Re: [asterisk-users] Confbridge GUI?
Interesting. Are you using the included cbend.php script to terminate conferences? I occasionally get questions about using WMM with Confbridge, and to date I have not had an answer . If you can provide details, even vague ones, about how you did it, I can update the WMM package. Dan -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Richard Kenner Sent: Friday, October 13, 2017 2:14 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Confbridge GUI? > I have a very old server that is used only for conferences on > Meetme. To manage the conference rooms we use Web Meetme. Now it is > time to upgrade everything but since Meetme is no longer available I > need to find a replacement GUI to manage the conference rooms. Anyone > know a solution that works with Confbridge? It's straightforward to use web-meetme with Confbridge; we've been doing it here for years. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 11 and H.323 trunk using OOH323 - is it stable?
Patrick Lists wrote: On 16-01-14 21:37, Gergely Kiss wrote: Dear List, I'm about to build an Asterisk 11.7 based PBX from scratch for our company. I'm in the middle of the planning phase and it turned out that our VoIP provider prefers H.323 protocol for handling voice calls (while SIP is also supported as plan B). It's SIP everywhere and anyone who requires you, in 2014, to use H.323 should get a clue. Avoid them or at least demand SIP Bah. There is nothing wrong with a working H.323 stack. Just assuming that they will have a working SIP stack because of the date can lead to heartache. As I never worked with H.323 channels in Asterisk earlier, I'm not sure if it's stable enough to be used in production. No idea. Maybe someone else with H.323 experience will respond. AFAIK it's a dead-end. The ooh323 channel has been fairly reliable in our use case, which involve connecting to a commercial IP PBX with crud SIP support. Only you can tell if it will work for you however, as sadly many times new core features only get tested against the SIP channel(s), or worse only implemented there as well. Our current Asterisk version is 11.5.1 Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] meetme conference password and time limitation
Look at Web-MeetMe ( http://sf.net/projects/web-meetme ) If you are on Asterisk 1.6.7 or later you have access to RealTime MeetMe conference storage, otherwise you need to use a script and Asterisk application included with the WMM download. Dan From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad Sent: Tuesday, October 01, 2013 12:43 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] meetme conference password and time limitation Hello; We need to have admin page, so the administrator can create passwords to be used to join the meetme conferences and can determine the allowed time .. Well, the admin interface can be done easy (I do not know if there is something ready), and the password and the time limitation can be added to the database (or even text file), but how asterisk can use it? Do I need to use the AGI to read/write from database and do the meetme conference within the AGI script it self, or there is simpler method? Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] meetme list concise
This list was accurate up to and including Asterisk 11 [0] = Caller # [1] = Callerid Number [2] = Callerid Name [3] = Channel: [4] = 1 for Admin, NULL for User [5] = 1 for Monitor, Null otherwise [6] = 1 for Muted, NULL for UnMuted [7] = 1 for Resquests Floor, 0 otherwise [8] = 1 for 'Is Talking', 0 otherwise [9] = Call duration Dan From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of [Digital^Dude] (r) Sent: Thursday, August 15, 2013 4:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] meetme list concise Hello, Can anyone tell me the format for meetme list concise command, so that I know what field is what (separated by '!'s) Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] meetme list concise
The only way that I know of, and it may not be in all of the 1.6 series, is to use the telephone menu (*5) I think, but would need to dig through the code. Dan From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of [Digital^Dude] (r) Sent: Thursday, August 15, 2013 12:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] meetme list concise Thanks Dan, I found the list arguments from app_meetme.c for asterisk 1.6.x There doesn't seem to be any interface for [8] = Requests Floor. How can we put initially muted users in the request to talk queue? The provision of this parameter in the meet-me source indicates this is doable... but I am unable to find an appropriate way to do it. Any hints would be great help. On Thu, Aug 15, 2013 at 11:03 PM, Dan Austin dan_aus...@phoenix.commailto:dan_aus...@phoenix.com wrote: This list was accurate up to and including Asterisk 11 [0] = Caller # [1] = Callerid Number [2] = Callerid Name [3] = Channel: [4] = 1 for Admin, NULL for User [5] = 1 for Monitor, Null otherwise [6] = 1 for Muted, NULL for UnMuted [7] = 1 for Resquests Floor, 0 otherwise [8] = 1 for 'Is Talking', 0 otherwise [9] = Call duration Dan From: asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com] On Behalf Of [Digital^Dude] (r) Sent: Thursday, August 15, 2013 4:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] meetme list concise Hello, Can anyone tell me the format for meetme list concise command, so that I know what field is what (separated by '!'s) Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Meetme and maxusers option
Thiago wrote: I'm trying to limit the number of participants in a conference room with the realtime option maxusers, but it doesn't work. Asterisk version? Any error messages? Is the conference you are attempting to limit stored in a db (Realtime)? Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Web Meetme module not loading
Rohit Mahajan wrote: Matt Riddell lists at venturevoip.com writes: Are you using the latest version of the app_cbmysql? It looks like it needs to be updated for the latest version. Alternatively it may say somewhere on their website which version of Asterisk this works with? I have been encountering error whenever i run make install to load cbmysql. Below is the error. app_cbmysql.c:529:38: error: macro ast_config_load requires 2 arguments, but only 1 given app_cbmysql.c: In function âload_configâ: How i can resolve this problem. The best way, as the author of app_cbmysql, is to not use app_cbmysql. If you are running Asterisk 1.6.7 or later and and Web-MeetMe 4.X you can use the realtime functionality in app_meetme. The ODBC and realtime setup is a bit more complicated than app_cbmysql, but the reliability will be much better, and you won't need the equally hacky cbend.php script to handle CDR or conference shutdown events. Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI (Primary-NTT)
Edwin wrote: i recently setup an Asterisk system in Hong Kong. their phone company told me that their T1 PRI switch type is Primary-NTT. however in chan_dahdi.conf there's no such option. i have it set to national. it worked fine for a while, but now suddenly stop working. in coming call just keep ringing and didn't even show up on console. out going call hang up immediately with cause code 27. (as usual, phone co. just said it's problem with our equipment without giving us any detail). anybody have any suggestions? The last PRI I setup in Hong Kong was configured as a Primary-Net5, which maps to euroisdn in DAHDI. That was eight years ago, so things may have changed, but it is worth a try. You should also collect some Q.931 logs, as I have seen silly things like caller-id formatting cause calls to be rejected. Sadly I cannot tell you how to accomplish that with DAHDI... Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and OpenLDAP
Giuseppe wrote: Yes, but i think that's better to open an LDAP connection with extensions user and password. Or not? Better is not the right way to look at it. You questions is about early or late binding. Early binding requires a dedicated username and password to connect to LDAP before it can perform a query, and late can use the user provided credentials. I find that many applications will support only one or the other, so the choice is made for you. I do not know if Asterisk supports only early binding, but I suspect that it would be a better long term match for you. Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] meetme identify user number
Daniel wrote: Hi Group, is in MeetMe any option to identify the own number (from the view of a caller)? I would like to write an option to set on the telephone an request for voice, if the room is muted. That request should display on our Conference Control Website and an Admin should unmute this person. If you have the user menu enabled, and the user is muted, then option 2 sets a 'Requests the Floor' flag. I know that the conference display feature in Web-MeetMe can interpret that flag and display a message that the caller would like to be unmated. I don't know of any other conference management apps that do, but I really have not looked into it. The request the floor feature was added in one of the early 1.6 releases, so unless you are on a truly ancient version, the backend support should be there. Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8.11.0 Debian Squeeze packages with T.38 gateway, queue hints and fixed RFC4235 (notifycid=yes)
Kevin P. Fleming wrote: This is a valid point, and we'll get this corrected. Our package repository should have packages for Asterisk 10, but it doesn't. How likely is it that a Centos 6 repo might be setup at the same time? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ISDN PRI configuration
Tony wrote: Kevin P. Fleming kpflem...@digium.com wrote: As I said before... an Ethernet cable will work nearly all the time, and at a 5m length it's probably fine. Kevin, under what circumstances would an Ethernet cable potentially not work with T1/E1? And in those circumstances, what should be used instead? I'm wondering because I had never realised it was an issue until you said. I've never had an issue with using Cat5 cable, but I have run into telco/techs that choose to use a pin out other than 1245, and of course defend it with 'That is our standard way to do it'. So a standard Ethernet cable would fail, but once one end was cut off an replaced with the required pin out it would work fine (but no longer be an Ethernet cable, semantics but important). Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Talk detection in meetme
Eyal Mahalal wrote: I create Chat room with MEETME and now I have a problem. I want that the host of the room could identify the participants in the room by their speech, so that if a participant uses language the host could kick him from the room. Is there a way to do it? This is one of the features of the monitor page in Web-MeetMe. The key components are: 1. A web page that refreshes every x seconds. 2. Configuring Asterisk Manager interface to allow connections. 3. Code to connect to the manager interface an list the callers. a. I use PHPAGI in WMM, but there are other libraries to choose from. Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco IP Phones and Skinny in asterisk
You do not need sccp.conf if you are not using chan_sccp. It has different features(bugs) than chan_skinny, but yes it would also reset the phones (if it supports reload, and I have no idea if it does). Also if the phone is in a call it will not reset until after the user hangs up. Reloading the channel triggers a soft reset that causes the phone to request its configuration, which may have changed. Dan -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad Sent: Tuesday, June 21, 2011 1:15 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Cisco IP Phones and Skinny in asterisk Dear Dan; I have to do something in the compilation to have chan_sccp? Because, I do not have this channel and I have only chan_skinny. Even in the /usr/lib/asterisk/module/, I did not find chan_sccp. Maybe that is the reason why I do not have the sccp.conf file? So, using the sccp channel, will also face the same problem that the phones will restarted if I did reload? Regards Bilal --- On Mon, 6/20/11, Dan Austin dan_aus...@phoenix.com wrote: From: Dan Austin dan_aus...@phoenix.com Subject: RE: Cisco IP Phones and Skinny in asterisk To: bilal ghayyad bilmar...@yahoo.com Date: Monday, June 20, 2011, 7:09 PM It would be best to keep this on the list, I just had not had a chance to reply yet. Your first issue is just how the SCCP protocol works. Every keypress is relayed to the server, so the phones must maintain an active connection to the PBX. You can avoid this by just reloading the modules you update and not the whole PBX- ie- sip reload or module reload chan_sip The second issue is likely a firmware issue on the phone, and Likely one where the phone software is too new. You might also Not have the correct definition in skinny.conf I did use chan_sccp years ago, but have not kept up with it. The configuration should be with the source package for that channel. The configuration is similar, but you cannot rename the files as there are key differences. Dan -Original Message- From: bilal ghayyad [mailto:bilmar...@yahoo.com] Sent: Monday, June 20, 2011 3:40 PM To: Dan Austin Subject: Re: Cisco IP Phones and Skinny in asterisk Dear Dan; Because you are using skinny with your Cisco IP Phones in the office, so I beleive you might help me really to resolve my problem (please). First of all, are u using skinny channel or sccp channel? Actually, I tried skinny and I faced two major problems (so if I am going to face same problems in sccp then no need to use sccp, so please advise). The two problems that I faced them are: 1) When I do reload then the skinny channel is reloaded and that will cause a restart for the Cisco IP Phones (that are registered to skinny channel). Is the same thing happening with u when u r using sccp channel? 2) When I called the Phone, it is ringing, when we pickup the handset to answer the call, we hear t and we do not hear what source is talking and source does not hear us even .. but if we select music on hold, then caller will hear the music. Also, when we tried to use the Ciscp IP Phone to place a call, while we are dialing, the too tone is always existed and it is ringing at destination but no voice (always t). So if I used sccp then I will not face these problems? From the other side, if I need to use sccp (if we assumed the above problems are not existed) then can u please help for below: 1) If i used sccp and I gave the IP Phone the IP address TFTP server, and no configuration files were existed on TFTP, then it will register on the asterisk sccp channel? 2) The sccp.conf file, where I can find it? Is it the same as the skinny.conf file? 3) To use sccp instead of the skinny channel, all what I need is to unload the skinny from the modules.conf file and load the sccp channel in the modules.conf, and I can use the skinny.conf file for the configuration? About the firmware on the Phone, it will stay the same? I appreciate the kindly help please. Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo
Re: [asterisk-users] Cisco IP Phones and Skinny in asterisk
The Asterisk version is 1.8.3.2 The Cisco IP Phone is 7942G and it is running now skinny. The used TFTP is tftp-server which is installed in fedora. I placed the following two files (but look like it was not taken from the TFTP, as nothing appeared in the messages), but I am able to to ping from the asterisk box to the vlan that the Phone is connected, so no problem in the reachability: SEPB8BEBF22AB62.cnf.xml xmlDefault.CNF.XML Are the files name correct? Or the Cisco IP Phone 7942G are not working fine with Asterisk or the tftp-server? Cisco has changed the file name format a few times, so you may want to copy xmlDefault.CNF.XML to XMLDefault.cnf.xml The more important steps is how have you configured the phone to locate the TFTP server? Are you using option 150 in DHCP, or manually setting the TFTP server address on the phone. Technically you do not need a TFTP server, since the Skinny phones will try to use the TFTP server address for registration, so you can just set that address to point to your asterisk server. A TFTP server is needed if you want custom ringtones or to manage software updates. For small setups or my home, I skipped setting up the TFTP server until I wanted to update the phone firmware. Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The new ConfBridge application is now in Asterisk Trunk!
Richard wrote: No, conference scheduling is not a feature that we have built directly into ConfBridge, and I'm debating on what it would look like. Scheduling isn't built into MeetMe either, but the fact that it dynamically reads from a database means that you can write external programs (such as Web-Meetme) that create conferences that MeetMe can read. For me, in order for ConfBridge to be at all interesting, it needs the same functionality. In the context David is referring to, yes scheduling is in MeetMe. Web-MeetMe is used to maintain/manipulate the database that MeetMe reads. Prior to 1.6.1.7 the scheduling logic was in an out of tree application that would validate the conference details and pass control to app_meetme. Ending a conference depended on a relatively unreliable PHP script. The RealTime/Scheduling features trickled in during the early 1.6 releases, with 1.6.1.7 being the first release with all of the basic features need to work with Web-MeetMe. I recommend 1.6.2 or 1.8 for use with Web-MeetMe. David- I have been watching the announcements about the new ConfBridge with mixed feelings. I am excited about the new mixing options and profiles, and dreading the flood of 'When will you update Web-MeetMe to support ConfBridge?' questions. If it had at a minimum the option to use RealTime to override pins and start/end time logic, I would likely have a version supporting ConfBridge and profiles during 1.10 RC timeframe. As it stands I am not planning to go back to using pre/post-call processing for scheduling in WMM. I've started and deleted this message several times since the first announcement, as it almost always reads as an ultimatum and it is absolutely not meant to be. I really do think it represents a major leap forward in Asterisk conferencing. I simply feel I must let the WMM users know that support for it is highly unlikely. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX Call token revisited
Kevin wrote: On 03/21/2011 06:49 PM, Dan Austin wrote: I just finished a fresh install of 1.8.3.2 at home using the packages Digium hosts. After correcting a number of typo/config'o error that had crept in over the years, I thought I had everything working. My wife just complained that she cannot call her mother (who is using an old IAX hardphone I left for her). After turning up the logging level I see- chan_iax2.c: Call rejected, CallToken Support required Which google cays can be fixed with: [general] calltokenoptional=0.0.0.0/0.0.0.0 maxcallnumbers=16384 or [peer] requirecalltoken=no (or auto) Either set of changes does suppress the error, but the remote device still fails to register. No other errors/warnings are present. If there aren't any errors or warnings appearing, then you must not have the logging verbosity set high enough. Ensure that you've used 'core set verbose 10' and 'core set debug 10', and that your 'console' channel in logger.conf has all the logger levels enabled. If you still don't see what you are looking for, use 'iax2 set debug' to enable IAX2-specific debugging for that phone's IP address. I should have said relevant errors/warnings. I see info about devastate and queues, but little else. That said I think the problem is unrelated to call token and an issue with the NAT firewall at my mother-in-laws. The incoming traffic is on a very high port and not 4569. I interpret the following log as her phone is not receiving the replies- Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 3ms SCall: 14807 DCall: 0 [120.xxx.xxx.12:22686] USERNAME: XXX REFRESH : 60 Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGAUTH Timestamp: 00013ms SCall: 03287 DCall: 14807 [120.xxx.xxx.12:22686] AUTHMETHODS : 3 CHALLENGE : \x34\x36\x37\x38\x33\x35\x33\x33 USERNAME: XXX Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 3ms SCall: 14807 DCall: 0 [120.xxx.xxx.12:22686] USERNAME: XXX REFRESH : 60 Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 3ms SCall: 03287 DCall: 14807 [120.xxx.xxx.12:22686] Tx-Frame Retry[001] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGAUTH Timestamp: 00013ms SCall: 03287 DCall: 14807 [120.xxx.xxx.12:22686] AUTHMETHODS : 3 CHALLENGE : \x34\x36\x37\x38\x33\x35\x33\x33 USERNAME: XXX Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 3ms SCall: 14807 DCall: 0 [120.xxx.xxx.12:22686] USERNAME: XXX REFRESH : 60 Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 3ms SCall: 03287 DCall: 14807 [120.xxx.xxx.12:22686] Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 3ms SCall: 14807 DCall: 0 [120.xxx.xxx.12:22686] USERNAME: XXX REFRESH : 60 Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 3ms SCall: 03287 DCall: 14807 [120.xxx.xxx.12:22686] Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: LAGRQ Timestamp: 10015ms SCall: 03287 DCall: 14807 [120.xxx.xxx.12:22686] I've asked my wife to have her mother reboot her router and phone, but that has not happened yet. Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX Call token revisited
I just finished a fresh install of 1.8.3.2 at home using the packages Digium hosts. After correcting a number of typo/config'o error that had crept in over the years, I thought I had everything working. My wife just complained that she cannot call her mother (who is using an old IAX hardphone I left for her). After turning up the logging level I see- chan_iax2.c: Call rejected, CallToken Support required Which google cays can be fixed with: [general] calltokenoptional=0.0.0.0/0.0.0.0 maxcallnumbers=16384 or [peer] requirecalltoken=no (or auto) Either set of changes does suppress the error, but the remote device still fails to register. No other errors/warnings are present. I'm hoping a downgrade to 1.6/1.4 is not required, but google is not proving to be helpful in this case. Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 (Web-meetme)
Manmohan wrote: I am really not sure if this is related to the meetme in asterisk OR this is something to do in web-meetme. I tried to google but didnt get any proper results. I am facing one issue in Web-meetme on the expiry of any conference that we create. I am having Web-meetme 4.0.2 over Asterisk 1.6 Any conference we create in web-meetme never expires. I am not sure if i am missing while configuring, though i didnt find anything in lib/define.php also checked in asterisk that can point me, which can help me in fixing this issue. Since you can join the conference you created with WMM, the Realtime settings are likely correct. You do not mention which version of 1.6 you are on, so I would guess that you are on 1.6.2.7 or older. For a variety of reasons the realtime feature, in particular the scheduling code, was added and tweaked over a wide range of 1.6 releases. The first one I would consider feature complete for use with Web-MeetMe is 1.6.2.7, and even that version has a trivial, but nasty bug that causes segfaults (fixed in a recent 1.6.2 release) Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 (Web-meetme)
Manmohan wrote: I am currently on Asterisk 1.6.2.14. Do you have schedule=yes in meetme.conf? I incorrectly remembered/thought that all of the Realtime features were controlled by that option, only a small number, such as end times and CDR logging On Sat, Dec 4, 2010 at 12:09 AM, Dan Austin dan_aus...@phoenix.com wrote: Manmohan wrote: I am really not sure if this is related to the meetme in asterisk OR this is something to do in web-meetme. I tried to google but didnt get any proper results. I am facing one issue in Web-meetme on the expiry of any conference that we create. I am having Web-meetme 4.0.2 over Asterisk 1.6 Any conference we create in web-meetme never expires. I am not sure if i am missing while configuring, though i didnt find anything in lib/define.php also checked in asterisk that can point me, which can help me in fixing this issue. Since you can join the conference you created with WMM, the Realtime settings are likely correct. You do not mention which version of 1.6 you are on, so I would guess that you are on 1.6.2.7 or older. For a variety of reasons the realtime feature, in particular the scheduling code, was added and tweaked over a wide range of 1.6 releases. The first one I would consider feature complete for use with Web-MeetMe is 1.6.2.7, and even that version has a trivial, but nasty bug that causes segfaults (fixed in a recent 1.6.2 release) Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 (Web-meetme)
The errors you posted do not point to a the problem. Did you build from source or are you using packages? If from source, grep for useropts in app_meetme.c and The second instance should be: char useropts[OPTIONS_LEN + 1] = ; If the line does not have the = , then the issue is that the bug I mentioned is still present. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Manmohan Singh Jandu Sent: Friday, December 03, 2010 1:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.6 (Web-meetme) Hi Dan, In meetme.conf the schedule=yes was commented, after removing its working fine. But one strange thing had started now. I started getting segmentation fault. following are the errors which i see in it: warning: difference appears to be caused by prelink, adjusting expectations warning: .dynamic section for /usr/lib/libodbcinst.so.1 is not at the expected address warning: difference appears to be caused by prelink, adjusting expectations warning: .dynamic section for /usr/lib/libogg.so.0 is not at the expected address warning: difference appears to be caused by prelink, adjusting expectations warning: .dynamic section for /usr/lib/libbluetooth.so.2 is not at the expected address warning: difference appears to be caused by prelink, adjusting expectations Reading symbols from /lib/libssl.so.6...(no debugging symbols found)...done. Loaded symbols for /lib/libssl.so.6 Reading symbols from /lib/libcrypto.so.6...(no debugging symbols found)...done. Loaded symbols for /lib/libcrypto.so.6 Reading symbols from /lib/libc.so.6...(no debugging symbols found)...done. Loaded symbols for /lib/libc.so.6 Reading symbols from /usr/lib/libxml2.so.2...(no debugging symbols found)...done. Loaded symbols for /usr/lib/libxml2.so.2 Reading symbols from /usr/lib/libz.so.1...(no debugging symbols found)...done. Loaded symbols for /usr/lib/libz.so.1 Thanks Regards Manmohan Singh On Sat, Dec 4, 2010 at 1:12 AM, Dan Austin dan_aus...@phoenix.com wrote: Manmohan wrote: I am currently on Asterisk 1.6.2.14. Do you have schedule=yes in meetme.conf? I incorrectly remembered/thought that all of the Realtime features were controlled by that option, only a small number, such as end times and CDR logging On Sat, Dec 4, 2010 at 12:09 AM, Dan Austin dan_aus...@phoenix.com wrote: Manmohan wrote: I am really not sure if this is related to the meetme in asterisk OR this is something to do in web-meetme. I tried to google but didnt get any proper results. I am facing one issue in Web-meetme on the expiry of any conference that we create. I am having Web-meetme 4.0.2 over Asterisk 1.6 Any conference we create in web-meetme never expires. I am not sure if i am missing while configuring, though i didnt find anything in lib/define.php also checked in asterisk that can point me, which can help me in fixing this issue. Since you can join the conference you created with WMM, the Realtime settings are likely correct. You do not mention which version of 1.6 you are on, so I would guess that you are on 1.6.2.7 or older. For a variety of reasons the realtime feature, in particular the scheduling code, was added and tweaked over a wide range of 1.6 releases. The first one I would consider feature complete for use with Web-MeetMe is 1.6.2.7, and even that version has a trivial, but nasty bug that causes segfaults (fixed in a recent 1.6.2 release) Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks Regards Manmohan Singh -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Adaptive CDR and default fields
I'm running 1.6.2.13 and need to record a small number of custom values use cdr_odbc and cdr_adaptive_odbc, and only the custom fields. The good news is that the custom records are being stored in the database as desired. The bad news is that I get three sets of warnings/notice about 'SQL Exec Direct failed' and dropping then reconnecting the database handle. I traced the SQL calls and found these occur when the CDR engine attempts to record all of standard CDR fields. The cdr_adaptive_odbc documentation suggests that it is safe to drop the standard fields, and while my system does continue to function the dropping of the db handle and extra logging is annoying. Have I missed an option to disable recording the standard CDR fields? Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Crashes - Segmentation Fault
Manmohan wrote: I commented locale.php in defines.php and it perfectly worked well. Now i am wondering what is this invite participants for, while adding conference. wherein it asks for first name, lastname, emailaddress telephone number.. The 'Invite Others' option is mostly for installs that do not have a consistent e-mail environment, and are using the SERVER mailer. This feature lets the server send invite emails to multiple parties. In my environments we have Exchange and Outlook, so I prefer the CLIENT mailer, and I can manage the invitations in my mail client Let me brief you how i had done this setup. I had created a SIP trunk between Cisco Call manager and Asterisk server. And i am using webmeetme for Audio conferencing. Sounds familiar. I put this package together after wasting too much money and time trying to make an expensive Cisco conferencing solution work. Other than the invite participants, while the conf call is going on we get couple of more options, when we click to the current ongoing conference number. End call -- To end the conference call Yes Extend -- I am sure this is to extend the time of the call for which its scheduled for, but not sure on how much time does it extends by default OR is there any way to define the customized time on whatever required. 10 minutes is the default. I thought I had made it configurable in lib/defines.php, but no I have it hard coded in conf_add (to be fixed in the next release now). You can search for +600 and change it to any value you like. Invite-- When i click this button it asks me telephone number. I assume this is any number which asterisk server can reach as per the dialplan configured in extension.conf in /etc/asterisk.. Though this invite button looks pretty much interesting to use but whenever i enter any phone number it says System error not sure if am understand this wrongly. You understand it correctly, but the default settings are likely not working. Check out the section 'Outcall defaults' in lib/defines.php. It is likely you need to change the OUT_CONTEXT at a minimum. Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Crashes - Segmentation Fault
Manmohan wrote: I had tried the new version of webmeetme i.e., 4.0.2 The recording works very well. Great! I see following php errors whenever i try to add in conference. [Wed Aug 04 16:29:02 2010] [error] [client 32.93.17.40] PHP Notice: Undefined variable: order in /var/www/html/web-meetme/meetme_control.php on line 278, referer: http://10.1.1.30/web-meetme/meetme_control.php?s=4 You can ignore the Notices. They are fairly harmless, and only mean that variable is not set by the code or being passed in on the URL. You can turn off notices in /etc/php.ini if they bother you. Also the Reports link doesnt display anything and in httpd error logs it gives me following php errors: [Wed Aug 04 16:30:22 2010] [error] [client 32.93.17.40] PHP Warning: include(locale.php) [a href='function.include'function.include/a]: failed to open stream: No such file or directory in /var/www/html/web-meetme/lib/defines.php on line 3, referer: http://10.1.1.30/web-meetme/daily.php? In lib/defines.php, either comment out the 3rd line or add ../ before locale.php- include(../locale.php); But that is not likely why you do not get the reports. The most likely cause is A PHP notice is being thrown while the GD code is rendering the graph, resulting in a corrupt image which your browser cannot display. Check these settings /etc/php.ini- error_reporting = E_ALL display_errors = Off Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Crashes - Segmentation Fault
Manmohan wrote: I did added the record option in user options as well. $Mod_Options = array(array(_(Announce), I), array(_(Record), r)); $User_Options = array(array(_(Announce), I), array(_(Listen Only), m), array(_(Wait for Leader), w), array(_(Record), r)); And the gre8 news is, it did worked this time. But it saved the recorded file in the following path: That is good to hear. /var/lib/asterisk/sounds/ with the name as meetme-conf-rec-74438-1280463795.8.wav Than i tried to move the file to /var/lib/asterisk/sounds/conf-recordings/ just to see that it gives me a speaker icon when i click to past conferences. Unfortunately i couldnt see this speaker icon to hear this recorded conference .wav file. I am not surprised. By default MeetMe creates unique file names by appending pin-uniqueid, but uniqueid is not know until the conference starts, so the web interface does not know what to look for. Part of the changes to app_meetme included setting the realtime filename to use. I tried to download the .wav file into my windows machine and the filed played well.. like i mentioned in my earlier mail that following line i had added in lib/define.php, please correct me if i am wrong: define (RECORDING_PATH, /var/lib/asterisk/sounds/conf-recordings/); Do you think This recording path is taking the effect here? That setting effect where the WMM interface looks for recordings and not where Asterisk puts them. Looking back at your email history, I see you are on 4.0.1. After all of the suggestions, I remembered that I too found problems with recordings and addressed them in 4.0.2 That version adds a field to the database and stores the recording names in the database. I recommend using that version instead of 4.0.1. You can move your copy of lib/defines.php to the 4.0.2 install and keep your changes. Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Crashes - Segmentation Fault
Manmohan wrote: There was on very silly mistake and i missed to check that properly. Really apologize for that. Following change was done to get the conf-recording into the proper path: chown -R asterisk:asterisk /var/lib/asterisk/sounds/conf-recordings following is the output: [r...@linuxtest sounds]# ll total 6416 drwxrwxr-x 2 asterisk asterisk 4096 Jul 30 08:29 conf-recordings [r...@linuxtest sounds]# ll conf-recordings/ total 4060 -rw-r--r-- 1 asterisk asterisk 4150124 Jul 30 08:27 meetme-conf-rec-74438-1280463795.8.wav The only thing now is no speaker icon onto the webpage when i click to past conference link. The web interface cannot find the recording. The reason it cannot is that the name is wrong. By wrong, I mean it contains information that the database and program is not aware of (1280463795.8). To make this clear, if this conference was the 3rd one you ever scheduled on this system the correct file name would be- meetme-conf-rec-74438-3.wav using the format meetme-conf-rec-%PIN%-%BOOKID%.wav The database knows the pin and bookid, so it can construct the file name and test if it exists. Do you say that if i shift from 4.0.1 to 4.0.2 will fix the issue (of getting speaker icon in past conference)? I was not able to get the change into app_meetme to use the bookid in the filename, even though it has access to bookid. I gave up and now store the filename in the database, which app_meetme will use if it exists. Other that a handful of bug-fixes, this is the major difference between 4.0.1 and 4.0.2 Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Crashes - Segmentation Fault
Manmohan wrote: Following is the output for core set verbose 5, also i am really not able to get on the admin pin thing? Do you mean, that with admin pin configured we cant use recording? You are actually running a version that has been fixed to support recording with pin-less or user pins. I should point out that the default settings in WMM is only to present the recording checkbox with the admin pin field. It is a fairly simple edit to add the recording checkbox to the user pin (and these options apply if no pin is set). Look in lib/defines.php for Mod_Options and User_Options to See how to add or remove MeetMe options from the GUI and database. Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Crashes - Segmentation Fault
Manmohan wrote: I can see the path does exists but i cant see any recordings happening inn there. There are no files in it Following is the output: /var/lib/asterisk/sounds drwxrwxrwx 2 asterisk apache 4096 Jun 27 20:54 conf-recordings I hope m understandly this correctly but m sure m missing something here ;-) You did understand, and we have eliminated another of the possible issues. Are you assigning an admin pin to these conferences? There is a patch that allows recording pinless concenferences, but is has oddly not been merged yet. Try setting an admin pin. If that does not work, send the CLI output with core set verbose 5 as you dial in to the conference. Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] chan_skinny still maintained?
Jonathan wrote: I've managed to acquire a few Cisco handsets (7905, 7920) and would like to use them with Asterisk. Rather than simply switching to the SIP firmware I thought I'd use these with chan_skinny - partly because this is Cisco's primary firmware and therefore the phones might be more stable, and partly to help test chan_skinny as it seems to be generally underused. (Is functionality identical across both firmwares?) However, I've come across a couple of showstoppers and am not really sure where to go from here. I've raised bugs for both of them (#17680, #17692) and had no response so far - have I perhaps overestimated how much chan_skinny is in use these days, or do I need to follow another route? The problem in 17680 has been worked on a couple of times and I believe the issue is not actually in chan_skinny, although it seems easiest to trigger from that channel. I had thought that the problem described in 17692 had also been put to rest, but the more I think about it, I seem to remember a potential fix was deferred pending a re-write of the subchannel handling code. I'll dig around in my archives to see if I can find my old patches for either of these. I'm not an Asterisk developer but am happy to spend some time this week resolving the problems. Unfortunately I need the phones next week, so may have to end up taking the defeatist approach of switching to the SIP firmware :( Regardless of whether the fixes were available, there is no way they would be reviewed, merged and released within the next week... Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Crashes - Segmentation Fault
Manmohan Singh Jandu wrote: Excellent! I finally got it working, it was ODBC drivers issue actually. Installed the proper compatible version and its working. I thought that might be the case. There are still few errors which i see on asterisk console: [Jul 19 13:58:51] WARNING[30213]: res_config_odbc.c:1032 require_odbc: Realtime table book...@meetme requires column 'members', but that column does not exist! WMM does not use that column. You can disable it by Setting logmembercount=no in meetme.conf Also when i try to click the conference to manage it realtime it gives me Error connection to the manager! Following are the database files which i used: /web-meetme/cbmysql/db-admin-user-create.txt /web-meetme/cbmysql/db-table-create-v6.txt /web-meetme/cbmysql/db-tables-v6.txt Am i missing something here now? The WMM web interface used the Asterisk manager interface to monitor and manage conferences. The readme file documents the required changes to manager.conf. Sorry for the delay responding, I was on vacation last week with no email access. Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Crashes - Segmentation Fault
Manmohan Singh Jandu wrote: OK, now i added the column members in the table booking manually. and disabled selinux to have this working. Now i am struggling with recording option in webmeetme. Not sure on how to enable it, though m checking the checkbox while creating the conference. But where does this save and how to retrieve it? The location of the recordings is set in lib/defines.php as RECORDING_PATH, which defaults to /var/lib/asterisk/sounds/conf-recordings/ You can listen to the recordings after the conferences scheduled stop time by looking at the Past conferences page and clicking on the speaker icon next to the conference number. A couple of items to note- 1. You may have to check the path to ensure it exists and that the asterisk process can write to it. 2. Your web service accounts needs read permissions for that path 3. The speaker icon only displays if a recording exists. Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] chan_skinny still maintained?
Jonathan wrote: On 26 July 2010 23:50, Dan Austin dan_aus...@phoenix.com wrote: I'll dig around in my archives to see if I can find my old patches for either of these. Many thanks - I'm happy to test patches if I can do so. At least I can contribute in that way, even if I'm not directly contributing wonderful code modules.. Regardless of whether the fixes were available, there is no way they would be reviewed, merged and released within the next week... I can hope, right? :-) The transfer issue is straight forward with two problems- 1. We always assume we have a second sub-channel (we don't, and When we don't we should create one) 2. We forget to tell the phone we have gone back on hook. The first is 16 lines copied from the redial softkey and the second is a simple callstate update. Very limited testing, but a patch will be on the bugtracker soon. The park issue is indeed very familiar, but I do not see any patches in my archive. The system is trying to playback the parked extension, but the parking channel has already been masqueraded away. I have a hack patch for that, but someone who understands the park/features code may have a better fix. Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Crashes - Segmentation Fault
Manmohan wrote: Unfortunately m not able to get rid of the below mentioned errors. not sure on where i am missing now. On Sat, Jul 10, 2010 at 9:41 AM, Manmohan Singh Jandu manmoha...@gmail.com wrote: Ahh here is the catch i was still using app_cbmysql for this. now i had removed and just followed the README of 4.0 for WMM and m getting following on ,my asterisk console. Verbosity is at least 3 == Using SIP RTP CoS mark 5 -- Executing [...@phones:1] MeetMe(SIP/492-, ) in new stack -- SIP/492- Playing 'conf-getconfno.ulaw' (language 'en') == Parsing '/etc/asterisk/meetme.conf': == Found [Jul 10 13:42:15] NOTICE[16906]: res_odbc.c:1427 odbc_obj_connect: Connecting meetme [Jul 10 13:42:15] WARNING[16906]: res_odbc.c:1452 odbc_obj_connect: res_odbc: Error SQLConnect=-1 errno=0 [unixODBC][Driver Manager]Data source name not found, and no default driver specified [Jul 10 13:42:15] WARNING[16906]: res_odbc.c:1273 ast_odbc_request_obj2: Failed to connect to meetme [Jul 10 13:42:15] ERROR[16906]: res_config_odbc.c:144 realtime_odbc: No database handle available with the name of 'meetme' (check res_odbc.conf) -- SIP/492- Playing 'conf-invalid.ulaw' (language 'en') -- SIP/492- Playing 'conf-getconfno.ulaw' (language 'en') -- SIP/492- Playing 'conf-getconfno.ulaw' (language 'en') == Spawn extension (phones, 493, 1) exited non-zero on 'SIP/492-' (Initially i installed using yum, i was getting the same issue. Than i scrapped everything and installed it manually.) The good news is that you are making progress. Do you have the package unixODBC installed? The hint to that would have been if you created a new /etc/odbc.ini instead of editing a sample that the package would have installed. Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Crashes - Segmentation Fault
Manmohan wrote: My Web-MeetMe_v4.0.1, i followed the instructions in the README File in the same package. Good. There are other instruction packages, but since I wrote the README it is the one I am most familiar with. Are you using RealTime enabled app_meetme or app_cbmysql from the WMM package? i didnt get this actually what do i need to check here? Please dont mind but m not so good in opensource world. I try to read and understand and on trial n error basis try to implement things. Though had very much interest in learning things. Before version 4 of Web-MeetMe (WMM) the scheduling logic for Asterisk was in a separate Asterisk application (app_cbmysql). With version 4 of WMM and Asterisk 1.6.2 and later the logic is not part of the MeetMe application. The README in 4.0.1 lists the steps to setup RealTime (database) support for Asterisk and MeetMe. This narrows down the possible problems, since we do not need to consider app_cbmysql. Did you install Asterisk from a package with yum, or did you compile it yourself? Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Crashes - Segmentation Fault
Manmohan wrote: I was looking for audio conferencing solution where i got Web-meetme. I had installed Asterisk 1.6.2.9 on Centos 5.4. Its perfecting working fine. I tried using Meetme even meetme app is working perfectly fine. I installed Webmeetme 4.0 and integrated with my asterisk. When i try to dial the conference number it take me to an IVR wherein it asks for the conference number. The time i provide the conference number, asterisk crashes giving segmentation fault. I have been trying to google up and checked lot of forums but didnt get any solution for this yet. Which instructions did you follow for the integration? Are you using RealTime enabled app_meetme or app_cbmysql from the WMM package? Which exact version of WMM? Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Web Meetme module not loading
Matt wrote: On 1/09/09 5:53 PM, Glen wrote: Matt Riddell wrote: In the latest readme for WebMeetMe (3.1.0) it states: * Compile and install CBMySQL App_cbmysql is now included in the web-meetme package, located in ./cbmysql. To install just run make; make install Copy the sample cbmysql.conf to /etc/asterisk and create a dialplan similar to the one in cb-extensions.conf.sample Modify the settings to suit your system. The location of the mysql.sock file is likely not correct, check /etc/my.conf for the correct location. That fixed it Matt, just compiling in the wrong directory. Thanks for all your help. No problems :) I haven't actually used it myself, but it looks pretty cool! Matt- Thanks for jumping in. I have been offline for close to four weeks recovering from oral surgery. Months go by without a single Web-MeetMe question, and as soon as I stop watching email a bunch show up... As was discovered, the app is compiled separately from Asterisk. Due to changes in the AMI interface over the years, there will soon be three versions of WMM- 2.X for Asterisk 1.2 (largely unmaintained, but problem reports are rare) 3.x for Asterisk 1.4 4.X for Asterisk 1.6 (recommend 1.6.0.7 or 1.6.1 or newer) Starting in 1.6 the scheduling logic that was in app_cbmysql is now native to app_meetme when using RealTime, so app_cbmysql has not been updated for 1.6. I need to get 4.X released. I normally like to run a new release in house for a couple of months before release, but between the surgery and changes at work it is not likely to happen. The good news is that the 4.X release has less to test since the scheduling logic moved into app_meetme, so I just need to confirm that nothing is seriously broken in the UI. Thanks, Dan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk to CCM
Make sure you are stripping the 8 on inbound calls to that H323 gateway
under CCM and that it has a valid search space to find your extensions...
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jimmy Ezell
Sent: Tuesday, June 09, 2009 3:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Asterisk to CCM
Hit another problem in my tutorial in converting over from Cisco CallManager to
Asterisk.
I have been following the instructions at :
http://voip-info.capatres.com/wiki/view/Asterisk+Cisco+CallManager+Integration.html
on intergrating Asterisk and Cisco CallManager.
I can make calls from CCM to Asterisk phones - and yes that felt good to get
that working.
My problem is that it does not work from the other direction. I cannot make
calls from CCM phones to Asterisk Phones.
I want to be able to dial 8 and the extension of the ccm phone.
I am using CCM 3.3.(5) so I do not have the option to use a SIP turnk because
it is not supported. I am also using h323 instead of ooh323. Not sure if that
might make a difference.
In Asterisk console I get:
-- Executing [8...@internal:1] Dial(SIP/207-08bd64c8,
H323/callman02/2...@172.16.200.10:1720mailto:H323/callman02/2...@172.16.200.10:1720)
in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called
callman02/2...@172.16.200.10:1720mailto:callman02/2...@172.16.200.10:1720
== Everyone is busy/congested at this time (1:0/0/1)
This is the contents of my h323.conf file:
=
[general]
port = 1720
bindaddr = 172.17.100.2
disallow=all
allow=gsm ; Always allow GSM, it's cool :)
allow=ulaw ; see doc/rtp-packetization for framing options
allow=alaw
dtmfmode=rfc2833
gatekeeper = DISABLE
context=default
[callman02]
type=friend
context=default
ip=172.16.200.10
port=1720
disallow=all
allow=gsm
allow=ulaw
allow=alaw
dtmfmode=rfc2833
nat=no
canreinvite=yes
qualify=yes
extensions.conf file
==
[globals]
CISCOTRUNK=H323/callman02
[cisco]
exten = _8XXX,1,Dial(${CISCOTRUNK}/${EXTEN:1...@172.16.200.10:1720)
exten = _8XXX,n,Congestion()
exten = _8XXX,n,Hangup()
Jimmy Ezell
Converting CCM to Asterisk
http://qvlweb.blogspot.com/2009/04/asterisk-pbx-install-index.html
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Re: [asterisk-users] howto set up persistent dynamic meetme
Sean wrote: Tilghman Lesher wrote: On Saturday 16 May 2009 08:21:43 sean darcy wrote: With 1.6.1, I'm trying to set up a test of meetme for creating dynamic conferences. Trimmed I don't want the conference to stay up forever, since I'd like new pin's each time. This should be a common use case. How do you do it? In Asterisk 1.6, user DEA contributed realtime capabilities to MeetMe, which allows the capability of scheduling conferences, with new pins each time. I believe this would meet the needs your question has posed. I using 1.6.1. 'core show application meetme' doesn't have anything on realtime. I found http://www.voip-info.org/wiki/view/Asterisk+RealTime+MeetMe but that's just a stub. Any references available. I should point out that I (DEA) did not contribute the basic RealTime support in app_meetme. I added the scheduling and resource limit features, and the option to store the conference flags in the database table. You will want to understand the basics of Asterisk's RealTime features to get started. Basic support for RealTime conferences can be had by using the database table defined in contrib/scripts/meetme.sql The scheduling features require a more complex database table that I was sure that I included in the contribution, but I do not see it in SVN. The correct db table is available in the Web-MeetMe package, which is a front-end to manage the database and active conferences. It is hosted at https://sf.net/projects/web-meetme I am behind schedule to release a package for 1.6.1, and I need to submit the database table to Mantis so it can be added to future release packages. Dan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Step-by-Step Asterisk and Cisco 1760 Help
Jimmy wrote:
Second Call out the asterisk console looks like
this-:
-- Executing [92952...@internal:1] Dial(SIP/222-09ab3588,
SIP/Cisco1760/2952210) in new stack
-- Called Cisco1760/2952210
[Apr 22 16:08:58] NOTICE[3450]: chan_sip.c:14489 handle_request_invite: Call
from '222' to extension '2952210' rejected because extension not found.
-- Got SIP response 486 Busy here back from 172.17.2.1
-- SIP/Cisco1760-09ab7cf8 is busy
== Everyone is busy/congested at this time (1:1/0/0)
-- Executing [92952...@internal:2] Congestion(SIP/222-09ab3588, ) in
new stack
== Spawn extension (internal, 92952210, 2) exited non-zero on
'SIP/222-09ab3588'
localhost*CLI
--sip.conf -
[general]
bindaddr=0.0.0.0
[Cisco1760]
context=incoming_calls
type=friend
host=172.17.2.1
dtmfmode=rfc2833
disallow=all
allow=ulaw
insecure=very
--extensions.conf
[globals]
OUTBOUNDTRUNK=SIP/Cisco1760
[outbound-local]
exten = _9NXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
exten = _9NXX,n,Congestion()
exten = _9NXX,n,Hangup()
---Cisco 1760 config --
dial-peer voice 100 pots (This line that is set to preference 2 does not work)
huntstop
preference 2
destination-pattern .T
port 0/0
forward-digits all
!
dial-peer voice 2212 pots(This line that is set to Preference 1 is the one
that works)
huntstop
preference 1
destination-pattern .T
port 0/1
forward-digits all
You do not want to use huntstop on the dialpeers in this situation.
The huntstop option tells the call routing function in the router to
stop search for a call route if it encounters a failure.
Call number 2 hits dialpeer 1, finds it busy and the huntstop causes
the processing to stop.
Dan
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Re: [asterisk-users] Step-by-Step Asterisk and Cisco 1760 Help
Jimmy wrote: Dan thank you, yes that seems to help. It looks like the bridging is happening now and I see the light come on in the second FXO port, but then I get a busy signal after that and the call still does not complete. If I set the second line as priority 1 it completes the first call on that line and second call gets the busy on the first line. I even tried moving the lines to a different FXO card and the result is the same. Here is my current config for the cisco dial-peers: dial-peer voice 2212 pots preference 2 destination-pattern .T port 2/0 forward-digits all ! dial-peer voice 2211 pots preference 1 destination-pattern .T port 0/0 forward-digits all Thanks again Dan, I think I am much closer now. I think the suggestion by Jonathan will help you finish off your problem, but what you have listed should also have worked. What does your SIP dial-peer look like? After the second call fails, try issuing this command on the cisco: #show call history voice brief Then identify the call id of the failed call and use this: #show call history voice id call-id That will at least tell you why the call failed. I have not worked a lot with the Cisco analog interfaces, but I have setup a healthy number of ISDN ports, with the type of roll-over that you are trying to setup. I can try to help with the Cisco debug logs if you want to take this off list. Dan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can Asterisk bridge between a SIP client and a Cisco Call Manager server?
Shocky wrote: This is probably outside what Asterisk is intended for, but I'm hoping it can help. I need to make and receive calls through a Cisco Call Manager server that I have no control over. I have to use a Cisco soft phone (Cisco IP Communicator), which only runs on Windows. But I'm on Linux. CCM is apparently capable of supporting SIP and H.323 interfaces, but they won't provide this option for me. Right now I'm using a VMWare XP guest to run the soft phone, but this is painful (especially with some VPN complications thrown in). It maybe a small nuance, but as a CCM administrator I can understand the refusal to support a roaming H323 or SIP endpoint on CCM. Perhaps if your asterisk box was not mobile, the CCM admins would consider a H323 trunk to your system? I've read that Asterisk supports SCCP, at least somewhat. I'm wondering if I could set up Asterisk on my desktop machine to route calls between a SIP client such as Kphone or Ekiga and the CCM server. Would this be possible? The SCCP support in Asterisk is currently limited to asking as a SCCP server, not as an SCCP client. So you cannot use Asterisk to register as a phone to CCM. The SCCP protocol does have a 'trunking' mode, but Cisco barely uses it themselves, and it is geared to low density situation, two-four channels. I am not aware on any effort to duplicate that in chan_skinny. It is conceivable that chan_skinny could be taught to emulate a Cisco endpoint (7965 for example), but the end result would be of limited value. It would have a limited number of lines/channels and the protocol in this use model would not support passing destination information, so it would require a 1-to-1 mapping of a CCM extension to an Asterisk extension. I heard that one of the problems in interfacing with CCM over SCCP is the use of proprietary codecs. Would this be a problem in my case? Not quite true. SCCP is a proprietary protocol, but the codecs supported match well with what Asterisk offers, at least the codecs you would likely choose to use. If there's a chance it can be made to work, I'll give it a try. If I'd be wasting my time, please let me know. There is a chance, but it depends on working with the CCM admins and how willing they are to create a one-off configuration for you... Dan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there a public blacklist of hackers' IPaddresses?
Gordon wrote: There are other more advanced things you can do with iptables which I've been looking at - but the esence is to count/time new connections to a particular service from each IP address and if more connections per unit of time happen, then apply a temporary block for a bigger period of time. This works for ssh when you know there are only a small number of people who might connect in, but for SIP, you need to check the timings carefully, although one thing I've had issues with is Snom phones which seem to be overly enthusiastic when the end-user has the wrong password in them - they keep trying to register 2 or 3 times a second )-: I few years ago I noticed and quickly became annoyed by the volume of dictionary attacks on my home server. No one broke in, but the logs were becoming useless. Since installing it my logs are once again readable, and I have a nice long list of naughty addresses in my iptables DROP table. I found a package called sshdfilter that can add and remove iptables rules based on a number of conditions- 1. Invalid username - block immediately 2. Valid username w/invalid password - block after x attempts It supports white-listing so that a slip of the finger does not lock you out from a trusted host. The setup is fairly simple and system load is minimal. The package works by parsing syslog messages, and it appears that it could be extended to cover VoIP attacks, as long as the system is logging failed authentication attempts. Dan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is it possible to get full callin number fromE1?
Steve wrote: Speaking from T1 PRI and E1 PRI in West Africa, you tell the telco how many digits to send. Often times, at least in my experience, if not specified, they will only send the last four providing there are no conflicts. They should be able to send however many digits you require, but maybe they wont. I have found that each telco, and in fact each CO, may have a different practice for how many digits they send. If you have a good sales person when you place the order you can specify, or at least they will provide that information. If you have an uninformed sales person, the engineer who sets up you circuit can usually provide that. If you end up with uninformed sales and engineering personnel, then it is time to test and debug. After dealing with telcos in China, Korea, Taiwan, Japan, India, Israel, Germany, The Netherlands and the USA, I have found *1* telco that provided all of the circuit details in advance without asking. Most will tell you they are using the standard values, but have no idea what those values are. Here is my check-list I ask for on new orders- Linecoding (If E1 I also ask if G.703 or not) Framing Switchtype Digits sent/outpulsed Digits expected (City or local code required?) Some telcos have been willing to change the number of digits sent, but I usually just rewrite the received number to prepend the missing digits. It is a little more work for me, but it reduces the chance that future maintenance by the telco will revert the circuit back to their normal values and bust inbound calls. Dan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dailplan code for holiday detection?
This has been on my ToDo list far too long. I have a small call-center setup, with basic time of day/day of week validation before putting callers in the queues. With the holidays upon us, I need to add check to see if 'today' is a holiday so I do not put callers in unmanned queues. Due to how the agents work, I have to allow joinwhenempty. Does anyone have a snippet of dialplan code, perhaps using Astdb, to check it 'today' is a listed holiday? Thanks, Dan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dailplan code for holiday detection?
Tilghman wrote:
Astdb is a nice idea. Something along the lines of:
GotoIf(0${DB(holiday/${STRFTIME(,,%Y-%m-%d)})}?holiday,s,1)
would work. Holidays are evaluated as 01, which is true.
Anything not in the database would be evaluated as 0, which
is false. This will work both for holidays where the date
changes every year (e.g. Thanksgiving, Labor Day), as well as
holidays where it doesn't (e.g. Christmas, Independence Day).
On one hand I am embarrassed that it is that simple, on the other
I am thrilled that it is that simple.
After the Holidays I guess I need to put together a cheesy
web page to allow for adding the dates to Astdb, but for now
this is awesome and much appreciated.
Thanks,
Dan
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Re: [asterisk-users] Meetme realtime table structure
Sergey wrote: Sorry if I'll be very very stupid but really I write to this conference first. I have problems with configuration of app_meetme in realtime environment. I use last stable release of asterisk 1.6.0.3 trimmed db table definition The issue is not in the database, but a problem with how the options are processed. There is a patch for this, but it was not applied to 1.6.0 or 1.6.1, only trunk. The patch is in Mantis under bug id 150384 and is named- rt-meetme-flag-fixes-v2.txt Conference work fine but without possibility to manage OPTIONS. Neither adminOpts nor UserOpts does not work. All other fields such as PINs, conference nomber, startime etc works fine. I think that the problem is in the database table format. I try to look to the source in C but really not competitive in programming. I chahged field type to varchar(28) etc, I tried reccord values in 'value' and in value but there was not result. I did not also find asterisk debug which could detect database errors. No errors in logs file. But is I configure static meetme conference over /etc/asterisk/*.conf file I get good result. Could any one explain database table structure should be and help in this issue? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Limit the number of users in a meetme conference?
Noah wrote: I found the maxusers defined in meetme.c, but I'm not sure how this value is set. Does anybody know if one can limit the number of users permitted in a meetme conference? I know there's MeetmeCount(), but I'd rather avoid the dialplan logic and just set maxusers instead. That feature is primarily used with RealTime conferences. The maxusers value is read from a database and enforced on RealTime enable conferences. This presumes you are looking at 1.6.X or Trunk code... Dan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities
Yehavi wrote: Our university has to upgrade soon its old Nortel PBX's which holds around 10,000 extensions tied to 5 PBXes. Up to now we thought about commercial solutions but now there is a window openning for open source solution. However, I need examples to convince that this solution is feasible, and preferably at other universities. Are there any pointers for such installations? Sam Houston University migrated from a Cisco CallManager and Nortel setup to Asterisk a couple years back. I do not know any of the specific details, but maybe you can track down someone involved in the project. Dan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Meetme talker optimization always on even when no o option present.
Bill wrote: After loading 1.6.0.1, I notice that I always have the VOX effect on Meetme conferences whether I have the o option set in the dial plan or not. Is anyone else seeing this? Can you describe the effect? I am seeing odd behavior when I have PSTN calls in a conference, oddly most noticeable if the calling party is on a blackberry, but it also impacts other cell phones and land lines. Although I'm now running 1.6.0.1, I'm also seeing this on a system still running 1.6.0beta9. My calls route through a Cisco voice gateway and one Hint is that Asterisk tells me to turn off Comfort Noise for that peer (not possible as far as I can tell) All callers are G711, with very low latency and QOS between the gateway and Asterisk. Dan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] changing the size of voice packets
John Todd wrote: There was discussion recently (on -dev? on -users? on IRC?) about how there are some shortcomings on RTP packetization/transcoding. It appears, though I have not confirmed this, that trying to move a 20ms G.711 stream from a client, though Asterisk, to a remote gateway using 40ms G.711 will NOT work correctly. The 20ms packet size is passed through without aggregating to 40ms, or vice versa - no change in packetization (though I don't know which side takes precedence.) Going the opposite directon for dis-aggregation (which is what you want to do) I assume would fail in similar ways. I don't recall if changing the codec made any difference on the packetization between two bridged channels. In the past (trunk pre-1.4 and 1.4) both handled aggregation properly, with one important caveat: 1. The media actually flows through Asterisk (no RTP re-invites) If the media is re-invited, it is up to the clients/peers To honor the packetization the remote end requested. If the media is not reinvited and is 100% compatible, codec and packetization, it will go through the packet-to-packet bridge. At one point the P2P bridge did not know about packetization differences and would just relay the RTP packets. I believe that was fixed a long time ago. For what it's worth, 10ms is the maximum rate for most codecs. This creates twice as many packets as 20ms, three times as many as 30ms, etc. - hopefully your network hardware has sufficient power or your call volumes are reasonably low so as not to produce an overwhelming number of Packets Per Second (PPS). Decreasing sampling interval also gets you closer to reaching your NIC's threshhold of PPS, which often is not huge. I seem to recall asking the person who reported that to open a bug in Mantis, but I can't find it, though I didn't look exhaustively. If you can verify this and/or it's relevant to you, please open a ticket so that it at least will be reviewed. I'd open it myself, but I'm a bit resource constrained at the moment in an airport lobby. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Wierd queue question
I have just setup a small queue implementation for one of my branch offices, replacing a 16 year old key system that had a hacked together pseudo call queuing feature. The 'agents' are not dedicated to the queues and want to be able to logon and get one call only from the queue. I know this is odd, but it is how my users want it to work. I have the login process setup using dynamic agents and set a wrap-up time long enough for the agent to logout. They have accepted this as a short term solution, but they really want to be automatically logged out after taking one and only one call. Any tips or hints on how to accomplish this would be greatly appreciated. Thanks, Dan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wierd queue question
Julian wrote: show application RemoveQueueMember -= Info about application 'RemoveQueueMember' =- [Synopsis] Dynamically removes queue members [Description] RemoveQueueMember(queuename[|interface[|options]]): Dynamically removes interface to an existing queue If the interface is NOT in the queue and there exists an n+101 priority then it will then jump to this priority. Otherwise it will return an error The option string may contain zero or more of the following characters: 'j' -- jump to +101 priority when appropriate. This application sets the following channel variable upon completion: RQMSTATUS The status of the attempt to remove a queue member as a text string, one of REMOVED | NOTINQUEUE | NOSUCHQUEUE Example: RemoveQueueMember(techsupport|SIP/3000) Julian I should have mentioned that I already added a method for the agents to logout using RemoveQueueMember. What I am looking for is a way to trigger it automatically, after the agent logs in and gets one call. I admit I have not tried the simple and crude method: exten = 123,1,Answer exten = 123,n,AddQueueMember($member) exten = 123,n,Wait($sometime); long enough for a call to be delivered exten = 133,n,RemoveQueueMember($member) I was hoping that someone might have a more elegant solution. Dan Austin wrote: I have just setup a small queue implementation for one of my branch offices, replacing a 16 year old key system that had a hacked together pseudo call queuing feature. The 'agents' are not dedicated to the queues and want to be able to logon and get one call only from the queue. I know this is odd, but it is how my users want it to work. I have the login process setup using dynamic agents and set a wrap-up time long enough for the agent to logout. They have accepted this as a short term solution, but they really want to be automatically logged out after taking one and only one call. Any tips or hints on how to accomplish this would be greatly appreciated. Thanks, Dan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Announcing the release of Web-MeetMe 3.0.4
This release primarily focuses on security. A number of problems involving SQL injection and XSS were identified and reported by Jean-Michel Besnard. Jean-Michel was kind enough to help with the testing as each vulnerability was addressed. The new release is available in the downloads section of http://sourceforge.net/projects/web-meetme Thank you and enjoy. Dan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie in China: Red alaram in Zaptel for E1
John wrote: Thanks Steve for your suggestions. In China you will generally get either MFC/R2 or EuroISDN. MFC/R2 is much more common. This is exactly my current problem. NETCOM in Shanghai just told my local contact it is an E1 and that's it. I have no idea whether it is MFC/R2 or EuroISDN and so there is a lot of trial and error, not to mention about communicating with the telco. Is there anyway I could find out from zaptel what the line signal is? International installs are always fun. I have had some luck getting a local employee to relay my questions about provisioning, but all to often the response is 'We use the standard settings...'. At that point I resort to trial and error. I have setup a circuit in Shanghai, it is an E1, CRC4/HDB3 with the telco switch being/or compatible with ATT 5ESS. You should be able to get Netcom to tell you if the circuit is ISDN or not. Asking if it is a PRI will just confuse them, but they do understand the question 'ISDN or not ISDN' The only oddity with EuroISDN is that it often provided without CRC4. That doesn't make a lot of sense, but there it is. MFC/R2 seems to be universally provided without CRC4 in China. That's great info, Steve. Dan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk With Web meetme
Ali wrote: I got localhost*CLI cb mysql status No such command 'cb mysql' (type 'help' for help) That means that app_cbmysql is no loaded. The possible reasons: 1. The module did not compile 2. The module compiled, but did not get installed 3. The module is installed, but has a problem Set verbose to 5 and try *CLI load app_cbmysql.so The output will tell us if the module does not exist, or why it cannot be loaded. Dan Asterisk 1.4 and Meetme is the latest version 3.0, ztdummy is working fine. Thanks On Thu, Jun 26, 2008 at 6:48 PM, Dan Austin [EMAIL PROTECTED] wrote: Ali wrote: I followed this howto http://www.voip-info.org/wiki/view/MeetMe-Web-Control and http://asteriskpr.blogspot.com/2005/09/guide-to-install-web-meetm_112614171575673316.html to install web meetme with asterisk, I know the meetme module is included however I need to be able to ban and mute users as well. All of the installation went fine however when I do call a conference number I create using the interface all I get is service unavailable, I did run asterisk in verbose mode that did not make me any smarter. I added to extensions.conf the following [confserv] ;Make sure you change 1199 to your conference bridge extension(s) ;more information on this can be found at the asterisk web site. exten = 121212,1,Answer exten = 121212,n,Wait(3) exten = 121212,n,CBMysql() exten = 121212,n,Hangup Where 121212 is an existing extension, I really dont get it this all of the documentation available but I surely missed something here..any hints please ? Let's start with the easy stuff, if confserv included in the context that the phone has access to? What is the output of the command CLIcb mysql status? What version of Asterisk and Web-MeetMe are you using? Do you have a timing source (ztdummy or PSTN interface card)? Dan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- -- With Regards Ali Jawad System Administrator http://www.alijawad.org Phone : +961-01-559031 Mobile : +961-03-041705 Confidentiality Notice: The contents of this E-mail are intended for the named recipient only. It may contain confidential and privileged information. If you received it in error, please notify us immediately and then destroy it. Internet communications are not secure and therefore I do we do not accept legal responsibility for the contents of this message. Also, and though we provide every effort to keep our network free from viruses, you would need to check this E-mail and any attachments for viruses as we can take no responsibility for any computer virus which might be transferred by way of this E-mail. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk With Web meetme
Ali wrote: I followed this howto http://www.voip-info.org/wiki/view/MeetMe-Web-Control and http://asteriskpr.blogspot.com/2005/09/guide-to-install-web-meetm_112614171575673316.html to install web meetme with asterisk, I know the meetme module is included however I need to be able to ban and mute users as well. All of the installation went fine however when I do call a conference number I create using the interface all I get is service unavailable, I did run asterisk in verbose mode that did not make me any smarter. I added to extensions.conf the following [confserv] ;Make sure you change 1199 to your conference bridge extension(s) ;more information on this can be found at the asterisk web site. exten = 121212,1,Answer exten = 121212,n,Wait(3) exten = 121212,n,CBMysql() exten = 121212,n,Hangup Where 121212 is an existing extension, I really dont get it this all of the documentation available but I surely missed something here..any hints please ? Let's start with the easy stuff, if confserv included in the context that the phone has access to? What is the output of the command CLIcb mysql status? What version of Asterisk and Web-MeetMe are you using? Do you have a timing source (ztdummy or PSTN interface card)? Dan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CCM and multiple trunks
Aaron wrote: Okay, another Cisco related issue (sorry!). Single Asterisk box at location 1. Single Cisco box at location 2, however the Cisco is also the PBX for location 3 (same physical machine, calls routed via VoIP). Trying to have Asterisk be able to call EITHER Call Manager location. The single SIP trunk in CCM (version 6.1 mind you) only allows a single device pool to be selected. So configuring calls to one location...no problem. One location at a time that is. All you are missing is a translation of CCM terms into Asterisk concepts. A CCM Device Pool is similar to a context, a logical grouping Of devices. CCM extends the concept in that Device Pool membership controls/aids device provisioning. There is one major difference, and that is membership in a common Device Pool DOES NOT mean devices can call each other. Call 'routing' is handled by Calling Search Spaces. You want to have a CSS that included both locations Device Pools and assign that CSS to the trunk. The major caveat there is that the two locations should have non-overlapping dialplans. If they do overlap, you will need Translation Patterns, which can strip/prepend digits and be configured to search a different CSS If I want to be able to call the 3rd location I have to select a separate device pool and restart the trunk. Then calls to the 2nd location stop working. Multiple trunks in CCM doesn't appear to fix the issue, and creating multiple SIP profiles isn't permitted apparently. Any thoughts? Hope that helps... ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Want to know Frequency and lenght of Frame
Mojo wrote: [EMAIL PROTECTED] wrote: I am planning to write a module to find if a Special Information was detected or not. Can anyone please help me to figure out the below fields? 1. The Frequency of a frame 2. Length of frame in milliseconds Aren't all the frames in asterisk 20ms long, no exceptions? 1.0 1.2 Yes (with the possible exception fo iLBC) 1.4 1.6 No The default in 20ms, but can be changed per user/peer/codec ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Cisco Unity?
Tony wrote: Has anyone here any experience in getting an Asterisk box to talk to a Cisco Unity system? I have a potential customer who would like to add a conference bridge to their existing Cisco Unity setup. The digging I have done so far suggests that it should be possible to talk SIP between them, but I'd be interested in any stories of success or failure. As Peder mentioned, Unity is only a VM platform. I actually started using Asterisk to replace a Cisco Conferencing package that never worked right. We have had it running internally for three+ years now, and have been very happy with the results. I am currently using chan_ooh323, but SIP is possible if you have CCM 4.2 or higher. You'll also want to run a later release of Asterisk 1.4 which has a work-around for an odd CCM hold implementation. Dan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Meetme voice quality problems
Franklin wrote: ztdummy can give you issues as a timing device. Yes and no. See below Any way you could try using a Digium card just as a timing device to see if this helps? Tomasz wrote: I am using Debian OS kernel 2.6.22-3-amd64 and zaptel driver 1.4 with ztdummy module for meetme application. I use meetme with SIP channels. Your kernel is new enough that you should be able to leverage hi-res timers (you might need to patch ztdummy), or at least a RTC set to 8192 ticks/sec. What does dmesg show after ztdummy is loaded? I have such problem that when one connects to the conference voice is cut. Each voice sequence is disturbed. Do you have internal_timing=yes in asterisk.conf? This option allows Asterisk to time the RTP stream based on zaptel/ztdummy clock and not on the received RTP stream. In a MeetMe, where callers might mute themselves, the received RTP stream is all but useless for timing. Dan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Open Asterisk Exchange Project
Tzafrir wrote: On Sun, Dec 09, 2007 at 03:30:13PM -0500, Michelle Dupuis wrote: Well, we can already integrate to major platforms via SMTP. The real value is in deep integration to the most popular email platform in business: Exchange. I would love to see smart Exchange integration, where deleting the VM attached email will delete the corresponding message from asterisk. My clients would eat that up. IMAP support for voicemail? That would be an option if Microsoft had not implimeted such a limited set of IMAP features in Exchange. Exchange does not support the concept of master-user when using IMAP, so you'd need to have the password of every Exchange account you wanted to integrate with Asterisk voicemail. I suppose that the addition of IMAP support to Asterisk's voicemail application might open the way for someone to look at using a MAPI library, such as the one being developed by the OpenChange folks. Dan ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Client lost on skinny
Paul wrote: I have six cisco 7911g connected on asterisk over chan_skinny. Four of them are working OK. two of them even the screen on the phone is indicating that is registered and has number loose connection to asterisk . On asterisk the message is Skinny Client was lost, unregistering. also this phones does not appear anymore in the skinny show devices list . If I dial the tone does not stop asterisk and i get a message like Asked to transmit on a non existent session . Can somebody help me ? What version of Asterisk? Registration tracking and recovery was reworked around version 1.4.7 or 1.4.8 If you have a version newer than that, what value are you using in skinny.conf for the keepAlive setting? Dan ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Client lost on skinny
Paul wrote: Thank you for your answer. I am using asterisk 1.4.13 and keepalive has a value of 120 in skinny.conf. You can try reducing the keepAlive. The phone will still loose registration, but will re-register faster. Other than that, I would look at the health of your network, especially the ports for the phones that are dropping off. Dan ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CISCO 7921G with asterisk
Jordi wrote: Any one have experience with this CISCO Wireless IP phone running with Asterisk?? It doesn't support SIP protocol I believe, so I need to know if the skinny channel can work with the 7921. The 7921 works fine with SVN trunk, and I think the trivial changes required to support it also made it into 1.4 around release 1.4.7 Dan ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Chan_SCCP vs. Chan_Skinny
Lacy's response in the thread 'Why does everyone seem to dislike *now?', has a small bit that caught my eye. Chan_Skinny made a lot of progress between 1.2 and 1.4, and even more in the later 1.4.X releases. I am curious as to which features/functions that chan_skinny might be lacking compared to chan_sccp. We (the community) now have a small, but active, group of volunteers working on the chan_skinny code. I'm not interested in re-igniting the flame-wars of the past about these channel drivers, but I would like to know what else needs to be addressed in chan_skinny before it users of chan_sccp would consider using it. Thanks, Dan ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Show Callee name on Display
Shane wrote: I don't think that Asterisk currently sends a remote-party-id to the called party. That would proably have to be added to the sip channel. It *does* work with Broadworks, another SIP based phone system. On a phone registered to Broadworks: Your phone invites the Broadworks system, Broadworks replies with a 180 (ringing) which includes a remote-party-id: field populated with the destination you are calling. That is what displays on the Polycom and Sipura 841 that I have tried. I had eneabled remote-party-id on a Cisco 7960, but something in the dialog caused the call to die. I never investigated further. Like Lacy wrote earlier, there is a patch in Mantis: http://bugs.digium.com/view.php?id=8824 That provides this feature. The author has a include support in a number of channels. I can confirm that it works with chan_skinny. My tests with chan_sip have not meet with success, but that is likely due to old Cisco firmware, or a configuration issue on my part. Dan ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Show Callee name on Display
Replying to myself: Shane wrote: I don't think that Asterisk currently sends a remote-party-id to the called party. That would proably have to be added to the sip channel. It *does* work with Broadworks, another SIP based phone system. On a phone registered to Broadworks: Your phone invites the Broadworks system, Broadworks replies with a 180 (ringing) which includes a remote-party-id: field populated with the destination you are calling. That is what displays on the Polycom and Sipura 841 that I have tried. I had eneabled remote-party-id on a Cisco 7960, but something in the dialog caused the call to die. I never investigated further. Like Lacy wrote earlier, there is a patch in Mantis: http://bugs.digium.com/view.php?id=8824 That provides this feature. The author has a include support in a number of channels. I can confirm that it works with chan_skinny. My tests with chan_sip have not meet with success, but that is likely due to old Cisco firmware, or a configuration issue on my part. After taking the time to upgrade my SIP firmware to a less ancient version, I now have chan_sip working with remote-party-id as provided by the patch mentioned. Testing and feedback will help move it along... Dan ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7960 Won'
Shawn wrote: I'm having a wierd problem with a Cisco 7960 (sccp2) and asterisk (1.4.2) If the call that I'm trying to make goes through, everything works fine. But if there's any sort of error (like me messing around in my extensions.conf, etc). I can't get the connection to drop. ie: If I get the conjestion tone and hang up the phone, I can do a sccp show channels I can see that the channel is still in use (even after several minutes). If I pick up the phone to attempt to make another call, I get an error that it can't put the current call on hold to start the new call. What am I missing? An upgrade. The sccp channel in early 1.4 had quite a number of problems, and it was completely broken in 1.4.3 to 1.4.6 Any version after 1.4.7 should work better, with the latest being the best choice. Dan ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7960 Won'
Jason wrote: Dan Austin wrote: Shawn wrote: I'm having a wierd problem with a Cisco 7960 (sccp2) and asterisk (1.4.2) If the call that I'm trying to make goes through, everything works fine. But if there's any sort of error (like me messing around in my extensions.conf, etc). I can't get the connection to drop. ie: If I get the conjestion tone and hang up the phone, I can do a sccp show channels I can see that the channel is still in use (even after several minutes). If I pick up the phone to attempt to make another call, I get an error that it can't put the current call on hold to start the new call. What am I missing? An upgrade. The sccp channel in early 1.4 had quite a number of problems, and it was completely broken in 1.4.3 to 1.4.6 Any version after 1.4.7 should work better, with the latest being the best choice. Dan Well, he's also using chan_sccp, so no amount of upgrading is going to help with that. In my opinion (and I think Dan and several others would agree), chan_skinny is far more stable (and active...) than chan_sccp. Bugger! I should have noted the 'sccp show channels' command. I tend to swap skinny/SCCP automatically, since Cisco uses both in the documentation, and had it in my head that he meant skinny Yes, chan_skinny in 1.4.7+ has had major love applied. I only have a couple test phones hooked up for development, so my impression of stability is not worth much, but I think we have managed to fix up the most hideous bugs. If we can keep up the pace, chan_skinny in 1.6 is going to rock. Sorry for the confusion. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Change Packetization Time
Dovid wrote: Does anyone know if it is possible to change the packetization time in Asterisk ? I was told by a client of mine that adjusting this with using G729 can greatly lower the amount of bandwidth used. Your client is correct. Configurable packetization was added introduced with the release of 1.4.0. For details look at the rtp-packetization.txt file in the doc directory for full details. The short answer is to append :size to any codec on your allow directive that you want to change from the default of 20ms. Ex. Allow=g729:40 Dan ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dropouts and echo
Tom Wrote: Hi all, We have recently implemented an Asterisk system using Trixbox (asterisk v1.4.4 at the moment, yet to move to 1.4.9) but are getting pressure to switch back to our old key system unless we fix two major issues. So please help me avoid switching back! Have you tried changing the RTP packet size on the phones from .30(default I believe) to .20? That may help the cut-out issue. I wouldn't bet on it helping with the echo issue, which I would approach by tweaking the phone volume levels and seeing if environmental issues my play a part. Dan ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Issue in insatlling addons-1.4.2
Something changed in the final linking of the channel, and it now produces libchan_h323.1.0.1 instead of libchan_h323.so.1.0.1 Either edit the Makefile to copy libchan_h323.1.0.1, or manually copy that file... Dan From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Keshav K. Sent: Wednesday, July 18, 2007 8:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Issue in insatlling addons-1.4.2 There is no libchan_h323.so.1.0.1 file in libs... See here all the files of .libs asterisk-ooh323c]# ls -l .libs/ total 3732 lrwxrwxrwx 1 root root 18 Jul 18 19:21 libchan_h323 - libchan_h323.1.0.1 lrwxrwxrwx 1 root root 18 Jul 18 19:21 libchan_h323.1 - libchan_h323.1.0.1 -rwxr-xr-x 1 root root 1820126 Jul 18 19:21 libchan_h323.1.0.1 -rw-r--r-- 1 root root 1985154 Jul 18 19:21 libchan_h323.a lrwxrwxrwx 1 root root 18 Jul 18 19:21 libchan_h323.la - ../libchan_h323.la -rw-r--r-- 1 root root 810 Jul 18 19:21 libchan_h323.lai ---Keshav Dovid B [EMAIL PROTECTED] wrote: This is a bug. Search for the file and move it over manually. - Original Message - From: Keshav K. mailto:[EMAIL PROTECTED] To: Asterisk-users Digium mailto:asterisk-users@lists.digium.com Sent: Wednesday, July 18, 2007 5:09 PM Subject: [asterisk-users] Issue in insatlling addons-1.4.2 Hi, I'm using Asterisk-1.4.7.1. Everything was working fine. Now I'm trying to Install Asterisk-addons-1.4.2. The procedure I followed is as... # cd asterisk-addons-1.4.2 #./configure #make menuselect #make #make install Everything is going fine except make install. I've tried many times, but the same error I'm gettiing--- The error is--- asterisk-addons-1.4.2]# make install make[1]: Entering directory `/usr/src/asterisk/asterisk-addons-1.4.2' gcc -g -c -fPIC -fPIC -o app_saycountpl.o app_saycountpl.c gcc -shared -Xlinker -x -o app_saycountpl.so app_saycountpl.o make[2]: Entering directory `/usr/src/asterisk/asterisk-addons-1.4.2/asterisk-ooh323c' make all-am make[3]: Entering directory `/usr/src/asterisk/asterisk-addons-1.4.2/asterisk-ooh323c' make[3]: Nothing to be done for `all-am'. make[3]: Leaving directory `/usr/src/asterisk/asterisk-addons-1.4.2/asterisk-ooh323c' make[2]: Leaving directory `/usr/src/asterisk/asterisk-addons-1.4.2/asterisk-ooh323c' make[2]: Entering directory `/usr/src/asterisk/asterisk-addons-1.4.2/format_mp3' make[2]: Nothing to be done for `all'. make[2]: Leaving directory `/usr/src/asterisk/asterisk-addons-1.4.2/format_mp3' make[1]: Leaving directory `/usr/src/asterisk/asterisk-addons-1.4.2' for x in app_saycountpl.so; do /usr/bin/install -c -m 755 $x /usr/lib/asterisk/modules ; done make[1]: Entering directory `/usr/src/asterisk/asterisk-addons-1.4.2/format_mp3' /usr/bin/install -c -m 755 format_mp3.so /usr/lib/asterisk/modules make[1]: Leaving directory `/usr/src/asterisk/asterisk-addons-1.4.2/format_mp3' make[1]: Entering directory `/usr/src/asterisk/asterisk-addons-1.4.2/asterisk-ooh323c' cp .libs/libchan_h323.so.1.0.1 /usr/lib/asterisk/modules/chan_ooh323.so cp: cannot stat `.libs/libchan_h323.so.1.0.1': No such file or directory make[1]: *** [install] Error 1 make[1]: Leaving directory `/usr/src/asterisk/asterisk-addons-1.4.2/asterisk-ooh323c' make: *** [install] Error 2 Have anyone any Idea how to solve this issue.. Please suggest me how to solve this problem, or this is a bug?? Regards, Keshav Park yourself in front of a world of choices in alternative vehicles. Visit the Yahoo! Auto Green Center. http://us.rd.yahoo.com/evt=48246/*http:/autos.yahoo.com/green_center/;_ ylc=X3oDMTE5cDF2bXZzBF9TAzk3MTA3MDc2BHNlYwNtYWlsdGFncwRzbGsDZ3JlZW4tY2Vu dGVy ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Regards, Kesh Lets change the future...lets change the world. Take the Internet to Go: Yahoo!Go puts the Internet in your pocket:
Re: [asterisk-users] North American voice BRI - Informal survey
David Wrote: On Thu, 2007-07-05 at 16:42 -0600, Stephen Bosch wrote: David Boyd wrote: I seem to remember that the wan Pipeline units supported BRI, and also provided a couple of analog phone jacks. I will dig around in the basement and try to find the one that I had, if I find it, who wants it for play? Well, whoever ends up with the simulator should get it. I'm not familiar with the Pipeline stuff. Got a link you can share? -Stephen- No link, it was something I used 8+ years ago, so I am surprised i pulled it out of my memory :) I will dig around this weekend and see if I can find it. Pretty easy to setup, used it for an ISP connection for centrex purposes. Hopefully I am not mis-remembering it capabilities. Ascend Pipeline 50/75 units were great remote access devices long before ADSL killed em off. Yes they could handle voice, with one or two FXS ports, and one to three BRI ports. I think I only recently threw away the units I had in the closet and maybe even the ones at work. Setup was not hard, at least the ISDN bits. We still use BRIs for our VC systems, so they can be ordered at least for businesses. I've also used BRI lines to setup small offices on out CCM installation (mostly outside of the U.S.) The hardest part of setting one up is getting the carrier to provide the provisioning details (switchtype, SPID or no SPID, PTP or MP). If you can get those details, it is no harder to setup than a PRI. Dan ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7941 localized menus with SIP firmware
Greg wrote: So, if you ever use a Cisco SIP Phone with an Asterisk server, it's not possible to localize menus, soft keys, and so on ? Not unless someone wants to add support for it in the SIP channel, which I doubt. I would be more than willing to provide the SIP messages that a CallManager sends to accomplish it though. Localization with CCM happens when the phone boots. The initial TFTP config download (xml or older .cfg) includes a setting to identify the local, which then TFTP downloaded. Setting up the initial config file (xml or older .cfg) is not difficult, but without copies of the localization files on your TFTP server, it will not help much. With the localization files, the channel driver can send the button templates and the phone will display the localized version of the button(s). Dan ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Realtime Meetme in 1.4
1.4 does have support for MeetMe RealTime, and the docs are a tad lacking. I have a patch up on Mantis that extends/cleans up the RT features in app_meetme I made the column names configurable, with optionally enabled scheduling features (start time, end time, maximum participant count per conf). The scheduler features/code grew out of my work on Web-MeetMe, and the code to enforce end times will likely need to be redone. I still need to work on the code that sets options for the moderator (if used) and normal callers. So it is still a work in progress. Dan From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tony Plack Sent: Tuesday, June 12, 2007 2:15 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Realtime Meetme in 1.4 Looking at the archives, it looks like MeetMe Realtime is in 1.4 but alas the documentation is lacking. Is it as simple as add the SQL table and placing a meetme family in the extconfig.conf? It also looks like Dan Austin at phoenix dot com was working on a scheduler for this. Any news on that? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] SIP Echo
Alex wrote: I tried with the ping ... all of the phones respond in ca. 0.3ms, so network seems to be OK. More than 90% of CPU on * box is idle even in peak times, so this shouldn't cause echoes either, right? Hmmm, so handset could be an issue, but did anyone ever experience any handset problems with Polycom IP SoundPoint 430 :-) ? I will check the headsets and any possibilities of acoustical echo. Besides that, if we rule out the network, and the CPU on the * box, is there anything else that could be causing echoes on internal SIP calls? As others have pointed out it is highly unlikely that a network issue is the source of the problem (unless the phone's firmware has a MAJOR bug). Acoustic issue is 99.9% likely to be the cause, but is can be less than obvious why. A certain vintage of Cisco phone firmware would introduce echo when the headset/handset/speaker volume was set above 65~75%. I spent about 6 months chasing that one on and off. After Cisco fixed that, then next two common causes were: 1. Enclosed offices/conference rooms without acoustic treatment 2. 3rd party amplified headsets (Echo was only one symptom of this one and not a common one, but it did happen) Some phones deal with item 1 better than others. last ditch efforts to fixup a room that a phone has problems with would include wall hangings, or even a cloth place mat (don't use the wife's Holiday mats) under the phone if the echo is most common on speakerphone calls. I've often wondered why phone designers put the mic on the bottom front of so many phones, where it is most likely to get acoustic reflection off the table/desk surface... Oh, one more cause that is a bear to correct. After first switching to the new system, my users felt the need to yell at their phones. Maybe a byproduct of poor experience with cell phones, which is how they expected the new phones to work like. Getting the yellers and loud talkers to bring it down a notch also helped. Dan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] [Announce] Web-MeetMe V3.0.1 released
Ondrej wrote: I finally got some time to test the SVN branches and here are my comments: Cool. One thing that does not work for sure - I had some problems to terminate the running conference from within the web page - I just clicked the button and nothing happened. This is likely a manager.conf security issue, but it could be a problem in the php code. I just tested branches/3.0 and trunk against 1.4.1 and it worked as expected. If you set core verbose to 10 and click on 'End Now' the console should display a message like this: app_meetme.c:941 meetme_cmd: Cmdline: 41251|k|1 At this point this would be a topic better suited for the support forums on SF. I browsed the php sources trying to understand and from what I see, the End Now button does nothing else than kicking all attendees - not exactly what I would expect. I would expect this action to terminate the conference immediately so, it could be seen in the Past conferences list. Also, some javascript popup dialog confirming this action would be nice. The same is valid for the Extend button - it works, but from the user prospective, nothing happens - I would expect some dialog box like The conference # has been extended by 10 minutes. This is the only missing piece, I would say - thanks :-) Oh! Those are great ideas and fairly easy to add. I'm about to be offline for two weeks, and need to get updated releases of 2.X and 3.X out before I go. Your ideas are now on the ToDo list and I'll try to get them integrated and released by mid-June. Dan ___ --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] Called party identification - where to take calledname?
Yehavi wrote: I am trying to apply the called party identification patch (patch 8824) and managed to make it work with a static data. Where do I take the name of the called person (the equivalent of CALLERID, but the other way...)? Short answer is that you cannot. Longer answer is that it is possible, but requires new functionality to be added to the core and a new API call be added that can check if the called party is a local endpoint and retrieve the caller-id values. At least that was what I found when working on the patch. If anyone knows a way to lookup a peer/friend from the dialplan and collect such details, it would be possible to use the existing patch without any more changes in the core. BTW, one note to the above patch: To make it work the device should have the parameter sendrpid set to true. Dan ___ --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] Semi-OT: useful things to do with XML browsers inphones
Chris wrote: It seems that more and more phones these days are coming with XML mini-browsers. I'd like to have a go at developing something useful to use on them, but in all honesty, most of our customers use their phones to make and take calls and very little else. So I'm open to suggestions. What useful applications are you developing for these mini-browsers? What sort of things do your customers want to use on them? I've been planning to write to app for joining scheduled conferences. It would be bundled with the Web-MeetMe suite. Users of the app would see a list of conferences scheduled for the current time and have one-button access to the conference (assuming no PINs) ___ --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] [Announce] Web-MeetMe 3.0.2 and 2.2.2 Released
Basic bug fix releases Both have updates to app_cbmysql to be thread-safe, reconnect to the database in case of timeout and to detect missing/mis-configured conference app/conference participant counting apps. The last one has caused Asterisk to crash. Now If it does not find MeetMe or MeetMeCount (the defaults) it posts a warning and exits back to the dialplan. Web-MeetMe 2.2.2 also has an a couple of small PHP Updates (2.2.1 shipped with a copy of one PHP file from the 3.X tree that broke the conference monitoring page) The new releases can be found at: http://sourceforge.net/projects/web-meetme/ Thanks go out to the users and testers who found these issues and who kept after me until I found and fixed them. Special thanks to hadefix on SF for identifying the threading issue and providing hints about the fix. Thanks, The Web-MeetMe development team ___ --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] Display Caller ID of called party
Alex wrote: It seems to me that what you are really talking about is manipulating the display features of the phone. Caller ID is unlikely to have this effect as the phone does not consider the From: URI in the SIP header unless the call is of an incoming nature. This feature is often referred to as 'Called Party Identification'. There is a patch on the bug tracker that implements it for chan_sip and chan_skinny. The solution to this is bound to be proprietary to the phone in some way or another--if there is one. I just wanted to point out that the mechanism for its delivery would almost certainly not be caller ID. Semi-proprietary. A good number of SIP endpoints support it. SCCP endpoints support it, and in theory some H323 endpoints support it. Of course, you COULD always set your dial plan in such a way that it never actually completes the outbound call leg, but instead hangs up, and then dials it, and rings you back (with the caller ID of the intended incoming leg). The current patch in the bug tracker does require dialplan edits. Deeper changes would be required in Asterisk to allow it to lookup a called party to see if it was a local extension and use its caller-id for this. Dan ___ --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] ADSL routers with integrated SIP QoS for otherdevices
Andrew wrote: On Saturday 28 April 2007 11:22 am, Chris Bagnall wrote: Thanks to all who replied to my thread a few days ago SIP devices with packet loss tolerance. One of the suggestions that came out of that thread was to replace routers at users' premises with ones that support QoS. Sangoma S518 (internal PCI) on a Linux box with iproute2/iptables/tc or BSD with pf. These are the best solutions, IMO. I was just about to reply with the same recommendation. A SFF chassis with 2 PCI slots could host one S518 and a PSTN interface. These units typically have built-in ethernet and some have built-in wireless. I still have my fingers crossed that Sangoma will offer an ADSL daughercard for the A200. That would make for a perfect combination in a SFF chassis... The latest Linux kernels also have SIP connection tracking/matching, so it should be possible to mark packets and prioritize based on iptables matching. I have not done this just yet, as the latest 2.6.20/2.6.21 kernels do not play nice with the wanrouter drivers. (note: there was a recent patch to 2.6.20.4 which apparently has much better SIP matching, and has been tested successfully with Asterisk. I have not tested it yet, and the iptables guys have rejected the patch as their direction for packet matching is shifting significantly in the near future. It can be found at http://thread.gmane.org/gmane.comp.security.firewalls.netfilter.devel/18 860.) I'm still looking for a miniPCI ADSL chipset that Linux can use, or an actual raw ADSL non-PCI chipset that I can design into an embedded system. If anyone has any leads, please don't hesitate to contact me! Good luck and let us know if you find one. The manufacturers of the XDSL chipsets seem to be even worse than the video card companies when it comes to OSS. There's a project on SF called OpenADSL that was working to make common XDSL chipsets work under Linux. The project appears almost dead with a developer post every 6~8 weeks, but that might be a good place to start Looking. If you're curious, I have my rc.tc script for Linux up on http://mixdown.ca/~andrew/rc.tc. It's loosely based off of wondershaper, but works much better, IMO. It does host-based prioritization for VOIP, puts mail just underneath bulk traffic, and P2P beyond that (if you have the p2p connmark stuff set). I can completely saturate DSL links with the S518 with this config without appreciable VOIP degradation. I'm using something similar. The missus can talk to her mother (in rural Japan) over IAX while I am using a IPSEC tunnel to work, and doing heavy downloads. Even without an S518, this script works well with external ADSL/cable modems. You may have to play with the upload rate; some cheap ADSL modems will start blocking your upstream traffic beyond as little as 50% of the upstream rate. ___ --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 1.4.3 segfaults on receiving calls.
The latest zaptel release has a bug that can cause segfaults. Did you upgrade zaptel at the same time? Dan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Thomas Kenyon Sent: Wednesday, April 25, 2007 9:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk 1.4.3 segfaults on receiving calls. On upgrading 2 machines (1 with a very simple configuration) from asterisk 1.4.2 to 1.4.3, I have found that on receiving a call (on either an IAX2 or SIP channel) the server process segfaults. Is anyone else having this trouble? ___ --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: [asterisk-announce] Asterisk-addons 1.4.1Released
Bill Wrote: On Wed, 25 Apr 2007 12:18:10 -0500, The Asterisk Development Team wrote The Asterisk.org development team has released Asterisk-addons version 1.4.1. When I run make install I get: [EMAIL PROTECTED] asterisk-ooh323c]# make install cp .libs/libchan_h323.so.1.0.1 /usr/lib/asterisk/modules/chan_ooh323.so cp: cannot stat `.libs/libchan_h323.so.1.0.1': No such file or directory make: *** [install] Error 1 [EMAIL PROTECTED] asterisk-ooh323c]# ls -la .libs/ total 4164 drwxr-xr-x 2 root root4096 Apr 26 10:26 . drwxr-xr-x 8 root root4096 Apr 26 10:26 .. lrwxrwxrwx 1 root root 18 Apr 26 10:26 libchan_h323 - libchan_h323.1.0.1 lrwxrwxrwx 1 root root 18 Apr 26 10:26 libchan_h323.1 - libchan_h323.1.0.1 -rwxr-xr-x 1 root root 2043862 Apr 26 10:26 libchan_h323.1.0.1 -rw-r--r-- 1 root root 2197238 Apr 26 10:26 libchan_h323.a lrwxrwxrwx 1 root root 18 Apr 26 10:26 libchan_h323.la - ../libchan_h323.la -rw-r--r-- 1 root root 807 Apr 26 10:26 libchan_h323.lai Have I done something wrong? Or is there a bug? Isn't it fun when old bugs come back? This is a Makefile issue where the compiled file has the wrong name. The 'make install' is looking for .libs/libchan_h323.so.1.0.1, but the compile produced .libs/libchan_h323.1.0.1 You can copy the file manually and it will work fine: cp .libs/libchan_h323.1.0.1 /usr/lib/asterisk/modules/chan_ooh323.so With luck Asterisk-Addons 1.4.2 will address this once again Dan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] SIP kpml DTMF support in *
Grigoriy wrote: I'm trying to connect Asterisk 1.4 and Cisco CallManager 5 using SIP Trunk without MTP (media termination point). Howerver, Cisco 79xx phones do not support RFC2833, they always notify CCM5 via SKINNY channel no matter where they send RTP to. If you are running the phone loads that shipped with CCM5, then your skinny phones have 'support' for RFC2833. CCM figures out during the call if the call will traverse a SIP trunk and instruct the phone to use RFC2833 for DTMF I have a CCM5-Asterisk trunk setup for MeetMe conferencing with NO MTP and DTMF works fine. For non-MTP trunk there's Out-of-band DTMF support in CCM5 called kpml. I wonder if Asterisk can support it. Interesting, will look it up... I found an intertnet-draft for kpml: http://tools.ietf.org/id/draft-ietf-sipping-kpml-07.txt, but it seems to be very old - Expires June 25, 2005. I know that using MTP in SIP Trunk at CCM5 makes DTMF work in RFC2833, but MTP resource is very limited and I don't want to proxy RTP via CCM5. I don't blame you, nut again as of CCM5 you are no longer required to use an MTP for SIP trunks. Please, do not offer to use H.323. OK, not an offer, but I have found that even as of the latest CCM5 release, the SIP stack is 'quirky'. I also maintain a H323 trunk between the same CCM cluster and Asterisk and in general it is much better behaved (using chan_ooh323). Either will work Dan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] SIP kpml DTMF support in *
Grigoriy wrote: Dan Austin wrote: If you are running the phone loads that shipped with CCM5, then your skinny phones have 'support' for RFC2833. CCM figures out during the call if the call will traverse a SIP trunk and instruct the phone to use RFC2833 for DTMF I have a CCM5-Asterisk trunk setup for MeetMe conferencing with NO MTP and DTMF works fine. Can you specify the version of the loads? Not specifically. I am already up on CCM 5.1 which ships with 8.0(4) for 7940/7960 phones. I seem to recall that CCM 5.0 had 8.0(1), but could be wrong. In any case I was using SIP trunks without MTP and with G729 25 days after 5.0 was released (I managed to get Cisco to actually release it to me when they announced it as available instead of the more normal 90 days after announcement) Dan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] [Announce] Web-MeetMe V3.0.1 released
Ondrej wrote: What version of Asterisk are you using? I've had recording working with SVN before 1.4, the 1.4 betas and currently 1.4.1. *** Update *** Recordings are tied to a moderator joining the conference at this time. I may need to change that based on feedback/requests to do so. *** Update *** Please include a note in the documentation for that (and maybe even note that in the web page for configuring conferences) !! It is really needed. Also please update the web page of each (past) conference with the link from where the recording could be downloaded The links to download a recording are already on the past conference page IF the conference was recorded. I will try to make time to update the README and installation How-To on SF. I also plan to add mouse-over help text to the UI, but I do not know when I will get to it (real work takes priority) I've never user the sql option for the user/participant. It was contributed by another user of the suite. Depending on the technology the caller used to call into the conference you should have their Caller-id number and possibly their Caller-id name. What additional Information would you like to see? Also - this is probably again a problem of the missing documentation, but let me clarify my problem in detail: If I create a conference, there is a button email participants. If I click that button, nothing happens (). How does the whole email procedure works? How does the web-meetme gather the email addresses of the participants? There is no way how to configure participants to the conference. My understanding was, that participants are informed about conference start/end/extended by this procedure. But since there is no way how the application could find their email addresses, I just do not know how it should work. OK, I get it now. This is a side effect of offering too much flexibility. I use and prefer the client-side mailer, and my users simply get an new message draft in their email client that they can add the participants to. If you use the server-side mailer, then there is currently no way to add participants to the notice other than to email the details to yourself and forward them. I'd happily integrate an AD address book function, but it is Not a feature I or my users would use, so I cannot dedicate too much time to writing it myself. From the sources I see that it uses SQL database users - but since I use AD, my users database is empty Contributions welcome. There is a new How-To up on SF that covers the installation on a step by step basis. I've tried to comment the configuration files to make it clear how each setting works. Some features have been contributed to the project, and I am sorry to say that beyond making sure they integrate cleanly, I have not taken enough time to document their setup and use. I guess I should ask for supporting documentation before merging the changes/features. I agree, because without any documentation is the feature de-facto unusable. I am happy to contribute to the project but at this stage it is (due to the bugs mentioned above) for me unfortunately still quite far from being promising. Lot of work has been done, but here are still some important pieces to be done. I'm sorry to hear that. I know it has some rough edges, but many people are using it. Some feature combinations work better/are better documented than others. If you are interested in following the development progress, I recommend monitoring the forums for the project on SF. I also hope I am not sounding if I do not care about the changes Or suggestions you are making. I agree will most if not all of them, but I need to focus on the problems that impact my users first and if anytime is left I can work on features that will not be used by them, but that others will enjoy. Thank you for the feedback. I am surprised almost daily how many people have found it useful. I did not really expect it to be as popular as it has become, and I am more than happy to try and address any problems. Glad to hear that :-) Ondrej Dan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] [Announce] Web-MeetMe V3.0.1 released
Ondrej wrote: Ok, I understand that now as well - you click that button and thunderbird should popup with the mail composer open, right? Yes. Does not happen to me - most likely problem w/ my firefox settings. Browser security settings most likely Now it all make a sense, sorry for being too pessimistic! No worries. One thing that does not work for sure - I had some problems to terminate the running conference from within the web page - I just clicked the button and nothing happened. This is likely a manager.conf security issue, but it could be a problem in the php code. I just tested branches/3.0 and trunk against 1.4.1 and it worked as expected. If you set core verbose to 10 and click on 'End Now' the console should display a message like this: app_meetme.c:941 meetme_cmd: Cmdline: 41251|k|1 At this point this would be a topic better suited for the support forums on SF. Anyway - thanks a lot for the explanation - I will give it a try! I just committed a simple set of mouse-over text popups to provide details about the options/settings in 'Add Conference' that might not be obvious to everyone. Since I know what the fields are for, I may have over/under thought which ones need more explanation, and the text I used to explain the fields may be poor. If you'd care to check svn branches/3.0, I'd love to know what needs more work. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] [Announce] Web-MeetMe V3.0.1 released
Ondrej wrote: Ok I had a chance to test web-meetme 3.0.1 and I have few comments here - the Makefile for CBmysql lacks procedure that verifies existence of /var/lib/asterisk/sounds/conf-recordings directory where the conference records should reside. You are right that this should be documented at least, and part of the make install process ideally. I had to go through .php files to find out where they are supposed to be and create the directory manually. Strange enough, the recording still does not work and the main web interface lack any support for the record files (I would expect some link in the past conference list). There will be a link if the conference is recorded. I received a report of the recording option not working just this weekend and I started Looking for the cause today. I was out of town for a week, otherwise I would have gotten a chance to respond earlier. What version of Asterisk are you using? I've had recording working with SVN before 1.4, the 1.4 betas and currently 1.4.1. - Active Directory integration works fine, but we should be able to gather email addreess for the participant from AD, too (avoid using the sql users table if web-meetme was configured to use AD). Actually this is still a big mystery to me - how do I add participants to the conference using the web-interface? It must be done via the web interface as otherwise we have no information about the participant except of his channel number. I've never user the sql option for the user/participant. It was contributed by another user of the suite. Depending on the technology the caller used to call into the conference you should have their Caller-id number and possibly their Caller-id name. What additional Information would you like to see? It is very promising project but it needs - a better documentation Contributions welcome. There is a new How-To up on SF that covers the installation on a step by step basis. I've tried to comment the configuration files to make it clear how each setting works. Some features have been contributed to the project, and I am sorry to say that beyond making sure they integrate cleanly, I have not taken enough time to document their setup and use. I guess I should ask for supporting documentation before merging the changes/features. - fix the conference recording backend I hope to have this resolved this week. - clear the confusion with users/email addresses/mail notifications. More details about what you would like the system to do please... If all that works, it would be just perfect... Thanks, Ondrej Thank you for the feedback. I am surprised almost daily how many people have found it useful. I did not really expect it to be as popular as it has become, and I am more than happy to try and address any problems. Thanks, Dan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] [Announce] Web-MeetMe V3.0.1 released
Ondrej wrote: Ok I had a chance to test web-meetme 3.0.1 and I have few comments here - the Makefile for CBmysql lacks procedure that verifies existence of /var/lib/asterisk/sounds/conf-recordings directory where the conference records should reside. You are right that this should be documented at least, and part of The make install process ideally. I had to go through .php files to find out where they are supposed to be and create the directory manually. Strange enough, the recording still does not work and the main web interface lack any support for the record files (I would expect some link in the past conference list). There will be a link if the conference is recorded. I received a report of the recording option not working just this weekend and I started Looking for the cause today. I was out of town for a week, otherwise I would have gotten a chance to respond earlier. What version of Asterisk are you using? I've had recording working with SVN before 1.4, the 1.4 betas and currently 1.4.1. *** Update *** Recordings are tied to a moderator joining the conference at this time. I may need to change that based on feedback/requests to do so. *** Update *** - Active Directory integration works fine, but we should be able to gather email addreess for the participant from AD, too (avoid using the sql users table if web-meetme was configured to use AD). Actually this is still a big mystery to me - how do I add participants to the conference using the web-interface? It must be done via the web interface as otherwise we have no information about the participant except of his channel number. I've never user the sql option for the user/participant. It was contributed by another user of the suite. Depending on the technology the caller used to call into the conference you should have their Caller-id number and possibly their Caller-id name. What additional Information would you like to see? It is very promising project but it needs - a better documentation Contributions welcome. There is a new How-To up on SF that covers the installation on a step by step basis. I've tried to comment the configuration files to make it clear how each setting works. Some features have been contributed to the project, and I am sorry to say that beyond making sure they integrate cleanly, I have not taken enough time to document their setup and use. I guess I should ask for supporting documentation before merging the changes/features. - fix the conference recording backend I hope to have this resolved this week. *** See update above *** - clear the confusion with users/email addresses/mail notifications. More details about what you would like the system to do please... If all that works, it would be just perfect... Thanks, Ondrej Thank you for the feedback. I am surprised almost daily how many people have found it useful. I did not really expect it to be as popular as it has become, and I am more than happy to try and address any problems. Thanks, Dan ___ --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] ztdummy and MOH
Wooi wrote: I have the similar problem on 1.4.1. I don't remember having it in 1.4.0, I could be wrong. I have a SIP provider, when calls come in, it play MOH while waiting for to be picked up. ztdummy is loaded. Another interesting thing I notice, exten = s,1,Zapateller(answer|nocallerid) exten = s,n,Background(PleaseWait) exten = s,n,Dial(100,30,r) Please note, if I use r (ring) instead of m in the Dial option, I have choppy ring too. If I rub my finger on the mouth piece, the ring/MOH is fine. Any solution to this problem? I'm using asterisk 1.4.1 with zaptel 1.4.0. Try adding this to the [options] section of /etc/asterisk/asterisk.conf: internal_timing = yes The restart asterisk. Let us know if it helps. Dan ___ --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] Packetization Rate
Matt wrote: To my knowledge, Asterisk's packetization rate is hard coded at 30ms. If I wanted to, where in the code could I go to change it to 20ms. Is there anything bad that might happen if I change it (asterisk related)? You don't mention what version you are using, but 1.4 does support alternate framing (packetization) options on a per codec basis. The feature originally was based on SVN trunk when it was still close to 1.2, but I would not want to try to backport and support it. Dan ___ --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] [Announce] Web-MeetMe V3.0.1 released
Minor bug-fix release, no new functionality. Bugs fixed: * app_cbmysql would fail to load * Incorrect handling of recurring conferences that spanned a DST transition Minor cleanup: * A couple image files were duplicated with both upper and lowercase names. The uppercase variants were deleted and the HTML code cleaned up to use just the remaining files. The new release can be found at: http://sourceforge.net/projects/web-meetme/ We do have a volunteer developer who will be maintaining the 2.X.X chain for Asterisk 1.2.X compatibility, so bug fixes and features that are not Asterisk version dependant will still be made available for older installations. The 2.X.X chain does not have the problem with app_cbmysql, but may suffer from the DST transition bug. Thanks, The Web-MeetMe development team... ___ --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] Fax with T.38
Ray wrote: Could anybody give me an authoritative answer on whether Asterisk can support T.38 pass-through when the clients are behind NAT? We have Asterisk servicing clients behind NAT (with nat=route, canreinvite=no) and would love to get T.38 going but have had no luck so far. The following case: http://bugs.digium.com/view.php?id=7844 Authoritative? Nope. But I'll try to help anyways... 1. t38pt_udptl must be set to yes in [general] in sip.conf ...suggests that T.38 *does* now work for clients behind NAT but I have the latest SVN trunk but still cannot get it to work? On the other side I have seen on this list only 2 weeks or so ago: http://www.mail-archive.com/asterisk-users@lists.digium.com/msg172556.ht ml This suggests that T.38 does *NOT* work behind NAT? So, can anybody save me the trouble and tell me how it is. Am I on a hiding to nothing trying to get T.38 going with NAT? Please put me out of my misery! :) Part of an age old issue that doesn't bear repeating, but is also not terribly accurate or relevant. Cheers, Ray Capture a debug log of a failed T.38 session and post it on Mantis. Make sure to set: core set verbose 4 core set debug 4 sip set debug Testing and (what little) feedback the developers have received indicate that it SHOULD work with the latest SVN. PS. Does anybody know whether OpenPBX would support T.38 and NAT configurations? This was my backup plan if I couldn't get it to go in Asterisk. No idea. Dan ___ --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] web-meetme cbmysql not registered
Ma write: HI, today I download Web-MeetMe-3.0.0 for asterisk 1.4.0 but when I call the extension which invoke cbmysql, a warning appears: What version of Asterisk? I ask because I have had Reports of problems against svn trunk and svn branches after 1.4.0 was released. WARNING[20225] pbx.c: No application 'CBMysql' for extension (default, 1995, 3) I check the application, it didn't registered CLI core show application CBMySQL Your application(s) is (are) not registered But I can see it use show module log snip this seems it was loaded successful. That portion of the log looks good. what's the matter? These steps will help identify the problem- 1. module unload app_cbmysql.so 2. core set verbose 10 3. core set debug 10 4. module load app_cbmysql.so Dan ___ --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] web-meetme cbmysql not registered
Ma wrote: WARNING[20225] pbx.c: No application 'CBMysql' for extension (default, 1995, 3) I check the application, it didn't registered CLI core show application CBMySQL Your application(s) is (are) not registered But I can see it use show module I made a small mess of supporting the new module loading process. The code attempts to determine if the config file was successfully loaded, and only then load the module and register the application. That is all fine and well, except I failed to properly flag a successful config load. How it ever worked for me, I don't know, but here is a quick fix: Find this section of the code- ast_log(LOG_NOTICE,Successfully connected to MySQL database.\n); connected = 1; records = 0; connect_time = time(NULL); } And add this: if (connected) return 1; else return 0; I'll get an update into svn if this works for you and release 3.0.1 Dan ___ --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] Which H323 module for asterisk
Pavel wrote: I prefer h323 included in asterisk tree, I have caller id issues with ooh323 and nobody answer to bugreports oh323 from inaccessible network is unmaintained/death project, incompatible with asterisk 1.4. PJ Response to ooh323c bugs is very slow, and patches can take some time to be applied if you manage to fix the issue for yourself. That said I prefer ooh323c, as it does not require OpenH323 or PWlib. I find building it easier. Michel wrote: Hello, I need your advice about H323 and asterisk! ;) Which one do you advice me to choose H323 (only gateway mode)? ooh323? ooh323c? Since you mention gateway mode, then ooh323c is worth testing. The bugs that I am aware of are mostly gatekeeper related (but not all). Since the channel doesn't have any external dependencies, it is the easiest to test. If it doesn't work for your setup, there's a very good chance that chan_h323 included with Asterisk will and then you can deal with getting the OpenH323 and PWlib dependencies meet. (Not a major issue, but one I have preferred to avoid) Which one is the best H323 module to use with asterisk? Which one did you choose and why? What is your return on experience? Bugs happen. I've found that the code for chan_ooh323c is reasonably easy to read and make patches for. The current release seems stable and I have it running on four light to moderately loaded servers. Dan ___ --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: [Announce] Web-MeetMe 3.0.0 released
Buki wrote: Sorry I forgot to change the subject line in my last posting! I have been using the Web Meet-Me 2.1.0 on Asterisk 1.2.12.1 for many months now and I am a big fan and I have been very happy with it. I'm glad it's working well for you, positive feedback is always welcome. I want to try the v3.0.0 but I would like to know if there are specific steps I need to carry out to upgrade to the v3.0.0 on my current Asterisk 1.2.X? There are a couple answers here. First is that version 3.0.0 is NOT compatible with Asterisk 1.2.X, so there is no way to test or use it in your installation. There is a plan to release version 2.2.0 soon that has the features and bug fixes from version 3.0.0 that do not have a dependancy on Asterisk's version. The second answer is about the upgrade it self. Since the package is mostly php pages, there is not an 'upgrade'. Just rename the directory where Web-MeetMe is installed and extract the latest package. With the 3.0.0 and 2.2.0 releases we have further seperated the configuration settings from the actual code, so future upgrades should be able to re-use the ./lib/defines.php. With the 3.0.0 and 2.2.0 release it will be easiest to just edit the new defines.php to match your settings. Lastly you may need to add a couple columns to your database to take advantage of the improved recurring conference support. Refer to the sample tables in the ./cbmysql directory for details. Dan current document root and extract the package to ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] [Announce] Web-MeetMe 3.0.0 released
Rob wrote: On 1/5/07, Dan Austin [EMAIL PROTECTED] wrote: Trunk has already moved on and code compatible with 1.4, may have problems on it. For a sanity check, I wiped out my test system and rebuilt it with fresh components for 1.4 (libpri, zaptel, asterisk, asterisk-addons), and I have no issues with unloading and re-loading the module, and of course the app does what it claims and works as intended. So I can either ask that you try 1.4.0, or I will need to setup a test against trunk. I'd prefer to wait a bit before coding against trunk, since it will break again, and likely before not too long. I guess I figured that trunk couldn't have gone far from 1.4 yet, so I'll move to 1.4. Nothing in particular on trunk I need. Thanks for your time, but sorry to have wasted it. It wasn't a waste though. I now know to expect more work once I look at Trunk closely. A quick review did not show anything exciting that has changed, but the APIs to register applications and CLI commands have changed a small bit. Thanks, Dan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] [Announce] Web-MeetMe 3.0.0 released
*CLI core set verbose 10 Verbosity was 0 and is now 10 *CLI module unload app_cbmysql.so Unable to unload resource app_cbmysql.so Command 'module unload app_cbmysql.so' failed. *CLI [Jan 5 11:09:04] WARNING[30610]: loader.c:465 ast_unload_resource:Firm unload failed for app_cbmysql.so So I added noload = app_cbmysql.so to modules.conf, and load manualy after restarting asterisk... *CLI module load app_cbmysql.so == Parsing '/etc/asterisk/cbmysql.conf': Found *CLI This *is* with verbosity set to 10, but this is all I was seeing before... I suspect a config file issue, but a log of the module loading will help peg down the problem. Doesn't look to me like this will be much help, but what do I know... It actually helps quite a bit, along with me taking the time to fully accept what version you are running (in one ear and out the other problem) Trunk has already moved on and code compatible with 1.4, may have problems on it. For a sanity check, I wiped out my test system and rebuilt it with fresh components for 1.4 (libpri, zaptel, asterisk, asterisk-addons), and I have no issues with unloading and re-loading the module, and of course the app does what it claims and works as intended. So I can either ask that you try 1.4.0, or I will need to setup a test against trunk. I'd prefer to wait a bit before coding against trunk, since it will break again, and likely before not too long. Dan ___ --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
