Re: [Asterisk-Users] CLI dial command
On Tue, Apr 26, 2005 at 04:31:59PM +0200, flavio patria wrote: I have just installed a release version 1.0.7 of asterisk: I already installed in past asterisk and in my previous installation I may find the dial command on CLI that now I haven't found: it is possible? The lack of dial CLI command is an upgrade?or Is there some problem in my installation? Dial command is provided by the OSS and/or ALSA modules. Check you're loading one of those... -- Martijn van Oosterhout Ecomtel Pty Ltd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP UA behind NAT and REINVITE ???
On Thu, Apr 07, 2005 at 03:57:11PM -0400, William M. Sandiford wrote: Hello: I've read through the list archives and found tonnes of threads on this topic but there has been no definitive answer, so hopefully someone can give me one. Can a proper 2-way audio call be established when the UA is behind a NAT firewall and REINVITE is enabled? Original Call Made SIP UA 1-- NAT FIREWALL --- Asterisk -- SIP UA 2 Then REINVITE occurs and SIP UA 1-- NAT FIREWALL SIP UA 2 Possible, yes. Whether it works depends on the firewall. Your problem is that UA2 is sending directly to the firewall and the firewall will block it because it knows nothing about UA2. Or not, if it supports partial matching on UDP ports. In theory a packet of UA1 to UA2 should open the back channel, except you run the risk of the firewall assigning a new port number, thus breaking everything. This is a problem uPNP was supposed to solve, the client can request an externally visible port on the router. Never seen any client that does this though. If you only have one UA you can get around it with port forwarding on the firewall... But you need to know in advance what ports SIP is going to use... I have tried and tried and tried to get this working but with no luck (well, I can get it to work with canreinvite=no, but thats not what I want. I want * out of the audio path) Good luck! -- Martijn van Oosterhout Ecomtel Pty Ltd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] User Regerstation, allowing non-registered users on *
On Fri, Apr 08, 2005 at 09:14:48AM +0200, Etienne Pretorius wrote: Hello *users, I would like to know how one would go about to allow every-one that wishes to connect to my * machine to connect without a registration being placed in the conf files. Would this be achieved through a Database that will lookup the UserName and Password and if it does not exist create a user with restricted access (only to call VoIP calls, with no voice mail). So what I am asking is has anyone done this and if so if they could give me a guideline... The config files for SIP and IAX both include examples of guest users, that don't need to login. No username, no password. Generally dropped to an incoming only context. After all, the idea is that anyone should be able to call you without having an account on your server. -- Martijn van Oosterhout Ecomtel Pty Ltd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How can I make base calls with X-Lite via Asterisk?
I beleive the Asterisk example config has an example, but this works for me (Xlite-linux-beta): [martijn] secret=secret type=friend context=from-sip; Where to start in the dialplan when this phone calls ;callerid=John Doe 1234 ; Full caller ID, to override the phones config host=dynamic; we have a static but private IP address ; No registration allowed nat=yes ; there is not NAT between phone and Asterisk canreinvite=no ; allow RTP voice traffic to bypass Asterisk ;dtmfmode=info ; either RFC2833 or INFO for the BudgeTone ;incominglimit=1; permit only 1 outgoing call at a time ; from the phone to asterisk ;[EMAIL PROTECTED] ; mailbox 1234 in voicemail context default disallow=all; need to disallow=all before we can use allow= allow=gsm ; Note: In user sections the order of codecs ; listed with allow= does NOT matter! allow=alaw ;allow=g723.1 ; Asterisk only supports g723.1 pass-thru! ;allow=g729 ; Pass-thru only unless g729 license obtained On Wed, Apr 06, 2005 at 03:25:56PM +0800, Abraham WEI wrote: I installed Asterisk in a default way. I ran over many manuals and FAQ's on asterisk.org. However, I found that many exaples included in them were equipment-dependent. I do not know how to configure my Asterisk for my X-Lite. Is anybody willing to help me? Regards, Abe ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Martijn van Oosterhout Ecomtel Pty Ltd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Incomming Call issues
On Tue, Apr 05, 2005 at 02:37:39AM -0600, Sascha Ferley wrote: Hi I am using asterisk at home, which seems to be messing up the extentions.conf file and a few other ones quite bad from the default. Well, it seems to me your problem is here. asterisk1*CLI -- Starting simple switch on 'Zap/3-1' -- Detected ring pattern: 0,0,0 -- Distinctive Ring matched context == Starting Zap/3-1 at ,s,1 failed so falling back to exten 's' == Starting Zap/3-1 at ,s,1 still failed so falling back to context 'default' You havn't got a defined context to go to. Somehow. So it keeps failing until it finds something that does exist, the default context, which hangs up on you. You need to set the context for incoming calls. Look for distinctive rings. -- Martijn van Oosterhout Ecomtel Pty Ltd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_iax2 stops listening to packets
I have two machines, running different versions of CVS HEAD. One with CVS-HEAD-02/15/05-19:41:22, the other with CVS-HEAD-03/22/05-22:15:16. Both are exhibiting the same problem. Basically, they stop listening to packets, so they queue up in the kernel. Proto Recv-Q Send-Q Local Address Foreign AddressState snip udp 107824 0 0.0.0.0:45690.0.0.0:* The number just keeps going up. I've tried lots of different configuration options. Just reloading doesn't unstuck it. Unloading the module causes strange hangs. Stopping and starting Asterisk works until the first IAX call attempt is made. Then it stops working. One is using a TDM card (wct4xxp driver), the other is using ztdummy. They are dual xeons, if it matters. How can I debug this? How do I identify the thread running the IAX stuff? I confirmed the socket is still open in lsof. SIP calls keep working, ZAP calls keep working, just IAX is stuffed. Perhaps I should just stick to SIP for the time being... I can't find any bug that might have something to do with this. Any known patches recently which might be related to this? Actually, it doesn't even need to be triggered by a call. The log looks like: Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 1ms SCall: 6 DCall: 0 [203.194.19.82:4569] Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: PONG Timestamp: 1ms SCall: 3 DCall: 6 [203.194.19.82:4569] Tx-Frame Retry[-01] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 1ms SCall: 6 DCall: 3 [203.194.19.82:4569] Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 3ms SCall: 4 DCall: 0 [203.194.19.82:4569] glenmorangie*CLI iax2 debug IAX2 Debugging Enabled At this point the other machine starts recording this one as UNREACHABLE. I've disabled all trunking, but that didn't help. Thanks in advance, -- Martijn van Oosterhout Ecomtel Pty Ltd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Authenticating username
Hi, From what I can see in the documentation the title of the section in sip.conf is the username that the user logs in as. Is there a way of seperating the names so that you can login with a normal username, but call them with SIP/extension. Like so: [904] authuser=john secret=password etc... Dial(SIP/904)calls whoever logged on as john. Any ideas? -- Martijn van Oosterhout Ecomtel Pty Ltd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Looking for res_config_pgsql
A google search shows exactly one reference, so it appears to exist somewhere. It's in somebodies CVS, any ideas? -- Martijn van Oosterhout Ecomtel Pty Ltd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Preserve g729 registration over reinstall??
On Thu, Mar 31, 2005 at 01:43:41PM -0800, Mike Matthews wrote: I purchased the g729 codec from Digium. Every time I reinstall Asterisk (or Linux) I naturally lose the registration. Digium only allows one reinstallation without calling them which is a nusance for both them and me. Is there any way to preserve the registration across a reinstall? Perhaps by backing up a directory or a file? Any help appreciated. Check the archives to be sure, but I beleive the relevent files are stored in /var/lib/asterisk/licences or something like that. Good luck. -- Martijn van Oosterhout Ecomtel Pty Ltd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] setting SIP to dial PSTN with TDM400P
Hi, I've never used fxs/fxo modules, only E1 cards so I'm not entirely sure. However, this log: *CLI -- Starting simple switch on 'Zap/1-1' -- Executing Dial(Zap/1-1, Zap/1/6998256) in new stack -- Called 1/6998256 -- Zap/1/6998256-busy-1013475805 is busy -- Hungup 'Zap/1/6998256-busy-1013475805' == Everyone is busy/congested at this time -- Timeout on Zap/1-1 == CDR updated on Zap/1-1 seems to indicate you're making the call from Zap/1 and trying to make the outgoing call on Zap/1 also. I think you need to figure out which Zap channel is your FXO and which is your FXS. Maybe the outgoing is Zap/2? zap show channels gives a list I beleive... Secondly, your config files only seem to mention one channel. Have you looked at [EMAIL PROTECTED] It seems to autodrtect your config somehow On Fri, Apr 01, 2005 at 01:08:24PM +0500, Muhammad Haris wrote: to dear martijn, i made every possible change i can make i have a TDM400P Zap card... i had connected PSTN line to FXO Kewlstart at channel 1. and analog phone to FXS Kewlstart at Channel 4. i can hear continous ring tone when i hook up the receiver. plz have a look at my confs. Have a nice day, -- Martijn van Oosterhout Ecomtel Pty Ltd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Are there online forums instead of this
On Thu, Mar 31, 2005 at 05:14:50PM -0500, Tim Bass wrote: I use procmail and know very well how to manage email. All asterisk mail goes to a folder,etc. Your point...because a few people don't understand how to manage e-mail is nonsense and shows why this list should be moderated. *BLINK* He expresses an opinion, you disagree with it and therefore what he says should be moderated out of existance? I'm sorry, but arguments that on this list not every post is as nice as possible are just not going to fly. This list is no worse than any other I'm on. This is the real world and the list reflects that. If you don't like what you read, ignore it. But the idea of moderation scares me because I might miss something useful just because someone else decided I shouldn't see it. I'll decide for myself thank-you-very-much. From my point of view, web based forums can never compete for me because: 1. The RTT to bring a new page in and render it takes at least a second, usually more. 2. Displaying more than one message at a time is irritating because then you have to scroll around and it can no longer track read/unread. 3. Finally, colours, pictures, odd fonts, etc slow down my reading speed. I prefer everything in a fixed width font, white text, black background, each message starting at exactly the same point on my screen. Keyboard control only. The combination means that I could only get through less than half as many forum posts as mailing list posts in a given period. Time being money completes the picture. -- Martijn van Oosterhout Ecomtel Pty Ltd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Australia and SetCallerID
On Thu, Mar 31, 2005 at 01:58:21PM +1130, Craig wrote: From my investigations, I can't find any carrier in Au that does allow it to be set outside the allocated range, can't even get one to set it to our 1300 number, have been told the ACA doesn't permit it in Au, but not certain on that. Correct. It's bordering on illegal to lie about your callerID (not quite though, because it's not legislated). Certainly you're not going to be able to do it on any basic commercial service. Carrier interconnects can do it obviously. The reason is related to emergency services (amongst other things), a 1300 number has no physical location. Origination numbers can be geographic or mobile but must be associated with a specific location or device. Technically, if you have a 100 number range that you split over multiple locations, you're required to indicate which numbers are where. Doesn't happen often though. It even goes so far as if you're transiting a call you must preserve the originating number. This means if you setup a calling card platform as a carrier you may have a few extra requirements. My understanding many pri in the US can be set to almost any number. So I've heard, can't quite understand the benefits, but this is the way things are. -- Martijn van Oosterhout Ecomtel Pty Ltd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] File permissions and ownership
On Wed, Mar 30, 2005 at 11:36:18AM -0800, Kenneth Porter wrote: Excellent, thanks for the info! I was mostly worried about opening privileged ports, but an initial test showed only high ports opened. I'd guess that only files asterisk needs to write need to be owned by the asterisk user, and the other files (eg. sounds) can be owned by root and made world-readable. The only issue is that as non-root, asterisk can't set the TOS bits (though iptables can do it for you) and it can't set its own priority (though nice/renice can help there). -- Martijn van Oosterhout Ecomtel Pty Ltd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Xten-lite for linux
On Thu, Mar 31, 2005 at 02:35:07AM -0500, Kris Edwards wrote: Well, I'm certainly not selling xten.. Perhaps my enthusiasm extends from my disgust with everything else. In particular, kphone, and sjphone. I have noticed latency with xten in meetme, but if I just dial somebody it works better than anything I've tried (so far.. I've only spend about 1 hour talktime). Anyway, I'm certainly more hip on open source, and can't wait to try gnomemeetings sip once I can actually get it to compile :/ I have to agree though, I tried a lot of softphones under linux and the xten was the first one that worked. Not just that, it worked *perfectly* first time, no whacky obscure problems. Now if only Firefly worked under linux, that's be really cool... -- Martijn van Oosterhout Ecomtel Pty Ltd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] setting SIP to dial PSTN with TDM400P
On Thu, Mar 31, 2005 at 05:05:21PM +0500, Muhammad Haris wrote:
The calls between SIPs and zap phone (fxs) are OK. But 2 issues
cannot be solved:
1. To dial to PSTN via zap phone, the setup in extensions.conf with
the following
exten = _Nxx, 1, zap/1
doesn't work.
I think you need to tell Asterisk what you actually want (Dial) and
tell it the phonenumber, perhaps:
exten = _Nxx, 1, Dial(zap/1/${EXTEN})
Check the wiki for more details about the dial command...
2. When trying using SIP phone to dial PSTN, I got no luck.
Probably same issue...
Hope this helps,
--
Martijn van Oosterhout
Ecomtel Pty Ltd
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Re: [Asterisk-Users] Are there online forums instead of this email
On Thu, Mar 31, 2005 at 01:19:40PM -0500, [EMAIL PROTECTED] wrote: Any decent on-line forum would be much better than these digium email lists. The lists are poorly formatted, there is no easy way to post code, you cannot neatly quote anyone, the s earch function in the archive is elementary at best, there is no possibility for active use rs to moderate their area of interest, there is no private messaging, the list goes on and on. Email has attachments, there's wiki for out-of-band stuff and you can use google to search. And email *is* private messaging, so I don't understand that point. A GOOD ON-LINE FORUM IS TO EMAIL LISTS WHAT THE WORLD WIDE WEB WAS TO GOPHER. Ok, basic use case. I today go to a forum and read all the messages. Next day I come along, how do I get a list of all the messages I havn't read in thread order in such a way that if I decide to go somewhere in the meantime, it knows what I've read and what I havn't. I also monitor several other projects all on mailing lists. With one mail box I can monitor six projects in one interface. I don't touch the mouse the whole time. I can whizz through a message every few seconds because every one is in the same font, same colour, same spacing (HTML all disabled). No forum is ever going to compete with that sorry. In other words, Digium is only hurting this community by not transitioning these lists to a good on-line community building software product. The list of major projects with mailing lists is longer than the list of projects with forums... Feel free to set one up, but if the mailing list is removed for a forum, many current people won't subscribe... -- Martijn van Oosterhout Ecomtel Pty Ltd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Are there online forums instead of this email
On Thu, Mar 31, 2005 at 12:15:54PM -0800, Kenneth Porter wrote: I'm not opposed to forums, but your arguments don't sway me. (I'd actually prefer a newsgroup to either a mailing list or forum, because I don't need to download the message bodies for stuff I'm not interested in. But I can't seem to post from the gmane gateway.) The suggestion I saw on the postgresql list is to subscribe nomail. Then read via a news gateway and when replying, reply via email to the mailing list. If the gateway preserves Message-IDs, people won't even notice the difference (threading is preserved) and you don't have to download all the messages... Whether this list can work like that I don't know, but there it certainly works. -- Martijn van Oosterhout Ecomtel Pty Ltd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can't get format_mp3 to work for music on hold
On Sun, Mar 27, 2005 at 01:41:31PM +0100, Umar Sear wrote: Hi Guys, I am having trouble trying to get format_mp3 working to play music on hold. I have followed the instructions in the read-me and the wiki however it seems after un-installing mpg123, asterisk is not even attempting to play MOH. I don't do anything with music-on-hold, but... [moh_files] default = /var/lib/asterisk/moh-native ^^^ That spacing looks really odd... Hope this helps, -- Martijn van Oosterhout Ecomtel Pty Ltd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Error cannot record voicemail
On Thu, Mar 24, 2005 at 01:30:25PM -0500, Joel Duffield wrote: I tried to share my spool directory so I could get monitored calls, and now this error comes up when I try to leave a message in any of my voicemail boxes. Mar 24 12:48:35 WARNING[344081]: file.c:906 ast_writefile: Unable to open file /var/spool/asterisk/v oicemail/default/300/INBOX/msg.WAV: No such file or directory When you get the error No such file or directory when opening a file for writing it generally means that one of the preceeding directories doesn't exist. So check if the directory /var/spool/asterisk/voicemail/default/300/INBOX/ exists. I beleive the Voicemail app creates the directories itself so if the directory doesn't exist, it can't create them. Make sure you're the same user as asterisk is running as and try: mkdir -p /var/spool/asterisk/voicemail/default/300/INBOX/ Maybe one of the components has been replaced by a file or directory with bad permissions Hope this helps, -- Martijn van Oosterhout Ecomtel Pty Ltd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk compare with Skype
On Fri, Mar 25, 2005 at 01:18:21PM -0800, Sys Admin wrote: if some one was to create a open source IAX client as good/better then skype, even then a asterisk IAX based network will not be able to compete with skype. Since asterix is a centralized server regitration network it can not grow as big as a skype P2P network can grow, One thing Skype has going for them, they have clients for Linux and Windows. I like Firefly, but no Linux interface. Until I found x-lite for linux I couldn't find a single Linux client that came remotly to Just Working. I downloaded Skype for Linux and it worked out of the box, first time, no hassles. How can you compete with that. Asterisk is a PBX, Skype is not, but Skype solves a problem simply and straight forwardly that Asterisk doesn't. -- Martijn van Oosterhout Ecomtel Pty Ltd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Permission issue with outgoing calling
On Tue, Mar 22, 2005 at 02:07:32PM +1200, Cameron Beattie wrote: I have created a call file which has been moved into the outgoing directory. However the log file displays the following message: Unable to open /var/spool/asterisk/outgoing/1.call: Permission denied, deleting I have executed chmod 777 1.call on the file prior to moving it to the outgoing directory but is there something else I need to do before the file can be used by Asterisk? It also needs to be the owner of the file (unless asterisk is root) so it can utime() it. Also check asterisk has rwx permissions on the spool directory and can read all the parent directories... Hope this helps, -- Martijn van Oosterhout Ecomtel Pty Ltd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Some IAX questions
On Sat, Mar 19, 2005 at 09:47:49PM -0700, Tim Pushor wrote: Hi, Is this a silly question? I am trying to come up with an elegant way of joining a few small * servers in a peer to peer arrangement, and I am just curious as to what algorithm * uses to determine which channel (and therefore context) an inbound call belongs to (IAX and SIP).. Also, knowing when name resolution happens would be beneficial if the peer * boxes had dynamic IP's and dynamic dns ... I googled for Asterisk iax authentication which returned, amongst others: http://www.voip-info.org/wiki-Asterisk+IAX+authentication It should tell you all you need to know... -- Martijn van Oosterhout Ecomtel Pty Ltd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Realtime does not work yet, ... *bug*
On Wed, Mar 16, 2005 at 03:25:17PM +0800, Ronald Wiplinger wrote: Mar 16 15:13:45 DEBUG[29502]: Raw Hangup 69.73.19.178:4569, src=14, dst=1259 Mar 16 15:13:45 DEBUG[29502]: MySQL RealTime: Update SQL: UPDATE sip_buddies SET name = '621' WHERE allow = 'g729' Mar 16 15:13:45 DEBUG[29502]: MySQL RealTime: Everything is fine. Mar 16 15:13:45 DEBUG[29502]: MySQL RealTime: Updated 0 rows on table: sip_buddies *ALARM* Where is that query fron? It's totally wrong! It just changed the name of anyone who is allowed to use g729. Looks like Realtime is not quite there yet for production... Have a nice day, -- Martijn van Oosterhout Ecomtel Pty Ltd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] live monitoring of SIP calls chan_spy
On Wed, Mar 16, 2005 at 11:06:08AM +, Atif Rasheed wrote: hello there, I have searched lists about an application chan_spy, people talked about it on lists that we can use it to monitor sip to sip calls. but I am unable to find any clue of it. can some one please tell me from where I can get this chan-spy application Maybe it's been replaced by the Monitor app? Or does it do something else? -- Martijn van Oosterhout Ecomtel Pty Ltd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Realtime does not work yet, ... *bug*
Hi, On Wed, Mar 16, 2005 at 08:47:49AM -0600, Matthew Boehm wrote: OK. I've been patient and kind up until now. Here comes the rudeness: Martijn, shut up! This is now the 3rd time you have stated that Realtime is not ready for production using baseless acquisations. I'm terribly sorry, It was a little hasty of me to lay the problem on Realtime as opposed to user error. I was just pointing out that the query was obviously wrong. However, I'm quite sure (and I grepped all my mail just to be sure) but this is the only time I have ever mentioned realtime in any email involving this list. Maybe you have me confused with somebody else... There is still a problem somewhere with somebodies mail gateway where one can receive multiple copies of the same email. Your email reached me three times for example. Again, my apologies, -- Martijn van Oosterhout Ecomtel Pty Ltd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ASTCC and NuFone billing is different!!
On Sat, Mar 12, 2005 at 03:26:40PM -0500, William Suffill wrote: NuFone service bills in industry standard billing increments, which are: six (6) seconds for the US48, sixty (60) seconds to Mexico and fifteen (15) seconds to the remainder of the world. From: http://www.nufone.net/tac.html Ah right, so this is where some dodgy Australian carriers got that idea. Fortunatly it's not common and per second billing is the norm for standard phone lines. Calling cards tend to fiddle around more here. The worst I saw was a calling card that advertised per minute rates and at the bottom of the page declared: 1 minute is defined as 57 seconds. -- Martijn van Oosterhout Ecomtel Pty Ltd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] possible bug in chan_capi concerning context handling
On Sun, Mar 13, 2005 at 06:44:52PM +0200, Dimitris Kounalakis wrote: Thank you for your response Marco. I do. The problem is that all incomings calls from ISDN are handled by the default s extension in the context [default] and not by an s extension in the context [isdn] or by the msm numbers as extensions in the context [isdn]. Looking at the line here: // == Starting CAPI[contr1/2810111694]/3 at ,2810111694,1 failed so // falling back to exten 's' It looks like the context is blank. What does the show command in asterisk show the context as being (paste output please). -- Martijn van Oosterhout Ecomtel Pty Ltd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unable to create channel of type 'IAX2'
On Sat, Mar 12, 2005 at 04:04:29PM +0100, Androtech wrote: Hi all, I'm a newbie and I have a configuration problem with Asterisk. Seems that I'm not able to call an outbound number. I'm quite sure that it is a configuration problem, but I'm not able to find out where is the mistake, even reading several docs to www.voip-info.org. You didn't read all the messages, see: -- Executing Dial(IAX2/[EMAIL PROTECTED]/2, IAX2/3479450772) in new stack Mar 12 16:15:36 WARNING[3149]: chan_iax2.c:2341 create_addr: No such host: 3479450800 Mar 12 16:15:36 NOTICE[3149]: app_dial.c:911 dial_exec_full: Unable to create channel of type 'IAX2' (cause 3) chan_iax2 says you have no section in your iax.conf telling it what 3479450772 is, *therefore* it couldn't create the channel. Always read *all* the messages, especially the first few since they may indicate something that may cause a failure later. FWIW, you probably wanted IAX2/target/3479450772. Hope this helps, -- Martijn van Oosterhout Ecomtel Pty Ltd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Should ICMP port unreachable generate a BYE request?
On Wed, Mar 09, 2005 at 01:51:09PM +0200, Dipole Moment wrote: I'm researching random call drops on our Asterisk and would like to make sure whether it's something wrong with our VoIP provider or with the Asterisk. I sniffed traffic between Asterisk and our VoIP provider's SIP gateway, and observed that in the middle of the conversation an RTP stream originating from Asterisk gets an ICMP port unreachable from provider's SIP gateway at random times and conversation seems to go on for a while, but after a while a few more port unreachables are observed and Asterisk sends BYE request to both parties. I wonder if it is it normal for Asterisk to send BYE requests to both parties once it gets a port unreach even though noone from the either end has hanged up the call? If yes, then why doesn't it send BYE request on the first unreach it sees? Or is it some tunable parameter that can be set via configuration files? Or should I mail the sniffer dump to my provider and ask them to fix their gateway? You'd have to trace the code to work it out properly. But ICMP packets aren't generally passed to userspace. What's more likely is that the kernel, upon receiving sufficient of these errors, decides the connection is dead and notifies asterisk. Although, with UDP (in Linux anyway) the error can be passed back. Strange problem though, how can only some packets generate Port Unreachable, but not all. Random routing problem? Have a nice day, -- Martijn van Oosterhout Ecomtel Pty Ltd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Upgrading Asterisk
On Wed, Mar 09, 2005 at 02:13:02PM -0600, Dennis Webb wrote: Question about the make linux26 command. I use a 2.6 kernel and always do a straight make. Does adding the linux26 do anything except help the makefile know that it's a 26 kernel if it has trouble detecting for some reason? I might go ahead and try 1.0.7 and need to know if make linux26 does anything special a simple make doesn't. There's a little chunk in the makefile that attempts to guess what your current kernel is and selects the right target. I've never specifically target 2.6 but I have made sure that the kernel headers on the system as the exact kernel headers I compiled the kernel and I'm running that kernel also. Has always worked for me... Hope this helps -- Martijn van Oosterhout Ecomtel Pty Ltd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Providing a dialtone
Hi, I see applications for signalling busy, congested, ringing, progress etc, which I understand can be provided either in or out of band. But all I want to do is generate a dialtone. This obviously can only be done in band. There is code for generating the tones when you have a physical line, like the alsa channel, or a zap channel. But I'm just thinking of if they've selected an option that allows them to dial a normal number, to also provide a normal dialtone. Should I just record one and use Background()? Thanks in advance, -- Martijn van Oosterhout Ecomtel Pty Ltd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] LiveVoIP Problems?
On Mon, Mar 07, 2005 at 12:10:48AM +, Mike Dent wrote: BT providing IAX2 and SIP termination? Hmmm, maybe one day. Telstra (BTs equiv in Australia) is trialling a VoIP service. Unfortunatly, it's not quite clear what services they'll be providing... -- Martijn van Oosterhout Ecomtel Pty Ltd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE110P module woes
On Mon, Mar 07, 2005 at 11:16:18PM +0100, Alfredo Sola wrote: Ok, I'll correct. I would like to be able to set in the Makefile which user I'll be using, so that permissions can be set accordingly. No problem AFAIAC if the default user is root. Perhaps time for an autoconf? Hardly. There's a wiki page on how to run non-root. The only thing that needs modification is the default varrundir. You do have to manually chown a bunch of stuff but that's not a big deal. The only thing I'd really like to see is a commandline option or config option that will cause it to die if you accedently start it as root. Doing it once by accident tends to alter permissions on files that then needs to be fixed manually... -- Martijn van Oosterhout Ecomtel Pty Ltd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Getting asterisk-addons installed on Debian?
On Sat, Mar 05, 2005 at 01:19:24PM -, C. Tomlinson wrote: Hi, I am having some trouble installing asterisk addons on Debian. I wish to do this to use mysql billing. snip make[1]: gcc: Command not found You need a C compiler, try apt-get install build-essential Hope this helps, -- Martijn van Oosterhout Ecomtel Pty Ltd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems with g729 codec
On Fri, Mar 04, 2005 at 04:37:06PM +, Asterisk guy wrote: G729 will not work without a licensecan't G729 work in passthrough mode without license? if yes, how to configure it work in passthrough mode? Passthrough means that the codec going in is the same as the codec going out. If you configure all your phones to be g729 then you can use passthrough. If you want to use prompts stored in something else (like wav or gsm) you'll need a licence for that... You said the other phone was using a different codec, hence the problem... On Fri, 04 Mar 2005 08:50:11 -0600, Steven Critchfield [EMAIL PROTECTED] wrote: G729 will not work without a license. The error message above told you that asterisk couldn't find a valid path to convert from gsm audio to g729 audio data. Seems that should have been very obvious from the error. It is well documented had you even decided to search. Hope this helps, -- Martijn van Oosterhout Ecomtel Pty Ltd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] country/city codes
On Thu, Mar 03, 2005 at 06:25:09AM -0800, VoIP Services wrote: Some country codes are three digits long. Some are two. e.g. UK 44 , Bermuda 441 Does anyone know a formula for determining which part of a dialled number is the country code and city code ? There is no formula, you need to make a list. There are lists around the place, telling you what each prefix means. It also changes over time, as countries change their numbering to deal with growth of population and services. http://www.wtng.info has a lot of useful information... Hope this helps, -- Martijn van Oosterhout Ecomtel Pty Ltd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] country/city codes
On Thu, Mar 03, 2005 at 11:38:12AM -0500, Gary Reuter wrote: Aren't country codes 1 or 2 digits? Area codes are 3 digits: 44 is a country code, 441 is an area code... the country code for Bermuda is '1', same as Canada, US, and most of the Carribean nations. Check this out for the solution you need: http://lists.digium.com/pipermail/asterisk-dev/2004-May/004151.html Country codes are between one and three digits. Anything with the country code +1 is part of the NANP (North American Numbering Plan) and has three digits area codes. Australia has one digits area codes, Holland has two digits. Every country decides for itself. Many countries had varying length area codes but these are slowly being standardized. Browse http://www.wtng.info/wtng-cod.html if you want the grizzly details. On Thu, 3 Mar 2005 06:25:09 -0800, VoIP Services [EMAIL PROTECTED] wrote: Some country codes are three digits long. Some are two. e.g. UK 44 , Bermuda 441 Does anyone know a formula for determining which part of a dialled number is the country code and city code ? -- Martijn van Oosterhout Ecomtel Pty Ltd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [OT] stupid firmware question...
On Wed, Mar 02, 2005 at 02:24:17PM -0600, Chris Wade wrote: PS: [* put on flame suit *] why won't any of the phone manufacturer's just open-source the firmware for their phones? [* ducks head back inside gopher hole just before a giant fireball hits *] Probably because they're afraid you might do a better job... That aside, if they'd acquired licences for codecs or things, it might be tricky. Then again, if you paid for the licence when you bought the phone why would they care. -- Martijn van Oosterhout Ecomtel Pty Ltd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Getting Polycom IP500 to talk to Asterisk - um... Newbie question :)
On Wed, Mar 02, 2005 at 12:42:11PM -0800, Don Murray wrote: Hmmm... I have this aweful feeling that I'm choosing the exact wrong time to ask a newbie question :) Oh well, here it goes. The quick question is : How do I dial an extension? (answer is probably - you don't in which case:) How do I dial my asterisk box? - I have no outside line, I just want to start testing things like voicemail internally. snip No stupid question here, you've obviously done your homework. You should look up breifly in the docs about contexts and extensions. According to the context line in your sip.conf, when those phones dial, they will be in context sip. Go to your extensions.conf and check you have something defined there. What you'd expect is something like: [sip] exten = 6000,1,Dial(SIP/175polycom) exten = 6001,1,Dial(SIP/175polycom) exten = 6010,1,Goto(demo,s,1) ; Just for fun... Then they can use 6000 and 6001 to call themselves and eachother. This should be enough to get you started. -- Martijn van Oosterhout Ecomtel Pty Ltd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] timing/clock problem
You checked the crc4 setting, right? And the protocols... On Wed, Mar 02, 2005 at 06:49:53PM -0300, Alex G Robertson wrote: But when I configure span4 to get clock source from telco they become unsynchronized. TElco bit rate stays in 2048000 bps, but asterisk stays on 2048443 pbs!! span=4,1,0,ccs,hdb3,crc4 Hope this helps, -- Martijn van Oosterhout Ecomtel Pty Ltd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How does the g.729 registration program work?
On Mon, Feb 28, 2005 at 12:35:29AM -0330, Paul Fielding wrote: You misunderstand. Ofcourse I need to run the register program on the machine itself. The point is I build them from images and every now and then I roll out a new image. My question is, what do I need to preserve from the previous image to keep the licences. Obviously reformatting the disk and reregistering is not going to work. I could be mistaken, but doesn't the license tie itself to the nics on the server? I believe the Digium server will allow you to reregister as much as you want as long as it's still got the same nics... It does, but if you're only upgrading asterisk but not changing any hardware, there's no need to reregister anything. In my case I just wanted to make sure that the registration wasn't going to write somewhere read-only (say /usr/lib) or in a ram-disk (in my case /etc). It's in /var/lib/asterisk which I have as a real disk... Even if you can reregister each time, it easier to remember the actual licence than it is to remember the key and reenter manually each time. -- Martijn van Oosterhout Ecomtel Pty Ltd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium Card Problems
On Mon, Feb 28, 2005 at 09:58:28AM +0200, Mark Kidd wrote: Hi all i need urgent help our entire switchboard is down only 5 days after it came up. Read the other email first, you seem to need to know a little more about linux also. In any case I do have one hint for you: [EMAIL PROTECTED] root]# modprobe wcfxo /lib/modules/2.4.20-8/misc/wcfxo.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including inva lid IO or IRQ parameters. You may find more information in syslog or the output from dmesg /lib/modules/2.4.20-8/misc/wcfxo.o: insmod /lib/modules/2.4.20-8/misc/wcfxo.o fa iled /lib/modules/2.4.20-8/misc/wcfxo.o: insmod wcfxo failed After you load a module and it fails, the error message is generally in the kernel logs. Type dmesg to see that. The lines near the end should be helpful. They are necessary for anyone to diagnose your problem... Have a nice day, -- Martijn van Oosterhout Ecomtel Pty Ltd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T.38 fax summary
On Sun, Feb 27, 2005 at 05:32:49PM -0800, Lee Howard wrote: Quite right. I'm sorry to have misled. What happens is this (as an example scenario): The receiver will, for an example, receive the post-page message. The sender expects a response to this. The receiver, however, is required to wait between 55 and 95 ms before transmitting the response. The sender will likely be looking for the post-page response immediately after transmitting the post-page message. Per spec the sender will only wait about 3 seconds (per-spec between 2550 and 3450 ms) before giving up wating and retransmitting the post-page message (and then re-expecting the response). Thank you. So the 1 second lag I suggested is too much, but the principle is sound. Say we change it to half a second you're well under the limit. The question then becomes, is a fixed half-a-second jitterbuffer good enough to remove all the problematic jitter from the signal. This is a testable assertion (though unfortunatly I don't have the necessary equipment), simulating jitter is possible and hopefully the jitterbuffer itself is tunable. A tunable jitterbuffer sounds like a good idea, anyone actually thinking of implementing it though? Have a nice day, -- Martijn van Oosterhout Ecomtel Pty Ltd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Possibility of getting someone to delete a user from the list???
On Mon, Feb 28, 2005 at 10:04:27AM +0100, Dave Cotton wrote: On Sun, 2005-02-27 at 17:54 -0500, Robert Webb wrote: This is getting VERY annoying. Is there anyone in here that has access to the list administration to delete the user below??? Pray tell me why. The list isn't being flooded by these messages as far as I see. Their mailserver is broken in that it sends bounces to the From address (ie the person who sent the email) rather than the Sender (the asterisk mail server). So you only get an message from them when you send something. That is, one email for *every* message you send. There's also a server somewhere sending each message back to me with this attached: --- Spam detection software, running on the system zeus.avanzada7.com, has identified this incoming email as possible spam. The original message has been attached to this so you can view it (if it isn't spam) or label similar future email. If you have any questions, see the administrator of that system for details. --- However, the content analysis tells me the score is 0.1 of the necessary 5.0. Unfortunatly it's not helpful enough in determining the email address with the problem. Have a nice day, -- Martijn van Oosterhout Ecomtel Pty Ltd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T.38 fax summary
On Sun, Feb 27, 2005 at 04:14:46PM +0800, Steve Underwood wrote: Questions keep comming up about this, so I started writing something at http://www.soft-switch.org/foip.html . I think I covered the FAX over VoIP issues fairly completely. T.37 is pretty simple to explain. There is rather more to say about T.38, but at least this is a start. If anyone wants to suggest corrections or additions, just blurt them out. Hi, I read it and found it very enlightening. I do have one question regarding Modems don't like relativity. It says modems need a constant delay; is there a limit to what it can handle. For example, would it be possible to configure a jitterbuffer right at the endpoint before the fax to put a constant delay of 1 second relative to the sender. This should be enough time to weed out any jitter. Basically, fix the jitterbuffer so the delay is constant. If a fax can handle a constant delay of up to a second you're home. Maybe also allow people to setup a jitterbuffer to do a special interpolation for fax, which might just amount to sending a special tone representing zeros. Is there a possiblity of creating a app allowing people to configure the jitterbuffer for a particular call? In any case, if it can't be done this way, could you explain why? -- Martijn van Oosterhout Ecomtel Pty Ltd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T.38 fax summary
On Sun, Feb 27, 2005 at 09:10:48AM -0800, Lee Howard wrote: Fax cannot handle a one-second delay. As Steve mentions in the article, per-spec fax has some timings (particularly silence in direction switching) set at 75 ms +/- 20 ms. So if the delay gets much larger than 75 ms, then there's likely to be trouble. Now, some fax machines may tolerate larger delays, but that tolerance is beyond the spec, and thus should not be used as a gauge. Something's not right here. In 75ms light has just made it from here to the other side of the world. Even a PSTN network will provide a longer delay than that calling across the world. And the time to get a response back will be around twice that, which is well beyond that tolerated range. If you're saying there is a limit to the round-trip-time then within the fax specification is a predefined maximum physical distance you can send faxes. Faxing across the world does work so there is something else going here... -- Martijn van Oosterhout Ecomtel Pty Ltd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How does the g.729 registration program work?
On Fri, Feb 25, 2005 at 09:15:19PM +0100, Martijn van Oosterhout wrote: You misunderstand. Ofcourse I need to run the register program on the machine itself. The point is I build them from images and every now and then I roll out a new image. My question is, what do I need to preserve from the previous image to keep the licences. Obviously reformatting the disk and reregistering is not going to work. To follow up for the rest of the list, the licences appear to be stored under /var/lib/asterisk/ so as long as you preserve everything in there when you upgrade you shouldn't lose your licences. -- Martijn van Oosterhout Ecomtel Pty Ltd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ERROR: compile asterisk(from CVS HEAD) and got an error
See bug #3639 http://bugs.digium.com/bug_view_page.php?bug_id=0003639 Nothing's been committed yet though I think... On Sat, Feb 26, 2005 at 10:58:11PM +0800, Charles Wang wrote: Dear ALL: I got an error message lists below. Does anyone have the same problem? How to solve it? Best Regard Charles In file included from config.c:34: include/asterisk/app.h:62: array size missing in `options' make: *** [config.o] Error 1 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Martijn van Oosterhout Ecomtel Pty Ltd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What is an E400P-SS7??
On Fri, Feb 25, 2005 at 09:40:00AM +0100, Roger Schreiter wrote: When I learned about that other project, Steve Underwood was talking here, I gave up looking after the asterisk-SS7 project by OpenSS7, and begun supporting that libisup project for asterisk. You mentioned my very old status reports. I think, I already wrote about that change in the early autumn, but then got silent, because Steve gave some statements, and he is more involved in that project than me. Could I clarify hereby the advances of my SS7 interest and the status reports you mentioned? I guess I asked the wrong question. I'm in the situation where being able to do SS7 with Asterisk would be *very* useful and have someone who may be in interested in spending money on equipment and/or programming time to realize it. But information about SS7 on Asterisk is very thin on the ground. At least it seems that the hardware doesn't require changing, this is good... Thanks in advance, -- Martijn van Oosterhout Ecomtel Pty Ltd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How does the g.729 registration program work?
I'm asking because I'm planning to install multiple machines from the same image and I need to know what file(s) I need to backup/restore to make sure I don't lose my licences in the process. The only options I can think of are: - There's a config file, though I've seen no mention of it - The actual binary shared library is modified - The system contacts Digium every time you start asterisk In the last case nothing is changed at all and I'm fine. Thanks in advance, -- Martijn van Oosterhout Ecomtel Pty Ltd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How does the g.729 registration program work?
On Fri, Feb 25, 2005 at 11:24:21AM -0600, Steven Critchfield wrote: It is based on a machine unique key created by querying your hardware. You will not be able to share your licenses between machines. You will need to buy licenses for each machine you deploy on. You misunderstand. Ofcourse I need to run the register program on the machine itself. The point is I build them from images and every now and then I roll out a new image. My question is, what do I need to preserve from the previous image to keep the licences. Obviously reformatting the disk and reregistering is not going to work. So what will??? Thanks in advance, -- Martijn van Oosterhout Ecomtel Pty Ltd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] What is an E400P-SS7??
Hi, Is this card the same as the T410P, after all, it's made by Digium. There's one prior reference on the mailint list[1] but it didn't answer the question. There was also an SS7 status report[2] last June but it's doesn't seem to have lead anywhere either. There was post saying an SS7 release was immenent last September[3], but then silence. Any info anyone would like to share? Thanks in advance, [1] http://lists.digium.com/pipermail/asterisk-users/2004-September/062882.html [2] http://lists.digium.com/pipermail/asterisk-users/2004-June/052198.html [3] http://lists.digium.com/pipermail/asterisk-users/2004-September/062872.html -- Martijn van Oosterhout Ecomtel Pty Ltd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip Notify PAP2-NA?
On Thu, Feb 17, 2005 at 09:44:49AM +0100, Olle E. Johansson wrote: It is time to check the CVS head (v1.1dev) version of Asterisk now, we are heading towards code freeze and production of a new stable release. We do need help testing all new features, finding bugs, reporting them, fixing them. The new realtime architecture is a major improvement and a good platform for a lot of new future technology in Asterisk. We need it tested and proven before we release version 1.2. Thank you for your support in creating a new version of Asterisk -the Open Source PBX! Awesome, I hope the new jitter buffer makes it in too.. -- Martijn van Oosterhout ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2: Connection rejected
On Wed, Feb 16, 2005 at 04:10:17PM -0500, Sergey Kuznetsov wrote: They are the same. That's what I've checked first. Have you restarted Asterisk? Not all changes picked up with a reload, sometimes you have unload/reload the module or do a full restart for all changes to take effect... Hope this helps, Peter Bowyer wrote: On Wed, 16 Feb 2005 15:40:19 -0500, Sergey Kuznetsov [EMAIL PROTECTED] wrote: Hi there, I am having a problem. It looks like this: Feb 16 15:01:10 WARNING[11122]: chan_iax2.c:5546 socket_read: Call rejected by XXX.XXX.XXX.XXX: No authority found -- Martijn van Oosterhout ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone having trouble with VoicePulse Connect?
On Thu, Feb 17, 2005 at 06:32:52PM -0800, beonice wrote: To answer my own question, at least partially, here is a quote from the Asterisk Configuration chapter in Paul Mahler's book VoIP Telephony With Asterisk: Table 1. Reserved Extension Names -- Character NameUsage - - -- s Start A call that does not have digits associated with it, for example a loopstart analog line, begins at the s extension Interesting. I don't understand it fully, but I'm sure I will if I stare at it long enough. :) I guess it implies that calls coming from DIDs have digits associated with them. Correct. On ISDN lines, E1, T1 and related digital protocols, details such as CallerID, Dialled Number, CLI Presentation, etc are passed as part of the call setup, before there is any discussion of ringing. So Asterisk can go straight into the part of the script that matches. However, on an analog line, you start with ringing and you still know nothing about the call. CallerID comes later and Dialled number is generally never sent at all. So you always start in s. Hope this helps, -- Martijn van Oosterhout ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] problem : undefined symbol.
On Thu, Feb 17, 2005 at 01:05:30PM +0200, Michael Manousos wrote: Did you try asterisk-oh323? http://www.inaccessnetworks.com/projects/asterisk-oh323 Is there any particular reason to prefer oh323 over the builtin h323? I can't find any feature comparison and I can't have both since they require completely different versions of OpenH323. Thanks in advance, -- Martijn van Oosterhout ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call termination database
On Thu, Feb 17, 2005 at 09:07:46PM +, Alistair Cunningham wrote: Gonzalo, Yes, pricing would be included, as would minimum call volumes. Providers could choose not to disclose these, but then they'd be shown at the bottom of the page. A feedback system is a good idea; I'll think about how to do it. I was thinking about letting people provide a quality rating, maybe 1-5 on call quality. This would allow people to compare price/quality and aim for where they feel comfortable. But I think it's an awesome idea... -- Martijn van Oosterhout ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems compiling pridump utility
-lzap means it's looking more libzap, presumably the zaptel library. Have you got it somewhere where the makefile will find it? Hope this helps, On Thu, Feb 17, 2005 at 05:54:42PM -0500, Arlen Raasch wrote: I do 'make pridump' from the libpri source directory and receive the following: # make pridump cc -o pridump pridump.o -L. -lpri -lzap -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g /usr/bin/ld: cannot find -lzap collect2: ld returned 1 exit status make: *** [pridump] Error 1 I am new to all of this, so I am sure I am missing something obvious, any help will be appreciated. I am using libpri verision 1.0.4 with Fedora Core version 2.6.5-1.358. Note: Asterisk and the kernel modules compiled fine, I would just like to try out this utility. Thanks, -Arlen Raasch -- Martijn van Oosterhout ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk-H323
On Thu, Feb 17, 2005 at 10:47:28PM -0800, kolo sos wrote: is there any version mismatch or path needed to have a succesful build? i got an error when i done MAKE to the asterisk-oh323. Obviously people have successfully built it, people here use it all the time. Perhaps you can post the actual error you're getting... -- Martijn van Oosterhout ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Codec Issue on IAX trunk? (Solved)
On Sat, Feb 12, 2005 at 10:44:11AM -0600, Rich Adamson wrote: I haven't tried to keep track of the code changes involving reloads (or cli restarts for that matter), but having been around * for a fair amount of time and having been caught with making changes that have had no affect, I'll usually play it very safe and simply stop / start asterisk for many different changes. Iax and sip def's in particular. Reloads are fine for lots of things, but experience is the only way to know what's acceptable at this point. I've noticed this myself. However, I have been able to acheive a similar effect by unloading and then reloading the module. In my case I was testing H323, it might be trickier if you're actually using what you're playing with... Hope this helps, -- Martijn van Oosterhout ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] i want to load chan_h323.so
On Fri, Feb 11, 2005 at 06:09:06PM +0900, ?? wrote: If you actually sent text instead of an HTML only email, you increase the chance someone will actually read your message... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400 Problem
On Mon, Feb 07, 2005 at 11:16:05PM -0600, Eric Rees wrote: Has anyone seen this message trying to install an TDM400.. spurious 8259A interrupt: IRQ7 Not sure what to has to do with your system, but I read somewhere that it is related to how the original interrupt controllers worked. If a card signalled an interrupt but then withdrew it before the host processor got around to reading the interrupt register, it would register as IRQ7. The kernel here is just pointing out that it got an IRQ7 but wasn't expecting it and it has now disabled it. If a module wants it it needs to register it. It's a harmless message... I'm fairly sure that IO-APIC only systems don't have this problem. -- Martijn van Oosterhout ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] choppy sound after 15 minutes in a call
On Tue, Feb 01, 2005 at 10:37:36PM +0100, Anders F Eriksson wrote: In my CLI I get NOTICE[32322]: RTP Transmission error to 85.xxx.xxx.xxx:35162: Operation not permitted. I get it on calls to the PSTN through my X100P (clone) as well as call connected through my IP telephony provider. I have also tried SJ-Phone and it happens with that as well. I don't know if you're using linux or not or exactly how the code is structured, but... if send() or write() to a network socket returns -EPERM it's generally that your firewall blocked it. Hope this helps, -- Martijn van Oosterhout Ecomtel Pty Ltd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Digium and Intel Chipset compatability
Hi, I'm going to be setting up some machines with 4 port E1 cards from Digium and I'm being told that TE410 is incompatable with several Intel chipsets including the ones in a lot of Dell server systems. Is this true? I can't find any confirmed details on the mailing list about it. Also, the email seems to imply that the TE405P will be fine, though it doesn't say that explicitly. Basically, is anyone using a 4 port E1 card successfully on an Intel® E7221 Chipset or similar? Thanks in advance, -- Martijn van Oosterhout Ecomtel Pty Ltd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: R: R: R: [Asterisk-Users] How to force G729
On Thu, Jun 24, 2004 at 09:52:45AM -0700, Chris A. Icide wrote: It sounds like what you are looking for is an Asterisk-wide (or perhaps channel-specific) preserve_codec option. Where preserve_codec=1 means that asterisk tries to preserve the originating codec if at all possible, and preserve_codec=0 lets asterisk freely choose any codec per whatever algorithm it chooses. As far as I know, this option doesn't exist, but depending upon the need, perhaps someone should issue a feature request. It seems like this might be an easy feature to add. I wonder, if you remove all the codec converters, or set their cost very high, would that help? Given that the cost of no conversion is by definition zero, it's a bit odd that asterisk is changing them at all unless absolutly required. There must be something else going on... -- Martijn van Oosterhout IT Manager Ecomtel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Calling the firefly network?
Is there a way to register with or call the firefly network from an Asterisk server. It would be pretty cool if you could gateway calls onto it. Have a nice day, -- Martijn van Oosterhout ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Compiling problem on Debian
On Thu, Jun 17, 2004 at 11:30:11PM +0200, Robin Calmeg?rd Siurua wrote: Hi, I can't compile Asterisk on a Debian machine. I couldn't get asterisk to compile with the default openh323 and libpt packages in debian so I went and grabbed the original source for: pwlib v1.5.2 openh323 v1.12.2 asterisk v1.0 And compiled them all from stratch. Make sure you don't have any old packages lying around to confuse things. Version skew is a problem with the openh323 stuff, if you're going to upgrade any of them to CVS versions you'll probably need to get CVS versions of all of them. Also, I had an issue with openh323 where it didn't use the right c++ compiler, but that's just my setup. Hope this helps, -- Martijn van Oosterhout IT Manager Ecomtel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk/netmeeting works, asterisk/ohphone doesn't?
I've been banging my head on this one for a few days and am quite stuck. I've got a gatekeeper running and everything works there. Netmeeting works calling other netmeeting clients. Netmeeting calling asterisk connects, but netmeeting can't generate the signals to make the demo do anything other than talk. But connection from ohphone always disconnects straight away. I can't seem to enable any kind tracing on the asterisk end: Asterisk Server: openh323 v1.12.2 pwlib v1.5.2 asterisk v1.0 asterisk-oh323 v0.5.10 inAccess Networks OpenH323 Wrapper OhPhone: (attempt one) openh323 v1.12.2 pwlib v1.5.2 ohphone v1.4.1 OhPhone 1.4.1 doesn't compile with CVS pwlib and openh323 so I pulled the latest CVS of all three and got the same result. This OhPhone doesn't connect properly with netmeeting either. The calls connects fine but all I hear from the ohphone end is misdecoded voice data, basically it plays back at twice the speed and then goes silent, alternating every few hundred milliseconds or so. Some kind of mismatch somewhere. So my question, has anyone had ohphone work for them and if so, what versions did they use at each end. Are any known compiler issues (g++ 3.0.4, gcc 2.95.4)? Are there any clients (other than ohphone) which one can use with asterisk to test it out? Thanks in advance, -- Martijn van Oosterhout IT Manager Ecomtel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk/netmeeting works, asterisk/ohphone doesn't?
On Wed, Jun 16, 2004 at 08:34:25PM +0300, Michael Manousos wrote: Martin List-Petersen wrote: The first channel driver i tried was asterisk-oh323, which gave me very poor results (asterisk core dumping, if i got a connect i had echo + 10 secs delay on a Lan connection, stuff like that). As the maintainer of asterisk-oh323, I would like to add that this is not true anymore. A lot of bugs have been eliminated and the latest release is functioning very well. Ok, I had no core-dumping problems with either, it just didn't connect. I've reverted to h323 and it works now but that's completely unrelated (see below). One major thing I've noticed so far is that the oh323 driver registers a number of aliases with the gatekeeper and the h323 driver just registers the username. I'm sure this is configurable, I just couldn't see it at quick glance. The actual problem was a combination of things: - OhPhone not complaining about the fact that it couldn't open the audio device - The fact that the soundcard didn't support full-duplex after all - Even after all that, OhPhone couldn't find the plugins and so wasn't negotiating a common codec. - Asterisk not noticing or complaining that no sound channels were configured for the connection. The end result being that asterisk was happily playing the demo, but not actually transmitting anything and OhPhone wasn't sending anything even though it could see the mixer settings. Now everything works. Sorry for the noise people. -- Martijn van Oosterhout IT Manager Ecomtel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
