[asterisk-users] Cubix / Firefly softphone and Asterisk
Hi All Has anyone used Cubix / Firefly successfully with Asterisk? When someone calls a Cubix softphone, Cubix never seems to answer the call correctly. The other person just hears ringing even though it has been answered. I am using IAX as the SIP support doesn't seem to 100% either. Idefisk works 100% on the same setup. Kind Regards Garth ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: SIP stuck channel soft hangup?
On 2006-10-10 20:25:44 -0700, Nic Bellamy [EMAIL PROTECTED] said: On 2006-10-08 21:28:08 -0700, Nic Bellamy [EMAIL PROTECTED] said: I am seeing occasional stuck SIP channels that seem to occur when the fricking Nokia E60 drifts out of WIFI range in the midst of a call. snipI wonder if there is some way to automatically soft hangup these channels when the qualify fails? Take a look at rtptimeout in sip.conf - that might do what you need. Thanks again for the idea Nic! This does seem like a great way to do what I need, but it doesn't seem to work! I have added the statement rtptimeout=60 Into my extension for the Nokia E60. Then I reloaded asterisk. I tried just now to call through my gateway and then walk out of wifi range. The console continues to show me 2 active channels 1 active call, even after the minute (or several minutes) have passed? Any thoughts on why this doesn't work in 1.2.12? Hmm, this should work in 1.2.12 (I think it has for me). I'd recommend watching with tcpdump while you try this, as it's possible that your AP is picking up packets from your E60, but the E60 isn't getting them from the AP - in this case, as Asterisk will still be seeing the RTP, it won't time it out - even though it's dead from a users perpective. I don't think that is the case since the e60 is off the network entirely at that point Can the other end still hear you at this point? No. There was a patch added a couple of months back, but this made it into 1.2.11: http://bugs.digium.com/view.php?id=7459 Depending on the state of the call, it won't always do the job - for instance if you're dialing but not connected, and the other end sends perpetual call progress tones. Asterisk isn't expecting any RTP at this point, so won't be able to do anything about it at this level. Hmmm, unlikely, but could still happen at some point. I think that scenario would timeout though? Even with this, if even one RTP packet gets through in that 60 seconds, it'll reset the timeout. Trying to make this more robust would get tricky, as we don't necessarily know what packetization interval the peer is using, so working on a % lost basis would be quite tricky. /braindump ;-) Thanks I appreciate your insight, and ideas that seem to be pretty close to what I need... Marty ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Understanding NAT Traversal
On 2006-10-10 18:12:23 -0700, hugolivude [EMAIL PROTECTED] said: An Internet browser uses port 80. I might have two or more behind a NAT both using port 80. Isn't that the same thing? Remember that the browser INITIATES all activity on the port 80 transfers. There is no data coming in out of the blue to you browser. This makes it MUCH simpler for you NAT to send the right data to the right machine. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Billing
Dear I am using a2billing accounting software, how can I charge on the destination target not at the caller side Ex: if user A have 10$ and user B have 10$ ,and the onnet call charge cost 1$ When user A call user B for 1 minute ,user A amount remains 10$ and user B amount be 9$ Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Psst... Top secret information: Codename Pineapple
Friends in the Asterisk community, I've been talking for years about the new version of the SIP channel. I've been trying to get funding and get going. Well, the funding part remains to be handled, but I have other news - if you kan keep it to yourself. ...I've began coding. Finally. With a happy smile on my face I removed pedantic=yes the other day. After years of disliking that option it's gone! And srvlookup now defaults to yes in the source code :-) So what is the chan_sip3 project (codename pineapple) about? -- The current SIP channel has many code relationships to the IAX2 channel. Concepts like users, peers and friends doesn't really fit the SIP architecture. The channel supports locally connected phones very well, but is having severe problems being part of a larger SIP infrastructure. Forking, branching and such is not handled, as well as multiple transactions at the same time. The new channel will have configurations for trunks, services and phones. It will be more domain-focused to support multihosting better. It will have a proper SIP state machine so we can handle TCP and TLS alongside UDP. It will have STUN support, like the current Google talk channel. And a lot of other changes... Can I test this now? -- Don't expect this work to be completed yesterday. Right now, I'm cleaning up stuff, moving around variables, splitting up the code in multiple files and grouping variables into structures. When all of that is done, the real work will start. I am expecting to have an experimental version ready for the release of Asterisk *after* the 1.4 release and a more production-ready version ready for the release a year from now. As always with Open Source, the final result depends a lot on the help from the community in testing, providing fixes, development time, funding and additions. Is it available for download? --- The code is hosted in the codename-pineapple branch in the svn server. In that branch, there's a chan_sip.c (version 1) and a chan_sip3.c. As I said: don't expect much yet and don't run this in production! Right now, downloading it is a good way of wasting the bytes on your hard disk drive and not much more. In Q1 2007 I will run an AstriSIPcon developer's meeting to be able to meet everyone that has interest in Asterisk and SIP to test, discuss and work with the new SIP channel. SIP greetings! /Olle PS. A big thank you to Voop AS, who keeps supporting my development work with Asterisk as well as all the students in my training classes that provide development funding by attending the classes. Thanks! --- * Olle E. Johansson - [EMAIL PROTECTED] * Asterisk Training http://edvina.net/training/ * Next class: Stockholm, Sweden November 13-17 2006 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.2.12.1 dies after asterisk -rx and never comes up again
Hi list! I recently upgraded to FreePBX 2.1.3 although I am not sure if this has something to do with it. I do a nightly restart of Asterisk, just in case. This has been working fine months but since a few days asterisk seems to die and I am not able to restart it again, I keep getting a socket in use message. This is on Asterisk 1.2.12.1, Zaptel 1.2.9.1 and Libpri 1.2.3 This is a snippet from the log (full logging enabled) Oct 11 01:05:01 VERBOSE[12923] logger.c: -- Remote UNIX connection Oct 11 01:05:01 VERBOSE[726] logger.c: Beginning asterisk restart Oct 11 01:05:01 VERBOSE[726] logger.c: Executing last minute cleanups Oct 11 01:05:01 VERBOSE[726] logger.c: == Destroying musiconhold processes Oct 11 01:05:01 VERBOSE[726] logger.c: Asterisk cleanly ending (0). Oct 11 01:05:01 VERBOSE[726] logger.c: Preparing for Asterisk restart... Oct 11 01:05:01 VERBOSE[726] logger.c: Restarting Asterisk NOW... Oct 11 01:05:01 VERBOSE[6546] logger.c: Asterisk Event Logger Started /var/log/asterisk/event_log Oct 11 01:05:01 VERBOSE[6546] logger.c: Asterisk Dynamic Loader loading preload modules: Oct 11 01:05:01 VERBOSE[6546] logger.c: == Parsing '/etc/asterisk/modules.conf': Oct 11 01:05:01 VERBOSE[6546] logger.c: = = Parsing '/etc/asterisk/modules.conf': Found Oct 11 01:05:01 VERBOSE[6546] logger.c: == Manager registered action Ping Oct 11 01:05:01 VERBOSE[6546] logger.c: == Manager registered action Events Oct 11 01:05:01 VERBOSE[6546] logger.c: == Manager registered action Logoff Oct 11 01:05:01 VERBOSE[6546] logger.c: == Manager registered action Hangup Oct 11 01:05:01 VERBOSE[6546] logger.c: == Manager registered action Status Oct 11 01:05:01 VERBOSE[6546] logger.c: == Manager registered action Setvar Oct 11 01:05:01 VERBOSE[6546] logger.c: == Manager registered action Getvar Oct 11 01:05:01 VERBOSE[6546] logger.c: == Manager registered action Redirect Oct 11 01:05:01 VERBOSE[6546] logger.c: == Manager registered action Originate Oct 11 01:05:01 VERBOSE[6546] logger.c: == Manager registered action Command Oct 11 01:05:01 VERBOSE[6546] logger.c: == Manager registered action ExtensionState Oct 11 01:05:01 VERBOSE[6546] logger.c: == Manager registered action AbsoluteTimeout Oct 11 01:05:01 VERBOSE[6546] logger.c: == Manager registered action MailboxStatus Oct 11 01:05:01 VERBOSE[6546] logger.c: == Manager registered action MailboxCount Oct 11 01:05:01 VERBOSE[6546] logger.c: == Manager registered action ListCommands Oct 11 01:05:01 VERBOSE[6546] logger.c: == Parsing '/etc/asterisk/manager.conf': Oct 11 01:05:01 VERBOSE[6546] logger.c: = = Parsing '/etc/asterisk/manager.conf': Found Oct 11 01:05:01 WARNING[6546] manager.c: Unable to bind socket: Address already in use Ideas anyone? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Strange FXS disconnection problem.
On Tue, Oct 10, 2006 at 11:43:09PM +0100, David Bath wrote: Hi All, I've tried to find the solution to this, but sadly met with failure. I've got an asterisk box with two X100P's in, and both cards have the same strange problem. After 2min 40seconds (always: within +/- 1sec) they drop an outbound call. Inbound calls are not affected... they stay up as long as required. Do you use busydetect? Any chance that there's a busy tone there? I've seen a fair bit of chatter about similar kinds of problems sometimes being related to callprogess detecting false hang-ups, so I've made sure this is disabled in Zapata.conf but it seems to have had no effect. I was running a slightly older version of zaptel/asterisk (1.2.0) and I've upgraded to the lastest build, but also with no success. Details, pleasse: zapata.conf logs from a call (set verbose to at least 3, and enable full in logger.conf). -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hangup or busy when the peer answer outgoing calls
Hi all!!, I haven't the 'r' options in the dial command. I also try to turn off busydetect and callprocess obtaining the same result.. If I turn off polarityswitch, I get hangup instead busy... The peer isn't busy because I'm trying with my movil phone, and whit known automatic operators from my telephony provider... when they answer my call, asterisk hangup the call.. Regards.. El mar, 10-10-2006 a las 14:39 -0800, Mojo with Horan Company, LLC escribió: If your Dial() cmd has an 'r' in the options, could it be that the ringing you're hearing is asterisk-generated, and the remote side actually is busy? Have you tried turning busydetect=no in zapata.conf? Moj Eloy Gomez wrote: Hi all.. I have a problem with my asterisk installation. I'm using a Wilcard X100P clone in Spain. Incoming calls work fine, but when I make a outgoing call, a hear the ringing, and the peer phone ring, when the peer answer, asterisk hangup the call, or say busy. This is my conf: zaptel.conf: - loadzone = es defaultzone=es fxsks=1 zapata.conf -- [channels] signalling=fxs_ks busydetect=yes answeronpolarityswitch=yes hanguponpolarityswitch=yes callprogress=yes progzone=es context = contexto group = 1 channel = 1 And this is the asterisk log: -- Executing Dial(SIP/200-4803, ZAP/1/966736800|90) in new stack -- Called 1/966736800 -- Hungup 'Zap/1-1' == Everyone is busy/congested at this time (1:0/0/1) -- Executing PlayTones(SIP/200-4803, busy) in new stack -- Executing Wait(SIP/200-4803, 10) in new stack == Spawn extension (indeos, 0966736800, 103) exited non-zero on 'SIP/200-4803' Thanks all Eloy. -- Indeos Consultoria Eloy Gomez ([EMAIL PROTECTED]) Tel: 966787431 www.indeos.es ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Extension and Voice Mail setup
Folks, I am absolutely new to asterisk for the Voice Over IP. I have set up my own server using asterisk, successfully connected and be in position to test the voice over IP by connecting to the digium server and testing the echo system working absolutely fine. My therefore comes, how to work with extensions and voicemail. The documentation that I am using does not provide me with concrete information on working on this. Any assistance will highly be appreciated. Ahmed Ndaula Technology Officer UgaBYTES Initiative http://www.ugabytes.org ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4.0 compile error on AMD64 Opteron server; recompile with -fPIC?
Hi, Installed 1.4.0 libpri and 1.4.0 zaptel and everything went smoothly. I configured asterisk 1.4.0 with no problems (./configure), but when I compile it (make), it fails with this error: [LD] res_snmp.o snmp/agent.o - res_snmp.so/usr/bin/ld: /usr/local/lib/libz.a(gzio.o): relocation R_X86_64_32 against `a local symbol' can not be used when making a shared object; recompile with -fPIC/usr/local/lib/libz.a: could not read symbols: Bad valuecollect2: ld returned 1 exit statusmake[1]: *** [res_snmp.so] Error 1make: *** [res] Error 2 How do I recompile with -fPIC? - Gabriel ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2.12.1 dies after asterisk -rx and never comes up again
On Wed, Oct 11, 2006 at 09:14:55AM +0200, Remco Barendse wrote: Hi list! I recently upgraded to FreePBX 2.1.3 although I am not sure if this has something to do with it. I do a nightly restart of Asterisk, just in case. Why? This has been working fine months but since a few days asterisk seems to die and I am not able to restart it again, I keep getting a socket in use message. = Parsing '/etc/asterisk/manager.conf': Found Oct 11 01:05:01 WARNING[6546] manager.c: Unable to bind socket: Address already in use Asterisk is already running. Probably the wonders , or a misuse of- safe_asterisk. Alternatively, netstat -lntp | grep 5038 -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2.12.1 dies after asterisk -rx and never comes up again
On Wed, 11 Oct 2006, Tzafrir Cohen wrote: On Wed, Oct 11, 2006 at 09:14:55AM +0200, Remco Barendse wrote: Hi list! I recently upgraded to FreePBX 2.1.3 although I am not sure if this has something to do with it. I do a nightly restart of Asterisk, just in case. Why? Sometimes the internet connection is dropped and asterisk doesn't do a dns lookup and provider re-rest quickly enough so all calls are going out via expensive ISDN. Also I sometimes seem to have some trouble after re-loading FreePBX too often I get things like extensions being marked as busy/not available. = Parsing '/etc/asterisk/manager.conf': Found Oct 11 01:05:01 WARNING[6546] manager.c: Unable to bind socket: Address already in use Asterisk is already running. I don't think so, asterisk -r will throw me an error that * is not running. If * would be running I would get console I guess? Probably the wonders , or a misuse of- safe_asterisk. Alternatively, netstat -lntp | grep 5038 Thanks, I'll try that when the problem occurs again Cheers! Remco ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Increase VoiceMail Messages Recording Gain - AudioCalls are Ok
Hi Aaron!Could you please provid me your patch for 1.2? I didn't get you, it was a problem for you to get the messages into mp3 format?Did you have any problem until now with this patch on *1.2 ? My box is 1.2.5 and still very stable until now:)Hope you can help me, i can't figure out why no one though about this has a serious request on *1.2 , as this seems to happen always when you have asterisk behind a legacy pbx with zapata in telephony interface. On 10/11/06, Aaron Daniel [EMAIL PROTECTED] wrote: That doesn't always work :)There's two options... either port the volgain patch from 1.4 to 1.2 (Ifanyone wants a copy, we've been using it for months... however it alsoconverts to mp3 so we'd have to strip that out)... or use 1.4 whichincludes the patch.Let me know if I should post a copy of the older code somewhere.The 1.4 patch is here:http://bugs.digium.com/view.php?id=6237 Aaron DanielComputer Systems TechnicianSam Houston State University[EMAIL PROTECTED](936) 294-4198On Tue, 2006-10-10 at 21:18 -0500, Shawn Kelley wrote: I had the same problem. Checking voicemail via the phone was perfectly normal but the email attachments were so quiet we had to turn the computer volume all the way up along with the speakers amps just to make the attachment understandable. Then just wait until someone forgets to turn the volume back down and a lovely windows message box pops up. Scares the (pick your word) out of everyone in the office! After much searching I found the solution: In the voicemail.conf file change the order in which the recording formats are specified. Asterisk will email the first format in the list. My original line: format=wav49|wav|gsm My new line: format=wav|wav49|gsm NOTE: My understanding is that the wav files are much larger attachments than the wav49 version. However, we haven't noticed much difference, still fairly small attachments. Definitely no problems on a LAN or Broadband connection. From: Marco Mouta [mailto:[EMAIL PROTECTED]] Sent: Tuesday, October 10, 2006 2:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Increase VoiceMail Messages Recording Gain - AudioCalls are Ok Hi all I'm deploying aVoiceMailserver with Asterisk behind a legacy pbx, providing Voicemail to email services for Lecagy PBX extensions. On busy or unanswered calls, Legacy pbx will dial a specific DID (one per extension) to asterisk, and the call is handled by Voicemail application. I've several SIP extensions on this Asterisk box, and calls between Asterisk extensions and legacy PBX are just fine, at least no complaining from users, seems good to me:) The problem is: Right now, and i'm referring only to calls directly handled by VoiceMail application, the users get their audio files in email but the audio is very very low. I've thought about changing RX gain on PRI interface between legacy pbx and asterisk, but until now no complaining with audio calls. I'm afraid that changing this parameter to solve voicemail issues will get me in troubles with Voice Calls . Any advice, or previous similar experience? ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Com os melhores cumprimentos, Marco Mouta ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hicom 150 -- BRI -- Asterisk
Hi,Is is possible to implement this:Hicom150 --- BRI (QSIG) AsteriskI've been reading Siemens documentation and they say:Digital nailed connectionsCorporate communication networks can be implemented over digital S0 or S2M nailed connections between several Hicom systems using the CorNet Nprotocol and between Hicom and non-Siemens systems using the QSig protocol.The systems are linked with each other via public and/or private lines. Does any one ever got this configuration working sucessfully?I'm wondering if it would be possible to communicate via BRI cards using QSIG.In the past i've made this successfully happened but using E1 PRI. thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voicemail app. not working...
Hi * guys, I had a perfectly working * (1.2.0 version). I updated it to 1.2.12 and now VoiceMail app doesn't find entries in voicemail.conf any more. I recompiled only * 1.2.0 and installed it again and now Voicemail is up again, with no configuration's change! Anybody knows anything about this? Regards Mauro ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Strange FXS disconnection problem.
I experienced a similar problem, but with AT-RG 623TX (ISDN BRA gateway). I can only tell you that there is no Asterisk problem. You should try to debug hardware / driver problems. Question: is(are) the user-agent(s) still authenticated with Asterisk after the call is dropped? You should also set the debug level top the highest value. -- Paul Ianas Programming Engineer Level 7 Software Timisoara, 59D Bucovinei phone: 0744137020 email: [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Wednesday, October 11, 2006 10:29 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Strange FXS disconnection problem. On Tue, Oct 10, 2006 at 11:43:09PM +0100, David Bath wrote: Hi All, I've tried to find the solution to this, but sadly met with failure. I've got an asterisk box with two X100P's in, and both cards have the same strange problem. After 2min 40seconds (always: within +/- 1sec) they drop an outbound call. Inbound calls are not affected... they stay up as long as required. Do you use busydetect? Any chance that there's a busy tone there? I've seen a fair bit of chatter about similar kinds of problems sometimes being related to callprogess detecting false hang-ups, so I've made sure this is disabled in Zapata.conf but it seems to have had no effect. I was running a slightly older version of zaptel/asterisk (1.2.0) and I've upgraded to the lastest build, but also with no success. Details, pleasse: zapata.conf logs from a call (set verbose to at least 3, and enable full in logger.conf). -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Extension and Voice Mail setup
Have a look at the book: Asterisk: The future of Telephony. It will teach you almost everything that you need to know. Also you have the wiki (http://voip-info.org) and remember google is your friend. - Original Message - From: Ahmed Ndaula [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, October 11, 2006 10:06 AM Subject: [asterisk-users] Extension and Voice Mail setup Folks, I am absolutely new to asterisk for the Voice Over IP. I have set up my own server using asterisk, successfully connected and be in position to test the voice over IP by connecting to the digium server and testing the echo system working absolutely fine. My therefore comes, how to work with extensions and voicemail. The documentation that I am using does not provide me with concrete information on working on this. Any assistance will highly be appreciated. Ahmed Ndaula Technology Officer UgaBYTES Initiative http://www.ugabytes.org ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] call takeover?
Hi, situation is the following: There's an inbound call, that rings on SIP/tel21 (ATA is PAP2). At the time, bobody there, but a lazy people sits by SIP/tel22 (about 5m distance) and he want to takeover the call. How can I do this whit asterisk? Ok. I can do with call parking, but with call parking on SIP/tel21 must I call the parking extension too, and if nobody picks up the phone, the fax machine (the SIP/tel21) must answer it, and the fax machine can not call the parking extension. ps.: sorry for starting new thread with reply, but I can not send mails to this list otherwise. -- Best regards, Csibra Gergomailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP fails when internet connection lost.
I have been seeing this problem for a long time and it occurs in 1.4.0b2 (as well as 1.2.0-1.2.12.1). If the internet connection is lost and I have SIP services that require me to register, any SIP devices attached to the system stop working. I have an IAX phone connected to one of my servers that I've been having this problem with which will work fine (and filover to the PSTN) the problem is that SIP handsets and softphones can no longer register or make calls. Is this normal behaviour or have I got something wrong with each server? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] WRT54GP2 provisioning
If you are an ITSP provider, you could do with SPC tools (provided by Linksys to ITSPs) Regards Curt Shaffer escribió: Can anyone point me to a good source for provisioning WRT54GP2 from a central server? Thanks Curt ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alberto Sagredo I+D Area (Asterisk // Cisco-Linksys) Peoplecall Email : [EMAIL PROTECTED] Blog: http://www.voipnovatos.es Tel./Ph. : +34 91 120 5080 Tel. Dir./Dir. Ph.: 700 757 139 / 91 120 50 39 Fax./Fax.: +34 91 661 9460 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Understanding NAT Traversal
On Tue, Oct 10, 2006 at 05:03:30PM -0400, hugolivude wrote: I understand how sitting behind a NAT could cause problems for a SIP UA. The SIP UA would create SIP mesages using IP addresses from inside the network (i.e. 192.#.#.# or 10.#.#.#) and these IP addresses are of course unnavigable for the recipient. What I don't get is why don't web browsers suffer the same problem? A web brower behind a NAT sends an HTTP request much the same way as a SIP UA might send an INVITE. Kind of, except: (1) HTTP runs over TCP, SIP runs over UDP. This is not in itself a major issue, because the NAT firewall will keep state open in both cases (so that inbound response packets are de-masqueraded back to the original host). But: (2) All the web content (whether it be HTML, embedded images etc) is pulled back down the same TCP session as requested it in the first place. With a SIP phone, one UDP exchange performs the INVITE signalling, but a separate (unrelated at the IP layer) UDP exchange is used for the actual audio traffic. (3) A web browser is not expected to receive inbound requests from a central server. A SIP client has to receive unsolicited INVITEs for inbound calls. (4) The HTTP request does not include any IP addresses within the request or response. SIP headers and SDP bodies do: e.g. Contact: sip:[EMAIL PROTECTED] This information is invalid on the other side of a NAT, since these addresses are not reachable by the other party. So SIP and NAT do not mix well. There are a host of half-baked solutions which sometimes work and sometimes don't, because even the concept of NAT itself is not well-defined, and NAT implementations differ widely (see RFC 3489 for the gorey details) Probably the most nearly-baked solution is to use a SIP and RTP proxy, such as siproxd, and give it a real public IP address. Roll on the day when all NAT routers have this built in. For more info see: * http://www.voip-info.org/wiki-NAT+and+VOIP * http://www.sipcenter.com/sip.nsf/html/WEBB5YN5GE/$FILE/SIPNATtraversal.pdf * http://siprouter.onsip.org/doc/gettingstarted/ch04s05.html HTH, Brian. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Understanding NAT Traversal
On Tue, Oct 10, 2006 at 05:03:30PM -0400, hugolivude wrote: Similarly, why do we need a timeout on a SIP registration? Does this work the same way as a heartbeat enabling disconnected UA to be unregistered? Yes, that's the purpose: so that if you unplug a SIP phone without giving it a chance to unregister itself, it will eventually be unregistered due to the timeout. (Additionally, some half-baked SIP NAT solutions require you to set a ludicrously short registration timeout, e.g. 20 seconds, just to keep UDP state open on the firewall) Regards, Brian. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Extension and Voice Mail setup
On Wed, Oct 11, 2006 at 04:06:06AM -0400, Ahmed Ndaula wrote: I am absolutely new to asterisk for the Voice Over IP. I have set up my own server using asterisk, successfully connected and be in position to test the voice over IP by connecting to the digium server and testing the echo system working absolutely fine. My therefore comes, how to work with extensions and voicemail. The documentation that I am using does not provide me with concrete information on working on this. Try Asterisk: The Future of Telephony. It has practical dialplan and voicemail examples (chapters 5 and 6). http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 (or buy the printed book) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] user address format
Hello everybody!
[Introduction]
This is a quite long message, but I think the problem is interesting.
[The problem]
Does anyone know how can I tell Asterisk that a certain user
has a certain telephone number (or address)? For example, I have some
registered users, but nor the client (X-lite) nor the server (Asterisk) specifies
what telephone number has the user. I dont want to specify fot each user
2 lines like this in extensions.conf because if I have lets say 200
user-agents, it will be quite time-consuming to introduce/change user info or
make some modifications to the dial plan:
This would be the current solution (user pianas must have
address 102):
=== Settings in sip.conf for user pianas ===
[pianas]
type=friend
username=pianas
secret=somepassword
context=input ;
see below the input context
callerid=Paul Ianas 102
host=dynamic
nat=no
canreinvite=yes
qualify=300
call-limit=10
=== Settings in extensions.conf (for specifying that user
pianas has address 102) ===
; input context
[input]
//other users
exten = 102,1,SetCallerId,${FWDCIDNAME}
exten = 102,2,Dial(SIP/pianas)
//other users
//other extensions
[Some logs from the console]
I have a media gateway (AT-RG 613 TX) where I define a user
(pianas) with address 102. That means user pianas must register with telephone
number 102. If the server is configured with another address (telephone number)
for user pianas, I should get an invalid number log (or something
like that). Here is the response from the server:
Oct 11 14:35:14 NOTICE[7877]: chan_sip.c:11084
handle_request_register: Registration from 'pianas
sip:[EMAIL PROTECTED]' failed for '10.56.74.245' - Username/auth name
mismatch
If I dont specify the number there is no problem in
registering the user (the same settings without the users address
specified):
-- Registered SIP 'pianas' at 10.56.74.245
port 56742 expires 60
And if I dont set the users address and I give
a wrong password, I get the following message:
Oct 11 14:42:36 NOTICE[7877]: chan_sip.c:11084
handle_request_register: Registration from pianas sip:[EMAIL PROTECTED]'
failed for '10.56.74.245' - Wrong password
Please observe the difference between these 2 users: sip:[EMAIL PROTECTED]
and sip:[EMAIL PROTECTED]
[My conclusion]
My conclusion is that Asterisk doesnt know to
register users with specified address (it doesnt recognize this user
address format) or there is some setting in sip.conf that I dont know.
Shouldnt the address be specified in [user]
definition?
[Please help]
Maybe someone had the same problem (Im a newby in
Asterisk) and can give me the solution; this should be a quite basic facility
that a SIP server provides.
10x, and sorry for this long message! J
PS: I didnt find any information about this problem in
Asterisk:
The Future of Telephony
--
Paul Ianas
Programming Engineer
Level 7 Software
Timisoara, 59D
Bucovinei
phone:0744137020
email: [EMAIL PROTECTED]
___
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[asterisk-users] MGCP stuff
Hello everybody! I have an Asterisk 1.2.12.1 server with SIP as the VoIP protocol. What I want to do: I want to talk to the outside world via MGCP. I suppose I must set an MGCP peer to route outgoing calls. So, I must set the endpoint syntax of the Asterisk server (Asterisk will act as an MGCP gateway and will talk with an MGCP Gatekeeper) and with other MGCP gateways via RTP. Ex: DALN/S1/SU0/0@my_address.mydomain.my_dns_suffix Where the part after @ is stored in BTS and contains my telephone number, etc (this is the providers problem). The question: is this possible with Asterisk? Where can I find some documentation for configuring mgcp.conf? The documentation (Asterisk: The Future of Telephony) says MGCP isnt completely developed. 10q! -- Paul Ianas Programming Engineer Level 7 Software Timisoara, 59D Bucovinei phone:0744137020 email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Guest SIP-Invites not accepted
Folks, I hope this is not a FAQ or some other kind of dumb question. I am currently running 1.2.10-BRIstuffed-0.3.0-PRE-1s using a straight-forward configuration mostly only for ISDN. However, I am also accepting anonymous SIP connections for external people calling me. This always worked until yesterday and I am currently trying to find out why it isnt working anymore. This is what is happening: The SIP invite UDP-packet is coming in. I can see it when doing a tcpdump. There is no Netfilter-rule set. Asterisk does not show the packet when doing sip debug nor does Asterisk send any reply. The packet is just repeated from the remote Asterisk for some time until its timing out. So I thought I should use an external SIP-client (softphone) that is authenticating with the Asterisk. This is working. The Asterisk replies properly. So short question: how to debug SIP packets that arrive but do not show up in Asterisk when doing sip debug? Any more details required? If yes, please ask. Thanks a bunch! Sascha ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sending fax with chan-capi
Hi! Has someone ever used the sendfax option of new chan-capi to send fax? I need some help regarding the sff format: How can I generate sff format? I found sfftobmp, not nothing the other way round. Is there a nice way to get the sff out of an Windows application (like virtual printers for hylafax) or at least some scripts which produce the sff and the asterisk call file out of an pdf? thanks klaus ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] user address format
Lets say that I could modify some
stuff in register_verify function (which returns -2 for my request), but I
would also need to modify the sip_request struct and this implies things I dont
know very well.
As I can see, struct sip_peer doesnt
contain any information about user number (telephone number). So, one can not
send number information when registering? L
--
Paul Ianas
Programming Engineer
Level 7 Software
Timisoara, 59D Bucovinei
phone:0744137020
email: [EMAIL PROTECTED]
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul Ianas
Sent: Wednesday, October 11, 2006
3:17 PM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: [asterisk-users] user
address format
Hello everybody!
[Introduction]
This is a quite long message, but I think the problem is
interesting.
[The problem]
Does anyone know how can I tell Asterisk that a certain user
has a certain telephone number (or address)? For example, I have some registered
users, but nor the client (X-lite) nor the server (Asterisk) specifies what
telephone number has the user. I dont want to specify fot each user 2
lines like this in extensions.conf because if I have lets say 200
user-agents, it will be quite time-consuming to introduce/change user info or
make some modifications to the dial plan:
This would be the current solution (user pianas must have
address 102):
=== Settings in sip.conf for user pianas ===
[pianas]
type=friend
username=pianas
secret=somepassword
context=input
; see below the input context
callerid=Paul Ianas 102
host=dynamic
nat=no
canreinvite=yes
qualify=300
call-limit=10
=== Settings in extensions.conf (for specifying that user
pianas has address 102) ===
; input context
[input]
//other users
exten = 102,1,SetCallerId,${FWDCIDNAME}
exten = 102,2,Dial(SIP/pianas)
//other users
//other extensions
[Some logs from the console]
I have a media gateway (AT-RG 613 TX) where I define a user (pianas)
with address 102. That means user pianas must register with telephone number
102. If the server is configured with another address (telephone number) for
user pianas, I should get an invalid number log (or something
like that). Here is the response from the server:
Oct 11 14:35:14 NOTICE[7877]: chan_sip.c:11084
handle_request_register: Registration from 'pianas
sip:[EMAIL PROTECTED]' failed for '10.56.74.245' - Username/auth name
mismatch
If I dont specify the number there is no problem in registering
the user (the same settings without the users address specified):
-- Registered SIP 'pianas' at
10.56.74.245 port 56742 expires 60
And if I dont set the users address and I give
a wrong password, I get the following message:
Oct 11 14:42:36 NOTICE[7877]: chan_sip.c:11084
handle_request_register: Registration from pianas
sip:[EMAIL PROTECTED]' failed for '10.56.74.245' - Wrong password
Please observe the difference between these 2 users:
sip:[EMAIL PROTECTED] and sip:[EMAIL PROTECTED]
[My conclusion]
My conclusion is that Asterisk doesnt know to
register users with specified address (it doesnt recognize this user
address format) or there is some setting in sip.conf that I dont know.
Shouldnt the address be specified in [user]
definition?
[Please help]
Maybe someone had the same problem (Im a newby in
Asterisk) and can give me the solution; this should be a quite basic facility
that a SIP server provides.
10x, and sorry for this long message! J
PS: I didnt find any information about this problem
in Asterisk:
The Future of Telephony
--
Paul Ianas
Programming Engineer
Level 7 Software
Timisoara, 59D
Bucovinei
phone:0744137020
email: [EMAIL PROTECTED]
___
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[asterisk-users] Digium TE405 card and Matra PBX
Hello asterisk-users, I have problem with E1 line between Asterisk computer and our PBX Matra: asta*CLI pri show span 1 Primary D-channel: 16 Status: Provisioned, Down, Active Switchtype: EuroISDN Type: CPE Window Length: 0/7 Sentrej: 0 SolicitFbit: 0 Retrans: 0 Busy: 0 Overlap Dial: 0 T200 Timer: 1000 T203 Timer: 1 T305 Timer: 3 T308 Timer: 4000 T313 Timer: 4000 N200 Counter: 3 My configuration: 2xPIII/1000, 1GB RAM, SCSI disk. Distro Ubuntu Dapper, asterisk version 1.2.7.1.dfsg-2ubuntu3.1, zapata driver version 1.2.5-1, kernel 2.6.15-26-686. My configuration files: /etc/zaptel.conf: span=1,1,0,ccs,hdb3,crc4,yellow bchan=1-15,17-31 dchan=16 loadzone= cz defaultzone = cz /etc/asterisk/zapata.conf: [channels] language=cz context=prichozi switchtype=euroisdn pridialplan=unknown prilocaldialplan=unknown signalling=pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=64 echotraining=800 echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=yes group = 1 channel = 1-15 channel = 17-31 When I tried in asterisk console 'pri intense debug span 1', I've only seen this packets: pri intense debug span 1 Enabled EXTENSIVE debugging on span 1 Sending Set Asynchronous Balanced Mode Extended [ 00 01 7f ] Unnumbered frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 M3: 3 P/F: 1 M2: 3 11: 3 [ SABME (set asynchronous balanced mode extended) ] 0 bytes of data Sending Set Asynchronous Balanced Mode Extended Please, have anyone similar configuration? How I can set this span to synchronize D-channel? PBX is source of time ticks now, but I planning to switch role and to have timesource on the computer... Is there any other possibility? Thanks for your advices. Sincerely Jan Marek -- Ing. Jan Marek | Nez mi poslete prilohu .doc, .xls University of South Bohemia | nebo .ppt, prectete si, prosim, Academic Computer Centre | WWW stranku uvedenou na poslednim Phone: +420-38-9032080 | radku signatury... http://www.gnu.org/philosophy/no-word-attachments.cs.html ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Redefinition of transfer
Hi, I redifined the transfer key in Asterisk 1.2.11 svn from the default # key to ** and when I do a show features in CLI I get: Builtin Feature Default Current --- --- --- Pickup*8 *8 Blind Transfer# ** Attended Transfer *2 One Touch Monitor *1 Disconnect Call * * Also, I have included: include = featuremap in my extensions.conf But when I try to use the transfer feature, I only works on the # key. And in other contexts the # key should be used to signify the end of a recording, but pressing that key activates the transfer. By the way, the attended transfer does not work at all Any ideas are more than welcomed. Thanks for the help John ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sending fax with chan-capi
How can I generate sff format? I found sfftobmp, not nothing the other way round. You can use ghostscript: gs -dNOPAUSE -dBATCH -sDEVICE=cfax -sOutputFile=output.sff input.ps Is there a nice way to get the sff out of an Windows application (like virtual printers for hylafax) or at least some scripts which produce the sff and the asterisk call file out of an pdf? Here's something I use (not Windoze, sorry): http://svn.dataflake.org/filedetails.php? repname=DataflakeSoftwarepath=%2Fasterisk-chancapi-faxscripts% 2Ffile2fax.pyrev=0sc=0 The script takes TIFF, PS or PDF as input, creates SFF and a call file. It is run out of cron and checks if suitable files have been dropped into a spool directory. The whole package at http://svn.dataflake.org/listing.php? repname=DataflakeSoftwarepath=%2Fasterisk-chancapi-faxscripts% 2Frev=0sc=0 contains some documentation and also a script that I use to handle incoming faxes (with capicommand receivefax). jens ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2.12.1 dies after asterisk -rx and never comes up again
I quits on my as well, when I try to make a second call. There is a bug report on it: http://bugs.digium.com/view.php?id=7972 -- #Joseph On Wed, 2006-10-11 at 09:14 +0200, Remco Barendse wrote: Hi list! I recently upgraded to FreePBX 2.1.3 although I am not sure if this has something to do with it. I do a nightly restart of Asterisk, just in case. This has been working fine months but since a few days asterisk seems to die and I am not able to restart it again, I keep getting a socket in use message. This is on Asterisk 1.2.12.1, Zaptel 1.2.9.1 and Libpri 1.2.3 This is a snippet from the log (full logging enabled) Oct 11 01:05:01 VERBOSE[12923] logger.c: -- Remote UNIX connection Oct 11 01:05:01 VERBOSE[726] logger.c: Beginning asterisk restart Oct 11 01:05:01 VERBOSE[726] logger.c: Executing last minute cleanups Oct 11 01:05:01 VERBOSE[726] logger.c: == Destroying musiconhold processes Oct 11 01:05:01 VERBOSE[726] logger.c: Asterisk cleanly ending (0). Oct 11 01:05:01 VERBOSE[726] logger.c: Preparing for Asterisk restart... Oct 11 01:05:01 VERBOSE[726] logger.c: Restarting Asterisk NOW... Oct 11 01:05:01 VERBOSE[6546] logger.c: Asterisk Event Logger Started /var/log/asterisk/event_log Oct 11 01:05:01 VERBOSE[6546] logger.c: Asterisk Dynamic Loader loading preload modules: Oct 11 01:05:01 VERBOSE[6546] logger.c: == Parsing '/etc/asterisk/modules.conf': Oct 11 01:05:01 VERBOSE[6546] logger.c: = = Parsing '/etc/asterisk/modules.conf': Found Oct 11 01:05:01 VERBOSE[6546] logger.c: == Manager registered action Ping Oct 11 01:05:01 VERBOSE[6546] logger.c: == Manager registered action Events Oct 11 01:05:01 VERBOSE[6546] logger.c: == Manager registered action Logoff Oct 11 01:05:01 VERBOSE[6546] logger.c: == Manager registered action Hangup Oct 11 01:05:01 VERBOSE[6546] logger.c: == Manager registered action Status Oct 11 01:05:01 VERBOSE[6546] logger.c: == Manager registered action Setvar Oct 11 01:05:01 VERBOSE[6546] logger.c: == Manager registered action Getvar Oct 11 01:05:01 VERBOSE[6546] logger.c: == Manager registered action Redirect Oct 11 01:05:01 VERBOSE[6546] logger.c: == Manager registered action Originate Oct 11 01:05:01 VERBOSE[6546] logger.c: == Manager registered action Command Oct 11 01:05:01 VERBOSE[6546] logger.c: == Manager registered action ExtensionState Oct 11 01:05:01 VERBOSE[6546] logger.c: == Manager registered action AbsoluteTimeout Oct 11 01:05:01 VERBOSE[6546] logger.c: == Manager registered action MailboxStatus Oct 11 01:05:01 VERBOSE[6546] logger.c: == Manager registered action MailboxCount Oct 11 01:05:01 VERBOSE[6546] logger.c: == Manager registered action ListCommands Oct 11 01:05:01 VERBOSE[6546] logger.c: == Parsing '/etc/asterisk/manager.conf': Oct 11 01:05:01 VERBOSE[6546] logger.c: = = Parsing '/etc/asterisk/manager.conf': Found Oct 11 01:05:01 WARNING[6546] manager.c: Unable to bind socket: Address already in use Ideas anyone? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2.12.1 dies after asterisk -rx and never comes up again
On 10/11/06 21:15 Joseph said the following: I quits on my as well, when I try to make a second call. There is a bug report on it: http://bugs.digium.com/view.php?id=7972 this seems like a configuration error within FreePBX and isnt really a bug in asterisk. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] NFAS Not Passing Audio on B-chan 48,72,96
I have NFAS setup on several quad port T1 cards (Sangoma). It mostly works well with the exception that calls coming in on channels 48,72, and 96 have no audio. I tried removing these channels from zapata.conf with hopes that the channels would not come up or be used. Now I get Ring requested on unconfigured channel. How can I busyout these these channels so that incoming calls are not sent to them, or how can I fix the real problem? I think it may be a Sangoma/Wanpipe configuration issue. Thanks, Steve Totaro ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sending fax with chan-capi
Hi Jens! Thanks for the script. Do you generate and notifications (succeeded, failed) or retransmit in case of failed sending? Or does that CAPI internally? regards klaus Jens Vagelpohl wrote: How can I generate sff format? I found sfftobmp, not nothing the other way round. You can use ghostscript: gs -dNOPAUSE -dBATCH -sDEVICE=cfax -sOutputFile=output.sff input.ps Is there a nice way to get the sff out of an Windows application (like virtual printers for hylafax) or at least some scripts which produce the sff and the asterisk call file out of an pdf? Here's something I use (not Windoze, sorry): http://svn.dataflake.org/filedetails.php?repname=DataflakeSoftwarepath=%2Fasterisk-chancapi-faxscripts%2Ffile2fax.pyrev=0sc=0 The script takes TIFF, PS or PDF as input, creates SFF and a call file. It is run out of cron and checks if suitable files have been dropped into a spool directory. The whole package at http://svn.dataflake.org/listing.php?repname=DataflakeSoftwarepath=%2Fasterisk-chancapi-faxscripts%2Frev=0sc=0 contains some documentation and also a script that I use to handle incoming faxes (with capicommand receivefax). jens ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] cisco 7960 not registering after * restart
Hello, When I restart asterisk the cisco 7960/7940 phones (sip fw 7.5) fail to re-register themselves with asterisk, even though I put timer_register_expires: 60 in SIPDefault.cnf Is there a way to have these phones register themselves every 60 seconds? Alternatively, can asterisk be made to remember its dynamic sip hosts' registration after a restart? Thanks, ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Increase VoiceMail Messages Recording Gain - AudioCalls are Ok
I've uploaded a patch to my host, it only does the volgain in int format (we use +7 which seems to work well). We've had no problems with it since we set it up back in February, and everyone seems to love it since nobody's blowing out their speakers anymore lol. The patch we use actually does a number of things. We convert from WAV to mp3 for better client support (i.e. my boss used his pda phone to listen to an mp3 voicemail), and we also change the From field of the email to come from the user leaving the voicemail instead of the server email. I think that's it, the file's posted here: http://asterisk.mdaniel.net/?p=5 Check it out, let me know if it works for ya'll. Aaron On Wed, 2006-10-11 at 10:49 +0100, Marco Mouta wrote: Hi Aaron! Could you please provid me your patch for 1.2? I didn't get you, it was a problem for you to get the messages into mp3 format? Did you have any problem until now with this patch on *1.2 ? My box is 1.2.5 and still very stable until now:) Hope you can help me, i can't figure out why no one though about this has a serious request on *1.2 , as this seems to happen always when you have asterisk behind a legacy pbx with zapata in telephony interface. On 10/11/06, Aaron Daniel [EMAIL PROTECTED] wrote: That doesn't always work :) There's two options... either port the volgain patch from 1.4 to 1.2 (If anyone wants a copy, we've been using it for months... however it also converts to mp3 so we'd have to strip that out)... or use 1.4 which includes the patch. Let me know if I should post a copy of the older code somewhere. The 1.4 patch is here: http://bugs.digium.com/view.php?id=6237 Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 On Tue, 2006-10-10 at 21:18 -0500, Shawn Kelley wrote: I had the same problem. Checking voicemail via the phone was perfectly normal but the email attachments were so quiet we had to turn the computer volume all the way up along with the speakers amps just to make the attachment understandable. Then just wait until someone forgets to turn the volume back down and a lovely windows message box pops up. Scares the (pick your word) out of everyone in the office! After much searching I found the solution: In the voicemail.conf file change the order in which the recording formats are specified. Asterisk will email the first format in the list. My original line: format=wav49|wav|gsm My new line: format=wav|wav49|gsm NOTE: My understanding is that the wav files are much larger attachments than the wav49 version. However, we haven't noticed much difference, still fairly small attachments. Definitely no problems on a LAN or Broadband connection. From: Marco Mouta [mailto:[EMAIL PROTECTED] Sent: Tuesday, October 10, 2006 2:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Increase VoiceMail Messages Recording Gain - AudioCalls are Ok Hi all I'm deploying a VoiceMailserver with Asterisk behind a legacy pbx, providing Voicemail to email services for Lecagy PBX extensions. On busy or unanswered calls, Legacy pbx will dial a specific DID (one per extension) to asterisk, and the call is handled by Voicemail application. I've several SIP extensions on this Asterisk box, and calls between Asterisk extensions and legacy PBX are just fine, at least no complaining from users, seems good to me:) The problem is: Right now, and i'm referring only to calls directly handled by VoiceMail application, the users get their audio files in email but the audio is very very low. I've thought about changing RX gain on PRI interface between legacy pbx and asterisk, but until now no complaining with audio calls. I'm afraid that changing this parameter to solve voicemail issues will get me in troubles with Voice Calls . Any advice, or previous similar experience? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users
Re: [asterisk-users] NFAS Not Passing Audio on B-chan 48,72,96
Steve Totaro wrote: I have NFAS setup on several quad port T1 cards (Sangoma). It mostly works well with the exception that calls coming in on channels 48,72, and 96 have no audio. I tried removing these channels from zapata.conf with hopes that the channels would not come up or be used. Now I get Ring requested on unconfigured channel. How can I busyout these these channels so that incoming calls are not sent to them, or how can I fix the real problem? I think it may be a Sangoma/Wanpipe configuration issue. Thanks, Steve Totaro Steve, I take it you have one D channel for all four spans? On 24? I think this should be pretty transparent to wanpipe. You should configure four spans, with one channel group (type TDM) per span (obvious). You should also probably disable any native D channel features. I always have nothing but problems with that. In zaptel.conf, only specify one d channel. The tricky stuff is in zapata.conf. Can you post that, and maybe zaptel.conf too? -- Kristian Kielhofner ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk + E1 with MFC/R2 in Argentina?
Hello, Has somebody installed this configuration: Asterisk + E1 with MFC/R2 (Telefónica Argentina) in Argentina before? I need to know if it´s possible with MFC/R2 argentine variation. Thanks in advance. R.R. Libera ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RE: Welcome to the asterisk-users mailing list
Most want the 2.0.1 firmware for a few reasons:A) They have the latest; although, this is a poor reason, it's still a reason people download and use the latest firmware - remember here always, "If it's not broken, DON'T fix it!"B) They are hoping to fix a previous problem they've had in the past (i.e. stability issues - usually caused by other factors besides just firmware)C) They are told to. IF you are needing to talk to a support professional, especially Polycom, you NEED to upgrade to the latest firmware or they simply will not help you (most of the times anyways)These are a few I can think of anyways; and unfortunately, it's going to be a problem sooner or later. Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products? Visit our VoIP store at http://voipstore.atacomm.com/ On Oct 10, 2006, at 4:56 PM, Eric ManxPower Wieling wrote:What I don't understand is why people MUST use the 2.0.x firmware.Jessee J Holmes wrote: A few of our technical support staff here at Atacomm are currently working on this issue with Polycom, Digium and one of our customer's (who posted in here earlier this week).There are some major differences from the 1.x.x firmware and the 2.01 firmware. Obviously, many on here, voip-info.org, and all over the Internet have reported troubles where their presence feature that stops working after they upgrade their Polycom phones to firmware revision 2.0.1 when the phones are configured with an Asterisk system. A couple new things have been added to the .cfg files that MUST now be set in order for presence to work again with an Asterisk system.In the sip.cfg file:feature feature.1.name="presence" feature.1.enabled="1"In the phone[mac].cfg file:attendant attendant.uri="[EMAIL PROTECTED]" attendant.reg=""/However, there are still confirmed problems with this setup (i.e. LEDs not working), which Polycom and ourselves are currently testing in our labs trying to fix. The reasoning for this is Digium doesn't seem to follow the "standards" for presence support and are currently working to change this functionality within Asterisk. Polycom designed their phones, and specifically their firmware, to work with the "standards" (more specifically, the Microsoft LCS - Live Communications Server).This issue has been reported on multiple instances to Polycom, Digium, and ourselves; but, no real resolution is completed yet. We'll continue working on this issue within our labs and post an official answer when one is available.Sorry for the bit of bad news, if anyone is willing to contribute working / half working code, we'd be more than happy to look at it and work with Polycom and Digium on getting this fixed for everyone AS ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cisco 7960 not registering after * restart
That's a bug with the 7.5 firmware. I would suggest upgrading to the 8.4 version, we've been running it for a few weeks in a test environment and everyone's been pretty satisfied with the new firmware (read: nobody's complained). If the server goes out, they re-register after the timeout without problems. Aaron On Wed, 2006-10-11 at 15:35 +0200, Louis-David Mitterrand wrote: Hello, When I restart asterisk the cisco 7960/7940 phones (sip fw 7.5) fail to re-register themselves with asterisk, even though I put timer_register_expires: 60 in SIPDefault.cnf Is there a way to have these phones register themselves every 60 seconds? Alternatively, can asterisk be made to remember its dynamic sip hosts' registration after a restart? Thanks, ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RE: Welcome to the asterisk-users mailing list
I'm not saying Microsoft is the standard (they usually aren't by FAR), but how Microsoft handles presence and interoperates with presence on various IP phones is what Polycom calls a "standard" (guess I should have quoted that word originally).I believe there is some RFC for presence out there that some people consider the "standard"; although, I'm not sure what this is... Saying the word standard to me is like saying that someone is "normal" . there is no such thing. It's normally just something that "most" people agree on as a standard. Anyways, some of us here at Atacomm are currently arguing with Polycom why they can't make their phones support BOTH "methods" of handling presence, we think that would be the simplest solution instead of just shutting out some of the systems like Asterisk. Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products? Visit our VoIP store at http://voipstore.atacomm.com/ On Oct 10, 2006, at 6:05 PM, Mike Clark wrote:Jessee J Holmes wrote:snipped However, there are still confirmed problems with this setup (i.e. LEDsnot working), which Polycom and ourselves are currently testing in ourlabs trying to fix. The reasoning for this is Digium doesn't seem tofollow the "standards" for presence support and are currently working tochange this functionality within Asterisk. Polycom designed theirphones, and specifically their firmware, to work with the "standards"(more specifically, the Microsoft LCS - Live Communications Server). What are the specific standards to which you refer? To my knowledge,Microsoft LCS is *not* an industry adopted standard.___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX2 outgoing calls delayed before they connect
Hi, everybody: I have just set up a system with a regional VOIP provider. I have two IAX channels to this provider. Incoming calls ring a configured SIP extension immediately, but outgoing calls are delayed for about 8 to 10 seconds before the remote PSTN end starts ringing: -- Called [IAX2 channel] -- Call accepted by [IAX2 provider IP] (format ulaw) -- Format for call is ulaw -- IAX2/[channel ID] is making progress passing it to SIP/polycom -- Hungup 'IAX2/[channel ID]' The first three steps happen instantly; between Format... and IAX2/[channel ID]... there is a delay of about 8 - 10 seconds. The calling party hears nothing during this time. Is the source of this problem local, or should I be bugging our provider about this? Thanks, -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RE: Welcome to the asterisk-users mailing list
Dean,Try obtaining the latest bootrom again, should be 3.2.2, we've seen this happen before for various odd reasons and Polycom's recommended fix is get the "non-engineering version" of the bootrom (don't ask please, just do it).So download the bootrom again and attempt it once more, while you're at it, be safe and get the 2.0.1 firmware again as well. Let me know if this fixes you're problem. By the way, are you using IP 500's? or 501's? 500's may not take, I think we had that discussion in this list before. Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products? Visit our VoIP store at http://voipstore.atacomm.com/ On Oct 10, 2006, at 7:10 PM, Dean Collins wrote:I've had problems loading 2.01 onto 2 of my 4 polycom 500's2 work great no probs, 2 I cant get it to upload without failing.Cheers,Dean -Original Message-From: [EMAIL PROTECTED] [mailto:asterisk-users-[EMAIL PROTECTED]] On Behalf Of Eric "ManxPower" WielingSent: Tuesday, 10 October 2006 5:56 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [asterisk-users] RE: Welcome to the "asterisk-users" mailing listWhat I don't understand is why people MUST use the 2.0.x firmware. ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RE: Welcome to the asterisk-users mailing list
Limit was increased in firmware 2.0.1.NOTE: a new Polycom Administrator's guide is now also available covering the 2.0.1 features. Re-obtain this manual if you haven't from your reseller or from Polycom direct if you're certified. Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products? Visit our VoIP store at http://voipstore.atacomm.com/ On Oct 11, 2006, at 12:01 AM, Douglas Garstang wrote:I think that limit was increased in 1.6.6 or 1.6.7. -Original Message- From: C F [mailto:[EMAIL PROTECTED]] Sent: Tue 10/10/2006 6:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [asterisk-users] RE: Welcome to the "asterisk-users" mailing list On 10/10/06, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: What I don't understand is why people MUST use the 2.0.x firmware. I believe it's because the limit of how many can be monitored at once, someone correct me if I'm wrong. Jessee J Holmes wrote: A few of our technical support staff here at Atacomm are currently working on this issue with Polycom, Digium and one of our customer's (who posted in here earlier this week). There are some major differences from the 1.x.x firmware and the 2.01 firmware. Obviously, many on here, voip-info.org, and all over the Internet have reported troubles where their presence feature that stops working after they upgrade their Polycom phones to firmware revision 2.0.1 when the phones are configured with an Asterisk system. A couple new things have been added to the .cfg files that MUST now be set in order for presence to work again with an Asterisk system. In the sip.cfg file: feature feature.1.name="presence" feature.1.enabled="1" In the phone[mac].cfg file: attendant attendant.uri="[EMAIL PROTECTED]" attendant.reg=""/ However, there are still confirmed problems with this setup (i.e. LEDs not working), which Polycom and ourselves are currently testing in our labs trying to fix. The reasoning for this is Digium doesn't seem to follow the "standards" for presence support and are currently working to change this functionality within Asterisk. Polycom designed their phones, and specifically their firmware, to work with the "standards" (more specifically, the Microsoft LCS - Live Communications Server). This issue has been reported on multiple instances to Polycom, Digium, and ourselves; but, no real resolution is completed yet. We'll continue working on this issue within our labs and post an official answer when one is available. Sorry for the bit of bad news, if anyone is willing to contribute working / half working code, we'd be more than happy to look at it and work with Polycom and Digium on getting this fixed for everyone AS ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] NFAS Not Passing Audio on B-chan 48,72,96
Kristian Kielhofner wrote: Steve Totaro wrote: I have NFAS setup on several quad port T1 cards (Sangoma). It mostly works well with the exception that calls coming in on channels 48,72, and 96 have no audio. I tried removing these channels from zapata.conf with hopes that the channels would not come up or be used. Now I get Ring requested on unconfigured channel. How can I busyout these these channels so that incoming calls are not sent to them, or how can I fix the real problem? I think it may be a Sangoma/Wanpipe configuration issue. Thanks, Steve Totaro Steve, I take it you have one D channel for all four spans? On 24? I think this should be pretty transparent to wanpipe. You should configure four spans, with one channel group (type TDM) per span (obvious). You should also probably disable any native D channel features. I always have nothing but problems with that. In zaptel.conf, only specify one d channel. The tricky stuff is in zapata.conf. Can you post that, and maybe zaptel.conf too? -- Kristian Kielhofner Yes the D chan is 24 with no backup. Before I had channel = 1-23,25-96. [trunkgroups] trunkgroup = 1,24 spanmap = 1,1,0 spanmap = 2,1,1 spanmap = 3,1,2 spanmap = 4,1,3 [channels] resetinterval=never callerid=asreceived usecallerid=yes hidecallerid=no callwaiting=no usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes callreturn=yes echocancel=no echocancelwhenbridged=no ;echotraining=800 group=0 callgroup=1 pickupgroup=1 immediate=no rxgain=0.0 txgain=0.0 context=from-pstn switchtype=5ess signalling=pri_cpe channel = 1-23,25-47,49-71,73-95 span=1,0,0,esf,b8zs span=2,0,0,esf,b8zs span=3,0,0,esf,b8zs span=4,0,0,esf,b8zs bchan=1-23,25-96 dchan=24 loadzone=us defaultzone=us ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Increase VoiceMail Messages Recording Gain -AudioCalls are Ok
externnotify=/opt/asterisk-1.2.7.1/sbin/vm-audio vm-audio uses 'sox -e' to determine how much to scale by without clipping and then Then 'sox -v' to scale the sound file. This happens after the email message is sent, but by changing the order of a few lines in the app_voicemail.c program you can have the externnotify run before the email message is sent. Cullin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Aaron Daniel Sent: Wednesday, October 11, 2006 12:49 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Increase VoiceMail Messages Recording Gain -AudioCalls are Ok That doesn't always work :) There's two options... either port the volgain patch from 1.4 to 1.2 (If anyone wants a copy, we've been using it for months... however it also converts to mp3 so we'd have to strip that out)... or use 1.4 which includes the patch. Let me know if I should post a copy of the older code somewhere. The 1.4 patch is here: http://bugs.digium.com/view.php?id=6237 Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 On Tue, 2006-10-10 at 21:18 -0500, Shawn Kelley wrote: I had the same problem. Checking voicemail via the phone was perfectly normal but the email attachments were so quiet we had to turn the computer volume all the way up along with the speakers amps just to make the attachment understandable. Then just wait until someone forgets to turn the volume back down and a lovely windows message box pops up. Scares the (pick your word) out of everyone in the office! After much searching I found the solution: In the voicemail.conf file change the order in which the recording formats are specified. Asterisk will email the first format in the list. My original line: format=wav49|wav|gsm My new line: format=wav|wav49|gsm NOTE: My understanding is that the wav files are much larger attachments than the wav49 version. However, we haven't noticed much difference, still fairly small attachments. Definitely no problems on a LAN or Broadband connection. From: Marco Mouta [mailto:[EMAIL PROTECTED] Sent: Tuesday, October 10, 2006 2:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Increase VoiceMail Messages Recording Gain - AudioCalls are Ok Hi all I'm deploying a VoiceMailserver with Asterisk behind a legacy pbx, providing Voicemail to email services for Lecagy PBX extensions. On busy or unanswered calls, Legacy pbx will dial a specific DID (one per extension) to asterisk, and the call is handled by Voicemail application. I've several SIP extensions on this Asterisk box, and calls between Asterisk extensions and legacy PBX are just fine, at least no complaining from users, seems good to me:) The problem is: Right now, and i'm referring only to calls directly handled by VoiceMail application, the users get their audio files in email but the audio is very very low. I've thought about changing RX gain on PRI interface between legacy pbx and asterisk, but until now no complaining with audio calls. I'm afraid that changing this parameter to solve voicemail issues will get me in troubles with Voice Calls . Any advice, or previous similar experience? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Fwd: Re: [asterisk-users] NFAS Not Passing Audio on B-chan 48, 72, 96]
Kristian Kielhofner wrote: Steve Totaro wrote: I have NFAS setup on several quad port T1 cards (Sangoma). It mostly works well with the exception that calls coming in on channels 48,72, and 96 have no audio. I tried removing these channels from zapata.conf with hopes that the channels would not come up or be used. Now I get Ring requested on unconfigured channel. How can I busyout these these channels so that incoming calls are not sent to them, or how can I fix the real problem? I think it may be a Sangoma/Wanpipe configuration issue. Thanks, Steve Totaro Steve, I take it you have one D channel for all four spans? On 24? I think this should be pretty transparent to wanpipe. You should configure four spans, with one channel group (type TDM) per span (obvious). You should also probably disable any native D channel features. I always have nothing but problems with that. In zaptel.conf, only specify one d channel. The tricky stuff is in zapata.conf. Can you post that, and maybe zaptel.conf too? -- Kristian Kielhofner Yes the D chan is 24 with no backup. Before I had channel = 1-23,25-96. [trunkgroups] trunkgroup = 1,24 spanmap = 1,1,0 spanmap = 2,1,1 spanmap = 3,1,2 spanmap = 4,1,3 [channels] resetinterval=never callerid=asreceived usecallerid=yes hidecallerid=no callwaiting=no usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes callreturn=yes echocancel=no echocancelwhenbridged=no ;echotraining=800 group=0 callgroup=1 pickupgroup=1 immediate=no rxgain=0.0 txgain=0.0 context=from-pstn switchtype=5ess signalling=pri_cpe channel = 1-23,25-47,49-71,73-95 span=1,0,0,esf,b8zs span=2,0,0,esf,b8zs span=3,0,0,esf,b8zs span=4,0,0,esf,b8zs bchan=1-23,25-96 dchan=24 loadzone=us defaultzone=us ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] RE: Welcome to the asterisk-users mailing list
We must have had the magic version of 1.6.x then, because we increased our buddy watch limit from 8 to 48 in that version. -Original Message-From: Jessee J Holmes [mailto:[EMAIL PROTECTED]Sent: Wednesday, October 11, 2006 8:18 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [asterisk-users] RE: Welcome to the "asterisk-users" mailing listLimit was increased in firmware 2.0.1. NOTE: a new Polycom Administrator's guide is now also available covering the 2.0.1 features. Re-obtain this manual if you haven't from your reseller or from Polycom direct if you're certified. Jessee Holmes Atacomm / Ataractic Corporation www.atacomm.com V: 1-877-700-VOIP [EMAIL PROTECTED] Looking for voice over IP products? Visit our VoIP store at http://voipstore.atacomm.com/ On Oct 11, 2006, at 12:01 AM, Douglas Garstang wrote: I think that limit was increased in 1.6.6 or 1.6.7. -Original Message- From: C F [mailto:[EMAIL PROTECTED]] Sent: Tue 10/10/2006 6:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [asterisk-users] RE: Welcome to the "asterisk-users" mailing list On 10/10/06, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: What I don't understand is why people MUST use the 2.0.x firmware. I believe it's because the limit of how many can be monitored at once, someone correct me if I'm wrong. Jessee J Holmes wrote: A few of our technical support staff here at Atacomm are currently working on this issue with Polycom, Digium and one of our customer's (who posted in here earlier this week). There are some major differences from the 1.x.x firmware and the 2.01 firmware. Obviously, many on here, voip-info.org, and all over the Internet have reported troubles where their presence feature that stops working after they upgrade their Polycom phones to firmware revision 2.0.1 when the phones are configured with an Asterisk system. A couple new things have been added to the .cfg files that MUST now be set in order for presence to work again with an Asterisk system. In the sip.cfg file: feature feature.1.name="presence" feature.1.enabled="1" In the phone[mac].cfg file: attendant attendant.uri="[EMAIL PROTECTED]" attendant.reg=""/ However, there are still confirmed problems with this setup (i.e. LEDs not working), which Polycom and ourselves are currently testing in our labs trying to fix. The reasoning for this is Digium doesn't seem to follow the "standards" for presence support and are currently working to change this functionality within Asterisk. Polycom designed their phones, and specifically their firmware, to work with the "standards" (more specifically, the Microsoft LCS - Live Communications Server). This issue has been reported on multiple instances to Polycom, Digium, and ourselves; but, no real resolution is completed yet. We'll continue working on this issue within our labs and post an official answer when one is available. Sorry for the bit of bad news, if anyone is willing to contribute working / half working code, we'd be more than happy to look at it and work with Polycom and Digium on getting this fixed for everyone AS ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk 1.2.12 lost phone registrations today... why?
I lost my internet connection today for a short time. During that time 1.2.12.1 stopped talking to my phones. Asterisk was still working as I got 2 voicemails. I have TDM analog cards for incoming calls. Anyway my cisco phones had X's (lost registration) and my uniden phones said Registration error. Why would phones loose registration to asterisk when the internet connection and DNS was lost. All phones have hardcoded IP addresses not DNS names. Any ideas? THanks, Jerry ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] NFAS Not Passing Audio on B-chan 48,72,96
Sangoma has excellent technical support, and usually pretty quick to respond IF you are sure it isn't a configuration issue, your best resource is Sangoma Please report back when it is resolved. John Novack Steve Totaro wrote: I have NFAS setup on several quad port T1 cards (Sangoma). It mostly works well with the exception that calls coming in on channels 48,72, and 96 have no audio. I tried removing these channels from zapata.conf with hopes that the channels would not come up or be used. Now I get Ring requested on unconfigured channel. How can I busyout these these channels so that incoming calls are not sent to them, or how can I fix the real problem? I think it may be a Sangoma/Wanpipe configuration issue. Thanks, Steve Totaro ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] RE: Welcome to the asterisk-users mailing list
Hi Jesse, 4 x ip500s Ive held off upgrading the bootrom past 2.62 as I understand this is a one way trip to 3.01 and above. As Im a second hand hardware user I dont have access to Polycoms direct firmware and have been upgrading from freedomphone.net Cheers, Dean From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jessee J Holmes Sent: Wednesday, 11 October 2006 10:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] RE: Welcome to the asterisk-users mailing list Dean, Try obtaining the latest bootrom again, should be 3.2.2, we've seen this happen before for various odd reasons and Polycom's recommended fix is get the non-engineering version of the bootrom (don't ask please, just do it). So download the bootrom again and attempt it once more, while you're at it, be safe and get the 2.0.1 firmware again as well. Let me know if this fixes you're problem. By the way, are you using IP 500's? or 501's? 500's may not take, I think we had that discussion in this list before. Jessee Holmes Atacomm / Ataractic Corporation www.atacomm.com V: 1-877-700-VOIP [EMAIL PROTECTED] Looking for voice over IP products? Visit our VoIP store at http://voipstore.atacomm.com/ On Oct 10, 2006, at 7:10 PM, Dean Collins wrote: I've had problems loading 2.01 onto 2 of my 4 polycom 500's 2 work great no probs, 2 I cant get it to upload without failing. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED]] On Behalf Of Eric ManxPower Wieling Sent: Tuesday, 10 October 2006 5:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] RE: Welcome to the asterisk-users mailing list What I don't understand is why people MUST use the 2.0.x firmware. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] zt_chanconfig failed
Hello to all, I have a question. I am installing te110p, when I give ztcfg him - v leaves the following error to meZT_CHANCONFIG failed on channel 25: No such device or address (6)- That means east error?- It is a physical damage of the card? Thank you very much/texto original en españolHola a todos, tengo una pregunta. Estoy instalando un te110p, cuando le doy un ztcfg -v me sale el siguiente error ZT_CHANCONFIG failed on channel 25: No such device or address (6)-Que significa este error?-Es un daño fisico de la tarjeta? Muchas gracias/-- //DiegoF Dichosos aquellos que no esperan nada de la vida, porque nunca seran defraudados ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sending fax with chan-capi
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi Klaus, The incoming fax script will generate an email with the fax attached, and there is another script, sendfax_status.py, which is run as a DeadAGI after the outgoing fax has been sent, it retrieves status information and sends it to a (hardcoded) email address. The call file created by the outgoing script file2fax.py specifies 3 retries in case of failure. This just retries it within Asterisk, I don't know if I could have chan_capi do that. jens On 11 Oct 2006, at 09:52, Klaus Darilion wrote: Hi Jens! Thanks for the script. Do you generate and notifications (succeeded, failed) or retransmit in case of failed sending? Or does that CAPI internally? regards klaus Jens Vagelpohl wrote: How can I generate sff format? I found sfftobmp, not nothing the other way round. You can use ghostscript: gs -dNOPAUSE -dBATCH -sDEVICE=cfax -sOutputFile=output.sff input.ps Is there a nice way to get the sff out of an Windows application (like virtual printers for hylafax) or at least some scripts which produce the sff and the asterisk call file out of an pdf? Here's something I use (not Windoze, sorry): http://svn.dataflake.org/filedetails.php? repname=DataflakeSoftwarepath=%2Fasterisk-chancapi-faxscripts% 2Ffile2fax.pyrev=0sc=0 The script takes TIFF, PS or PDF as input, creates SFF and a call file. It is run out of cron and checks if suitable files have been dropped into a spool directory. The whole package at http://svn.dataflake.org/listing.php? repname=DataflakeSoftwarepath=%2Fasterisk-chancapi-faxscripts% 2Frev=0sc=0 contains some documentation and also a script that I use to handle incoming faxes (with capicommand receivefax). jens ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (Darwin) iD8DBQFFLQ6qRAx5nvEhZLIRAh3SAKCBt6XOf98C2IfoPjkIGms8AbTO3ACglmU5 iyx3xR0dijuk0VnrK3bggCg= =/XV9 -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Where is the PlayDTMF command?
http://bugs.digium.com/view.php?id=6682 Thanks I patch my installation with the patch on the above URL. It works fine now. Thanks Moises. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cisco 7960 not registering after * restart
On Wed, Oct 11, 2006 at 09:11:43AM -0500, Aaron Daniel wrote: That's a bug with the 7.5 firmware. I would suggest upgrading to the 8.4 version, we've been running it for a few weeks in a test environment and everyone's been pretty satisfied with the new firmware (read: nobody's complained). If the server goes out, they re-register after the timeout without problems. Thanks for your helpful answer, What is the cisco part number for the appropriate smartnet contract required to obtain 79XX firmware? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom 2.01 sip issues
Jessee, The reason for me upgrading to 2.01 is we wanted to add some 430s to our system which from what I understand have a problem with 1.67, at this point we will just go with more 501s instead. What is the procedure to go back to 1.67? Will you be adding 1.67 to your FTP site? currently you only have 2.01. Will the sip.cfg and phone.cfg from 2.01 work on 1.67? Thanks, Issac From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jessee J Holmes Sent: Wednesday, October 11, 2006 10:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] RE: Welcome to the asterisk-users mailing list Most want the 2.0.1 firmware for a few reasons: A) They have the latest; although, this is a poor reason, it's still a reason people download and use the latest firmware - remember here always, If it's not broken, DON'T fix it! B) They are hoping to fix a previous problem they've had in the past (i.e. stability issues - usually caused by other factors besides just firmware) C) They are told to. IF you are needing to talk to a support professional, especially Polycom, you NEED to upgrade to the latest firmware or they simply will not help you (most of the times anyways) These are a few I can think of anyways; and unfortunately, it's going to be a problem sooner or later. Jessee Holmes Atacomm / Ataractic Corporation www.atacomm.com V: 1-877-700-VOIP [EMAIL PROTECTED] Looking for voice over IP products? Visit our VoIP store at http://voipstore.atacomm.com/ On Oct 10, 2006, at 4:56 PM, Eric ManxPower Wieling wrote: What I don't understand is why people MUST use the 2.0.x firmware. Jessee J Holmes wrote: A few of our technical support staff here at Atacomm are currently working on this issue with Polycom, Digium and one of our customer's (who posted in here earlier this week). There are some major differences from the 1.x.x firmware and the 2.01 firmware. Obviously, many on here, voip-info.org, and all over the Internet have reported troubles where their presence feature that stops working after they upgrade their Polycom phones to firmware revision 2.0.1 when the phones are configured with an Asterisk system. A couple new things have been added to the .cfg files that MUST now be set in order for presence to work again with an Asterisk system. In the sip.cfg file: feature feature.1.name=presence feature.1.enabled=1 In the phone[mac].cfg file: attendant attendant.uri=[EMAIL PROTECTED] attendant.reg=/ However, there are still confirmed problems with this setup (i.e. LEDs not working), which Polycom and ourselves are currently testing in our labs trying to fix. The reasoning for this is Digium doesn't seem to follow the standards for presence support and are currently working to change this functionality within Asterisk. Polycom designed their phones, and specifically their firmware, to work with the standards (more specifically, the Microsoft LCS - Live Communications Server). This issue has been reported on multiple instances to Polycom, Digium, and ourselves; but, no real resolution is completed yet. We'll continue working on this issue within our labs and post an official answer when one is available. Sorry for the bit of bad news, if anyone is willing to contribute working / half working code, we'd be more than happy to look at it and work with Polycom and Digium on getting this fixed for everyone AS ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk + E1 with MFC/R2 in Argentina?
Has somebody installed this configuration: Asterisk + E1 with MFC/R2 (Telefónica Argentina) in Argentina before? I need to know if it´s possible with MFC/R2 argentine variation. I have not tested in Argentina, but support is included in the code, so I suppose it should work. Regards -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 1.2.12 lost phone registrations today... why?
Jerry Geis wrote: I lost my internet connection today for a short time. During that time 1.2.12.1 stopped talking to my phones. Asterisk was still working as I got 2 voicemails. I have TDM analog cards for incoming calls. Anyway my cisco phones had X's (lost registration) and my uniden phones said Registration error. Why would phones loose registration to asterisk when the internet connection and DNS was lost. All phones have hardcoded IP addresses not DNS names. Any ideas? THanks, Jerry Was the time correct on the phones? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Increase VoiceMail Messages Recording Gain -AudioCalls are Ok
Would you be able to tell me which lines must be reordered in app_voicemail.cOn 10/11/06, Cullin J. Wible [EMAIL PROTECTED] wrote:externnotify=/opt/asterisk-1.2.7.1/sbin/vm-audio vm-audio uses 'sox -e' to determine how much to scale by without clippingand thenThen 'sox -v' to scale the sound file.This happens after the email message is sent, but by changing the order of a few lines in the app_voicemail.c program you can have the externnotify runbefore the email message is sent.Cullin-Original Message-From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED]] On Behalf Of Aaron DanielSent: Wednesday, October 11, 2006 12:49 AM To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [asterisk-users] Increase VoiceMail Messages Recording Gain-AudioCalls are OkThat doesn't always work :)There's two options... either port the volgain patch from 1.4 to 1.2 (Ifanyone wants a copy, we've been using it for months... however it alsoconverts to mp3 so we'd have to strip that out)... or use 1.4 which includesthe patch.Let me know if I should post a copy of the older code somewhere. The 1.4 patch is here:http://bugs.digium.com/view.php?id=6237Aaron DanielComputer Systems TechnicianSam Houston State University [EMAIL PROTECTED](936) 294-4198On Tue, 2006-10-10 at 21:18 -0500, Shawn Kelley wrote: I had the same problem. Checking voicemail via the phone was perfectly normal but the email attachments were so quiet we had to turn the computer volume all the way up along with the speakers amps just to make the attachmentunderstandable. Then just wait until someone forgets to turn the volume back down and a lovely windows message box pops up. Scares the (pick your word) out of everyone in the office! After much searching I found the solution: In the voicemail.conf file change the order in which the recording formats are specified. Asterisk will email the first format in the list. My original line: format=wav49|wav|gsm My new line: format=wav|wav49|gsm NOTE: My understanding is that the wav files are much larger attachments than the wav49 version. However, we haven't noticed much difference, still fairly small attachments. Definitely no problems on a LAN or Broadband connection. From: Marco Mouta [mailto: [EMAIL PROTECTED]] Sent: Tuesday, October 10, 2006 2:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Increase VoiceMail Messages Recording Gain - AudioCalls are Ok Hi all I'm deploying aVoiceMailserver with Asterisk behind a legacy pbx, providing Voicemail to email services for Lecagy PBX extensions. On busy or unanswered calls, Legacy pbx will dial a specific DID (one per extension) to asterisk, and the call is handled by Voicemail application. I've several SIP extensions on this Asterisk box, and calls between Asterisk extensions and legacy PBX are just fine, at least no complaining from users, seems good to me:) The problem is: Right now, and i'm referring only to calls directly handled by VoiceMail application, the users get their audio files in email but the audio is very very low. I've thought about changing RX gain on PRI interface between legacy pbx and asterisk, but until now no complaining with audio calls. I'm afraid that changing this parameter to solve voicemail issues will get me in troubles with Voice Calls . Any advice, or previous similar experience?___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Com os melhores cumprimentos, Marco Mouta ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Strange FXS disconnection problem.
Hi Tzafrir, Many thanks for reply. Busydetect is also disabled. There's no chance of an actual busy signal, as it happens exactly 2m 40 seconds (give or take 1s) into an active call with both parties connected and talking away. Zapata.conf copied below: [channels] signalling=fxs_ks echocancel=64 echocancelwhenbridged=yes echotraining=400 cidstart=polarity hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes busydetect=no callprogress=no progzone=uk pulsedial=no answeronpolarityswitch=yes musiconhold=default ringtimeout=1000 userincomingcalledidonzaptransfer=yes usercallerid=yes cidsignalling=v23 cidstart=usehist language=en rxgain=3 txgain=3 context=bt_pstn channel = 1 Thanks for your time. Dave ~ -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: 11 October 2006 08:29 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Strange FXS disconnection problem. On Tue, Oct 10, 2006 at 11:43:09PM +0100, David Bath wrote: Hi All, I've tried to find the solution to this, but sadly met with failure. I've got an asterisk box with two X100P's in, and both cards have the same strange problem. After 2min 40seconds (always: within +/- 1sec) they drop an outbound call. Inbound calls are not affected... they stay up as long as required. Do you use busydetect? Any chance that there's a busy tone there? I've seen a fair bit of chatter about similar kinds of problems sometimes being related to callprogess detecting false hang-ups, so I've made sure this is disabled in Zapata.conf but it seems to have had no effect. I was running a slightly older version of zaptel/asterisk (1.2.0) and I've upgraded to the lastest build, but also with no success. Details, pleasse: zapata.conf logs from a call (set verbose to at least 3, and enable full in logger.conf). -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 1.2.12 lost phone registrations today... why?
On Wed, 2006-10-11 at 11:19 -0400, Jerry Geis wrote: I lost my internet connection today for a short time. During that time 1.2.12.1 stopped talking to my phones. Asterisk was still working as I got 2 voicemails. I have TDM analog cards for incoming calls. Anyway my cisco phones had X's (lost registration) and my uniden phones said Registration error. Why would phones loose registration to asterisk when the internet connection and DNS was lost. All phones have hardcoded IP addresses not DNS names. Any ideas? THanks, Jerry ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Ditto here. I run Trixbox 1.1 with the latest updates. We had a power failure that took down the internet connection and local DNS server. My local Cisco phones could not register (IP addresses are hard-coded) and, because of the DNS failure I could not register with my SIP provider. I have not had a chance to sort through the logs, but I had to reset the Asterisk box, after the DNS server was restored. In my case, inbound and outbound PSTN calls (via a TDM11b) were failing. The local analog phone rang (on an inbound PSTN call), but did not recognize the analog answering machine taking the line off-hook. Once the caller hung up, the local (analog) phones would ring again, but no call was present, as reported by my wife. BTW: The Asterisk box is on UPS and did not go down and I do not have voicemail enabled for my local extensions. This sounds similar, possibly: http://lists.digium.com/pipermail/asterisk-users/2006-October/168910.html Bob... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zt_chanconfig failed
Hi DiegoF, I had a similar problem, it was a zaptel.conf misconfiguration. Maybe for you is the same. Post your zaptel.conf to give more details. Giorgio Incantalupo DiegoF wrote: Hello to all, I have a question. I am installing te110p, when I give ztcfg him - v leaves the following error to me ZT_CHANCONFIG failed on channel 25: No such device or address (6) - That means east error? - It is a physical damage of the card? Thank you very much / texto original en español Hola a todos, tengo una pregunta. Estoy instalando un te110p, cuando le doy un ztcfg -v me sale el siguiente error ZT_CHANCONFIG failed on channel 25: No such device or address (6) -Que significa este error? -Es un daño fisico de la tarjeta? Muchas gracias / -- // DiegoF // // Dichosos aquellos que no esperan nada de la vida, porque nunca seran defraudados ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] GPL Softphones
Hi, Im searching for GPLed softphones. I found WengoPhone but actually not available for Asterisk PBX, only for Wengo network. I found Kiax but only for IAX protocol. Did you know a good GPLed softphones which works on Windows ? Thanks Greg ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] NFAS Not Passing Audio on B-chan 48,72,96
This reply helps me how? Of course I am pursuing the issue through their support channel. Thanks, Steve John Novack wrote: Sangoma has excellent technical support, and usually pretty quick to respond IF you are sure it isn't a configuration issue, your best resource is Sangoma Please report back when it is resolved. John Novack Steve Totaro wrote: I have NFAS setup on several quad port T1 cards (Sangoma). It mostly works well with the exception that calls coming in on channels 48,72, and 96 have no audio. I tried removing these channels from zapata.conf with hopes that the channels would not come up or be used. Now I get Ring requested on unconfigured channel. How can I busyout these these channels so that incoming calls are not sent to them, or how can I fix the real problem? I think it may be a Sangoma/Wanpipe configuration issue. Thanks, Steve Totaro ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: asterisk-users Digest, Vol 27, Issue 49
-- Forwarded message --From:Doug Lytle [EMAIL PROTECTED]To:Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.comDate:Tue, 10 Oct 2006 16:25:11 -0400 Subject:Re: [asterisk-users] How big is *your* dialplan??Steve Murphy wrote: Hello! In my relentless quest for knowledge, I pose this question: who's got the biggest dialplans, and how big are these monsters? Sounds interesting. Small facility of 60 users:-= 161 extensions (597 priorities) in 59 contexts. =---Ben Franklin quote:Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. Single server stats, 50 user system,-= 238 extensions (870 priorities) in 57 contexts. =-- Buki ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How big is *your* dialplan??
-- Forwarded message --From:Doug Lytle [EMAIL PROTECTED]To:Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date:Tue, 10 Oct 2006 16:25:11 -0400 Subject:Re: [asterisk-users] How big is *your* dialplan??Steve Murphy wrote: Hello! In my relentless quest for knowledge, I pose this question: who's got the biggest dialplans, and how big are these monsters? Sounds interesting. Small facility of 60 users:-= 161 extensions (597 priorities) in 59 contexts. =---Ben Franklin quote:Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. Single server stats, 50 user system,-= 238 extensions (870 priorities) in 57 contexts. =-- Buki ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Segmentation fault asterisk realtime problem
Hi to all, I've a segmentation fault while using asterisk relatime conf with mysql db. I've cretate sip_buddies and extensions tables into db and edit res_mysql.conf, extconf.conf without any issues. So when I start asterisk and my phone try to register using sip user configured in my db, asterisk stops with Segmentation fault error. Follow post gdb backtrace 0 0x400337c0 in pthread_setcanceltype () from /lib/libpthread.so.0 #21 0x0805d8de in ast_load_realtime (family=0x666d7464 Address 0x666d7464 out of bounds) at config.c:994 #22 0x4047cdad in realtime_peer (peername=0xbe7f8891 101, sin=0x730) at chan_sip.c:1696 #23 0x4046cf67 in find_peer (peer=0xbe7f8891 101, sin=0x0, realtime=1) at chan_sip.c:1776 #24 0x40485dfd in register_verify (p=0x81944d8, sin=0xbe7fe79c, req=0xbe7fe7ac, uri=0xbe7fe9cd sip:192.168.1.2, ignore=1718449252) at chan_sip.c:6514 #25 0x404839b7 in handle_request (p=0x81944d8, req=0xbe7fe7ac, sin=0xbe7fe79c, recount=0x666d7464, nounlock=0x666d7464) at chan_sip.c:11083 #26 0x4048150d in sipsock_read (id=0x813ed80, fd=15, events=1, ignore=0x0) at chan_sip.c:11377 #27 0x080558dd in ast_io_wait (ioc=0x8162320, howlong=1718449252) at io.c:284 #28 0x404776a9 in do_monitor (data=0x0) at chan_sip.c:11536 #29 0x40034cc4 in pthread_detach () from /lib/libpthread.so.0 #30 0x40201037 in clone () from /lib/libc.so.6 any suggestion about? Thanks 4 all, -- * (o ing. Patria Flavio * //\ phone 0823451358 * V_/_ mobile 3407873357 * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Strange FXS disconnection problem.
Hi Paul, Thanks for reply. It is only recently that I have added an X100P - the asterisk server has been doing purely SIP and IAX2 (to an ISDN gateway) before and everything is perfect. There are no agents dropped etc. It is purely that the zap channel (to X100P) gets released with no errors (that I can currently see!), even with -v 50. Cheers, Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Ianas Sent: 11 October 2006 11:14 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Strange FXS disconnection problem. I experienced a similar problem, but with AT-RG 623TX (ISDN BRA gateway). I can only tell you that there is no Asterisk problem. You should try to debug hardware / driver problems. Question: is(are) the user-agent(s) still authenticated with Asterisk after the call is dropped? You should also set the debug level top the highest value. -- Paul Ianas Programming Engineer Level 7 Software Timisoara, 59D Bucovinei phone: 0744137020 email: [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Wednesday, October 11, 2006 10:29 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Strange FXS disconnection problem. On Tue, Oct 10, 2006 at 11:43:09PM +0100, David Bath wrote: Hi All, I've tried to find the solution to this, but sadly met with failure. I've got an asterisk box with two X100P's in, and both cards have the same strange problem. After 2min 40seconds (always: within +/- 1sec) they drop an outbound call. Inbound calls are not affected... they stay up as long as required. Do you use busydetect? Any chance that there's a busy tone there? I've seen a fair bit of chatter about similar kinds of problems sometimes being related to callprogess detecting false hang-ups, so I've made sure this is disabled in Zapata.conf but it seems to have had no effect. I was running a slightly older version of zaptel/asterisk (1.2.0) and I've upgraded to the lastest build, but also with no success. Details, pleasse: zapata.conf logs from a call (set verbose to at least 3, and enable full in logger.conf). -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call takeover?
Hi C., Check out the pickupgroup and callgroup options in sip.conf -- these should accomplish what you're looking for: http://www.voip-info.org/wiki/index.php?page=Asterisk+config+sip.conf More about this feature is defined here: http://www.voip-info.org/wiki/view/Asterisk+callgroups+and+pickupgroups If you need to be more specific in what to pickup, you could likely use the Asterisk Manager API's Redirect action to redirect the call to another device: http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+Redirect Hope that helps -samy Csibra Gergo wrote: Hi, situation is the following: There's an inbound call, that rings on SIP/tel21 (ATA is PAP2). At the time, bobody there, but a lazy people sits by SIP/tel22 (about 5m distance) and he want to takeover the call. How can I do this whit asterisk? Ok. I can do with call parking, but with call parking on SIP/tel21 must I call the parking extension too, and if nobody picks up the phone, the fax machine (the SIP/tel21) must answer it, and the fax machine can not call the parking extension. ps.: sorry for starting new thread with reply, but I can not send mails to this list otherwise. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] average waiting time in a queue
Hello all, we want to use asterisk queues for a call center application. Depending on the average waiting time in a queue, we want to make a decision to either enqueue a call or transfer it to another site. Are the applications available to query the average waiting time of a queue, if possible for a configurable time frame? Thanks and Regards Markus ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] NFAS Not Passing Audio on B-chan 48,72,96
Steve Totaro wrote: Kristian Kielhofner wrote: Steve Totaro wrote: I have NFAS setup on several quad port T1 cards (Sangoma). It mostly works well with the exception that calls coming in on channels 48,72, and 96 have no audio. I tried removing these channels from zapata.conf with hopes that the channels would not come up or be used. Now I get Ring requested on unconfigured channel. How can I busyout these these channels so that incoming calls are not sent to them, or how can I fix the real problem? I think it may be a Sangoma/Wanpipe configuration issue. Thanks, Steve Totaro Steve, I take it you have one D channel for all four spans? On 24? I think this should be pretty transparent to wanpipe. You should configure four spans, with one channel group (type TDM) per span (obvious). You should also probably disable any native D channel features. I always have nothing but problems with that. In zaptel.conf, only specify one d channel. The tricky stuff is in zapata.conf. Can you post that, and maybe zaptel.conf too? -- Kristian Kielhofner Yes the D chan is 24 with no backup. Before I had channel = 1-23,25-96. [trunkgroups] trunkgroup = 1,24 spanmap = 1,1,0 spanmap = 2,1,1 spanmap = 3,1,2 spanmap = 4,1,3 [channels] resetinterval=never callerid=asreceived usecallerid=yes hidecallerid=no callwaiting=no usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes callreturn=yes echocancel=no echocancelwhenbridged=no ;echotraining=800 group=0 callgroup=1 pickupgroup=1 immediate=no rxgain=0.0 txgain=0.0 context=from-pstn switchtype=5ess signalling=pri_cpe channel = 1-23,25-47,49-71,73-95 span=1,0,0,esf,b8zs span=2,0,0,esf,b8zs span=3,0,0,esf,b8zs span=4,0,0,esf,b8zs bchan=1-23,25-96 dchan=24 loadzone=us defaultzone=us Steve, Shouldn't your channel line from zapata.conf look like this: channel = 1-23,25-96 -- Kristian Kielhofner ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk users help
Hi, I had a question. I am installing Asterisk on a windows machine Astwind. I was wondering if it works with Dialogic card or if it needed only digium card. Is there anyway Asterisk can work with a Dialogic card or a Pika board? Thanks in advance. Vijay Naidu Never Interrupt your enemy when he is making a mistake -Napolean Bonaparte (1769 - 1821) -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.408 / Virus Database: 268.13.2/471 - Release Date: 10/10/2006 Naidu, Vijay.vcf Description: Binary data ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom 2.01 sip issues
Dear Issac,Makes sense.We got asked about moving back to firmware 1.6.7 as well and the official answer from Polycom is "not a problem"! Put the firmware on your server and remove the 2.0 firmware from this server and when the phone reboots it will grab the 1.6.7 firmware and load it on the phone. Polycom says its as easy as that.We can surely get 1.6.7 on our ftp site. We haven't done this due to a major server system upgrade we've been working on for our website. The current server we have is a temporary location for this firmware, we have just the necessities on this server as this particular server isn't within our main cluster of servers (there isn't much bandwidth or power here until we move to the new server farm). I'll have one of our techs post this firmware in a little bit here on the temp ftp server.Good question on the .cfg files. I don't know ... 1.6.7 .cfg files won't work "correctly" with 2.0.1 firmware since the files changed. Not sure about the other way around, I would assume they'd work, but wouldn't recommend it as you may experience stability issues or glitches from the phone not knowing what to do with some the parameters in these newer files. It's always best to use the .cfg files given with the firmware on your phones.Hope that helps.As far 1.6.7 firmware supporting multiple presences (48 i think), maybe I was wrong on that; however, I remember reading the 2.0.1 firmware release notes and they mentioned that feature was fixed within the 2.0 firmware. Maybe they fixed it before that and just never documented it or maybe I misread it. If it works through in 1.6.7, great! Thanks Douglas. Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products? Visit our VoIP store at http://voipstore.atacomm.com/ On Oct 11, 2006, at 10:36 AM, Issac Simchayof wrote: Jessee, The reason for me upgrading to 2.01 is we wanted to add some 430’s to our system which from what I understand have a problem with 1.67, at this point we will just go with more 501’s instead. What is the procedure to go back to 1.67? Will you be adding 1.67 to your FTP site? currently you only have 2.01.Will the sip.cfg and phone.cfg from 2.01 work on 1.67? Thanks, Issac From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Jessee J Holmes Sent: Wednesday, October 11, 2006 10:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] RE: Welcome to the "asterisk-users" mailing list Most want the 2.0.1 firmware for a few reasons: A) They have the latest; although, this is a poor reason, it's still a reason people download and use the latest firmware - remember here always, "If it's not broken, DON'T fix it!" B) They are hoping to fix a previous problem they've had in the past (i.e. stability issues - usually caused by other factors besides just firmware) C) They are told to. IF you are needing to talk to a support professional, especially Polycom, you NEED to upgrade to the latest firmware or they simply will not help you (most of the times anyways) These are a few I can think of anyways; and unfortunately, it's going to be a problem sooner or later. Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED] Looking for voice over IP products? Visit our VoIP store at http://voipstore.atacomm.com/ On Oct 10, 2006, at 4:56 PM, Eric ManxPower Wieling wrote:What I don't understand is why people MUST use the 2.0.x firmware. Jessee J Holmes wrote: A few of our technical support staff here at Atacomm are currently working on this issue with Polycom, Digium and one of our customer's (who posted in here earlier this week). There are some major differences from the 1.x.x firmware and the 2.01 firmware. Obviously, many on here, voip-info.org, and all over the Internet have reported troubles where their presence feature that stops working after they upgrade their Polycom phones to firmware revision 2.0.1 when the phones are configured with an Asterisk system. A couple new things have been added to the .cfg files that MUST now be set in order for presence to work again with an Asterisk system. In the sip.cfg file: feature feature.1.name="presence" feature.1.enabled="1" In the phone[mac].cfg file: attendant attendant.uri="[EMAIL PROTECTED]" attendant.reg=""/ However, there are still confirmed problems with this setup (i.e. LEDs not working), which Polycom and ourselves are currently testing in our labs trying to fix. The reasoning for this is Digium doesn't seem to follow the "standards" for presence support and are currently working to change this functionality within Asterisk. Polycom designed their phones, and specifically their firmware, to work with the "standards" (more specifically, the Microsoft LCS - Live Communications Server). This issue has been reported on multiple instances to Polycom, Digium, and ourselves; but, no real
Re: [asterisk-users] RE: Welcome to the asterisk-users mailing list
Issac Simchayof wrote: Polycom 601 with Sip 2.01 Anyone using Sip 2.01? I have upgraded my phones and now presence no longer functions. Buddy list shows all phones online but status does not change when someone is on a call. Also blf does not function. I am using trixbox, 1.67 was working fine on the same box. Any ideas? Yes. Don't use such a useless subject for your queries to the list, and you might find them better received. . . The archives of this list is a valuable resource for those doing due diligence before bothering list members. A subject like yours hides the intent and content of your message totally, making it worthless as a subject search target. Why not SIP 2.01 on Polycom? Too late now, though :-) B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RE: Welcome to the asterisk-users mailing list
Dean,Tough call ... I haven't played with an IP 500 in a long time now and all that I know is Polycom officially doesn't support them.I'm sure the 2.0.1 firmware wasn't designed to ever work with bootroms 2.xx. I'm sure the problem lies with either the phone not supporting it or the bootrom not accepting the firmware, but I'd be very weary of upgrading an IP 500 to a 3.xx bootrom.Maybe someone else has better experience with this that has some of the older phones. Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products? Visit our VoIP store at http://voipstore.atacomm.com/ On Oct 11, 2006, at 10:29 AM, Dean Collins wrote: Hi Jesse,4 x ip500’s I’ve held off upgrading the bootrom past 2.62 as I understand this is a one way trip to 3.01 and above. As I’m a second hand hardware user I don’t have access to Polycom’s direct firmware and have been upgrading from freedomphone.net Cheers, Dean From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Jessee J Holmes Sent: Wednesday, 11 October 2006 10:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] RE: Welcome to the "asterisk-users" mailing list Dean, Try obtaining the latest bootrom again, should be 3.2.2, we've seen this happen before for various odd reasons and Polycom's recommended fix is get the "non-engineering version" of the bootrom (don't ask please, just do it). So download the bootrom again and attempt it once more, while you're at it, be safe and get the 2.0.1 firmware again as well. Let me know if this fixes you're problem. By the way, are you using IP 500's? or 501's? 500's may not take, I think we had that discussion in this list before. Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED] Looking for voice over IP products? Visit our VoIP store at http://voipstore.atacomm.com/ On Oct 10, 2006, at 7:10 PM, Dean Collins wrote:I've had problems loading 2.01 onto 2 of my 4 polycom 500's 2 work great no probs, 2 I cant get it to upload without failing. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED]] On Behalf Of Eric "ManxPower" Wieling Sent: Tuesday, 10 October 2006 5:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] RE: Welcome to the "asterisk-users" mailing list What I don't understand is why people MUST use the 2.0.x firmware. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] compiling libunicall
hola a todos de nuevo, tengo el siguiente error cuando compilo el libunicall despues de compilar spandsp y libsupertone. esto es en fedora 5hello to all, I have the following error again when I compile libunicall after compiling spandsp and libsupertone. this is in fedora 5testcall.o: In function `handle_uc_event':/root/asterisk/mfc/libunicall/testcall.c:515: undefined reference to `dtmf_put'/root/asterisk/mfc/libunicall/testcall.c:500: undefined reference to `dtmf_put' testcall.o: In function `channel_read_file':/root/asterisk/mfc/libunicall/testcall.c:141: undefined reference to `dtmf_get'/root/asterisk/mfc/libunicall/testcall.c:192: undefined reference to `dtmf_put' -- //DiegoF Dichosos aquellos que no esperan nada de la vida, porque nunca seran defraudados ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How big is *your* dialplan??
Douglas, it seems to me that you don't understand how the extensions
of an asterisk dialplan relate to real life. As an example:
-= 135 extensions (657 priorities) in 31 contexts. =-
This from a box (yes one box) that has just 10 phones, and 6 lines.
Every s extension is considered an extension. Which makes every macro
a context and at least one extension. If one has:
exten = s,n,Dial(whatever)
exten = s,n,Goto(s-${DIALSTATUS},1)
Then that context (macro) has at least 2 extensions.
Calling Voicemail in my dialplan has 7 extensions (yes just pressing
the message button). For real life it's only 1 extension.
Another example:
This is for a system with around 75 different offices hosted on the
same box, using 3 T1s, and each office with at least 2 extensions, the
biggest one being around 15 extensions.
-= 1110 extensions (2279 priorities) in 138 contexts. =-
That's for around 90 phones and 150 published active phone numbers
(some of the phone numbers are just IVRs). Why would the fact that
it's on one box matter? If the main incoming T1 is down (which
happens), there is no incoming calls anyhow. What would clustering
help in this case?
Why would someone have to build a new box if a system went down? A
system should never be built with a single point of failure. The only
thing that should be allowed to bring down a system is a fire. The CPU
fan should be noticed making noise way before it dies, which gives
enough time for a planned shutdown, in any case that doesn't require
(if/when the CPU dies) rebuilding the whole box.
Any asterisk system that has more than 50-60 users should NEVER be
built in a way that if it doesn't get physically damaged it needs to
be rebuilt if/when it goes down.
On 10/11/06, Douglas Garstang [EMAIL PROTECTED] wrote:
I see some awefully large dialplans here. Are people putting all this on one
box or clustering it amongst a number of boxes? I think any business is going
to be pretty annoyed if they suddenly lost access to 16,000+ extensions, and
had to wait for a new box to be built and configured.
-Original Message-
From: George Pajari [mailto:[EMAIL PROTECTED]
Sent: Tue 10/10/2006 10:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc:
Subject: Re: [asterisk-users] How big is *your* dialplan??
Single server, dual P3 866Mhz, 1.5Gb, TE407P, two PRIs to telco, one PRI
to fax server, one PRI to T.38 gateway:
1791 extensions (4378 priorities) in 240 contexts
--
George Pajari, netVOICE communications604 484 VOIP (484 8647 x102)
Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102)
Hosted IP PBX Services for SOHO Small Businesses - www.ip-centrex.ca
VoIP Service, Equipment, Systems, and Consulting - www.netvoice.ca
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[asterisk-users] 1.4 beta2 on intel mac
Has anyone built and run asterisk 1.4 beta2 on an intel mac? Did it work? I've got it building ok (once I installed Xcode, wget and bison) However Asterisk hangs on startup (halfway through loading the modules). I have not (yet) had time to debug it, but I wondered if anyone else had done this before me ? Tim Panton www.mexuar.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2.12.1 dies after asterisk -rx and never comes up again
On Wed, Oct 11, 2006 at 11:25:08AM +0200, Remco Barendse wrote: On Wed, 11 Oct 2006, Tzafrir Cohen wrote: On Wed, Oct 11, 2006 at 09:14:55AM +0200, Remco Barendse wrote: Hi list! I recently upgraded to FreePBX 2.1.3 although I am not sure if this has something to do with it. I do a nightly restart of Asterisk, just in case. Why? Sometimes the internet connection is dropped and asterisk doesn't do a dns lookup and provider re-rest quickly enough so all calls are going out via expensive ISDN. So detect a connection change and then restart, by the way of 'asterisk -rx restart now' (or 'restart when convinient', depending on whether you care about local calls or remote calls). Assuming a restart is really needed, rather than a reload. A simple nightly restart means that on the avarage you'll be half a day too late. How can you detect that both (a) DNS lookup failed with Asterisk and (b) DNS lookup is already OK elsewhere? Also I sometimes seem to have some trouble after re-loading FreePBX too often I get things like extensions being marked as busy/not available. = Parsing '/etc/asterisk/manager.conf': Found Oct 11 01:05:01 WARNING[6546] manager.c: Unable to bind socket: Address already in use Asterisk is already running. I don't think so, asterisk -r will throw me an error that * is not running. If * would be running I would get console I guess? Maybe asterisk is in some sort of restart loop? Probably the wonders , or a misuse of- safe_asterisk. Alternatively, netstat -lntp | grep 5038 Thanks, I'll try that when the problem occurs again -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GPL Softphones
On Wed, Oct 11, 2006 at 06:23:48PM +0200, Gregory Duchatelet wrote: Hi, I'm searching for GPLed softphones. I found WengoPhone but actually not available for Asterisk PBX, only for Wengo network. Have you actually tried it? Were you actually able to build it? I found Kiax but only for IAX protocol. For which platform? IAX: kiax, iaxcomm, mozphone SIP: kphone, linhone, minisip, twinkle, ekiga (Try twinkle) Just a partial list of free phones that work on Linux. Did you know a good GPLed softphones which works on Windows ? IAXcomm should. So should wengophone and mozphone. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GPL Softphones
The X-Ten is probably the most know free soft-phone availible. You can find it at http://www.xten.com/index.php?menu=Productssmenu=xlite Gregory Duchatelet wrote: Hi, I’m searching for GPLed softphones. I found WengoPhone but actually not available for Asterisk PBX, only for Wengo network. I found Kiax but only for IAX protocol. Did you know a good GPLed softphones which works on Windows ? Thanks Greg ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zt_chanconfig failed
On Wed, Oct 11, 2006 at 10:29:44AM -0500, DiegoF wrote: Hello to all, I have a question. I am installing te110p, when I give ztcfg him - v leaves the following error to me ZT_CHANCONFIG failed on channel 25: No such device or address (6) - That means east error? - It is a physical damage of the card? Is the span E1 or T1? What signalling? Please post zaptel.conf and the output of 'cat /proc/zaptel/*' -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] max users
Whats the max headcount you can have in a conference bridge using ztdummy...since it is all sip based incomming? Don ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: sending fax with chan-capi
On Wed, Oct 11, 2006 at 11:32:57AM -0400, Jens Vagelpohl wrote: The call file created by the outgoing script file2fax.py specifies 3 retries in case of failure. Fax may fail even if the phone call was successfull. This just retries it within Asterisk, I don't know if I could have chan_capi do that. chan_capi 0.7 does set some variables which can / should be used in the dialplan (FAXSTATUS, ) -- Stefan Tichy [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Understanding NAT Traversal
H, hugolivude wrote: For various reasons, I'm not too partial to UPnP, but maybe there needs to be a SIP UA that uses UPnP to configure a NAT router for it, when an RTP stream is begun? Not following this part... While I could probably never bring myself to enjoy (Microsoft's?) Universal Plug-n-Play features, they would be helpful for the rtp streams, although not the signalling. Conceivably, if only one SIP UA were in use behind a NAT router, then when it constructed a call and needed to receive RTP streams, it would configure port mappings in the router via the UPnP protocol, so external port 10xxx is forwarded to the internal IP of the SIP UA. It could remove this port mapping when the call was deconstructed. The problem of course happens when two SIP UAs need to work behind a NAT router, because, as Cullin mentioned, It is very difficult to track a a many-to-one NAT (technically port address translation (PAT)) when you can't change the source or destination ports. Thanks Cullin! Moj ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] How big is *your* dialplan??
Title: Re: [asterisk-users] How big is *your* dialplan?? No one's system is redundant? :O -Original Message-From: Douglas Garstang [mailto:[EMAIL PROTECTED]On Behalf Of Douglas GarstangSent: Tuesday, October 10, 2006 10:58 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [asterisk-users] How big is *your* dialplan?? I see some awefully large dialplans here. Are people putting all this on one box or clustering it amongst a number of boxes? I think any business is going to be pretty annoyed if they suddenly lost access to 16,000+ extensions, and had to wait for a new box to be built and configured. -Original Message- From: George Pajari [mailto:[EMAIL PROTECTED] Sent: Tue 10/10/2006 10:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [asterisk-users] How big is *your* dialplan?? Single server, dual P3 866Mhz, 1.5Gb, TE407P, two PRIs to telco, one PRIto fax server, one PRI to T.38 gateway:1791 extensions (4378 priorities) in 240 contexts--George Pajari, netVOICE communications 604 484 VOIP (484 8647 x102)Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102)Hosted IP PBX Services for SOHO Small Businesses - www.ip-centrex.caVoIP Service, Equipment, Systems, and Consulting - www.netvoice.ca___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with ZAPTEL-1.4.0-beta1 and WCT100P card
Hello,
I'm trying to upgrade an Asterisk 1.2 linux box to Asterisk 1.4. I
installed the following
-rw-r--r-- 1 root root 10908541 Sep 21 13:25 asterisk-1.4.0-
beta2.tar.gz
-rw-r--r-- 1 root root 993921 Sep 21 13:25 asterisk-addons-1.4.0-
beta1.tar.gz
-rw-r--r-- 1 root root80019 Sep 21 13:25 libpri-1.4.0-beta1.tar.gz
-rw-r--r-- 1 root root 1523413 Sep 21 13:25 zaptel-1.4.0-beta1.tar.gz
I get an error when I run ztcfg. (see below). Any ideas? Is the
ztcfg issue with unable to read version info a problem?
[EMAIL PROTECTED] src]# ztcfg -d 99 -v
Line 11: span=1,1,1,esf,b8zs
Line 12: bchan=1-23
Line 13: dchan=24
Line 17: loadzone = us
Line 18: defaultzone= us
End of File
Notice: Configuration file is /etc/zaptel.conf
line 18: Unable to read Zaptel version information.
Zaptel Version: môy
Echo Canceller:
Configuration
==
SPAN 1: ESF/B8ZS Build-out: 133-266 feet (DSX-1)
24 channels configured.
ZT_CHANCONFIG failed on channel 1: Inappropriate ioctl for device (25)
[EMAIL PROTECTED] src]# strace ztcfg
execve(/sbin/ztcfg, [ztcfg], [/* 25 vars */]) = 0
uname({sys=Linux, node=asterisk-1, ...}) = 0
brk(0) = 0x8437000
open(/etc/ld.so.preload, O_RDONLY)= -1 ENOENT (No such file or
directory)
open(/etc/ld.so.cache, O_RDONLY) = 3
fstat64(3, {st_mode=S_IFREG|0644, st_size=38549, ...}) = 0
old_mmap(NULL, 38549, PROT_READ, MAP_PRIVATE, 3, 0) = 0xf6ff6000
close(3)= 0
open(/lib/tls/libm.so.6, O_RDONLY)= 3
read(3, \177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\0\203
\206..., 512) = 5
12
fstat64(3, {st_mode=S_IFREG|0755, st_size=214796, ...}) = 0
old_mmap(0x865000, 139424, PROT_READ|PROT_EXEC, MAP_PRIVATE, 3, 0) =
0x865000
old_mmap(0x886000, 8192, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_FIXED,
3, 0x2
) = 0x886000
close(3)= 0
open(/lib/tls/libc.so.6, O_RDONLY)= 3
read(3, \177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\300ku
\000..., 512) = 5
12
fstat64(3, {st_mode=S_IFREG|0755, st_size=1455084, ...}) = 0
old_mmap(NULL, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS,
-1, 0) = 0
xf6ff5000
old_mmap(0x742000, 1158124, PROT_READ|PROT_EXEC, MAP_PRIVATE, 3, 0) =
0x742000
old_mmap(0x857000, 16384, PROT_READ|PROT_WRITE, MAP_PRIVATE|
MAP_FIXED, 3, 0x1150
00) = 0x857000
old_mmap(0x85b000, 7148, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_FIXED|
MAP_ANONYMO
US, -1, 0) = 0x85b000
close(3)= 0
mprotect(0x857000, 8192, PROT_READ) = 0
mprotect(0x886000, 4096, PROT_READ) = 0
mprotect(0x73e000, 4096, PROT_READ) = 0
set_thread_area({entry_number:-1 - 6, base_addr:0xf6ff5820, limit:
1048575, seg_
32bit:1, contents:0, read_exec_only:0, limit_in_pages:1,
seg_not_present:0, usea
ble:1}) = 0
munmap(0xf6ff6000, 38549) = 0
open(/dev/zap/ctl, O_RDWR)= 3
brk(0) = 0x8437000
brk(0x8458000) = 0x8458000
brk(0) = 0x8458000
open(/etc/zaptel.conf, O_RDONLY) = 4
fstat64(4, {st_mode=S_IFREG|0644, st_size=330, ...}) = 0
mmap2(NULL, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS,
-1, 0) = 0xf6
fff000
read(4, # Autogenerated by ./genzaptelco..., 4096) = 330
read(4, , 4096) = 0
close(4)= 0
munmap(0xf6fff000, 4096)= 0
ioctl(3, 0x40244a12, 0x8078560) = 0
ioctl(3, 0x80844a05, 0xfeec3140)= -1 EINVAL (Invalid argument)
ioctl(3, 0x404c4a13, 0x80797ac) = -1 ENOTTY (Inappropriate
ioctl for dev
ice)
write(2, ZT_CHANCONFIG failed on channel ..., 71ZT_CHANCONFIG
failed on channe
l 1: Inappropriate ioctl for device (25)
) = 71
close(3)= 0
exit_group(1) = ?
[EMAIL PROTECTED] src]# cat /etc/zaptel.conf
# Autogenerated by ./genzaptelconf -- do not hand edit
# Zaptel Configuration File
#
# This file is parsed by the Zaptel Configurator, ztcfg
#
# It must be in the module loading order
# Span 1: WCT1/0 Digium Wildcard T100P T1/PRI Card 0
span=1,1,1,esf,b8zs
bchan=1-23
dchan=24
# Global data
loadzone= us
defaultzone = us
[EMAIL PROTECTED] src]# cat /etc/asterisk/zapata.conf
switchtype=national
context=default
signalling=pri_net
group=1
channel = 1-23
--
Matthew S. Crocker
Vice President
Crocker Communications, Inc.
Internet Division
PO BOX 710
Greenfield, MA 01302-0710
http://www.crocker.com
___
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Re: [asterisk-users] zt_chanconfig failed
hola, este lo copie de internethello, this it copies it of Internetspan=1,1,0,cas,hdb3cas=1-15:1101dchan=16cas=17-31:1101 loadzone = usdefaultzone=usthanksOn 10/11/06, Giorgio Incantalupo [EMAIL PROTECTED] wrote:Hi DiegoF,I had a similar problem, it was a zaptel.conf misconfiguration. Maybe for you is the same. Post your zaptel.conf to give more details.Giorgio IncantalupoDiegoF wrote: Hello to all, I have a question. I am installing te110p, when I give ztcfg him - v leaves the following error to me ZT_CHANCONFIG failed on channel 25: No such device or address (6) - That means east error? - It is a physical damage of the card? Thank you very much / texto original en español Hola a todos, tengo una pregunta. Estoy instalando un te110p, cuando le doy un ztcfg -v me sale el siguiente error ZT_CHANCONFIG failed on channel 25: No such device or address (6) -Que significa este error? -Es un daño fisico de la tarjeta? Muchas gracias / -- //DiegoF// // Dichosos aquellos que no esperan nada de la vida, porque nunca seran defraudados ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- //DiegoF Dichosos aquellos que no esperan nada de la vida, porque nunca seran defraudados ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 1.2.12 lost phone registrations today... why?
On Wed, 2006-10-11 at 11:19 -0400, Jerry Geis wrote: I lost my internet connection today for a short time. During that time 1.2.12.1 stopped talking to my phones. Asterisk was still working as I got 2 voicemails. I have TDM analog cards for incoming calls. Anyway my cisco phones had X's (lost registration) and my uniden phones said Registration error. Why would phones loose registration to asterisk when the internet connection and DNS was lost. All phones have hardcoded IP addresses not DNS names. Any ideas? THanks, Looking deep in sip.conf there is ;registertimeout=20 ;retry registration calls every 20 seconds (default) ;registerattempts=10 ;Number of registration attempts before we give up ; 0 = continue forever, hammering the other server until it ; accepts the registration ; Default is 0 tries, continue forever If you have the default i.e. forever this would cause a block at this point for the external registrations and the internals would also become blocked on their reregistration. I had a system try 1547 times to register to it's outside provider (all night) causing the internals to go to No service. So when someone came in in the morning ... -- Dave Cotton Directeur Linux Autrement 193 rue Marcel Cerdan 84270 Vedene FRANCE +33 (0)4 90 23 30 81 http://www.linuxautrement.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GPL Softphones
Hi, I'm searching for GPLed softphones. I found WengoPhone but actually not available for Asterisk PBX, only for Wengo network. I found Kiax but only for IAX protocol. Did you know a good GPLed softphones which works on Windows ? Thanks Greg Apparently (from what I gathered from #openwengo at irc.freenode.net)Wengo's own network runs on a combination of Asterisk and OPENSer. To get Wengophone working with your asterisk you will need to do some code hackingz...so download the source code and change it. You will need to change the authentication procedure in Wengo phone so that your server ip and port numbers are used. Check the gmane mailing lists, I've posted the format of the XML messages used. So basically all the work you will have to do is hardcode your website's url in place of wengo.fr's and make sure your website sends back the right type of XML and bazzam! Oh just a note - its not a simple compile, so you will be messing around with it for a while, trying to get it compile. But its awesome... and IMHO one of the best softphones in the world. Maybe even the galaxy Also, to make things a bit better, some devs are in fact developing the server agnostic version of Wengophone. Hope that helped a bit ok bye :) -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MGCP stuff
Asterisk can only be the proxy/server for MGCP, you connect other devices to it. Asterisk can not be a user agent connecting to other MGCP server. On 10/11/06, Paul Ianas [EMAIL PROTECTED] wrote: Hello everybody! I have an Asterisk 1.2.12.1 server with SIP as the VoIP protocol. What I want to do: I want to talk to the outside world via MGCP. I suppose I must set an MGCP peer to route outgoing calls. So, I must set the endpoint syntax of the Asterisk server (Asterisk will act as an MGCP gateway and will talk with an MGCP Gatekeeper) and with other MGCP gateways via RTP. Ex: DALN/S1/SU0/0@my_address.mydomain.my_dns_suffix Where the part after @ is stored in BTS and contains my telephone number, etc (this is the provider's problem). The question: is this possible with Asterisk? Where can I find some documentation for configuring mgcp.conf? The documentation (Asterisk: The Future of Telephony) says MGCP isn't completely developed. 10q! -- Paul Ianas Programming Engineer Level 7 Software Timisoara, 59D Bucovinei phone: 0744137020 email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Echo problems on ISDN. (mainly incoming calls)
OK I have been battling with echo problems with asterisk on ISDN for a few weeks now, and still can't solve it (although I think I have tried everything I can find.) I will try a post everything I think is possibly relevant that I can remember with the hope someone can point me in the right direction. current spec I am using (Trixbox 1.0 was still in beta at the time an didn't detect the TE110 card, and modules didn't load properly when I tried manually either ) [EMAIL PROTECTED] 2.8 Wildcard TE110P ISDN PRI card (set to E1, I live in the UK) Telewest Q931 ISDN connection (currently 8 lines 100 phone numbers) dell SC1425 server (3.2 Ghz, 512MB, 80GB) SIP phones aastra 9133i run 1.4.0 firmware. This is on a live system with about 50 users. (I have a identical system for DR so can easily test out of hours) Most the time there is no echo, If I phone my wife (on a normal telewest analog line) I get a slight echo (fairly quiet) that she does not noticed. This happens on other phone calls but the user can tolerate this (would prefer to solve it) If a big customer (one with several thousand employees) phones me, I can hear a very loud echo with an annoying delay (0.5-1.0 sec, ish), which the users cannot tolerate. The customer does not get the echo, currently the user phones back? Any suggestion on How to solve the echo problems? I have tried with echo cancellation at 800 and changing the RX and TX gains to no effect. I have read that ISDN should not have echo problems and that may I sould ask about the gains on the line provided by the telco. Can anyone give me more information about this as if I phone Telewest I want to pretend that my PBX supplier has told me to ask XXX. As the first thing they ask me is to contact my my PBX supplier (I had a slight problem getting the ISDN card to initially connect) Thanks John ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] How big is *your* dialplan??
-= 1967 extensions (2838 priorities) in 285 contexts. =- Shared services PBX with a dozen or so customers. -ejay -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Murphy Sent: Tuesday, October 10, 2006 3:17 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] How big is *your* dialplan?? Hello! In my relentless quest for knowledge, I pose this question: who's got the biggest dialplans, and how big are these monsters? What's the biggest dialplan in use right now? If you feel you are a competitor, let me know how many contexts/extensions/priorities you are dealing with. Maybe the context with the most extensions, the extension with the most priorities would be interesting... For example: Digium's dialplan is roughly 50 contexts, 304 total extensions, 870 total priorities. My home system has 100 contexts, 400 total extensions, 935 total priorities. My biggest extension has 129 priorities... no inflation or useless cruft there, either... mostly. These would seem small compared to some dialplans out there, I'll bet. murf -- Steve Murphy Software Developer Digium ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] 1.4 beta2 on intel mac
Lol - use a real PC maybe :P Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Tim Panton Sent: Wednesday, 11 October 2006 1:02 PM To: asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] 1.4 beta2 on intel mac Has anyone built and run asterisk 1.4 beta2 on an intel mac? Did it work? I've got it building ok (once I installed Xcode, wget and bison) However Asterisk hangs on startup (halfway through loading the modules). I have not (yet) had time to debug it, but I wondered if anyone else had done this before me ? Tim Panton www.mexuar.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GPL Softphones
Xlite is not GPL! Joe Dennick wrote: The X-Ten is probably the most know free soft-phone availible. You can find it at http://www.xten.com/index.php?menu=Productssmenu=xlite Gregory Duchatelet wrote: Hi, I’m searching for GPLed softphones. I found WengoPhone but actually not available for Asterisk PBX, only for Wengo network. I found Kiax but only for IAX protocol. Did you know a good GPLed softphones which works on Windows ? Thanks Greg ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2.12.1 dies after asterisk -rx and never comes up again
On 10/11/06, Dinesh Nair [EMAIL PROTECTED] wrote: On 10/11/06 21:15 Joseph said the following: I quits on my as well, when I try to make a second call. There is a bug report on it: http://bugs.digium.com/view.php?id=7972 this seems like a configuration error within FreePBX and isnt really a bug in asterisk. It might be a config issue, but I think you'd agree that a config issue should never segfault the app, and in that respect, we're going to need to do something to get this fixed. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] NFAS Not Passing Audio on B-chan 48,72,96
Kristian Kielhofner wrote: Steve Totaro wrote: Kristian Kielhofner wrote: Steve Totaro wrote: I have NFAS setup on several quad port T1 cards (Sangoma). It mostly works well with the exception that calls coming in on channels 48,72, and 96 have no audio. I tried removing these channels from zapata.conf with hopes that the channels would not come up or be used. Now I get Ring requested on unconfigured channel. How can I busyout these these channels so that incoming calls are not sent to them, or how can I fix the real problem? I think it may be a Sangoma/Wanpipe configuration issue. Thanks, Steve Totaro Steve, I take it you have one D channel for all four spans? On 24? I think this should be pretty transparent to wanpipe. You should configure four spans, with one channel group (type TDM) per span (obvious). You should also probably disable any native D channel features. I always have nothing but problems with that. In zaptel.conf, only specify one d channel. The tricky stuff is in zapata.conf. Can you post that, and maybe zaptel.conf too? -- Kristian Kielhofner Yes the D chan is 24 with no backup. Before I had channel = 1-23,25-96. [trunkgroups] trunkgroup = 1,24 spanmap = 1,1,0 spanmap = 2,1,1 spanmap = 3,1,2 spanmap = 4,1,3 [channels] resetinterval=never callerid=asreceived usecallerid=yes hidecallerid=no callwaiting=no usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes callreturn=yes echocancel=no echocancelwhenbridged=no ;echotraining=800 group=0 callgroup=1 pickupgroup=1 immediate=no rxgain=0.0 txgain=0.0 context=from-pstn switchtype=5ess signalling=pri_cpe channel = 1-23,25-47,49-71,73-95 span=1,0,0,esf,b8zs span=2,0,0,esf,b8zs span=3,0,0,esf,b8zs span=4,0,0,esf,b8zs bchan=1-23,25-96 dchan=24 loadzone=us defaultzone=us Steve, Shouldn't your channel line from zapata.conf look like this: channel = 1-23,25-96 -- Kristian Kielhofner It did but I was getting no audio on those channels so I removed them in hopes that the telco would not try to send call to those channels as a temporary fix while I track down the cause of the problem. How can I just busyout those channels (48,72,96) so that calls are not sent to them from Global Crossing? I have a DS3 and seven servers running NFAS, I do not care about the 3 lost channels per trunkgroup but I do care about customers calling in and getting dead air. Global Crossing charges us $100 per D chan so NFAS is saving us alot of money. Thanks Steve Totaro ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Strange FXS disconnection problem.
As further info, here's the tail of the verbose logging (as enabled in
logger.conf). I have the complete log (but there are lots of irrelevant
SIP transactions for other phones/providers) which I can send if it
becomes helpful.
NB. The mysql server was down for maintenance at the time, so the cdr
errors are expected.
Oct 11 19:46:52 DEBUG[19438]: chan_zap.c:2342 zt_hangup: Hangup:
channel: 2 index = 0, normal = 20, callwait = -1, thirdcall = -1
Oct 11 19:46:52 DEBUG[19438]: chan_zap.c:1435 zt_disable_ec: disabled
echo cancellation on channel 2
Oct 11 19:46:52 DEBUG[19438]: chan_zap.c:2782 zt_setoption: Set option
TDD MODE, value: OFF(0) on Zap/2-1
Oct 11 19:46:52 DEBUG[19438]: chan_zap.c:1372 update_conf: Updated
conferencing on 2, with 0 conference users
-- Hungup 'Zap/2-1'
Oct 11 19:46:52 DEBUG[19438]: app_dial.c:1635 dial_exec_full: Exiting
with DIALSTATUS=CANCEL.
== Spawn extension (davesextensions, 517070, 1) exited non-zero on
'SIP/101-09d925a8'
Oct 11 19:46:52 ERROR[19438]: cdr_addon_mysql.c:144 mysql_log:
cdr_mysql: cannot connect to database server localhost.
Oct 11 19:46:52 DEBUG[19438]: cdr_addon_mysql.c:206 mysql_log:
cdr_mysql: inserting a CDR record.
Oct 11 19:46:52 DEBUG[19438]: cdr_addon_mysql.c:222 mysql_log:
cdr_mysql: SQL command as follows: INSERT INTO cdr
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,dura
tion,billsec,disposition,amaflags,accountcode) VALUES ('2006-10-11
19:43:52','\Dave Bath\ 101','101','517070','davesextensions',
'SIP/101-09d925a8','Zap/2-1','Dial','ZAP/2/17070',180,0,'NO
ANSWER',3,'')
Oct 11 19:46:52 DEBUG[19438]: pbx.c:1522
pbx_substitute_variables_helper_full: Function result is 'Dave Bath
101'
Oct 11 19:46:52 DEBUG[19438]: pbx.c:1522
pbx_substitute_variables_helper_full: Function result is '101'
Oct 11 19:46:52 DEBUG[19438]: pbx.c:1522
pbx_substitute_variables_helper_full: Function result is '517070'
Oct 11 19:46:52 DEBUG[19438]: pbx.c:1522
pbx_substitute_variables_helper_full: Function result is
'davesextensions'
Oct 11 19:46:52 DEBUG[19438]: pbx.c:1522
pbx_substitute_variables_helper_full: Function result is
'SIP/101-09d925a8'
Oct 11 19:46:52 DEBUG[19438]: pbx.c:1522
pbx_substitute_variables_helper_full: Function result is 'Zap/2-1'
Oct 11 19:46:52 DEBUG[19438]: pbx.c:1522
pbx_substitute_variables_helper_full: Function result is 'Dial'
Oct 11 19:46:52 DEBUG[19438]: pbx.c:1522
pbx_substitute_variables_helper_full: Function result is 'ZAP/2/17070'
Oct 11 19:46:52 DEBUG[19438]: pbx.c:1522
pbx_substitute_variables_helper_full: Function result is '2006-10-11
19:43:52'
Oct 11 19:46:52 DEBUG[19438]: pbx.c:1522
pbx_substitute_variables_helper_full: Function result is '(null)'
Oct 11 19:46:52 DEBUG[19438]: pbx.c:1522
pbx_substitute_variables_helper_full: Function result is '2006-10-11
19:46:52'
Oct 11 19:46:52 DEBUG[19438]: pbx.c:1522
pbx_substitute_variables_helper_full: Function result is '180'
Oct 11 19:46:52 DEBUG[19438]: pbx.c:1522
pbx_substitute_variables_helper_full: Function result is '0'
Oct 11 19:46:52 DEBUG[19438]: pbx.c:1522
pbx_substitute_variables_helper_full: Function result is 'NO ANSWER'
Oct 11 19:46:52 DEBUG[19438]: pbx.c:1522
pbx_substitute_variables_helper_full: Function result is 'DOCUMENTATION'
Oct 11 19:46:52 DEBUG[19438]: pbx.c:1522
pbx_substitute_variables_helper_full: Function result is '(null)'
Oct 11 19:46:52 DEBUG[19438]: pbx.c:1522
pbx_substitute_variables_helper_full: Function result is '1160592232.0'
Oct 11 19:46:52 DEBUG[19438]: pbx.c:1522
pbx_substitute_variables_helper_full: Function result is '(null)'
Oct 11 19:46:52 DEBUG[19438]: chan_sip.c:2433 sip_hangup:
update_call_counter(101) - decrement call limit counter
Cheers,
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David Bath
Sent: 11 October 2006 17:08
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Strange FXS disconnection problem.
Hi Tzafrir,
Many thanks for reply.
Busydetect is also disabled. There's no chance of an actual busy
signal, as it happens exactly 2m 40 seconds (give or take 1s) into an
active call with both parties connected and talking away.
Zapata.conf copied below:
[channels]
signalling=fxs_ks
echocancel=64
echocancelwhenbridged=yes
echotraining=400
cidstart=polarity
hidecallerid=no
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
busydetect=no
callprogress=no
progzone=uk
pulsedial=no
answeronpolarityswitch=yes
musiconhold=default
ringtimeout=1000
userincomingcalledidonzaptransfer=yes
usercallerid=yes
cidsignalling=v23
cidstart=usehist
language=en
rxgain=3
txgain=3
context=bt_pstn
channel = 1
Thanks for your time.
Dave
~
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir
Cohen
Sent: 11 October 2006 08:29
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Strange FXS disconnection problem.
On Tue, Oct
Re: [asterisk-users] average waiting time in a queue
[EMAIL PROTECTED] wrote: Hello all, we want to use asterisk queues for a call center application. Depending on the average waiting time in a queue, we want to make a decision to either enqueue a call or transfer it to another site. Are the applications available to query the average waiting time of a queue, if possible for a configurable time frame? Thanks and Regards Markus You can use the queue timeout feauture to go to the next priority in your dialplan which would then be a dial to another site. I have never tried to use variables in the timeout. I figured if an agent does not pickup in two minutes on average, they abandon so we send the calls to the other site at a timeout of 1min 50sec. This was a nasty way of doing what we needed until we implemented much better queue functionality. Thanks, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
