Hi! I am proving Asterisk 1.2.24 in realtime with MySQL 5.0.27 using Idefisk softphones. I followed the steps of how to of voip-org but always have this error: Sep 25 20:29:07 WARNING[12000]: res_config_mysql.c:360 update_mysql: MySQL RealTime: Failed to query database. Check debug for more info. Sep 25 20:29:07 WARNING[12000]: res_config_mysql.c:360 update_mysql: MySQL RealTime: Failed to query database. Check debug for more info. Sep 25 20:29:07 WARNING[12000]: res_config_mysql.c:360 update_mysql: MySQL RealTime: Failed to query database. Check debug for more info. Sep 25 20:29:07 NOTICE[12000]: chan_iax2.c:5252 register_verify: Host 127.0.0.1 failed MD5 authentication for '101' (9a43a82001dfa49d84e8facb765f7de2 != 31610d29241e861816b83998501ee223) I configure extconfig.conf as: [settings] iaxusers = mysql,asterisk,iax_buddies iaxpeers = mysql,asterisk,iax_buddies sipusers = mysql,asterisk,sip_buddies sippeers = mysql,asterisk,sip_buddies res_mysql.conf as: [general] dbhost = localhost dbname = asterisk dbuser = root dbpass = asterisk dbport = 3306 dbsock = /var/lib/mysql/mysql.sock My table as: CREATE TABLE iax_buddies ( name varchar(30) primary key NOT NULL, username varchar(30), type varchar(6) NOT NULL, secret varchar(50), callerid varchar(100), context varchar(100), host varchar(31) NOT NULL default 'dynamic', disallow varchar(100), allow varchar(100) ); I'm running asterisk on Fedora 6. Plz help thanks in advance Renzzo ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk audits
Hi, Some company asked me to do audits with there asterisk boxes. Is there a standard that i should be following in auditing? anyway can give me a start what to do with asterisk audits? thanks! -- Regards, Mark Quitoriano, CCNA Fan the flame... http://www.spreadfirefox.com/?q=user/registerr=19441 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk realtime error
Could be a mysql permission issue. Try this from the local box: mysql -u root -p enter asterisk as the password use asterisk; select * from sip_buddies; select * from iax_buddies; If you get that far and can see the entries in iax_buddies and sip_buddies, you know it isn't a permissions issue. If you can't, then you know where to look. RENZZO SOTOMAYOR wrote: Hi! I am proving Asterisk 1.2.24 in realtime with MySQL 5.0.27 using Idefisk softphones. I followed the steps of how to of voip-org but always have this error: Sep 25 20:29:07 WARNING[12000]: res_config_mysql.c:360 update_mysql: MySQL RealTime: Failed to query database. Check debug for more info. Sep 25 20:29:07 WARNING[12000]: res_config_mysql.c:360 update_mysql: MySQL RealTime: Failed to query database. Check debug for more info. Sep 25 20:29:07 WARNING[12000]: res_config_mysql.c:360 update_mysql: MySQL RealTime: Failed to query database. Check debug for more info. Sep 25 20:29:07 NOTICE[12000]: chan_iax2.c:5252 register_verify: Host 127.0.0.1 http://127.0.0.1/ failed MD5 authentication for '101' (9a43a82001dfa49d84e8facb765f7d e2 != 31610d29241e861816b83998501ee223) I configure extconfig.conf as: [settings] iaxusers = mysql,asterisk,iax_buddies iaxpeers = mysql,asterisk,iax_buddies sipusers = mysql,asterisk,sip_buddies sippeers = mysql,asterisk,sip_buddies res_mysql.conf as: [general] dbhost = localhost dbname = asterisk dbuser = root dbpass = asterisk dbport = 3306 dbsock = /var/lib/mysql/mysql.sock My table as: CREATE TABLE iax_buddies ( name varchar(30) primary key NOT NULL, username varchar(30), type varchar(6) NOT NULL, secret varchar(50), callerid varchar(100), context varchar(100), host varchar(31) NOT NULL default 'dynamic', disallow varchar(100), allow varchar(100) ); I'm running asterisk on Fedora 6. Plz help thanks in advance Renzzo ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk realtime error
Is your mysql.sock actually in /var/lib/mysql/ ? RENZZO SOTOMAYOR wrote: Hi! I am proving Asterisk 1.2.24 in realtime with MySQL 5.0.27 using Idefisk softphones. I followed the steps of how to of voip-org but always have this error: Sep 25 20:29:07 WARNING[12000]: res_config_mysql.c:360 update_mysql: MySQL RealTime: Failed to query database. Check debug for more info. Sep 25 20:29:07 WARNING[12000]: res_config_mysql.c:360 update_mysql: MySQL RealTime: Failed to query database. Check debug for more info. Sep 25 20:29:07 WARNING[12000]: res_config_mysql.c:360 update_mysql: MySQL RealTime: Failed to query database. Check debug for more info. Sep 25 20:29:07 NOTICE[12000]: chan_iax2.c:5252 register_verify: Host 127.0.0.1 http://127.0.0.1/ failed MD5 authentication for '101' (9a43a82001dfa49d84e8facb765f7d e2 != 31610d29241e861816b83998501ee223) I configure extconfig.conf as: [settings] iaxusers = mysql,asterisk,iax_buddies iaxpeers = mysql,asterisk,iax_buddies sipusers = mysql,asterisk,sip_buddies sippeers = mysql,asterisk,sip_buddies res_mysql.conf as: [general] dbhost = localhost dbname = asterisk dbuser = root dbpass = asterisk dbport = 3306 dbsock = /var/lib/mysql/mysql.sock My table as: CREATE TABLE iax_buddies ( name varchar(30) primary key NOT NULL, username varchar(30), type varchar(6) NOT NULL, secret varchar(50), callerid varchar(100), context varchar(100), host varchar(31) NOT NULL default 'dynamic', disallow varchar(100), allow varchar(100) ); I'm running asterisk on Fedora 6. Plz help thanks in advance Renzzo ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX gsm bandwith calls
If you've got a bandwidth of something that low you'll probably want to use g723.1 or g729 on this line. If your lucky you'll be able to place two calls at once over this link. You won't be able to do anything else though. Tom _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dario Mendez Sent: Wednesday, September 26, 2007 12:10 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] IAX gsm bandwith calls Hi everybody I have 2 asterisk server connected by iax trunk using gsm over a 64Kbps Frame relay circuit, my questions are:whats is bandwith of each call?, and how to limit this on asterisk? Thanks.. No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.488 / Virus Database: 269.13.30/1030 - Release Date: 9/25/2007 8:02 AM No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.488 / Virus Database: 269.13.30/1030 - Release Date: 9/25/2007 8:02 AM ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ChanSpy issue
I am not an expert on chanspy, but it seems to me spying on the trunk would not work very well, would not you hear multiple conversations mixed if more than one extension were calling? Seems best to me to spy on an extension. YOu also can do a show channels to see who is talking to whom. on Wednesday 09/26/2007 Wai Wu([EMAIL PROTECTED]) wrote The parameter to Chanspy should be the whole or part of the channel name. I do not understand what you mean by sip trunk. It make perfect sense that you can hear both streams of voice when you use the phone's extension as Asterisk usually uses SIP/extension+xxx as the channel name of the call. -Original Message- From: [EMAIL PROTECTED] on behalf of Ed Nuñez Sent: Wed 9/26/2007 4:48 PM To: [EMAIL PROTECTED] Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] ChanSpy issue Hello list I am having an issue with Chanspy/SIP that I'm hoping someone has come across and resolved in the past. I am sending calls that come in TDM through T1 ZAP channels and go out to a SIP trunk. If I spy on the SIP channel, I can hear the person on the SIP side of the call just fine, but the person on the ZAP channel fades in and out. If I spy on the ZAP channel, and can hear both sides just fine, but I don't know who I am spying on since I have other calls coming in on the same T1. If I spy on a SIP extension instead of a SIP trunk, I hear both sides just fine. I am using a recent version of Asterisk 1.2 and I am using g729 licenses. This is the command I am using to spy. exten = 8011,1,ChanSpy(Sip/SIPTRUNK|bqv(4)) !DOCTYPE HTML PUBLIC -//W3C//DTD HTML 3.2//EN HTML HEAD META HTTP-EQUIV=Content-Type CONTENT=text/html; charset=iso-8859-1 META NAME=Generator CONTENT=MS Exchange Server version 6.5.7638.1 TITLERE: [asterisk-users] ChanSpy issue/TITLE /HEAD BODY !-- Converted from text/plain format -- PFONT SIZE=2The parameter to Chanspy should be the whole or part of the channel name. I do not understand what you mean by quot;sip trunkquot;. It make perfect sense that you can hear both streams of voice when you use the phone's extension as Asterisk usually uses quot;SIP/extension+xxxquot; as the channel name of the call.BR BR BR -Original Message-BR From: [EMAIL PROTECTED] on behalf of Ed NuñezBR Sent: Wed 9/26/2007 4:48 PMBR To: [EMAIL PROTECTED]BR Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion'BR Subject: Re: [asterisk-users] ChanSpy issueBR BR BR BR Hello listBR BR BR BR I am having an issue with Chanspy/SIP that I'm hoping someone has comeBR across and resolved in the past.BR BR BR BR I am sending calls that come in TDM through T1 ZAP channels and go out to aBR SIP trunk.BR BR BR BR If I spy on the SIP channel, I can hear the person on the SIP side of theBR call just fine, but the person on the ZAP channel fades in and out.BR BR If I spy on the ZAP channel, and can hear both sides just fine, but I don'tBR know who I am spying on since I have other calls coming in on the same T1.BR BR BR BR If I spy on a SIP extension instead of a SIP trunk, I hear both sides justBR fine.BR BR BR BR I am using a recent version of Asterisk 1.2 and I am using g729 licenses.BR BR BR BR This is the command I am using to spy.BR BR BR BR exten =gt; 8011,1,ChanSpy(Sip/SIPTRUNK|bqv(4))BR BR BR BR BR BR BR BR BR BR BR /FONT /P /BODY /HTML___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.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] ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] h.323 out of media path
Hi folks !!! Is there a way to have asterisk out of the media path, when using H.323 ? I mean, it would be better to have something like sip's REINVITE... is that possible? Thanks in advance... -lars ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ChanSpy issue
There is no such thing as a SIP Trunk in Asterisk. Nope. It does not exist. Some people (seems to me mostly GUI people) use the term SIP trunk to mean SIP friend/user/peer. John covici wrote: I am not an expert on chanspy, but it seems to me spying on the trunk would not work very well, would not you hear multiple conversations mixed if more than one extension were calling? Seems best to me to spy on an extension. YOu also can do a show channels to see who is talking to whom. on Wednesday 09/26/2007 Wai Wu([EMAIL PROTECTED]) wrote The parameter to Chanspy should be the whole or part of the channel name. I do not understand what you mean by sip trunk. It make perfect sense that you can hear both streams of voice when you use the phone's extension as Asterisk usually uses SIP/extension+xxx as the channel name of the call. -Original Message- From: [EMAIL PROTECTED] on behalf of Ed Nuñez Sent: Wed 9/26/2007 4:48 PM To: [EMAIL PROTECTED] Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] ChanSpy issue Hello list I am having an issue with Chanspy/SIP that I'm hoping someone has come across and resolved in the past. I am sending calls that come in TDM through T1 ZAP channels and go out to a SIP trunk. If I spy on the SIP channel, I can hear the person on the SIP side of the call just fine, but the person on the ZAP channel fades in and out. If I spy on the ZAP channel, and can hear both sides just fine, but I don't know who I am spying on since I have other calls coming in on the same T1. If I spy on a SIP extension instead of a SIP trunk, I hear both sides just fine. I am using a recent version of Asterisk 1.2 and I am using g729 licenses. This is the command I am using to spy. exten = 8011,1,ChanSpy(Sip/SIPTRUNK|bqv(4)) !DOCTYPE HTML PUBLIC -//W3C//DTD HTML 3.2//EN HTML HEAD META HTTP-EQUIV=Content-Type CONTENT=text/html; charset=iso-8859-1 META NAME=Generator CONTENT=MS Exchange Server version 6.5.7638.1 TITLERE: [asterisk-users] ChanSpy issue/TITLE /HEAD BODY !-- Converted from text/plain format -- PFONT SIZE=2The parameter to Chanspy should be the whole or part of the channel name. I do not understand what you mean by quot;sip trunkquot;. It make perfect sense that you can hear both streams of voice when you use the phone's extension as Asterisk usually uses quot;SIP/extension+xxxquot; as the channel name of the call.BR BR BR -Original Message-BR From: [EMAIL PROTECTED] on behalf of Ed NuñezBR Sent: Wed 9/26/2007 4:48 PMBR To: [EMAIL PROTECTED]BR Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion'BR Subject: Re: [asterisk-users] ChanSpy issueBR BR BR BR Hello listBR BR BR BR I am having an issue with Chanspy/SIP that I'm hoping someone has comeBR across and resolved in the past.BR BR BR BR I am sending calls that come in TDM through T1 ZAP channels and go out to aBR SIP trunk.BR BR BR BR If I spy on the SIP channel, I can hear the person on the SIP side of theBR call just fine, but the person on the ZAP channel fades in and out.BR BR If I spy on the ZAP channel, and can hear both sides just fine, but I don'tBR know who I am spying on since I have other calls coming in on the same T1.BR BR BR BR If I spy on a SIP extension instead of a SIP trunk, I hear both sides justBR fine.BR BR BR BR I am using a recent version of Asterisk 1.2 and I am using g729 licenses.BR BR BR BR This is the command I am using to spy.BR BR BR BR exten =gt; 8011,1,ChanSpy(Sip/SIPTRUNK|bqv(4))BR BR BR BR BR BR BR BR BR BR BR /FONT /P /BODY /HTML___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ChanSpy issue
You are technically correct, its just a shorthand. on Wednesday 09/26/2007 Eric \ManxPower\ Wieling([EMAIL PROTECTED]) wrote There is no such thing as a SIP Trunk in Asterisk. Nope. It does not exist. Some people (seems to me mostly GUI people) use the term SIP trunk to mean SIP friend/user/peer. John covici wrote: I am not an expert on chanspy, but it seems to me spying on the trunk would not work very well, would not you hear multiple conversations mixed if more than one extension were calling? Seems best to me to spy on an extension. YOu also can do a show channels to see who is talking to whom. on Wednesday 09/26/2007 Wai Wu([EMAIL PROTECTED]) wrote The parameter to Chanspy should be the whole or part of the channel name. I do not understand what you mean by sip trunk. It make perfect sense that you can hear both streams of voice when you use the phone's extension as Asterisk usually uses SIP/extension+xxx as the channel name of the call. -Original Message- From: [EMAIL PROTECTED] on behalf of Ed Nuñez Sent: Wed 9/26/2007 4:48 PM To: [EMAIL PROTECTED] Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] ChanSpy issue Hello list I am having an issue with Chanspy/SIP that I'm hoping someone has come across and resolved in the past. I am sending calls that come in TDM through T1 ZAP channels and go out to a SIP trunk. If I spy on the SIP channel, I can hear the person on the SIP side of the call just fine, but the person on the ZAP channel fades in and out. If I spy on the ZAP channel, and can hear both sides just fine, but I don't know who I am spying on since I have other calls coming in on the same T1. If I spy on a SIP extension instead of a SIP trunk, I hear both sides just fine. I am using a recent version of Asterisk 1.2 and I am using g729 licenses. This is the command I am using to spy. exten = 8011,1,ChanSpy(Sip/SIPTRUNK|bqv(4)) !DOCTYPE HTML PUBLIC -//W3C//DTD HTML 3.2//EN HTML HEAD META HTTP-EQUIV=Content-Type CONTENT=text/html; charset=iso-8859-1 META NAME=Generator CONTENT=MS Exchange Server version 6.5.7638.1 TITLERE: [asterisk-users] ChanSpy issue/TITLE /HEAD BODY !-- Converted from text/plain format -- PFONT SIZE=2The parameter to Chanspy should be the whole or part of the channel name. I do not understand what you mean by quot;sip trunkquot;. It make perfect sense that you can hear both streams of voice when you use the phone's extension as Asterisk usually uses quot;SIP/extension+xxxquot; as the channel name of the call.BR BR BR -Original Message-BR From: [EMAIL PROTECTED] on behalf of Ed NuñezBR Sent: Wed 9/26/2007 4:48 PMBR To: [EMAIL PROTECTED]BR Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion'BR Subject: Re: [asterisk-users] ChanSpy issueBR BR BR BR Hello listBR BR BR BR I am having an issue with Chanspy/SIP that I'm hoping someone has comeBR across and resolved in the past.BR BR BR BR I am sending calls that come in TDM through T1 ZAP channels and go out to aBR SIP trunk.BR BR BR BR If I spy on the SIP channel, I can hear the person on the SIP side of theBR call just fine, but the person on the ZAP channel fades in and out.BR BR If I spy on the ZAP channel, and can hear both sides just fine, but I don'tBR know who I am spying on since I have other calls coming in on the same T1.BR BR BR BR If I spy on a SIP extension instead of a SIP trunk, I hear both sides justBR fine.BR BR BR BR I am using a recent version of Asterisk 1.2 and I am using g729 licenses.BR BR BR BR This is the command I am using to spy.BR BR BR BR exten =gt; 8011,1,ChanSpy(Sip/SIPTRUNK|bqv(4))BR BR BR BR BR BR BR BR BR BR BR /FONT /P /BODY /HTML___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.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
[asterisk-users] voip hacking article
http://www.informationweek.com/news/showArticle.jhtml?articleID=20210178 1 Nothing deep and meaningful in the article but worth a read. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Doesn't seem to want to transcode.
Hi all, I've got a .wav file on my asterisk server and I've got an extension that plays it back. When I dial the extension on the local server, it plays back just fine. When I create a call file that calls a (remote) pstn phone number and plays that file, it works just fine, also. But, when I change that call file to use a different voip provider, it tries to play the file, but the person only hears silence. I'm assuming that the second provider is wanting a different codec from the local asterisk server and the first provider. However, I was also expecting asterisk to transcode and do the right thing. What am I missing? TIA, -- Mike Diehl ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk - Spandsp Fax not working?
marco britannio wrote: Hi all, I'm trying to setup an asterisk based fax receiving machine. i'm using asterisk 1.2.18 and app_rxfax with spandsp 0.0.4pre9 I have no problems with a modem-fax, but with the fax machines i have tried almost every fax fails, both in sending and receive. the machines are sending a receiving a lot of faxes every day and working well, so i think the problem is on the spandsp side. i have tried almost every spandsp version from 0.0.2 to the current one, both with and without ECM, but without luck. has anybody succeeded in receiving faxes with asterisk app_rxfax and spandsp? I'm noticicing a lot of different behaviours: sending w ECM gave me an OK, and the second half of the page was missing, other faxes fail with Sep 26 17:26:18 DEBUG[4741] app_rxfax.c: == Sep 26 17:26:18 DEBUG[4741] app_rxfax.c: Fax receive not successful - result (11) Unexpected message received. Sep 26 17:26:18 DEBUG[4741] app_rxfax.c: == can anybody help me? thank you in advance, marco ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Marco, First off, do not use any version over 0.0.3. I am using 0.0.3 on centos 4.5, asterisk 1.2.24 and freepbx 2.3 and it is working very well. One very important thing to keep in mind is that faxing over voip will only work reliably with ulaw or alaw and your internet connection MUST be able to sustain a constant data stream with low jitter. If your interested I have a shell script to install asterisk 1.2.24 and freepbx-2.3 with rxfax and txfax on centos 4 and working on centos 5. Jonn http://jonnt.users.taylortelephone.com/asterisk/centos-asterisk-install.sh and hylafax / iaxmodem http://jonnt.users.taylortelephone.com/asterisk/ ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk - Spandsp Fax not working?
Jonn R Taylor wrote: marco britannio wrote: Hi all, I'm trying to setup an asterisk based fax receiving machine. i'm using asterisk 1.2.18 and app_rxfax with spandsp 0.0.4pre9 I have no problems with a modem-fax, but with the fax machines i have tried almost every fax fails, both in sending and receive. the machines are sending a receiving a lot of faxes every day and working well, so i think the problem is on the spandsp side. i have tried almost every spandsp version from 0.0.2 to the current one, both with and without ECM, but without luck. has anybody succeeded in receiving faxes with asterisk app_rxfax and spandsp? I'm noticicing a lot of different behaviours: sending w ECM gave me an OK, and the second half of the page was missing, other faxes fail with Sep 26 17:26:18 DEBUG[4741] app_rxfax.c: == Sep 26 17:26:18 DEBUG[4741] app_rxfax.c: Fax receive not successful - result (11) Unexpected message received. Sep 26 17:26:18 DEBUG[4741] app_rxfax.c: == can anybody help me? thank you in advance, marco ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Marco, First off, do not use any version over 0.0.3. I am using 0.0.3 on centos 4.5, asterisk 1.2.24 and freepbx 2.3 and it is working very well. One very important thing to keep in mind is that faxing over voip will only work reliably with ulaw or alaw and your internet connection MUST be able to sustain a constant data stream with low jitter. If your interested I have a shell script to install asterisk 1.2.24 and freepbx-2.3 with rxfax and txfax on centos 4 and working on centos 5. Jonn http://jonnt.users.taylortelephone.com/asterisk/centos-asterisk-install.sh and hylafax / iaxmodem http://jonnt.users.taylortelephone.com/asterisk/ ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Oopps, missed the file name. http://jonnt.users.taylortelephone.com/asterisk/iax-hylafax-setup.sh ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] help with channelbank audiocodes MP-124
Hi: We're offering some sort of reward to that one who can help us For this site we are using trixbox and Asterisk 1.2 More info off list. Many thanks, Carlos ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk realtime error
Peder, I have all the permissions in mysql user. I can query my database from the local box. Mik Cheez, yes, it is. mysql.sock is in /var/lib/mysql/ Asterisk and Mysql are in the same PC I still have the same error and don't know what to do. help plz! thanks in advance, Renzzo Mik Cheez wrote: Is your mysql.sock actually in /var/lib/mysql/ ? Peder wrote: Could be a mysql permission issue. Try this from the local box: mysql -u root -p enter asterisk as the password use asterisk; select * from sip_buddies; select * from iax_buddies; If you get that far and can see the entries in iax_buddies and sip_buddies, you know it isn't a permissions issue. If you can't, then you know where to look. RENZZO SOTOMAYOR wrote: Hi! I am proving Asterisk 1.2.24 in realtime with MySQL 5.0.27 using Idefisk softphones. I followed the steps of how to of voip-org but always have this error: Sep 25 20:29:07 WARNING[12000]: res_config_mysql.c:360 update_mysql: MySQL RealTime: Failed to query database. Check debug for more info. Sep 25 20:29:07 WARNING[12000]: res_config_mysql.c:360 update_mysql: MySQL RealTime: Failed to query database. Check debug for more info. Sep 25 20:29:07 WARNING[12000]: res_config_mysql.c:360 update_mysql: MySQL RealTime: Failed to query database. Check debug for more info. Sep 25 20:29:07 NOTICE[12000]: chan_iax2.c:5252 register_verify: Host 127.0.0.1 http://127.0.0.1/ failed MD5 authentication for '101' (9a43a82001dfa49d84e8facb765f7d e2 != 31610d29241e861816b83998501ee223) I configure extconfig.conf as: [settings] iaxusers = mysql,asterisk,iax_buddies iaxpeers = mysql,asterisk,iax_buddies sipusers = mysql,asterisk,sip_buddies sippeers = mysql,asterisk,sip_buddies res_mysql.conf as: [general] dbhost = localhost dbname = asterisk dbuser = root dbpass = asterisk dbport = 3306 dbsock = /var/lib/mysql/mysql.sock My table as: CREATE TABLE iax_buddies ( name varchar(30) primary key NOT NULL, username varchar(30), type varchar(6) NOT NULL, secret varchar(50), callerid varchar(100), context varchar(100), host varchar(31) NOT NULL default 'dynamic', disallow varchar(100), allow varchar(100) ); I'm running asterisk on Fedora 6. Plz help thanks in advance Renzzo ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk audits
On Wednesday 26 September 2007 18:39:31 Mark Quitoriano wrote: Some company asked me to do audits with there asterisk boxes. Is there a standard that i should be following in auditing? anyway can give me a start what to do with asterisk audits? Have you considered the ethics of getting yourself hired to do something you don't know how to do? Worse, have you considered the ramifications of posting to a publically archived list that you got yourself hired to do a job you're unqualified for? -- Tilghman ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to busy out zap channels
Brian: Maybe the CLI command stop gracefully is what are you looking for. Basically, Asterisk will stop receiving incoming calls (of any channel type) and stop itself when all the current calls finish. I hope this help you. Best regards, Tomás. 2007/9/26, Brian Roy [EMAIL PROTECTED]: I know this topic came up many months back and some discussions were being had on how to do this within the Zaptel drivers. However, I'm looking for even a crude hack that someone has put together to get this done. We have PRI's and LD T1's that are load balanced on two boxes. The hunt order goes from box to box as far as the spans are concerned. There are times that I would like to busy one out so that calls gradually role to the new box and I can eventually take one out of service. What I was thinking is to create a script that I could tell the specific channels and it would go through and initiate zap calls to an empty meetme. Basically bridging all of the available zap channels on a given span together. Then the trick is monitoring the hangups so that it can initiate a subsequent call immediately following. Once all of the channels in a span have been bridged, I can then bring the box down. Nasty huh? Anyone have a better idea? Or do they have anything like this so I'm not putting it together? Thanks, -Brian ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk audits
On 9/27/07, Tilghman Lesher [EMAIL PROTECTED] wrote: On Wednesday 26 September 2007 18:39:31 Mark Quitoriano wrote: Some company asked me to do audits with there asterisk boxes. Is there a standard that i should be following in auditing? anyway can give me a start what to do with asterisk audits? Have you considered the ethics of getting yourself hired to do something you don't know how to do? Worse, have you considered the ramifications of posting to a publically archived list that you got yourself hired to do a job you're unqualified for? I agree with what you're saying (personally I wouldn't accept the job), however I think that it's his business whether he accepts it or not. The one who will face the client will be him, not you, so the replies should probably stick to the technical aspects and not ethical matters. - Gonzalo ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
