Re: [Asterisk-Users] iax2 config sanity check
Brian May wrote:
Hello,
Based on my reading and understanding of the documentation, in
extensions.conf all I need is:
exten = _5XXX,1,Dial(IAX2/ivt/${EXTEN})
As asterisk will look up the rest of the configuration in iax.conf:
--- cut ---
[ivt]
username=microcomaustralia
type=friend
host=dynamic
context=default
host=202.91.207.49
permit=0.0.0.0/0.0.0.0
auth=rsa
inkeys=ivt
outkey=microcomaustralia
--- cut ---
However this doesn't work - I get no packets whatsoever getting sent to
202.91.207.49. In fact no packets I have observed look related in
anyway.
Asterisk displays:
-- Starting simple switch on 'Zap/1-1'
-- Executing Dial(Zap/1-1, IAX2/ivt/5999) in new stack
-- Called ivt/5999
[ pause until I hang up ]
-- Hungup 'IAX2/ivt/1'
== Spawn extension (international, 5999, 1) exited
non-zero on 'Zap/1-1'
It seems that I have to include the full IP address and key in the Dial
instruction. Then it works.
From memory if I wait long enough it will timeout, but the timeout error
doesn't help track the problem down.
What am I doing wrong?
The username and the peer name aren't the same thing. There is some
ambiguity floating around as to just how the syntax parses out fully.
Use the username, microcomaustralia (ugh. that name is too long) in
front of the peer name, e.g. IAX2/[EMAIL PROTECTED]/5999 and see how
that works out. Assuming 5999 is the extension you want to reach at the
other end.
Any help before I pull all my hair out would be much appreciated.
You shouldn't be pulling your hair out even if nobody answers your
emails. Just play around with the different parts of the dialstring and
watch the CLI. It's fun.
B.
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Re: [Asterisk-Users] Playtone on answering the phone
Quoting Matt Riddell [EMAIL PROTECTED]: Is there a way of converting the play tone to a gsm file which can be played using the A option? Obelix wrote: Is it possible to get Asterisk to issue a Playtones when an outgoing call is answered? The examples indicate what happens when an incoming call is answered. It would have to be done by the remote machine. Unless you want to play a sound to callee once connected: Some Dial options: 'A(x)' -- play an announcement to the called party, using x as file 'D([called][:calling])' -- Send DTMF strings *after* called party has answered, but before the call gets bridged. The 'called' DTMF string is sent to the called party, and the 'calling' DTMF string is sent to the calling party. Both parameters can be used alone. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Intel Desktop MotherBoards Unsuitable for Di gium Boards
Colin Anderson wrote: Onboard LAN with an un-movable IRQ would mess that up good Only if you had just one pci slot. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Intel Desktop MotherBoards Unsuitable for Digium Boards
Nir Simionovich - CTO wrote: Over the course of the past 3 years, I've used the following boards with high success rates: Intel, Tyan, GigaByte, HP/Compaq and some IBM machines. I also integrated on some Asus and SuperMicro, but I wouldn't call those a tier-1 installation, as they were mainly done in a lab condition for a lab test. We're running supermicro rack mounts (1U) in high load production environments and have been nothing but impressed. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] maximum concurrent conference peers in asterisk
nr k wrote: hi generally we describe the bandwidth in kilobits per second only. Cool, just checking, it seemed pretty low. According to http://www.voip-info.org/wiki-Asterisk+bandwidth+iax2 you should be able to do 4 calls with g729. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] strange tone is droping calls
Anton Krall wrote: I'll need to run a test, Ill remove busydetect from zapata.conf tomorrow and see if the line drops after a hangup, if so, then we should be set, if not, then we are in trouble? Unless you can change the PBX's cadences to not be the same as your disconnect cadences... -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] MeetMe invite another user
If I use meetme conference room, can I invite another user during a conversation? In which way? Matteo === Matteo Piazza, Junior Researcher CREATE-NET Via Solteri, 38 - 38100 Trento - Italy email: [EMAIL PROTECTED] Tel: +39-0461-408400ext:308 www.create-net.it === ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] MeetMe invite another user
Matteo Piazza wrote: If I use meetme conference room, can I invite another user during a conversation? In which way? Use a .call file (search for sample.call in the asterisk source directory for an example). You can then copy the file into /var/spool/asterisk/outgoing to make the call. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Playtone on answering the phone
Obelix wrote: Quoting Matt Riddell [EMAIL PROTECTED]: Is there a way of converting the play tone to a gsm file which can be played using the A option? Sure, if you send me the dtmf tones you need and I'll mail you some gsm files. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Asterisk Fax support using T.38
Hi, Just wondering whether anyone has done fax relaying or pass-through using Asterisk T.38 Please let me know your thoughts as I need to come up with a fax server using Asterisk with T.38 possible? Cheers! Lilantha ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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: SNOM360 Monitoring Extension States
Peter Dean wrote: I have now been successful in getting the notification lights working. Then asterisk extensions hint required a reference to the extension being monitored and the extension monitoring the call status. i.e. _226,hint,SIP/226SIP/101 So with this change the asterisk hint registary now looks like this; -= Registered Asterisk Dial Plan Hints =- _226: SIP/226SIP/101 State:IdleWatchers 1 where the state went from unknown to Idle. But it appears the SNOM360 is not able to pickup the call when you press the flashing light on the extension that is monitoring the call status - despite the SNOM360 indicating that is connected, whilst the phone that is being called continues to ring. That is not supported yet. There is a patch in the issue tracker that does this, but it's a proof-of-concept code. It will burden your asterisk quite a lot if you put it to use in larger production sites. /O ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] maximum concurrent conference peers in asterisk
On Wed, 2005-11-09 at 21:32 +1300, Matt Riddell wrote: nr k wrote: hi generally we describe the bandwidth in kilobits per second only. Cool, just checking, it seemed pretty low. According to http://www.voip-info.org/wiki-Asterisk+bandwidth+iax2 you should be able to do 4 calls with g729. Bandwidth is a tricky issue. You have your IP + UDP + RTP + whatever headers (iax2 combines stuff so potentially that skews this a bit) but something most often forgotten is link layer framing. Take ATM (DSL uses ATM as do many other links). ATM transmitted data is chopped up into 53 byte cells. Each cell has a 5 byte header. This leaves 48 bytes for payload per ATM cell. Lets say your total packet IP header on down is 80 bytes. This means that on the ATM layer you have 2 ATM cells with 16 bytes of padding. This is really only 83% efficient network wise. This is per RTP packet. By adjusting your sample size you can try to fill the cell completly so you dont waste extra bandwidth on padding (ATM cells can contain no more than 1 packet and they are padded to fill the cell. so every 48 bytes of payload is another cell). You dont want your sample size too small however because that causes more IP overhead, too large and it can degrade call quality (imagine a 30ms jitter buffer with 30ms sample sizes, that means only 1 packet goes in the jitter buffer, with only one packet you have the effect of no buffer at all, reordering packets is impossible, delayed packets cant be normalized timewise, etc). Its a really fine balance and something you should consider if you really want to tune your VoIP to your network. Obviously once its handed off to another network it becomes hard to create packets tuned for a network you dont control, but its fair to assume that the majority of backbone providers are doing ATM so by tweaking this you may find that your voice traffic works better over the net at large too ... YMMV I didnt pay attention to what type of link the 64Kbps links were (I dont think it was specified initially) so I dont know what framing is used, but this is something to consider. By not paying attention to this fine detail you can waste a lot of bandwidth then wonder why you start to have lossy performance when raw bandwidth meters suggest you shouldnt have any loss. This was something that I presented to the Sacramento Asterisk Users Group last friday, although my power point presentation doesnt give the subject the coverage it needs, most of that was audible. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] SNOM 360 Unknown SIP command 'PUBLISH'
[EMAIL PROTECTED] wrote: Hi List I’m getting this notification from my one and only SNOM 360 every time a number button is pushed. I know that it’s only a notification, but it really irritates me. Is it anything I can/should do anything about ?? Not really. We do not support PUBLISH in Asterisk 1.0.9. We don't support it in 1.2 either, but we tell the phone so properly. /O ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] MeetMe invite another user
On Wed, 2005-11-09 at 21:59 +1300, Matt Riddell wrote: Matteo Piazza wrote: If I use meetme conference room, can I invite another user during a conversation? In which way? Use a .call file (search for sample.call in the asterisk source directory for an example). You can then copy the file into /var/spool/asterisk/outgoing to make the call. For anyone doing this it may not always be /var/spool/asterisk/outgoing, especially with non linux installs. check your asterisk.conf file for astspooldir. It should be in that directiory/outgoing :) -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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-1.2-bêta2 | pre sence/subscription support in the SIP channel driver
Does asterisk support RFC3265 ? Harry --- Matt Riddell [EMAIL PROTECTED] a écrit : harry gaillac wrote: nobody has an answer here! Actually someone asked for you config details. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] PRI pass-through
Hi, I want to build a PRI pass-through with a Cisco 2600, with two VWIC E1 cards, is this possible ? and do i need any other modules except for the E1 modules ? What i want to do is connect the asterisk to the PRI through the Cisco router, and let my legacy PBX utilize some of the PRI channels while testing Asterisk, Anyone with experience, sample configs or idea, please contribute. Thanks. Marco. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] maximum concurrent conference peers in asterisk
On Wed, 2005-11-09 at 01:16 -0800, trixter aka Bret McDanel wrote: Bandwidth is a tricky issue. You have your IP + UDP + RTP + whatever headers (iax2 combines stuff so potentially that skews this a bit) but something most often forgotten is link layer framing. I should have added to this that for home users using DSL with PPPoE (which seems increasingly popular) the framing issues can be compounded because PPPoE has 6 bytes of headers and PPP has 2 bytes, this is added to the total frame sent over a DSL + PPPoE link. But if you are running max connections on VoIP over DSL you are either doing special purpose stuff or a very bad provider and have other issues. DSL, especially with PPPoE is not a 'carrier class' link :) Addiitonally ATM has a 8 byte SAR trailer, and I forgot to subtract that from the 16 bytes of padding so that is really only 8 bytes of padding (not nearly as bad). To clarify, if you have a straight ATM link, then you have 5 bytes per cell padding and 8 bytes at the end of all the cells that make the packet. So 40 bytes then add in increments of 48 bytes of usable data, to avoid waste. A little isnt that big of a deal (you shouldnt be trying to run your links at 100% anyway), but if you transmit 41 bytes of packet data for example you have 47 bytes of padding, which puts you about 50% efficiency, which is really bad. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] MeetMe invite another user
trixter aka Bret McDanel wrote: On Wed, 2005-11-09 at 21:59 +1300, Matt Riddell wrote: Matteo Piazza wrote: If I use meetme conference room, can I invite another user during a conversation? In which way? Use a .call file (search for sample.call in the asterisk source directory for an example). You can then copy the file into /var/spool/asterisk/outgoing to make the call. For anyone doing this it may not always be /var/spool/asterisk/outgoing, especially with non linux installs. check your asterisk.conf file for astspooldir. It should be in that directiory/outgoing :) Or alternatively in that directory/outgoing if we're nitpicking! :D -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] listening on multiple port #'s
You can set up two *. One that will only interact with your VoIP provider and another that will be POST gateway and will run on 5060. Connect them with IAX2. -- Tomislav Parcina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)393447 e-mail: tparcina#lama.hr http://www.lama.hr -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of PC Sent: 3. studeni 2005 1:12 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] listening on multiple port #'s Can two instances of asterisk coexist peacefully on a single box sharing the same ztdummy driver, amongst other things? The sip and iax channels would be completely separate. I'm also open to any other ideas that might work. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 Consultant
The only pointer I got is a $50/hr Mark phillip offered. I can make VOIP calls between my Asterisk server and my VOIP provider using sip channel without a problem. But when I attempt to make a call using IAX, the call get accepted and then get a hangup message: This is the message I get when I attempt to make an IAX call: Executing Dial(OSS/dsp, IAX2/callshopcompany/0017046872001) in new stack -- Called callshopcompany/0017046872001 -- Call accepted by 213.61.187.150 (format gsm) -- Format for call is gsm -- Hungup 'IAX2/callshopcompany/1' == No one is available to answeer at this time The call get accepted, but it seems there is no acknowledgement from my server to receive the call from the provider. Thanks; --- Mark Phillips [EMAIL PROTECTED] wrote: He did. And he got pointers to the relevant howto's. Matt Riddell wrote: chawki hammoud wrote: Hi: I posted my problem several times about being unable to make IAX calls from my Asterisk box to another IAX server without luck. So, what's your problem? Post some details. -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 __ Yahoo! FareChase: Search multiple travel sites in one click. http://farechase.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Music-on-Hold problem
Alex, thanks so much, that was it - I don't know how I missed it. I guess I was looking for more complicated reasons :-). Cheers, Alex -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Alexander O. Lopez Sent: Tuesday, November 08, 2005 7:54 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Music-on-Hold problem Have you tried adding an answer before playing MOH??? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex Epshteyn Sent: Tuesday, November 08, 2005 11:10 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Music-on-Hold problem Hi, We are experiencing a strange problem playing music-on-hold - or perhaps it is a problem with the configuration of a Zap channel. When a call comes in from PSTN (FXO card) and MusicOnHold application is executed, the music on hold starts (Asterisk reports that the moh has started - and you can see that the mpg123 process is running) but the caller continues hearing ringing and no moh. Also, and possibly related, Zap channel stays in Offhook state after the caller hangs up. We tried a variety of options to make Asterisk detect hangup (busydetect, callprogress, etc) with no success. Could it be a hardware problem? Does anyone know of any bugs/issues/configuration errors that are likely to cause this? It appears that somehow the music being played is not delivered to the channel (could it be device configuration?). We are running Red Hat with 2.6 kernel, with udev configured as specified in README.udev, mpg123-0.59r. I apologize for not describing the whole environment, software versions, etc - I am not sure what info would be relevant. Help would be very much appreciated. Thanks, Alex ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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_message_support.patch
Hello Matt, In fact I look for messaging an presence between sip phones . http://www.voip-forum.com/news.php?p=184c=1 I use polycom ip phone with presence (rfc3265) and IM (SIMPLE). Do you you think the job of Joshua Colp could help me to use presence/IM with asterisk ? Regards Harry http://www.voip-forum.com/news.php?p=184c=1 --- Matt Riddell [EMAIL PROTECTED] a écrit : harry gaillac wrote: Hello, Does sip_message_support.patch is available for asterisk-1.2-bêta2 ? Is there an other solution for Sip message ? pabx*CLI show agi send text Usage: SEND TEXT text to send Sends the given text on a channel. Most channels do not support the transmission of text. Returns 0 if text is sent, or if the channel does not support text transmission. Returns -1 only on error/hangup. Text consisting of greater than one word should be placed in quotes since the command only accepts a single argument. pabx*CLI show agi receive text Usage: RECEIVE TEXT timeout Receives a string of text on a channel. Specify timeout to be the maximum time to wait for input in milliseconds, or 0 for infinite. Most channels do not support the reception of text. Returns -1 for failure or 1 for success, and the string in parentheses. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] groupware + unified messagerie +Asterisk
Somebody would be interested in a such project ? Harry --- Kristof Hardy [EMAIL PROTECTED] a écrit : harry gaillac wrote: Is it possible to add a frontend groupware with All is possible, you're only limited by your imagination. (always wanted to say this :p) I'm not sure there's a(n Open-source) project like this already. Cheers.. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] A2Billing
Hi all, I am having an issue with individual access vs simultaneous access. If I set a card for individual access, make a call with that card the counter goes to 1. If the call complets normally shouldnt the counter reset to 0? Second call tells me that card is already in use. simultaneous access will only allow the counter to go to 40. Suggestions please John Fraser ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] queue_log and mysql support
Hello List, I'm glad to announce that we have released the first version of QueueMetrics that supports MySQL storage of queue_log data. It is still experimental, so if you run such a setup and would like to give it a try, you are welcome. The MySQL adapter should adapt to any existing table format, so you don't have to convert your existing data. See http://queuemetrics.loway.it/news.jsp QueueMetrics is free for personal / SOHO usage. Yours, l. -- Loway Research - Home of QueueMetrics http://queuemetrics.loway.it ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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_message_support.patch
harry gaillac wrote: Hello Matt, In fact I look for messaging an presence between sip phones . http://www.voip-forum.com/news.php?p=184c=1 Should work with current CVS HEAD version. I use polycom ip phone with presence (rfc3265) and IM (SIMPLE). Do you you think the job of Joshua Colp could help me to use presence/IM with asterisk ? Should also do :) -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 Consultant
chawki hammoud wrote: The only pointer I got is a $50/hr Mark phillip offered. Put notransfer=yes in the iax.conf section for that account. Then try adding trunk=yes or trunk=no (try both), and if you use trunk=yes make sure there is a (t) by the peer in iax2 show peers. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] groupware + unified messagerie +Asterisk
harry gaillac wrote: Somebody would be interested in a such project ? I think quite a few people do this kind of thing in house - it's kinda one of those personal preferences things. Does anyone want to make one that fits everyone's setup? I don't know. But I don't have enough time at the moment to do one. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] bysy tone when dialing out via SPA-3000 in the netherlands????
Hi,
I want to use an SPA-3000 connect my dutch kpn PSTN line to and from the
Asterisk VOIP network.
Dialing in (via my kpn pstn line) is functioning oke, with a great sound
quality.
Dialing out to the pstn line produces after a verry short ring a busy signal.
If I connect the pstn to my Internal phonecentrale (kpn homevox) I will get a
distorted dial signal (bothe when dialing directly (8) out or via a complete
number). With that signal I can dial my homevox extensions.
Doe anyone know how to solve this problem??? I guess the SPA-3000 is not
interpreting the phone signals correctly or there is some timing problem?
Any help greatly appreciated.
extensions.conf
[pstn]
ignorepat = 8
exten = _8.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60,)
sip.conf
[pstn-spa3k]
type=peer
auth=md5
host=10.1.1.41
port=5061
secret=geheim
username=asterisk
fromuser=asterisk
dtmfmode=rfc2833
context=from-sip
insecure=very
;reinvite=no
;canreinvite=no
Output from the Asterisk console.
-- Executing Dial(SIP/1001-5748, SIP/[EMAIL PROTECTED]|60|) in new stack
-- Called [EMAIL PROTECTED]
-- SIP/pstn-spa3k-c2e5 is ringing
-- SIP/pstn-spa3k-c2e5 answered SIP/1001-5748
-- Attempting native bridge of SIP/1001-5748 and SIP/pstn-spa3k-c2e5
== Spawn extension (from-sip, 8, 1) exited non-zero on 'SIP/1001-5748'
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Re: [Asterisk-Users] Asterisk Consultant
i can fix that please contact me off list, i have setup now that same as yours and i encountered that problem. On 11/9/05, chawki hammoud [EMAIL PROTECTED] wrote: The only pointer I got is a $50/hr Mark phillip offered. I can make VOIP calls between my Asterisk server and my VOIP provider using sip channel without a problem. But when I attempt to make a call using IAX, the call get accepted and then get a hangup message: This is the message I get when I attempt to make an IAX call: Executing Dial(OSS/dsp, IAX2/callshopcompany/0017046872001) in new stack -- Called callshopcompany/0017046872001 -- Call accepted by 213.61.187.150 (format gsm) -- Format for call is gsm -- Hungup 'IAX2/callshopcompany/1' == No one is available to answeer at this time The call get accepted, but it seems there is no acknowledgement from my server to receive the call from the provider. Thanks; --- Mark Phillips [EMAIL PROTECTED] wrote: He did. And he got pointers to the relevant howto's. Matt Riddell wrote: chawki hammoud wrote: Hi: I posted my problem several times about being unable to make IAX calls from my Asterisk box to another IAX server without luck. So, what's your problem? Post some details. -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 __ Yahoo! FareChase: Search multiple travel sites in one click. http://farechase.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 -- Best Regards, Angelito Manansala www.voicefidelity.net Mobile: +639175425807 DID: (+63) 44 7906770 msn: [EMAIL PROTECTED] skype: bulcrack ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] extension and overlap
Hi all; Ihave on my zapata.conf overlap=yes. In my extension i have: exten = _X.,1,Dial(SIP/[EMAIL PROTECTED]) I want to let user more than 5s to dial, i want to let him 3s by digits. Can you help me!!! Thanks //vador ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] MeetMe invite another user
On Wed, 2005-11-09 at 23:00 +1300, Matt Riddell wrote: trixter aka Bret McDanel wrote: For anyone doing this it may not always be /var/spool/asterisk/outgoing, especially with non linux installs. check your asterisk.conf file for astspooldir. It should be in that directiory/outgoing :) Or alternatively in that directory/outgoing if we're nitpicking! :D Just symlink it that way its both :) -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] ast_streamfile failed
Hi all, weird problem, this seems to happen without any rhyme nor reason yesterday from /var/log/asterisk/full Nov 8 18:07:02 VERBOSE[3270]: -- Executing BackGround(IAX2/[EMAIL PROTECTED]/4, crim/main-menu) in new stack Nov 8 18:07:02 VERBOSE[3270]: -- Playing 'crim/main-menu' (language 'en') today after a reboot Nov 9 10:37:29 VERBOSE[2083]: -- Executing BackGround(SIP/2004-6bca, crim/main-menu.mp3) in new stack Nov 9 10:37:29 WARNING[2083]: File crim/main-menu.mp3 does not exist in any format Nov 9 10:37:29 WARNING[2083]: Unable to open crim/main-menu.mp3 (format ulaw): Permission denied Nov 9 10:37:29 WARNING[2083]: ast_streamfile failed on SIP/2004-6bca for crim/main-menu.mp3 Output of # file main-menu.mp3 main-menu.mp3: MP3, 192 kBits, 44.1 kHz, Mono Any Ideas? Bails ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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_message_support.patch
the asterisk's answer ! // Connected to Asterisk 1.2.0-beta2 currently running on serveur1 (pid = 2729) Remote UNIX connection Verbosity is at least 3 Nov 9 11:48:21 WARNING[2926]: chan_sip.c:7251 receive_message: Received message to sip:[EMAIL PROTECTED] from bob sip:[EMAIL PROTECTED];tag=F71E8D2E-67C04697, dropped it... Content-Type:text/plain Message: Call me. serveur1*CLI // a part of my extension.conf file exten = 84,hint,Sip/84 exten = 84,1,Answer exten = 84,2,SendText() exten = 84,3,Dial(Sip/84,10) exten = 84,4,VoiceMail(u84) exten = 84,103,VoiceMail(b84) exten = 85,hint,Sip/85 exten = 85,1,Answer exten = 85,2,SendText() exten = 85,3,Dial(Sip/85,10) exten = 85,4,VoiceMail(u85) exten = 85,103,VoiceMail(b85) exten = 86,hint,Sip/86 exten = 86,1,Answer exten = 86,2,SendText() exten = 86,3,Dial(Sip/86,10) exten = 86,4,VoiceMail(u86) exten = 86,103,VoiceMail(b86) exten = 87,hint,Sip/87 exten = 87,1,Answer exten = 87,2,SendText() exten = 87,3,Dial(Sip/87,10) exten = 87,4,VoiceMail(u87) exten = 87,103,VoiceMail(b87) / neither SUBSCRIBE, NOTIFY, MESSAGE sip method are ok :( Harry --- Matt Riddell [EMAIL PROTECTED] a écrit : harry gaillac wrote: Hello Matt, In fact I look for messaging an presence between sip phones . http://www.voip-forum.com/news.php?p=184c=1 Should work with current CVS HEAD version. I use polycom ip phone with presence (rfc3265) and IM (SIMPLE). Do you you think the job of Joshua Colp could help me to use presence/IM with asterisk ? Should also do :) -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] groupware + unified messagerie +Asterisk
On Wed, 2005-11-09 at 23:39 +1300, Matt Riddell wrote: harry gaillac wrote: Somebody would be interested in a such project ? I think quite a few people do this kind of thing in house - it's kinda one of those personal preferences things. Does anyone want to make one that fits everyone's setup? I don't know. But I don't have enough time at the moment to do one. Unified Messaging was really hot 5+ years ago, now I dont see the advertisements for service providers and such that I once saw, but then I also dont work for a UM company anymore and I did at that time so maybe that has something to do with it. Many millions in VC were flying around at one time, of course that was late 90s early 2000 and any crackpot could get VC money (and many did :) Doing it totally free is going to be the trick because the tools that are available arent quite what they could be. So you would either have to invest a lot of time yourself into making this all work well or buy some commercial products. TTS is one of the areas that free isnt always better, although rumors abound that you can tweak festival to be much better than its default settings, it may be worth it to invest a few hundred into some other program so you dont have to play as much. Time vs Money issue. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] ast_streamfile failed
bails napisał(a): Nov 9 10:37:29 WARNING[2083]: Unable to open crim/main-menu.mp3 (format ulaw): Permission denied ^ Any Ideas? Maybe this is a problem with permisions to this file? -- Best regards, Bartosz Piec ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] dell and digium hardware
Hi! I read in the archive a lot of problems using the Dell 1850 servers and digium TE4xxP and TE2xxP hardware. I wonder if ever anybody has tried the Dell Poweredge 850 series and can report some experiences? btw: Does somebody knows why there are problems with 1850 but not with 2850 (digium recommends the 2850 for their Business Edition)? AFAIK both have the same chipset and both use Intel onboard NICs. Thank's for any hints. Regards Klaus ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Stopping Asterisk from forwarding calls?
You don't have this problem when using AstBill to manage Asterisk. We are doing call forwarding from the database to single or multiple extensions. As the Dial command is managed from the MySQL database we ignore voicemail forwarding when ringing multiple extensions.Are Casillahttp://astartelecom.com - Independent VOIP Telecoms Broker. Asterisk Consultants http://astbill.com - Open Source Billing, Routing and Management software for Asterisk and VOIPAstBill DEMO: http://demo.astbill.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 Consultant
Sounds like a good deal to me. If you want free answers don't sound so irritated that you haven't got a reply in $0 time. :)RobOn 11/9/05, chawki hammoud [EMAIL PROTECTED] wrote:The only pointer I got is a $50/hr Mark phillip offered.I can make VOIP calls between my Asterisk server andmyVOIP provider using sip channel without a problem. Butwhen I attempt to make a call using IAX, the call getaccepted and then get a hangup message: This is the message I get when I attempt to make anIAX call: Executing Dial(OSS/dsp,IAX2/callshopcompany/0017046872001) in new stack-- Called callshopcompany/0017046872001 -- Call accepted by 213.61.187.150 (format gsm)-- Format for call is gsm-- Hungup 'IAX2/callshopcompany/1'== No one is available to answeer at this time The call get accepted, but it seems there is noacknowledgement from my server to receive the callfrom the provider.Thanks;--- Mark Phillips [EMAIL PROTECTED] wrote: He did. And he got pointers to the relevant howto's. Matt Riddell wrote: chawki hammoud wrote: Hi: I posted my problem several times about being unable to make IAX calls from my Asterisk box to another IAX server without luck.So, what's your problem? Post some details. -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 __Yahoo! FareChase: Search multiple travel sites in one click.http://farechase.yahoo.com___ --Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] dell and digium hardware
The very problem is that DELL in the small one, block the IRQ. And this can make conflict to the cards. Bruno. Klaus Darilion wrote: Hi! I read in the archive a lot of problems using the Dell 1850 servers and digium TE4xxP and TE2xxP hardware. I wonder if ever anybody has tried the Dell Poweredge 850 series and can report some experiences? btw: Does somebody knows why there are problems with 1850 but not with 2850 (digium recommends the 2850 for their Business Edition)? AFAIK both have the same chipset and both use Intel onboard NICs. Thank's for any hints. Regards Klaus ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 -- BRUNO DE LUCA Tel. +39 02 9350 4780 (102) FGA Software 20017 Rho - Via Puccini, 8 E-Mail : [EMAIL PROTECTED] Internet: http://www.fgasoftware.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] codecs
Hi all, We use asterisk as a local pbx and we connect to a pstn/sip provider for calls to pstn. Since the messages on asterisk are on gsm format, we need gsm, but to call pstn, we need g729 or g723. How can we enable both codecs to be able to call pstn and hearing voicemail messages for example? Any idea is welcome. Olivier ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] codecs
i think gsm you mention is gsm sound files not gsm codecs. On 11/9/05, Olivier Taylor [EMAIL PROTECTED] wrote: Hi all, We use asterisk as a local pbx and we connect to a pstn/sip provider for calls to pstn. Since the messages on asterisk are on gsm format, we need gsm, but to call pstn, we need g729 or g723. How can we enable both codecs to be able to call pstn and hearing voicemail messages for example? Any idea is welcome. Olivier ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 -- Best Regards, Angelito Manansala www.voicefidelity.net Mobile: +639175425807 DID: (+63) 44 7906770 msn: [EMAIL PROTECTED] skype: bulcrack ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] codecs
Right, I must suppose I need gsm codec to hear gsm files, I miss? olivier -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Angelito Manansala Envoyé : mercredi 9 novembre 2005 12:28 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [Asterisk-Users] codecs i think gsm you mention is gsm sound files not gsm codecs. On 11/9/05, Olivier Taylor [EMAIL PROTECTED] wrote: Hi all, We use asterisk as a local pbx and we connect to a pstn/sip provider for calls to pstn. Since the messages on asterisk are on gsm format, we need gsm, but to call pstn, we need g729 or g723. How can we enable both codecs to be able to call pstn and hearing voicemail messages for example? Any idea is welcome. Olivier ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 -- Best Regards, Angelito Manansala www.voicefidelity.net Mobile: +639175425807 DID: (+63) 44 7906770 msn: [EMAIL PROTECTED] skype: bulcrack ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] A2Billing
John Fraser a écrit : Hi all, I am having an issue with individual access vs simultaneous access. If I set a card for individual access, make a call with that card the counter goes to 1. If the call complets normally shouldnt the counter reset to 0? Second call tells me that card is already in use. If you're using CVS/1.2, A2Billing is broken and don't recognize hangup. It's ok with 1.0 branch. Other solution is not to hangup and let A2Billing do the stuff ;-) [...] -- Daniel ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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: RE : [Asterisk-Users] codecs
You simply need to have g729/g723 codecs. Asterisk comes with gsm by default. Regards, Sahil Gupta VoiceValley On Wed, 9 Nov 2005, Olivier Taylor wrote: Right, I must suppose I need gsm codec to hear gsm files, I miss? olivier -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Angelito Manansala Envoyé : mercredi 9 novembre 2005 12:28 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [Asterisk-Users] codecs i think gsm you mention is gsm sound files not gsm codecs. On 11/9/05, Olivier Taylor [EMAIL PROTECTED] wrote: Hi all, We use asterisk as a local pbx and we connect to a pstn/sip provider for calls to pstn. Since the messages on asterisk are on gsm format, we need gsm, but to call pstn, we need g729 or g723. How can we enable both codecs to be able to call pstn and hearing voicemail messages for example? Any idea is welcome. Olivier ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 -- Best Regards, Angelito Manansala www.voicefidelity.net Mobile: +639175425807 DID: (+63) 44 7906770 msn: [EMAIL PROTECTED] skype: bulcrack ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] MP3 or OGG
On Wed, Nov 09, 2005 at 11:43:34AM +1100, Mark Edwards wrote: Hey Waldo. AFAIK there is quite a lot of scope for tuning the compression of speex - just as there is for mp3. I have no doubt that if you tune complexity, quality and bitrate parameters you will be able to get that filesize down even further. Can't see any reason at all why you shouldn't be able to whack mp3 for filesize. One thing off the top of my head: streams from Asterisk as 8mhz. Is there any use in recording in a higher bit-rate? (any anti-aliasing-like effect?) -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] MP3 or OGG
User-agents have g729, g723.1 and gsm, isn't it possible to force user-agent to use gsm for voicemail and g729 for outbound calls? Olivier -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Tzafrir Cohen Envoyé : mercredi 9 novembre 2005 12:35 À : asterisk-users@lists.digium.com Objet : Re: [Asterisk-Users] MP3 or OGG On Wed, Nov 09, 2005 at 11:43:34AM +1100, Mark Edwards wrote: Hey Waldo. AFAIK there is quite a lot of scope for tuning the compression of speex - just as there is for mp3. I have no doubt that if you tune complexity, quality and bitrate parameters you will be able to get that filesize down even further. Can't see any reason at all why you shouldn't be able to whack mp3 for filesize. One thing off the top of my head: streams from Asterisk as 8mhz. Is there any use in recording in a higher bit-rate? (any anti-aliasing-like effect?) -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] New asterisk web gui for small/medium sizedbusinesses
I can't open your on-line demo. (9. 11. 2005. at 12:49 GMT+2) Tomislav From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of snacktimeSent: 3. studeni 2005 2:22To: Asterisk Users Mailing List - Non-Commercial Discussion; Commercial and Business-Oriented Asterisk DiscussionSubject: [Asterisk-Users] New asterisk web gui for small/medium sizedbusinesses I posted last week that I would get out a release asap, so here it is. Before I start in on putting up an actual website for it I thought I would put out a beta release to get things going.At this point there isn't a name for this project yet, as it's primarily an internal piece of software that we have been developing. For now I'll call it Asterisk web gui (AWG).AWG has a particular focus, which is to provide an easy to use interface for managing and monitoring asterisk, as well as a nice web interface for voicemail users as well. We are trying to make it as easy to install as possible. It plays nice with existing asterisk installations, and it won't overwrite any of your existing asterisk configuration. If you already have ruby installed on your system the setup time should be around 15 minutes. Once you have done one or two installations and know what the steps are, installation on a new system should average 5-10 minutes at the most. We have tested it on Freebsd and Debian. It should work on windows also.AWG is not intended to help you install asterisk and do your basic configuration. There are other software packages that do everything from start to finish such as AMP. AWG is dictatorial software. We will not include features by consensus unless they also fit our vision of what a tool like this should include. That said we want all the feedback we can get, particularly from businesses who are looking at it as a solution they might deploy for their clients. Just realize that it has a particular focus, and that's not going to change. There are also a few features not currently present that are on our todo list to get done asap. A basic interface for viewing CDR records, zap channel configuration, and a page to monitor real time information such as channels, peers, queues, etc.. There is a basic online demo at http://69.25.136.214:3000. The administrative login is user admin, password 'changeme'. The user login is username demo, password 'changeme'. At the moment there are not a lot of script templates installed, but the ones that are there will give you an idea of what you can do with the provisioning features. At the moment the demo is running in development mode where the errors are verbose and the code is reloaded for every page, so it's not as fast as you would see in a production environment.The setup guide and distribution file are at http://catalog1.paymentonline.com/~chrisYou can send any feedback to my directly at [EMAIL PROTECTED] or on the list.The whole thing is licensed under the BSD license. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 : RE : [Asterisk-Users] codecs
User-agents have g729, g723.1 and gsm, isn't it possible to force user-agent to use gsm for voicemail and g729 for outbound calls? Olivier -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Sahil Gupta Envoyé : mercredi 9 novembre 2005 12:33 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: RE : [Asterisk-Users] codecs You simply need to have g729/g723 codecs. Asterisk comes with gsm by default. Regards, Sahil Gupta VoiceValley On Wed, 9 Nov 2005, Olivier Taylor wrote: Right, I must suppose I need gsm codec to hear gsm files, I miss? olivier -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Angelito Manansala Envoyé : mercredi 9 novembre 2005 12:28 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [Asterisk-Users] codecs i think gsm you mention is gsm sound files not gsm codecs. On 11/9/05, Olivier Taylor [EMAIL PROTECTED] wrote: Hi all, We use asterisk as a local pbx and we connect to a pstn/sip provider for calls to pstn. Since the messages on asterisk are on gsm format, we need gsm, but to call pstn, we need g729 or g723. How can we enable both codecs to be able to call pstn and hearing voicemail messages for example? Any idea is welcome. Olivier ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 -- Best Regards, Angelito Manansala www.voicefidelity.net Mobile: +639175425807 DID: (+63) 44 7906770 msn: [EMAIL PROTECTED] skype: bulcrack ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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-1.2-bêta2 | presen ce/subscription support in the SIP channel driver
Some parts of it, yes. On 11/9/05, harry gaillac [EMAIL PROTECTED] wrote: Does asterisk support RFC3265 ? Harry --- Matt Riddell [EMAIL PROTECTED] a écrit : harry gaillac wrote: nobody has an answer here! Actually someone asked for you config details. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Re: libbluetooth
Thanks Colin and Mark for your answers. I finally
manage to start up asterisk bycopying libbluetooth to
/usr/lib/.
Now the final step comes, make this bluetooth
thing works.
These are my configuration
files:
/etc/bluetooth/rfcomm.conf:
rfcomm0
{ bind yes;
device
00:0E:6D:34:BD:B1;
channel
3;
comment
"Bluetooth device";}
/etc/asterisk/bluetooth.conf:
[general]rfchannel_ag = 3interface =
0
;; A Nokia
6310i[00:0E:6D:34:BD:B1]name =
Nokiatype =
AGchannel = 3autoconnect = yes
/etc/asterisk/extensions.conf:
[general]static=yeswriteprotect=yesautofallthrough=yes
[globals]DEFAULT_RINGING_TIME=20
[default]exten =
_01,1,Dial(SIP/${EXTEN})exten = _02,1,Dial(SIP/${EXTEN})exten =
_03,1,Dial(BLT/test/XXX)
(where XXX is the mobile phone number)
And the result starting up asterisk:
[chan_bluetooth.so] = (Bluetooth Channel Driver) ==
Parsing '/etc/asterisk/bluetooth.conf': FoundJan 9 11:44:14
NOTICE[3401]: /usr/src/chan_bluetooth/chan_bluetooth.c:1825 rfcomm_listen:
Listening for RFCOMM channel 3 connections on FD 11Jan 9 11:44:14
NOTICE[3401]: /usr/src/chan_bluetooth/chan_bluetooth.c:1825 rfcomm_listen:
Listening for RFCOMM channel 6 connections on FD 12Jan 9 11:44:14
ERROR[3401]: /usr/src/chan_bluetooth/chan_bluetooth.c:1841 sco_listen: Can't
create SCO socket: No such file or directory (errno: 2)Jan 9 11:44:14
WARNING[3401]: loader.c:345 ast_load_resource: chan_bluetooth.so: load_module
failed, returning -1 == Unregistered channel type 'BLT'Jan 9
11:44:14 WARNING[3401]: loader.c:440 load_modules: Loading module
chan_bluetooth.so failed!
More:
- There is a bug reported regarding SCO links on Theo'ssite
(http://www.crazygreek.co.uk/content/chan_bluetooth)
- rfcomm, l2cap,hci_usb and bluez are sucesfully loaded into the
system.
- If I start up asterisk without bluetooth.conf, I don't have bluetooth
command from the console.
-Theo doesn't mention rfcomm in his documentation. Does rfcomm.conf
file have to be configured and the module loaded?
Does anybody know anything about this bugor justhow to
sort this out?
Thank you very much,
Victor.
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[Asterisk-Users] SIP/H.323 suggestion
HI all, Is Asterisk able to work as SIP and H.323 Gatekeeper same time? If it has the capability to work which i should open? Yours suggestion will be high appriciated. Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 Tel: +974 - 4883068 ICQ: 276994704 YM!: abdul_zu Fax: +974 - 4883063 Doha Qatar http://www.hatif.com __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] ast_streamfile failed
Bartosz Piec wrote: bails napisał(a): Nov 9 10:37:29 WARNING[2083]: Unable to open crim/main-menu.mp3 (format ulaw): Permission denied ^ Any Ideas? Maybe this is a problem with permisions to this file? It was indeed a permissions problem /var/lib/asterisk/sounds/crim/ was a symlink to /home/user/audio/ I moved the files to a new folder /var/lib/asterisk/sounds/new/ and all is well, I'm just trying to work out why it worked yesterday and stopped this morning. This is not logical Thanks Bails ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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-1.2-bêta2 | presen ce/subscription support in the SIP channel dri ver
I'm not a developper ! What do you mean Some parts of it, yes. harry --- BJ Weschke [EMAIL PROTECTED] a écrit : Some parts of it, yes. On 11/9/05, harry gaillac [EMAIL PROTECTED] wrote: Does asterisk support RFC3265 ? Harry --- Matt Riddell [EMAIL PROTECTED] a écrit : harry gaillac wrote: nobody has an answer here! Actually someone asked for you config details. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Zaptel: chan_zap.c:6514 mkintf: Unable to open channel 1 : Operation not supported by device
I get this when I boot asterisk. I have a Wildcard TDM400P REV H with 1 FXO board on it. [chan_zap.so] = (Zapata Telephony w/PRI) == Parsing '/usr/local/etc/asterisk/zapata.conf': Found Nov 8 22:11:51 WARNING[24698]: chan_zap.c:935 zt_open: Unable to specify channel 1: Operation not supported by device Nov 8 22:11:51 ERROR[24698]: chan_zap.c:6514 mkintf: Unable to open channel 1: Operation not supported by device here = 0, tmp-channel = 1, channel = 1 Nov 8 22:11:51 ERROR[24698]: chan_zap.c:10344 setup_zap: Unable to register channel '1' Nov 8 22:11:51 WARNING[24698]: loader.c:345 ast_load_resource: chan_zap.so: load_module failed, returning -1 my ztcfg all looks good though... Keyword: [loadzone], Value: [uk] Keyword: [defaultzone], Value: [uk] Keyword: [fxoks], Value: [1] Zaptel Configuration == Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) 1 channels configured. Any ideas? ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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-1.2-bêta2 | presence/ subscription support in the SIP channel driver
Salut Harry, Tu quittes Ser pour asterisk? Olivier -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de harry gaillac Envoyé : mercredi 9 novembre 2005 13:22 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [Asterisk-Users] asterisk-1.2-bêta2 | presence/subscription support in the SIP channel driver I'm not a developper ! What do you mean Some parts of it, yes. harry --- BJ Weschke [EMAIL PROTECTED] a écrit : Some parts of it, yes. On 11/9/05, harry gaillac [EMAIL PROTECTED] wrote: Does asterisk support RFC3265 ? Harry --- Matt Riddell [EMAIL PROTECTED] a écrit : harry gaillac wrote: nobody has an answer here! Actually someone asked for you config details. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 to setup Agent dialing in multiple asterisk servers
not so complicated.. use IAX trunking to share dialplans On 11/9/05, KRTorio [EMAIL PROTECTED] wrote: Our Setup: In our company we run multiple asterisk servers, and agents login (using AgentCallbackLogin) to any of these. One person, one agent number ID. The Problem: Dialing an agent number from within one pbx is easy, but if you want to dial an agent logged in another pbx, its more complicated. Our current dialplan performs guessing which pbx the agent is logged, by dialing all of them. We have to redesign this everytime we add another pbx, and we're looking for a more efficient method of doing this. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 -- Best Regards, Angelito Manansala www.voicefidelity.net Mobile: +639175425807 DID: (+63) 44 7906770 msn: [EMAIL PROTECTED] skype: bulcrack ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] groupware + unified messagerie +Asterisk
Hi Harry We are doing some of it with AstBill and love to work with you to include your requirements. We have Click to Dial from address book, SMS and Fax is not released but working with clients. We want to intergrate AstBill with a Groupeware or CRM but want input what people will prefeer. On our list today we have http://www.sugarcrm.com/crm/ http://www.vtiger.com/ http://www.egroupware.org/ So what is the best CRM/Groupeware to intergreate with AstBill? -- Are Casilla http://astartelecom.com - Independent VOIP Telecoms Broker. Asterisk Consultants http://astbill.com - Open Source Billing, Routing and Management software for Asterisk and VOIP AstBill DEMO: http://demo.astbill.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Music on hold
Hi all, I have some problems with music on hold. It works with the default category but not with additional ones. When I start Asterisk, 2 mpg123 processes are started with the default moh but none with additional ones. Does someone already had this problem and could help me? Thanks, Amaury ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] dell and digium hardware
Works well. I am running 1.0.9 stable on this with FC2 on kernel 2.6.9 The kernel needs patching to pick up the onboard SATA (ICH7), or we use a pci express SATA raid controller with a TE110p. The only real hassle is the single 'standard' pci slot in it. Remote access is via SOL and the embedded third nic. Very nice little server, even cheaper than the equivalent poweredge 750 as we no longer have to buy a drac card. Craig - Original Message - From: Klaus Darilion [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, November 09, 2005 7:06 PM Subject: [Asterisk-Users] dell and digium hardware Hi! I read in the archive a lot of problems using the Dell 1850 servers and digium TE4xxP and TE2xxP hardware. I wonder if ever anybody has tried the Dell Poweredge 850 series and can report some experiences? btw: Does somebody knows why there are problems with 1850 but not with 2850 (digium recommends the 2850 for their Business Edition)? AFAIK both have the same chipset and both use Intel onboard NICs. Thank's for any hints. Regards Klaus ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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: SNOM360 Monitoring Extension States
On Wed, 9 Nov 2005, Olle E. Johansson wrote: That is not supported yet. There is a patch in the issue tracker that does this, but it's a proof-of-concept code. It will burden your asterisk quite a lot if you put it to use in larger production sites. Which issue are you refering to? -- -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- - - - Jason Pyeron PD Inc. http://www.pdinc.us - - Partner Sr. Manager 7 West 24th Street #100 - - +1 (443) 269-1555 Baltimore, Maryland 21218 - - - -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- This message is for the designated recipient only and may contain privileged, proprietary, or otherwise private information. If you have received it in error, purge the message from your system and notify the sender immediately. Any other use of the email by you is prohibited. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Sipura 2000
Hi, Thanks for your response. I checked the setting, and indeed it was set to yes. However, once I change it to no and click on apply but after rebooting it's enabled again (with all settings reverted to factory defaults, as usual). Maxi. 2005/11/8, Rusty Dekema [EMAIL PROTECTED]: It's possible that your SPA-2000 is set up to read a configuration file from a remote host every time it boots up, which would overwrite any changes you make. If you log in as admin and go to the advanced view, there is an option under the Provisioning tab called Provision Enable. Make sure that this is set to no and your changes should remain in place. -Rusty On 11/8/05, Maximiliano J. Goldsmid [EMAIL PROTECTED] wrote: Hello, I have a problem with my Sipura 2000. The problem is that it does not accept any change in the configuration. When I access to it, via browser or phone, and make any change, after clicking submit all changes all the changes I made dissapear and teh configuration remains with the original parameters. So I need to know how can I work it out. Thank you very much. Maxi ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Extension Ring on Multiple Phones
Nonetheless .. Thanks everyone for the responses! I think I have it now! You guys are great! David A. Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodata.net Tel: (519) 951-6079 Fax: (519) 451-6615 Poor planning on your part does not necessarily constitute an emergency on my part! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jonathan k. Creasy Sent: Tuesday, November 08, 2005 4:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Extension Ring on Multiple Phones I guess I should have read up further before I posted a response. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Mojo with Horan Company, LLC Sent: Tuesday, November 08, 2005 2:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Extension Ring on Multiple Phones Like instead of exten = s,1,Dial(SIP/110,20,tr) you must mean exten = s,1,Dial(SIP/110SIP/112,20,tr) ? Just append all extensions you wish to ring, separated by ampersands (). The first one to answer will be winner. That's what I think you're asking, at least. Moj Dave Morrow wrote: Hi all. I wonder if anyone out there has a dial-plan which will ring an extension on multiple phones. David A. Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] _http://www.autodata.net_ Tel: (519) 951-6079 Fax: (519) 451-6615 Poor planning on your part does not necessarily constitute an emergency on my part! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at_ [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]_ ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 -- Mojo [EMAIL PROTECTED] Office Manger, Horan Company, LLC (907) 747- x112 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Sipura 2000
If you unplug the ethernet cable on a Sipura SPA and then reset the power it'll boot up in a diagnostic mode. When you pick up the phone that's connected to it you'll get a dialtone and there are speical codes you can dial to do various things. Reset it to factory defaults by dialing followed by 73738# full instructions are here: http://www.sipura.com/Documents/faq/Section_3.html#4 Once you do that the provisioning enable should be no and you can reconfigure the device however it needs to be. Hi, Thanks for your response. I checked the setting, and indeed it was set to yes. However, once I change it to no and click on apply but after rebooting it's enabled again (with all settings reverted to factory defaults, as usual). Maxi. 2005/11/8, Rusty Dekema [EMAIL PROTECTED]: It's possible that your SPA-2000 is set up to read a configuration file from a remote host every time it boots up, which would overwrite any changes you make. If you log in as admin and go to the advanced view, there is an option under the Provisioning tab called Provision Enable. Make sure that this is set to no and your changes should remain in place. -Rusty On 11/8/05, Maximiliano J. Goldsmid [EMAIL PROTECTED] wrote: Hello, I have a problem with my Sipura 2000. The problem is that it does not accept any change in the configuration. When I access to it, via browser or phone, and make any change, after clicking submit all changes all the changes I made dissapear and teh configuration remains with the original parameters. So I need to know how can I work it out. Thank you very much. Maxi ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] dell and digium hardware
On 11/9/05, Craig Guy [EMAIL PROTECTED] wrote: Works well.I am running 1.0.9 stable on this with FC2 on kernel 2.6.9Thekernel needs patching to pick up the onboard SATA (ICH7), or we use a pci express SATA raid controller with a TE110p. Which pci-e SATA controller are you using? The one that shipped with my dell was pci-x -Brian ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Options for 3-way or Conference Calling
Yes. I believe the Cisco phones do conferencing in the same fashion. I'm not 100% on whether or not the SPA-841 or the new SPA-941 does it. The SPA-941 does conferencing and it works exactly like the transfer. a soft button you hit twice, Conf once to dial the invited 3rd party and once to Join ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Intel Desktop MotherBoards Unsuitable for Digium Boards
On Tuesday 08 November 2005 18:20, George Pajari wrote: To make a long story short, according to Intel Dealer Technical Support (we became Intel dealers in order to get answers to our questions) there is no Intel motherboard that permits the IRQs to be configured uniquely. They are all hardwired and shared. This information applies to both the Intel Desktop Board and Server Board product lines. I find this almost impossible to believe. In XT-PIC mode, absolutely. However every modern chipset utilizes an IOAPIC now and every device has its own IRQ line. When the IOAPIC is in emulation (XT-PIC) mode, then yes many of the interrupts get merged into the standard 16 interrupts. However, if your Linux kernel is utilizing the IOAPIC's native mode things change drastically: # cat /proc/interrupts CPU0 0: 942314955IO-APIC-edge timer 1: 10IO-APIC-edge i8042 8: 1IO-APIC-edge rtc 9: 0 IO-APIC-level acpi 12:111IO-APIC-edge i8042 14: 496236IO-APIC-edge ide0 177: 211098355 IO-APIC-level eth0 185: 2 IO-APIC-level ehci_hcd:usb1 193: 0 IO-APIC-level ohci_hcd:usb2 201: 0 IO-APIC-level ohci_hcd:usb3 209: 86 IO-APIC-level ohci_hcd:usb4 217: 3769265646 IO-APIC-level wct4xxp As you can see on this particular system (not an Intel reference board, granted, but my Intel boards do work similarly) everything is on its own interrupt, and the interrupt numbers don't stop at 15. I'd really like some clarification on that... Do Intel reference boards actually tie the physical INT# signals of peripherals together, or are they just stating that unless you use the native IO-APIC mode you will have shared interrupts due to the emulation? Hopefully someone from Digium will step in and give the official word, because I have it on good authority that Digium hardware on Intel motherboards work well together. Hell, I've had my old P4 Intel reference board (with RamBus memory) work just fine without shared interrupts. -A. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Intel Desktop MotherBoards Unsuitable for Di gium Boards
On Wednesday 09 November 2005 03:29, Matt Riddell wrote: Colin Anderson wrote: Onboard LAN with an un-movable IRQ would mess that up good Only if you had just one pci slot. With 1U systems that is often all you get. -A. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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-1.2-bêta2 | presen ce/subscription support in the SIP channel driver
You don't need to be a developer to understand my statement. The current chan_sip does support some of the behaviors and methods described in RFC3265 to support the presence functionality that is currently part of Asterisk and chan_sip. Does this help? On 11/9/05, harry gaillac [EMAIL PROTECTED] wrote: I'm not a developper ! What do you mean Some parts of it, yes. harry --- BJ Weschke [EMAIL PROTECTED] a écrit : Some parts of it, yes. On 11/9/05, harry gaillac [EMAIL PROTECTED] wrote: Does asterisk support RFC3265 ? Harry --- Matt Riddell [EMAIL PROTECTED] a écrit : harry gaillac wrote: nobody has an answer here! Actually someone asked for you config details. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Sipura 2000
I followed your steps to the letter but after resetting to factory defaults unfortunately it still doesn't record the configuration changes I do. 2005/11/9, Adam Moffett [EMAIL PROTECTED]: If you unplug the ethernet cable on a Sipura SPA and then reset the power it'll boot up in a diagnostic mode. When you pick up the phone that's connected to it you'll get a dialtone and there are speical codes you can dial to do various things. Reset it to factory defaults by dialing followed by 73738# full instructions are here: http://www.sipura.com/Documents/faq/Section_3.html#4 Once you do that the provisioning enable should be no and you can reconfigure the device however it needs to be. Hi, Thanks for your response. I checked the setting, and indeed it was set to yes. However, once I change it to no and click on apply but after rebooting it's enabled again (with all settings reverted to factory defaults, as usual). Maxi. 2005/11/8, Rusty Dekema [EMAIL PROTECTED]: It's possible that your SPA-2000 is set up to read a configuration file from a remote host every time it boots up, which would overwrite any changes you make. If you log in as admin and go to the advanced view, there is an option under the Provisioning tab called Provision Enable. Make sure that this is set to no and your changes should remain in place. -Rusty On 11/8/05, Maximiliano J. Goldsmid [EMAIL PROTECTED] wrote: Hello, I have a problem with my Sipura 2000. The problem is that it does not accept any change in the configuration. When I access to it, via browser or phone, and make any change, after clicking submit all changes all the changes I made dissapear and teh configuration remains with the original parameters. So I need to know how can I work it out. Thank you very much. Maxi ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] ast_streamfile failed
bails wrote: Hi all, weird problem, this seems to happen without any rhyme nor reason yesterday from /var/log/asterisk/full Nov 8 18:07:02 VERBOSE[3270]: -- Executing BackGround(IAX2/[EMAIL PROTECTED]/4, crim/main-menu) in new stack Nov 8 18:07:02 VERBOSE[3270]: -- Playing 'crim/main-menu' (language 'en') today after a reboot Nov 9 10:37:29 VERBOSE[2083]: -- Executing BackGround(SIP/2004-6bca, crim/main-menu.mp3) in new stack Nov 9 10:37:29 WARNING[2083]: File crim/main-menu.mp3 does not exist in any format Nov 9 10:37:29 WARNING[2083]: Unable to open crim/main-menu.mp3 (format ulaw): Permission denied Nov 9 10:37:29 WARNING[2083]: ast_streamfile failed on SIP/2004-6bca for crim/main-menu.mp3 Output of # file main-menu.mp3 main-menu.mp3: MP3, 192 kBits, 44.1 kHz, Mono 1) I don't believe Background or Playback support MP3 (CVS-HEAD/1.2 may support MP3 with these apps) 2) Never put the extension when using Background or Playback ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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: RE : RE : [Asterisk-Users] codecs
Olivier Taylor wrote: User-agents have g729, g723.1 and gsm, isn't it possible to force user-agent to use gsm for voicemail and g729 for outbound calls? No. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Avaya 4612 IP phones with Asterisk?
Thank you very much for trying it for me, Dave. I really appreciate it. Paul _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Rahn Sent: Tuesday, November 08, 2005 8:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Avaya 4612 IP phones with Asterisk? I have gotten 4620's to work ( convert to sip ) It works ok... at best. I have a 4612 at work I will try tomorrow. good luck with yours . Dave _ From: [EMAIL PROTECTED] on behalf of Paul Mahler Sent: Tue 11/8/2005 2:06 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Avaya 4612 IP phones with Asterisk? Has anyone been able to make these phones work with *? If you have, what does it take? Thanks! Paul Paul Mahler [EMAIL PROTECTED] www.signate.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 attachment: winmail.dat___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Intel Desktop MotherBoards Unsuitable for Digium Boards
Andrew Kohlsmith wrote: On Tuesday 08 November 2005 18:20, George Pajari wrote: To make a long story short, according to Intel Dealer Technical Support (we became Intel dealers in order to get answers to our questions) there is no Intel motherboard that permits the IRQs to be configured uniquely. They are all hardwired and shared. This information applies to both the Intel Desktop Board and Server Board product lines. I find this almost impossible to believe. In XT-PIC mode, absolutely. However every modern chipset utilizes an IOAPIC now and every device has its own IRQ line. When the IOAPIC is in emulation (XT-PIC) mode, then yes many of the interrupts get merged into the standard 16 interrupts. However, if your Linux kernel is utilizing the IOAPIC's native mode things change drastically: # cat /proc/interrupts CPU0 0: 942314955IO-APIC-edge timer 1: 10IO-APIC-edge i8042 8: 1IO-APIC-edge rtc 9: 0 IO-APIC-level acpi 12:111IO-APIC-edge i8042 14: 496236IO-APIC-edge ide0 177: 211098355 IO-APIC-level eth0 185: 2 IO-APIC-level ehci_hcd:usb1 193: 0 IO-APIC-level ohci_hcd:usb2 201: 0 IO-APIC-level ohci_hcd:usb3 209: 86 IO-APIC-level ohci_hcd:usb4 217: 3769265646 IO-APIC-level wct4xxp As you can see on this particular system (not an Intel reference board, granted, but my Intel boards do work similarly) everything is on its own interrupt, and the interrupt numbers don't stop at 15. I'd really like some clarification on that... Do Intel reference boards actually tie the physical INT# signals of peripherals together, or are they just stating that unless you use the native IO-APIC mode you will have shared interrupts due to the emulation? Here's a list form an Intel Server Board, Dual CPU support (only 1 CPU installed): CPU0 0: 120413974IO-APIC-edge timer 1: 2434IO-APIC-edge keyboard 2: 0 XT-PIC cascade 8: 1IO-APIC-edge rtc 14: 642794IO-APIC-edge ide0 15: 51IO-APIC-edge ide1 18: 1204255212 IO-APIC-level wctdm 19: 1198491079 IO-APIC-level t1xxp 21:3395482 IO-APIC-level eth0 22: 1198502476 IO-APIC-level wcte11xp ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Sipura 2000
Well then you have me stumped. There's a Sipura forum at voxilla.com here: http://voxilla.com/forum-viewforum-f-14.html Maybe someone there will know more about it. I followed your steps to the letter but after resetting to factory defaults unfortunately it still doesn't record the configuration changes I do. 2005/11/9, Adam Moffett [EMAIL PROTECTED]: If you unplug the ethernet cable on a Sipura SPA and then reset the power it'll boot up in a diagnostic mode. When you pick up the phone that's connected to it you'll get a dialtone and there are speical codes you can dial to do various things. Reset it to factory defaults by dialing followed by 73738# full instructions are here: http://www.sipura.com/Documents/faq/Section_3.html#4 Once you do that the provisioning enable should be no and you can reconfigure the device however it needs to be. Hi, Thanks for your response. I checked the setting, and indeed it was set to yes. However, once I change it to no and click on apply but after rebooting it's enabled again (with all settings reverted to factory defaults, as usual). Maxi. 2005/11/8, Rusty Dekema [EMAIL PROTECTED]: It's possible that your SPA-2000 is set up to read a configuration file from a remote host every time it boots up, which would overwrite any changes you make. If you log in as admin and go to the advanced view, there is an option under the Provisioning tab called Provision Enable. Make sure that this is set to no and your changes should remain in place. -Rusty On 11/8/05, Maximiliano J. Goldsmid [EMAIL PROTECTED] wrote: Hello, I have a problem with my Sipura 2000. The problem is that it does not accept any change in the configuration. When I access to it, via browser or phone, and make any change, after clicking submit all changes all the changes I made dissapear and teh configuration remains with the original parameters. So I need to know how can I work it out. Thank you very much. Maxi ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Asterisk 1.0.9 + TE210 + SpanDSP
Hi all,
I need help with
Asterisk. Recently I setup an Asterisk with TE210 and spandsp to check the fax
capabilities of Asterisk + Spandsp. The two E1 are connected back to back. I
created a script that opens a channel from the first E1 and calls a channel in
the second in order to send a fax file. The result is Nothing. The script reach
a point that txfax send the file but nothing further. Find attached the config
files and the script.
Thanks in advance
George
--- SCRIPT ---
#!/usr/bin/perl
use Asterisk::Manager;
$|++;
my $astman = new Asterisk::Manager;
$astman-user('admin');
$astman-secret('secret');
$astman-host('localhost');
$astman-connect || die $astman-error . \n;
$astman-setcallback('Hangup', \hangup_callback);
$astman-setcallback('DEFAULT', \default_callback);
print $astman-sendcommand( Action = 'Originate',
Callerid = SLOT1,
Channel = 'Zap/g1/getfax',
Exten = 'sendfax',
Context = 'Outgoing',
Priority = '1' );
$astman-eventloop;
$astman-disconnect;
sub hangup_callback {
printf(hangup callback\n);
}
sub default_callback {
my (%stuff) = @_;
foreach (keys %stuff) {
printf(%s: %s\n, $_, $stuff{$_});
}
printf(\n);
}
--- RESULT WHEN RUNNING THE SCRIPT ---
EventNewchannelChannelZap/3-1StateRsrvdCallerIDunknownUniqueid1131547210.4CallerID:
SLOT1
Event: Newcallerid
Uniqueid: 1131547210.4
Channel: Zap/3-1
CallerID: SLOT1
Event: Newcallerid
Uniqueid: 1131547210.4
Channel: Zap/3-1
CallerID: SLOT1
Event: Newstate
Uniqueid: 1131547210.4
Channel: Zap/3-1
State: Dialing
CallerID: unknown
Event: Newchannel
Uniqueid: 1131547210.5
Channel: Zap/34-1
State: Ring
Event: Newexten
Channel: Zap/34-1
Context: Incoming
Extension: getfax
Application: SetVar
Uniqueid: 1131547210.5
AppData: FAXFILE=/tmp/1131547210.5.tiff
Priority: 1
Event: Newexten
Channel: Zap/34-1
Context: Incoming
Extension: getfax
Uniqueid: 1131547210.5
Application: RxFAX
AppData: /tmp/1131547210.5.tiff
Priority: 2
CallerID: unknown
Event: Newstate
Channel: Zap/34-1
State: Up
Uniqueid: 1131547210.5
CallerID: SLOT1
Event: Newstate
Channel: Zap/3-1
State: Up
Uniqueid: 1131547210.4
Event: Newexten
Channel: Zap/3-1
Context: Outgoing
Extension: sendfax
Uniqueid: 1131547210.4
Application: SetVar
AppData: SENDFAX=/tmp/sendfax.tiff
Priority: 1
Event: Newexten
Channel: Zap/3-1
Context: Outgoing
Extension: sendfax
Uniqueid: 1131547210.4
Application: TxFAX
AppData: /tmp/sendfax.tiff|caller
Priority: 2
* Stays there
--- ZAPATA.CONF ---
[trunkgroups]
; define any trunk groups
[channels]
switchtype=euroisdn
;pridialplan=national
signalling=pri_cpe
usecallerid=yes
hidecallerid=no
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
echocancel=no
rxgain=0.0
txgain=0.0
;faxdetect=both
; Span 1
context=Outgoing
group=1
;signalling=pri_net
signalling=pri_cpe
channel = 1-15
channel = 17-31
; Span 2
context=Incoming
group=2
signalling=pri_net
;signalling=pri_cpe
channel = 32-46
channel = 48-62
--- ZAPTEL.CONF ---
#
# Zaptel Configuration File
#
# This file is parsed by the Zaptel Configurator, ztcfg
#
#
# First come the span definitions, in the format
# span=span num,timing,line build out (LBO),framing,coding[,yellow]
#
# The timing parameter determines the selection of primary, secondary, and
# so on sync sources. If this span should be considered a primary sync
# source, then give it a value of 1. For a secondary, use 2, and so on.
# To not use this as a sync source, just use 0
#
# The line build-out (or LBO) is an integer, from the following table:
# 0: 0 db (CSU) / 0-133 feet (DSX-1)
# 1: 133-266 feet (DSX-1)
# 2: 266-399 feet (DSX-1)
# 3: 399-533 feet (DSX-1)
# 4: 533-655 feet (DSX-1)
# 5: -7.5db (CSU)
# 6: -15db (CSU)
# 7: -22.5db (CSU)
#
# The framing is one of d4 or esf for T1 or cas or ccs for E1
#
# Note: d4 could be referred to as sf or superframe
#
# The coding is one of ami or b8zs for T1 or ami or hdb3 for E1
#
# E1's may have the additional keyword crc4 to enable CRC4 checking
#
# If the keyword yellow follows, yellow alarm is transmitted when no
# channels are open.
#
#span=1,0,0,esf,b8zs
#span=2,1,0,esf,b8zs
#span=3,0,0,ccs,hdb3,crc4
#
# Next come the dynamic span definitions, in the form:
# dynamic=driver,address,numchans,timing
#
# Where driver is the name of the driver (e.g. eth), address is the
# driver specific address (like a MAC for eth), numchans is the number
# of channels, and timing is a timing priority, like for a normal span.
# use 0 to not use this as a timing source, or prioritize them as
# primary, secondard, etc. Note that you MUST have a REAL zaptel device
# if you are not using external timing.
#
# dynamic=eth,eth0/00:02:b3:35:43:9c,24,0
#
# Next come the definitions for using the
[Asterisk-Users] TDM400 FXO Screech
A nasty screech. That's what callers here sometimes when they dial into my FXO port from the PSTN. But usually, it works OK. Is this common? ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] [Asterisk-User] Festival help
I have install Festival in Asterisk, but I don't listen to anything, but Asterisk show this message Parsing '/etc/asterisk/cdr_custom.conf': Found -- Executing Answer(SIP/101-35e3, ) in new stack -- Executing Festival(SIP/101-35e3, Hello asterisk user| how are you today?) in new stack == Parsing '/etc/asterisk/festival.conf': Found therefore don't show error. Why? Thanks ___ Yahoo! Mail: gratis 1GB per i messaggi e allegati da 10MB http://mail.yahoo.it ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Sending DTMF tones after answering on an IAX channel
Hi, I'm trying to send some DTMF dialtones (for an extension on the other end). My call is done from a Zap channel, to Asterisk, throught an IAX provider, to a PSTN line in some university. The phone number I am trying to reach is 555-555- exten 1234. What I did is an Exten = 201,1,Dial(IAX2/provider/55||D(1234)) Well, that doesnt work. The other end doesnt seem to accept the DTMF tones. Maybe Asterisk is sending them too quickly (how to I put a pause in between?) I also tried using pauses (Dial/IAX2/provider/55ww1234) But my outgoing provider tells me it's circuit busy) Any clue or me? Just calling 555-555- is fine, but I have to put in the extension manually. No good for my needs. Mick ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 FXO Screech
I just installed FXO module in an older TDM400 card in port 1 and had problems. Moved it to port 2 and everything is fine now. - Dustin - Bill Michaelson wrote: A nasty screech. That's what callers here sometimes when they dial into my FXO port from the PSTN. But usually, it works OK. Is this common? ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] problem with g729 and CME-Asterisk
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi folks, my topology is: CME (Cisco) -- [sip trunk] -- Asterisk -- [sip trunk] -- ISP Services I need to connect my phones registered on CME to ISP Services using g729 codec. Well, on cisco I set the codec preference with a voice class: voice class codec 1 codec preference 1 g729r8 codec preference 2 g711alaw codec preference 3 g722ulaw On asterisk (if this is a right example of pass-thru utilization), I download the codec from http://kvin.lv/pub/Linux/Asterisk/freebsd/ (my processor is a Sempron 2.2, then I download codec_g729-gcc-athlon-sse.so and codec_g729-gcc-debug.so files) and put it in my codec directory /usr/local/lib/asterisk/modules/. I remove the dummy codec first, then on sip.conf: disallow=all allow=g729 allow=alaw allow=ulaw The ISP sip services have support of g729. When I try to make a call from cisco phone to ISP, I see something on CME that seems codec g729 doesn't work: barahir#sh voice call summary PORT CODECVAD VTSP STATEVPM STATE == === == 2/0.1 - - - 2/0.2 - - - 2/1.1 - - - 2/1.2 - - - 50/0/1 .1 g711alaw n S_CONNECT EFXS_CONNECT 50/0/1 .2 - - - EFXS_ONHOOK 50/0/2 .1 - - - EFXS_INIT 50/0/2 .2 - - - EFXS_INIT 50/0/3 .1 - - - EFXS_ONHOOK 50/0/4 .1 - - - EFXS_ONHOOK 50/0/4 .2 - - - EFXS_ONHOOK Where is my mistake? Any advice will be appreciated Thanks for your support Regards Andrea -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (Darwin) iD8DBQFDchFRMakHrsrHP9wRAoElAKDxrAxtMOOyRLO6kWaG/hvLVwAj8QCfW/TO LkuPpXb7DVpjUkoi6uV1PNU= =qwXR -END PGP SIGNATURE- ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Intel Desktop MotherBoards Unsuitable for DigiumBoards
Hi George, I run an Intel D865GBF Desktop board with Digium's TDM400P with 4 FXOs just fine. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of George Pajari Sent: Tuesday, November 08, 2005 6:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Intel Desktop MotherBoards Unsuitable for DigiumBoards FYI: We're trying to standardise on a tier one motherboard for the Asterisk boxes we build for customers and thought we'd try to use a low-end Intel Desktop Board since even a low-end Celeron has more than enough horsepower to handle a typical 8x32 PBX. To make a long story short, according to Intel Dealer Technical Support (we became Intel dealers in order to get answers to our questions) there is no Intel motherboard that permits the IRQs to be configured uniquely. They are all hardwired and shared. This information applies to both the Intel Desktop Board and Server Board product lines. Please let me know if your experience differs from what I've been told by Intel. Otherwise, you've been warned -- Intel mobos appear to be unsuitable for use with Digium hardware. -- George Pajari, netVOICE communications604 484 VOIP (484 8647 x102) Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102) www.netvoice.ca www.ip-centrex.ca www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Intel Desktop MotherBoards Unsuitable for Digium Boards
On Wednesday 09 November 2005 09:34, Eric ManxPower Wieling wrote: 18: 1204255212 IO-APIC-level wctdm 19: 1198491079 IO-APIC-level t1xxp 22: 1198502476 IO-APIC-level wcte11xp Holy shit and you've got three Digium cards in there... all on their own IRQ. -A. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 with g729 and CME-Asterisk
Do a debug voip ccapi on the CME and look through it. It will have detailed codec negotiations, etc in it. -Greg On Wed, 2005-11-09 at 16:10 +0100, Andrea Riela wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi folks, my topology is: CME (Cisco) -- [sip trunk] -- Asterisk -- [sip trunk] -- ISP Services I need to connect my phones registered on CME to ISP Services using g729 codec. Well, on cisco I set the codec preference with a voice class: voice class codec 1 codec preference 1 g729r8 codec preference 2 g711alaw codec preference 3 g722ulaw On asterisk (if this is a right example of pass-thru utilization), I download the codec from http://kvin.lv/pub/Linux/Asterisk/freebsd/ (my processor is a Sempron 2.2, then I download codec_g729-gcc-athlon-sse.so and codec_g729-gcc-debug.so files) and put it in my codec directory /usr/local/lib/asterisk/modules/. I remove the dummy codec first, then on sip.conf: disallow=all allow=g729 allow=alaw allow=ulaw The ISP sip services have support of g729. When I try to make a call from cisco phone to ISP, I see something on CME that seems codec g729 doesn't work: barahir#sh voice call summary PORT CODECVAD VTSP STATEVPM STATE == === == 2/0.1 - - - 2/0.2 - - - 2/1.1 - - - 2/1.2 - - - 50/0/1 .1 g711alaw n S_CONNECT EFXS_CONNECT 50/0/1 .2 - - - EFXS_ONHOOK 50/0/2 .1 - - - EFXS_INIT 50/0/2 .2 - - - EFXS_INIT 50/0/3 .1 - - - EFXS_ONHOOK 50/0/4 .1 - - - EFXS_ONHOOK 50/0/4 .2 - - - EFXS_ONHOOK Where is my mistake? Any advice will be appreciated Thanks for your support Regards Andrea -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (Darwin) iD8DBQFDchFRMakHrsrHP9wRAoElAKDxrAxtMOOyRLO6kWaG/hvLVwAj8QCfW/TO LkuPpXb7DVpjUkoi6uV1PNU= =qwXR -END PGP SIGNATURE- ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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-User] Festival help
asterisk183 napisał(a): therefore don't show error. Test the Festival server console (festival --server). I had permision denied for localhost.localdomain. You must change it in festival.smd file (maybe the name is a bit different). -- Best regards, Bartosz Piec ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Intel Desktop MotherBoards Unsuitable for Digium Boards
Andrew Kohlsmith wrote: On Wednesday 09 November 2005 09:34, Eric ManxPower Wieling wrote: 18: 1204255212 IO-APIC-level wctdm 19: 1198491079 IO-APIC-level t1xxp 22: 1198502476 IO-APIC-level wcte11xp Holy shit and you've got three Digium cards in there... all on their own IRQ. APIC rocks my world. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] PRI pass-through
This will not work. The PRI uses a single D channel for signalling. It can only go to one PBX, either * or legacy. Yes the cisco can map DS0 between the E1, but I believe you need the VWIC-MFT series to do so (may be wrong on that) but that will definately break the PRI. Either run the PRI directly to the legacy as is and add another from legacy to *, or connect the PRI to * and add a second from * to legacy. Good Luck On Nov 9, 2005, at 3:53 AM, Marco Supino wrote: Hi, I want to build a PRI pass-through with a Cisco 2600, with two VWIC E1 cards, is this possible ? and do i need any other modules except for the E1 modules ? What i want to do is connect the asterisk to the PRI through the Cisco router, and let my legacy PBX utilize some of the PRI channels while testing Asterisk, Anyone with experience, sample configs or idea, please contribute. Thanks. Marco. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Intel Desktop MotherBoards Unsuitable for Digium Boards
On Wed, 2005-11-09 at 09:37 -0600, Eric ManxPower Wieling wrote: Andrew Kohlsmith wrote: On Wednesday 09 November 2005 09:34, Eric ManxPower Wieling wrote: 18: 1204255212 IO-APIC-level wctdm 19: 1198491079 IO-APIC-level t1xxp 22: 1198502476 IO-APIC-level wcte11xp Holy shit and you've got three Digium cards in there... all on their own IRQ. APIC rocks my world. Am I right in thinking you need a 2.6.x kernel for this? Pete ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Intel Desktop MotherBoards Unsuitable for Digium Boards
No. APIC was in 2.4 as well, but you need an Intel CPU in there (I think) in order to be able to take advantage of it. AMD's don't have this option available. On 11/9/05, Pete Barnwell [EMAIL PROTECTED] wrote: On Wed, 2005-11-09 at 09:37 -0600, Eric ManxPower Wieling wrote: Andrew Kohlsmith wrote: On Wednesday 09 November 2005 09:34, Eric ManxPower Wieling wrote: 18: 1204255212 IO-APIC-level wctdm 19: 1198491079 IO-APIC-level t1xxp 22: 1198502476 IO-APIC-level wcte11xp Holy shit and you've got three Digium cards in there... all on their own IRQ. APIC rocks my world. Am I right in thinking you need a 2.6.x kernel for this? Pete ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 with g729 and CME-Asterisk
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Nov 9, 2005, at 4:33 PM, Greg Oliver wrote: Do a debug voip ccapi on the CME and look through it. It will have detailed codec negotiations, etc in it. thanks for your answer, Greg. Could you help me? http://www.nesys.it/snap/debug_voice_ccapi.txt thanks for your support Regards Andrea -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (Darwin) iD8DBQFDch8XMakHrsrHP9wRAkO2AJ9W15cGdtnWF+oWl0Yd/ai7HTHs+wCg1oUD X8BxszRaAVFpPkQzd1w5jEg= =Jsnv -END PGP SIGNATURE- ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Intel Desktop MotherBoards Unsuitable for Digium Boards
Pete Barnwell wrote: On Wed, 2005-11-09 at 09:37 -0600, Eric ManxPower Wieling wrote: Andrew Kohlsmith wrote: On Wednesday 09 November 2005 09:34, Eric ManxPower Wieling wrote: 18: 1204255212 IO-APIC-level wctdm 19: 1198491079 IO-APIC-level t1xxp 22: 1198502476 IO-APIC-level wcte11xp Holy shit and you've got three Digium cards in there... all on their own IRQ. APIC rocks my world. Am I right in thinking you need a 2.6.x kernel for this? No. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] groupware + unified messagerie +Asterisk
harry gaillac a écrit :
Hello,
Is it possible to add a frontend groupware with
asterisk in order to Provide send receive fax to mail,
sms to mail, voice messages .
Asterisk or openpbx could be the server of the unified
messagerie .
click to dial contact in address book ,...
[Shameless plug]
Using MozPhone (available at http://moziax.mozdev.org/), adding click to
dial is just a matter of using tel: URLs (tel:123123) in web pages
(address book of your groupware).
And adding URL in your estension.conf dial string will pop up a web page:
exten = 105,1,Dial(${jdg},45,tr,
http://taina.sysnux.pf/crm?cid=${CALLERIDNUM})
Thanks,
--
Jean-Denis Girard
SysNux Systèmes Linux en Polynésie française
http://www.sysnux.pf/ Tél: +689 483 527 / GSM: +689 797 527
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Re: [Asterisk-Users] problem with g729 and CME-Asterisk
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I've forgotten my dial-peer config: dial-peer voice 500 voip description ext destination-pattern .T voice-class codec 1 session protocol sipv2 session target ipv4:192.168.17.10 dtmf-relay rtp-nte no vad 192.168.17.10 is *, .1 is CME. Regards Andrea -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (Darwin) iD8DBQFDciEJMakHrsrHP9wRArwvAJ9/lz+D1xVL8WnU3dyNLfpkh62nJwCgm8DD /9HE2UKACZ/OOJkZpC8c6Ss= =+5Iw -END PGP SIGNATURE- ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 with g729 and CME-Asterisk
Post up your dial-peer 500 config as well. It is doing codec 0x2 (g.711Alaw) from the get go. Also post relevant config for the phone from asterisk and dialplan entry used. -Greg On Wed, 2005-11-09 at 17:08 +0100, Andrea Riela wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Nov 9, 2005, at 4:33 PM, Greg Oliver wrote: Do a debug voip ccapi on the CME and look through it. It will have detailed codec negotiations, etc in it. thanks for your answer, Greg. Could you help me? http://www.nesys.it/snap/debug_voice_ccapi.txt thanks for your support Regards Andrea -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (Darwin) iD8DBQFDch8XMakHrsrHP9wRAkO2AJ9W15cGdtnWF+oWl0Yd/ai7HTHs+wCg1oUD X8BxszRaAVFpPkQzd1w5jEg= =Jsnv -END PGP SIGNATURE- ___ --Bandwidth and Colocation sponsoreby Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Call forward to cell phone and X100P
I am running an Asterisk server (which has gone from 1.x to 1.2b2 at the moment) that has 3 X100P cards and around 10 SIP phones in my office and I have a problem when I want to redirect my desk phone to my cell phone. I have a Polycom 600 phone on my desk (I have also tried this with Aastra and Grandstream phones). If I choose the forward option and enter my cell number, the next call will ring my cell but I will get no audio on my side most of the time. After the call ends both incoming and outgoing Zap interfaces will report that the call is still on and will continue that way until I manually destroy the channel. Why doesn't either interface detect the end of the call? The result is that I cannot forward my phone because 2 lines will be engaged untill manually reset. -- Carlos Chavez Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 with g729 and CME-Asterisk
Just put codec g729(whatever version you need) in your dialpeer. I do not see what the voice-class codec 1 is without that section. -Greg On Wed, 2005-11-09 at 17:17 +0100, Andrea Riela wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I've forgotten my dial-peer config: dial-peer voice 500 voip description ext destination-pattern .T voice-class codec 1 session protocol sipv2 session target ipv4:192.168.17.10 dtmf-relay rtp-nte no vad 192.168.17.10 is *, .1 is CME. Regards Andrea -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (Darwin) iD8DBQFDciEJMakHrsrHP9wRArwvAJ9/lz+D1xVL8WnU3dyNLfpkh62nJwCgm8DD /9HE2UKACZ/OOJkZpC8c6Ss= =+5Iw -END PGP SIGNATURE- ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] force to expire a sip registration
take for example a phantom SIP/400b from a previos phone config, without restarting * how can I purge only 400b? testserver*CLI sip show peers Name/username HostDyn Nat ACL Port Status 400c/400c (Unspecified)D 0Unmonitored 400b/400b 192.168.1.106D 2051 Unmonitored 400/400192.168.1.106D 2051 Unmonitored 302/302192.168.1.106D 2051 Unmonitored 301/301192.168.1.106D 2051 Unmonitored 333/333192.168.1.106D 2051 Unmonitored -- -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- - - - Jason Pyeron PD Inc. http://www.pdinc.us - - Partner Sr. Manager 7 West 24th Street #100 - - +1 (443) 269-1555 Baltimore, Maryland 21218 - - - -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- This message is for the designated recipient only and may contain privileged, proprietary, or otherwise private information. If you have received it in error, purge the message from your system and notify the sender immediately. Any other use of the email by you is prohibited. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Receptionist phones
Ive been playing with Asterisk for a few weeks and its working great. I have a question about getting multi-line receptionist phones working. I was thinking about getting one of these expansion ports: http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheet09186a008008883d.html What are people using for receptionist phones that show all the extensions in use, etc? Is that even possible with Asterisk right now? Anyone play around with this thing yet? Bill ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Codecs problem
That's a call to pstn Callee and caller have 9729 but asterisk (astlinux and soekris) tell me that there is no match and give me an error :( Any idea? Kind regards, Olivier 9 headers, 11 lines Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 82.146.123.246:38098 Found description format G729 Found description format telephone-event Capabilities: us - 0x70f (g723|gsm|ulaw|alaw|g729|speex|ilbc), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) Nov 9 16:46:13 NOTICE[402]: channel.c:1763 ast_set_read_format: Unable to find a path from g729 to gsm Nov 9 16:46:13 NOTICE[402]: channel.c:1730 ast_set_write_format: Unable to find a path from ilbc to g729 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Receptionist phones
Nevermind I found a note about Hint which can be used for this purpose. Bill From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bill Gibbs Sent: Wednesday, November 09, 2005 11:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Receptionist phones Ive been playing with Asterisk for a few weeks and its working great. I have a question about getting multi-line receptionist phones working. I was thinking about getting one of these expansion ports: http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheet09186a008008883d.html What are people using for receptionist phones that show all the extensions in use, etc? Is that even possible with Asterisk right now? Anyone play around with this thing yet? Bill ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
