[Asterisk-Users] realtime queue not working realtime in asterisk versions above 1.2.0
Heya, I upgraded from 1.2.0 to 1.2.4 now and I still have the same problem, a problem which didn't exist in 1.2.0: When a new call comes in on a realtime queue, the queue settings and members are not updated anymore! Only a reload of Asterisk seems to update the settings. Is this a bug or is there some way to solve this? Cheers, Frank ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: [asterisk-biz] iDEFISK (mac iax2 softphone) release
On 02/01/06 16:00 trixter aka Bret McDanel said the following: are you running the linux mozilla? That may be the problem where you are trying to mix given that there is IPC stuff going on between flash and mozilla.. perhaps, but then as i said in another post in this thread, the native versions wont give you such problems. Right why I said its more a matter of recompiling. Even from linux to BSD is generally trivial for most applications (some network stuff can exactly, which is why i'd hoped that freebsd native versions could be released. heck, we'd even assist in porting if that were to be the case. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 5,000 concurrent calls system rollout question
On 02/02/06 06:13 [EMAIL PROTECTED] said the following: On Wed, 1 Feb 2006, Kristian Larsson wrote: Indeed, a FreeBSD machine doing just routing lookups can handle somewhere around 600Kpps. Not to nitpick, but freebsd has routed 1M+pps using commodity hardware. thanx, i wanted to point this out but didnt want to inadverntly start a linux vs freebsd flame war. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] (newby) IAX Trunk on low bandwidth connection
Brrghhh: Bandwidth calculation is really foggy for me: Using the calculator Im getting about 23 kbps for both incoming and outgoing. What does this mean: Is a 64kbit link used at 71% capacity ((23+23):64) or is it used at only 35% (23:64)? Will this vary over time (i.e: does the codec generate more then average data at times? How about less then average?) Thanks. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rob Lith Sent: Wednesday, February 01, 2006 11:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] (newby) IAX Trunk on low bandwidth connection What codec is that using. G.729 will give you 10 calls at best over 256k unless you're trunking with IAX2? I don't know anyone using lpc10... Remember a G.729 8k codec turns into 23.63 Kbps with all the overheads... Regards Rob On 2/1/06, Garth van Sittert [EMAIL PROTECTED] wrote: Hi Cosmin You should be able to get about 12 simultaneous calls on a 128k line and about 28 on a 256k line according to asteriskguru's bandwidth calculator http://www.asteriskguru.com/tools/bandwidth_calculator.php. Kind Regards Garth BitCo Data Communications http://www.bitco.co.za Cosmin Prund wrote: Hello everyone, this is my first post to the list, so hello again. We're a small company in Romania and we're trying to set up a really small version of call center. That is, we want to get a few land-lines from our telco in different countys and bridge all calls to our HQ, in order to make it cheeper for our clients to call us. Unfortunatelly there's no ISP in our area that can deliver a broadband connection for anything less then an arm and a leg, so we're considering runing an * - * connection using VoIP over a low bandwidth connection (we're considering 128kbit but we might be able to go to 256kbit). The bandwidth price is not a problem for our satelite installations, we cand get acceptably priced broadband (~256kbit) so the distant *'s will have propper connections. My question: Is 128kbit a wide enough connection for 1 simultaneous conversation, using IAX protocol with the comercial version of the g729 codec? I'm expecting this to be engough for more then 1 conversation (after all a single line analog connection is rated at 64kbit and I'm getting double that bandwidth) Cosmin Prund ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users From - Wed -- Garth van Sittert BSc (Physics Computer Science) - Mobile: +27 (0)83 791 6662 Email:[EMAIL PROTECTED] Phone:08600 BITCO Web:www.bitco.co.za ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Streaming MOH
Title: Streaming MOH Hi, I'm having some problems getting this to work with Asterisk 1.2.4. Does it work for anyone? Does anyone have a site I can test this with? Regards Lee ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Setting MSN for outgoing ISDN calls
Hi all, I have a problem setting the MSN for outgoing calls. I'm using a HFC PCI card together with zaptel and bristuff. All my outgoing calls are using the same (first|default|main) MSN. In my zapata.conf I tried different values for pridialplan, prilocaldialplan, nationalprefix, etc but without any success. In my extension.conf I'm setting the MSN with: exten = _X.,1,SetCIDNum(MSN) exten = _X.,2,Dial(Zap/g2/Number,60, T) What are the right values for pridialplan for Germany? Is setting the MSN with SetCIDNum the right way? Would be fine if someone could provide me a working zapata.conf. Thanks in advance, Henry -- Hi! I'm a .signature virus! Copy me into your ~/.signature to help me spread! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Swapping lines using dtmf
At 04:59 AM 02/01/2006, you wrote:
If anyone has any ideas if and
how this can be done I would apreciate any
info.
You can search for flash() on the wiki, start here
http://www.voip-info.org/wiki-Asterisk+cmd+Flash . I hope you have
better luck than me, I've no gotten it to work yet except by the transfer
trick, something like this:
exten = _6[0-2][0-4],1,Flash()
exten = _6[0-2][0-4],2,Dial(SIP/1${EXTEN:1},,rtT)
Ira
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[Asterisk-Users] OT O'Reilly Asterisk TFOT
I went to the Linux Solutions exhibition in Paris yesterday, visited the well stocked O'Reilly stand and saw a nice pile of Asterisk TFOT, 6 hours later there was only one left. It must say something, also it was the English version. -- Dave Cotton [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] (newby) Is PING a good indicator of latency?
On 02/02/06 00:06 Olle E Johansson said the following: Damon Estep wrote: Not really enough sample points to determine if the network will support RTP and no provision for jitter measurements and packet loss. I really like the statistics on the cheap Linksys ATAs! - latency, jitter, packet loss during an actual call. For a newbie, it's a start, but you are absolutely right. The work we are doing with RTCP support will help in this, measuring quality per i've usually just used ethereal to capture packets between asterisk and the SIP phones. the box running ethereal is plugged into the ethernet switch's monitor port, obviously. works like a charm. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Receiving faxes with spandsp - strange problem
Bartosz Jozwiak wrote: Check if rxfax actually receives anything... How? -- Best regards, Bartosz Piec ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] meetme and dtmf
Imran Ahmed wrote:
On 2/1/06, Kevin P. Fleming [EMAIL PROTECTED] wrote:
Imran Ahmed wrote:
Even though no IAX client supports inband dtmf, An IAX client can send
inband dtmf which would have corrected your problem.
No, it won't. No IAX2 client will start a DSP to listen for inband DTMF,
because IAX2 is defined to always send out-of-band DTMF.
At best, if the receiving IAX2 system is just passing the audio along to
another protocol that does support inband DTMF, then sending it in the
audio stream would work. If the application receiving the DTMF is on the
other IAX2 end, though (like MeetMe in this case), then it will never
'see' the DTMF, because Asterisk will not look in the audio stream for DTMF.
I agree, but the other ends of the conference were zap channels in
this case, at least that is what I figured by the first email.
Maybe if a paint better my scenario it would help the discussion.
Step 1: A IAX client make a call executing the following command
Dial(ZAP/g1/${EXTEN})
If aswered this call is tranfered to a conference room.
Step 2: The IAX client make a second call executing again
Dial(ZAP/g1/${EXTEN})
an IVR answer this call and the IAX client have to send some
DTMF stil now everything works very well.
At this point call is transfered to the previous conference room
and The IAX client reach the conference too.
Step 3 The Iax client heve to send some other DTMF to the IVR.
NOW THE IVR DOES NOT HEAR DTMF SENDED BY THE IAX CLIENT, EVEN IF
IT CAN HEAR DTMF SENDED BY THE FIRST ZAP CHANNEL.
Hoping to be clear enough
thank yuo very much for any help or suggestion.
Accursio Avona
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Re: [Asterisk-Users] RE: Euro-ISDN
On Wed, February 1, 2006 22:12, Armin Schindler said: On Wed, 1 Feb 2006, Aldo Bergamini wrote: [EMAIL PROTECTED] is believed to have said: chan_capi does not set the NT-mode. Your cards driver need to do that. E.g. for Eicon DIVA Server cards, you just set the '-x' option with divactrl or set NT-mode in the config wizard. chan_capi does not (need) to know anything about what protocol the card is doing. CAPI is independent here. Ok. Anyway, if the card is set to NT mode, you should specify ntmode=yes in the capi.conf to tell chan_capi to handle the progress better (get progress tones). Fine! One last related subpoint: while Eicon Diva cards have their own setup application, is there anything standard to control the basic setup of generic HFC-S cards? (something similar to the ztconfig tool for analog cards) Sorry, I cannot answer that one. I don't know enough about these cards and their drivers. With BRIstuff you get to use ztcfg, etc. Cannot say anything about mISDN, CAPI... -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] (newby) IAX Trunk on low bandwidth connection
On 2 Feb 2006, at 08:09, Cosmin Prund wrote: Brrghhh: Bandwidth calculation is really foggy for me: Using the calculator I’m getting about 23 kbps for both incoming and outgoing. What does this mean: Is a 64kbit link used at 71% capacity ((23+23):64) or is it used at only 35% (23:64)? Will this vary over time (i.e: does the codec generate more then average data at times? How about less then average?)It depends on what sort of link you have. Most links are full duplex (leased lines etc) which would be 35%but some radio based links are half duplex which would be 71%So for a 64k link you will (just about) get 3 729 calls. If all the calls between are between the same two servers, you can use IAX trunking, which would pushyou up to 5 calls. (What that tells you is that for 729 and gsm, the headers are as big as the data).You talk about satellite stations, if you are going for a hub and spoke, you should put the hubon the highest bandwidth link. Thanks. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Rob Lith Sent: Wednesday, February 01, 2006 11:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] (newby) IAX Trunk on low bandwidth connection What codec is that using. G.729 will give you 10 calls at best over 256k unless you're trunking with IAX2? I don't know anyone using lpc10... Remember a G.729 8k codec turns into 23.63 Kbps with all the overheads... Regards Rob On 2/1/06, Garth van Sittert [EMAIL PROTECTED] wrote:Hi Cosmin You should be able to get about 12 simultaneous calls on a 128k line and about 28 on a 256k line according to asteriskguru's bandwidth calculator http://www.asteriskguru.com/tools/bandwidth_calculator.php. Kind Regards Garth BitCo Data Communications http://www.bitco.co.za Cosmin Prund wrote: Hello everyone, this is my first post to the list, so hello again.We're a small company in Romania and we're trying to set up a really small version of "call center". That is, we want to get a few land-lines from our telco in different countys and "bridge" all calls to our HQ, in order to make it cheeper for our clients to call us. Unfortunatelly there's no ISP in our area that can deliver a broadband connection for anything less then an arm and a leg, so we're considering runing an * - * connection using VoIP over a low bandwidth connection (we're considering 128kbit but we might be able to go to 256kbit). The bandwidth price is not a problem for our "satelite" installations, we cand get acceptably priced broadband (~256kbit) so the distant *'s will have propper connections. My question: Is 128kbit a wide enough connection for 1 simultaneous conversation, using IAX protocol with the comercial version of the g729 codec? I'm expecting this to be engough for more then 1 conversation (after all a single line analog connection is rated at 64kbit and I'm getting double that bandwidth) Cosmin Prund___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users From - Wed -- Garth van Sittert BSc (Physics Computer Science) - Mobile: +27 (0)83 791 6662 Email: [EMAIL PROTECTED] Phone: 08600 BITCO Web:www.bitco.co.za ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://www.westhawk.co.uk/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Receiving faxes with spandsp - strange problem
Pierre Burton wrote: What's your cisco conf ? how did you transfert between Cisco and asterisk ? A-law, U-law ?? This is part of my Cisco config: voice-card 0 no dspfarm ! ! ! voice service voip sip ! ! voice class codec 1 codec preference 1 g711ulaw codec preference 2 g711alaw codec preference 3 g729r8 bytes 40 codec preference 4 g723r63 bytes 96 codec preference 5 g726r16 bytes 80 codec preference 6 g726r24 codec preference 7 g726r32 codec preference 8 g728 codec preference 9 gsmefr codec preference 10 gsmfr voice vad-time 65536 ! voice translation-rule 1 rule 1 /^0?/ // ! voice translation-rule 2 rule 2 /^1?2?/ // voice translation-profile CutTwelve translate called 2 ! voice translation-profile CutZero translate calling 3 translate called 1 voice-port 0/1/0:15 echo-cancel coverage 32 no comfort-noise music-threshold -70 dial-peer voice 1 pots translation-profile outgoing CutZero destination-pattern ^0 direct-inward-dial port 0/1/0:15 ! dial-peer voice 2 voip description Route calls starting with 293 to centile translation-profile outgoing CutTwelve application session destination-pattern 1229339[60-79] voice-class codec 1 session protocol sipv2 session target ipv4:62.111.174.79 dtmf-relay rtp-nte h245-signal h245-alphanumeric ! sip-ua retry invite 3 retry response 3 retry bye 3 retry cancel 3 timers trying 1000 sip-server ipv4:asterisk.ip.add.ress As I understand, the preferred codec is ulaw. Should I change something in this configuration? -- Best regards, Bartosz Piec ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 5,000 concurrent calls system rollout question
On Thu, 2 Feb 2006, Dinesh Nair wrote: On 02/02/06 06:13 [EMAIL PROTECTED] said the following: On Wed, 1 Feb 2006, Kristian Larsson wrote: Indeed, a FreeBSD machine doing just routing lookups can handle somewhere around 600Kpps. Not to nitpick, but freebsd has routed 1M+pps using commodity hardware. thanx, i wanted to point this out but didnt want to inadverntly start a linux vs freebsd flame war. 1Mpps is no longer only the realm of 'big iron'. linux can do it on commodity hardware too. there's no magic in 1Mpps anymore. of course thats just routing the packets. actually doing something with the contents is a different matter entirely. i doubt theres any hardware which can handle 5,000 simultaneous voip calls on a single box. -Dan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] callback script?
How do I setup a Callback script? This script does what I want to do. But how do I set it up? http://www.junghanns.net/en/callback.html I see it uses PHP for scriptlanguage. So where do I place it (the .agi)? Any tips would be appreciated. Thanks, Arne Morten Johansen. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Pri Hang up outgoing calls
Hi All,
the * is working rigth for incoming calls and internal calls, but when trying
to call out we got
hanged up. The hangup reason is AST_CAUSE_INVALID_IE_CONTENTS
I've been searching in the mailing list archive as I thing that some thing
similar happens to
someone else but did not find.
We are runnig asterisk 1.2.4
extensions.conf
[default]
exten = _0.,1,Set(CALLERID(Number)=971288612)
exten = _0.,2,Set(CALLERID(Name)='Gil Estarellas')
exten = _0.,3,Dial(Zap/g1/${EXTEN:1}||rf)
exten = _0.,4,NoOp(${HANGUPCAUSE})
exten = _0.,5,NoOp(${PRI_CAUSE})
__
LLama Gratis a cualquier PC del Mundo.
Llamadas a fijos y móviles desde 1 céntimo por minuto.
http://es.voice.yahoo.com
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RE: [Asterisk-Users] Caller ID patches - updated
Marc, The links to the patches on the site seem to be broken... can you supply correct links? Adam Hatia -Original Message- From: Marc McLaughlin (LUSYN) [mailto:[EMAIL PROTECTED] Sent: 01 February 2006 18:58 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Caller ID patches - updated Hello all, The Caller ID patches have been updated to work with X100P and TDM400P cards. There is also a patch that should fix distinctive ring on TDM400P cards when using polarity reversal check for Caller ID. It may be required for the history buffer method too. http://www.lusyn.com/resources/asterisk/index.htm contains links to the pages describing the patches, including details on how to apply them. Thanks go to Ian Plain of cyber-cottage.co.uk for donating a TDM card. Without it I would not have been able to work on these patches. If any of you are interested in helping me continue to provide these patches, clink on the Support link in the footer of any page on www.lusyn.com. Thanks, Marc Eur Ing Marc McLaughlin BSc (Hons) CEng CITP FIAP MBCS MIEE MIMIS ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk video conference
Hi , I just wanted to know , how would be asterisk work with video calls ? What are the hardware do we have to buy ? Who are the providers of particular harwares ? Can we use video calls / video conferenceing in the LAN perfectly ? How it would be depends on the WAN ?please reply me soon , very urgent to grab the info and buy the equipments .I really appreciate , If , you guys can provide few examples , web sites ... Thank you, Shaine. To help you stay safe and secure online, we've developed the all new Yahoo! Security Centre.___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] XLite dtmf issue?
Thanks changing the dtmfmode to rfc2833 did the trick. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of kevin ling Sent: 02 February 2006 01:10 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] XLite dtmf issue? set dtmfmode=rfc2833 in sip.confand try again. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Aisling Sent: Wednesday, February 01, 2006 11:03 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] XLite dtmf issue? Hi, Im wondering if anyone has experienced an issue with the XLite softphone and asterisk accepting dtmf? I can listen to my voicemail perfectly from my hardphone. However when I dial the voicemail number from my XLite softphone and enter the password at the voicemail prompt, an error appears vm-incorrect and I get an Unable to read password message on the asterisk console. Has anyone experienced issues with XLite dtmf? Many thanks, Aisling. ---Legal Disclaimer--- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt. ---Legal Disclaimer--- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt. ---Legal Disclaimer--- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE:[Asterisk-Users] Pri Hang up outgoing calls
Here is the debuging information when trying to call out __ LLama Gratis a cualquier PC del Mundo. Llamadas a fijos y móviles desde 1 céntimo por minuto. http://es.voice.yahoo.com error.log Description: 2522182428-error.log ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] fax possibilities
Hi James I would consider Hylaxfax if you are going to do purely faxing. Garth James Harper wrote: I am trying to set up a linux based faxing solution for a client, and have found that the modem they have (ancient dataplex external unit) just isn't up to the job. It talks to some remote fax machines but not others. A new external modem ranges from AUD$75 to AUD$400, which got me thinking of other possibilities... #1 FXO PCI card (more expensive for 1 port, probably cheaper for 2+) #2 Sipura SPA3000 #3 Grandstream ATA488 I assume there will be no problem getting #1 working as a fax modem, but what about #2 and #3? Has anyone done this before? Some brief googling shows that it is possible, but not that it has been done... James ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Garth van Sittert BSc (Physics Computer Science) - Mobile: +27 (0)83 791 6662 Email: [EMAIL PROTECTED] Phone: 08600 BITCO Web:www.bitco.co.za ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] (newby) IAX Trunk on low bandwidth connection
At my HQ Im instaling a 128kbit leased line connection, with guaranteed bandwidth to the Internet; The telco promises less then 20 ms to the internet (to ronix.ro), no jitter and no packet loss. So Im hoping for 40 ms times to net and small jitter J This is my hub. For my satelite instalations Im planning on grabing a connection from a different provider (as this telco provider is expensive) but Im also considering a 64kbit leased line from the same provider, just in case my VoIP doesnt work with the cheeper providers. My remote instalations will never have more then one conversation load, and this conversation would be ZAP to IAX or SIP. That is, the distant instalation will need to forward all calls coming in on the zap chanel to my HQ Asterisk. Thats all it will ever do J. Im not sure trunking woud provide anything in this case as there will never be more then one concurent conversation from the remote * to my HQ *. Im expecting IAX to provide better performance over SIP but not by much. Considering my remote * instalations will never have more then one concurent conversation with my HQ and considering I can get a really good 64kbit line I guess Im OK. As for my HQ, Im sure Im OK because Ill get a 128 kbit line and Ill be able to afford an upgrade to 256kbit. I can actually go all the way to 2048 kbit, but that would no longer be economically viable. So Ill see how it goes, and I hope Ill have the time to put in a comment on the low bandwidth wiki on voip-info.org. Thanks to everyone for your help. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of tim panton Sent: Thursday, February 02, 2006 11:05 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] (newby) IAX Trunk on low bandwidth connection On 2 Feb 2006, at 08:09, Cosmin Prund wrote: Brrghhh: Bandwidth calculation is really foggy for me: Using the calculator Im getting about 23 kbps for both incoming and outgoing. What does this mean: Is a 64kbit link used at 71% capacity ((23+23):64) or is it used at only 35% (23:64)? Will this vary over time (i.e: does the codec generate more then average data at times? How about less then average?) It depends on what sort of link you have. Most links are full duplex (leased lines etc) which would be 35% but some radio based links are half duplex which would be 71% So for a 64k link you will (just about) get 3 729 calls. If all the calls between are between the same two servers, you can use IAX trunking, which would push you up to 5 calls. (What that tells you is that for 729 and gsm, the headers are as big as the data). You talk about satellite stations, if you are going for a hub and spoke, you should put the hub on the highest bandwidth link. Thanks. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Rob Lith Sent: Wednesday, February 01, 2006 11:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] (newby) IAX Trunk on low bandwidth connection What codec is that using. G.729 will give you 10 calls at best over 256k unless you're trunking with IAX2? I don't know anyone using lpc10... Remember a G.729 8k codec turns into 23.63 Kbps with all the overheads... Regards Rob On 2/1/06, Garth van Sittert [EMAIL PROTECTED] wrote: Hi Cosmin You should be able to get about 12 simultaneous calls on a 128k line and about 28 on a 256k line according to asteriskguru's bandwidth calculator http://www.asteriskguru.com/tools/bandwidth_calculator.php. Kind Regards Garth BitCo Data Communications http://www.bitco.co.za Cosmin Prund wrote: Hello everyone, this is my first post to the list, so hello again. We're a small company in Romania and we're trying to set up a really small version of call center. That is, we want to get a few land-lines from our telco in different countys and bridge all calls to our HQ, in order to make it cheeper for our clients to call us. Unfortunatelly there's no ISP in our area that can deliver a broadband connection for anything less then an arm and a leg, so we're considering runing an * - * connection using VoIP over a low bandwidth connection (we're considering 128kbit but we might be able to go to 256kbit). The bandwidth price is not a problem for our satelite installations, we cand get acceptably priced broadband (~256kbit) so the distant *'s will have propper connections. My question: Is 128kbit a wide enough connection for 1 simultaneous conversation, using IAX protocol with the comercial version of the g729 codec? I'm expecting this to be engough for more then 1 conversation (after all a single line analog connection is rated at 64kbit and I'm getting double that bandwidth) Cosmin Prund ___ --Bandwidth and Colocation provided by Easynews.com --
Re: [Asterisk-Users] Setting MSN for outgoing ISDN calls
Answering my own question. It worked with prilocaldialplan=local, pridialplan=unknown and running CallerPres before every Dial command. Henry -- Hi! I'm a .signature virus! Copy me into your ~/.signature to help me spread! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Outbound Caller ID number on E1
Hi All I am having a problem setting the outbound callerid number on a PRI E1 in South Africa. The outbound number keeps on appearing as the main PRI number. How does it work between Asterisk and the Telko? More importantly how do I get it working? Kind Regards Garth -- Garth van Sittert BSc (Physics Computer Science) - Mobile: +27 (0)83 791 6662 Email: [EMAIL PROTECTED] Phone: 08600 BITCO Web:www.bitco.co.za ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Outbound Caller ID number on E1
I have this problem in the UK on BT too. Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Garth van Sittert Sent: 02 February 2006 11:37 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Outbound Caller ID number on E1 Hi All I am having a problem setting the outbound callerid number on a PRI E1 in South Africa. The outbound number keeps on appearing as the main PRI number. How does it work between Asterisk and the Telko? More importantly how do I get it working? Kind Regards Garth -- Garth van Sittert BSc (Physics Computer Science) - Mobile: +27 (0)83 791 6662 Email: [EMAIL PROTECTED] Phone: 08600 BITCO Web:www.bitco.co.za ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help with sip setup because can't receive calls!!!!!!
Hi all, i am still challanged to get the SIP phones to work. this is the output of debug for one of the SIP phones. I called the other sip phone and collected the output. if you look through the debug post (below), you'll notice the bolded text "address incomplete", what is the cause of this sort of error? and how can one eliminate this problem? my * server: 10.47.200.136 phone1:10.47.200.137 (2173 is the number to dial to reach this phone) phone2:10.47.200.141 (2172 is the number to dial to reach this phone)thank you in advance for any help. Ama- --- (16 headers 24 lines)---Using INVITE request as basis request - [EMAIL PROTECTED]Sending to 10.47.200.137 : 5060 (non-NAT)Found user 'stargate3'Found RTP audio format 0Found RTP audio format 18Found RTP audio format 96Found RTP audio format 102Found RTP audio format 107Found RTP audio format 104Found RTP audio format 105Found RTP audio format 106Found RTP audio format 4Found RTP audio format 97Found RTP audio format 98Found RTP audio format 2Found RTP audio format 99Found RTP audio format 8Found RTP audio format 101Peer audio RTP is at port 10.47.200.137:3000Found description format PCMUFound description format G729Found description format BV16Found description format BV32Found description format L16Found description format PCMUFound description format PCMAFound description format L16Found description format G723Found desc ription format G726-16Found description format G726-24Found description format G726-32Found description format G726-40Found description format PCMAFound description format telephone-eventCapabilities: us - 0x40e (gsm|ulaw|alaw|ilbc), peer - audio=0x55d (g723|ulaw|alaw|g726|slin|g729|ilbc)/video=0x0 (nothing), combined - 0x40c (ulaw|alaw|ilbc)Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)Looking for 2172 in default (domain 10.47.200.136)Reliably Transmitting (no NAT) to 10.47.200.137:5060:SIP/2.0 484 Address IncompleteVia: SIP/2.0/UDP 10.47.200.137;branch=z9hG4bKaa5252205;received=10.47.200.137From: stargate3 sip:[EMAIL PROTECTED]:5060;tag=95f353525026b9dTo: 2172 sip:[EMAIL PROTECTED]:5060;tag=as529cb545Call-ID: [EMAIL PROTECTED]CSeq: 775968348 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYMax-Forwards: 70Contact: sip:[EMAIL PROTECTED]Content-Length: 0 ---VoIP-*CLI -- SIP read from 10.47.200.137:5060: ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0Via: SIP/2.0/UDP 10.47.200.137;branch=z9hG4bKaa5252205Max-Forwards: 70Content-Length: 0To: 2172 sip:[EMAIL PROTECTED]:5060;tag=as529cb545From: stargate3 sip:[EMAIL PROTECTED]:5060;tag=95f353525026b9dCall-ID: [EMAIL PROTECTED]CSeq: 775968348 ACKProxy-Authorization:Digest response="a94c49745a7cde16ebf111a12426493c",username="stargate3",realm="asterisk",nonce="366925e5",uri="sip:[EMAIL PROTECTED]:5060"User-Agent: Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 ---abc def [EMAIL PROTECTED] wrote:not sure but this is the output from the pbx:sip show registryHost Username Refresh State local_sip:5060 stargate3 105 Registered local_sip:5060 stargate2 105 Registered local_sip:5060 stargate1 105 Registered from sip phone I can any other phone (cisco with sccp or iax protocol) but I can't call any other sip phone, or receive phone calls. Facundo Ameal [EMAIL PROTECTED] wrote: are you sure your sip phone is registering ok?2006/2/1, abc def <[EMAIL PROTECTED]>: Thanks Facundo for instruction but it didn't work. there is nothing new in your suggestion compare to my conf files nevertheless I tried it but it didn't work. I can make call from my sip phone but can't receive any phone call. I am sure some one had had the same problem an d solved it. as always I appreciate for your suggestion, advice and/or correction to my config files. if you know how to solve this problem please give me some hint. thank you Facundo Ameal <[EMAIL PROTECTED]>wrote: i've tested it with this config files and i worked: extensions.conf exten = 55,1,Dial(SIP/2271,20) sip.conf [2271] type=friend host=dynamic secret=sip allow=all qualify=200 nat=no Instead of 2271 you can put whatever you want. good luck. 2006/1/31, Facundo Ameal : Are you using a SIP Softphone or an ATA? 2006/1/31, Facundo Ameal : does it registers well? although i think you have to add "context=default" to the stargate1 section. ; try that and see what happens. 2006/1/31, abc def :Hi all, I am resending this message, so far no one has helped me with thisincoming call issue. there is no problem with outbound call but there is noinbound call to my sip phone. the only message I get when I call from pstnis "una ble to create local channel for call forward to'Local/[EMAIL
[Asterisk-Users] DeadAGI variables confusion
Hi * users,
We're using calls to external scripts through AGI at various points
throughout our IVR system. We use these scripts to log certain events
and to make certain choices that I wasn't sure would be possible in
the dial plan. The problem comes with with the final call to our
script. We use this line:
exten = h,1,deadagi(log.php|{$service}|Hung up|${UNIQUEID})
I know there are some issues with getting variables through DeadAGI,
but I just wanted some clarification, because I haven't seen it
explained clearly. Certainly the value of UNIQUEID was being
successfully passed to log.php in earlier versions of * but isn't now
(I just installed version 1.2.4)
Any advice welcome. Even if it is telling me we've beein doing this
all wrong the whole time!
Regards,
DaveB
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SV: [Asterisk-Users] Outbound Caller ID number on E1
How do you set the CallerID? Have you checked with your provider that they've enabled callerid? If yes, are you using a correct number that the provider allows? Regards, Jan -Ursprungligt meddelande- Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För Garth van Sittert Skickat: den 2 februari 2006 12:37 Till: Asterisk Users Mailing List - Non-Commercial Discussion Ämne: [Asterisk-Users] Outbound Caller ID number on E1 Hi All I am having a problem setting the outbound callerid number on a PRI E1 in South Africa. The outbound number keeps on appearing as the main PRI number. How does it work between Asterisk and the Telko? More importantly how do I get it working? Kind Regards Garth -- Garth van Sittert BSc (Physics Computer Science) - Mobile: +27 (0)83 791 6662 Email: [EMAIL PROTECTED] Phone: 08600 BITCO Web:www.bitco.co.za ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Outbound Call SIP Results
We make outbound calls via our Asterisk (*1), then via SIP toa 3rd party Asterisk (*2), which then routes to PSTN, again via SIP. If the called number is invalid or out of service, *2 getsa404 Not Found, which seems appropriate. However, *2 then passes on a 403 Forbidden to *1, which is not really the right response. *1then returns a 486 Busy Here, which also seems wrong, as it would generally mean the called number was busy (engaged), I think.
Is it possible to vary this behaviour? The ${DIALSTATUS} variable doesn't seem to be fine-grained enough to help.
Thanks for any thoughts,
David
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Re: [Asterisk-Users] Outbound Caller ID number on E1
Garth van Sittert wrote: Hi All I am having a problem setting the outbound callerid number on a PRI E1 in South Africa. The outbound number keeps on appearing as the main PRI number. How does it work between Asterisk and the Telko? More importantly how do I get it working? Kind Regards Garth Telcos usually arrange outgoing CLI in one of 3 ways: - a free for all - you can put what you like as your CLI, and no checks are made - a rigid arrangement - no matter what you give as the CLI, the telco will replace it with a fixed value before passing the message on - a constrained arrangement - if you give a CLI within the range that is valid for you, it will be passed on. If you give something which is not allocated to you, the telco wil replace it with a fixed value. Sounds like you do not have the first arrangement. You might have the third, though. It could be you just aren't specifying your number correctly - either the digits themselves or the TON/NPI pair might not be right. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Delaying media stream by short period after 183 is sent
Hi, anyone know of a way that I can delay the RTP stream a little bit once the 183 is sent, I just want to delay it by around 100ms or so for some troubleshooting. Also, I always see a RTP packet before the 183 is sent for each call, it is just a single packet, is something wrong here as my firewall won't open a connection entry until 183 is processed. Mark ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: RE: [Asterisk-Users] Blocked Callerid
YES Asterisk will support this. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Thursday, February 02, 2006 1:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: RE: [Asterisk-Users] Blocked Callerid I think they have a 1-800 number so you might be right. But the important question is still - will Asterisk support this? PaulH Alexander Lopez [EMAIL PROTECTED] wrote: They are using ANI instead of CallerID. If they have an 800 number thya have the right to know who is calling them because they are paying for the call. the *ANI*DNIS* format is known as Feature Grooup D. Alex From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joe Pukepail Sent: Wednesday, February 01, 2006 3:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Blocked Callerid Do they have an 800 number? If so perhaps their 800 number provider is doing it via DTMF. Search around on the internet, I believe the standard format for the DTMF is *CALLERID*CALLEDNUMBER* (or perhaps reversed). On 2/1/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: I have been discussing an asterisk solution with a company that has a custom written dialogic based solution. The issue is that their dialogic solution can read callerid from incoming calls, even if the callerid is blocked. I have read before that Asterisk can do this, and they want me to make sure that their new system will be able to do this. A quick poke around inside the zaptel source code was unproductive... Any ideas? PaulH ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/ -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Anyone know a good ITSP in Canada that supports *?
Hi, I'm looking for a new Internet Telephony Service Provider for my company in Canada to terminate calls from my Asterisk PBX. Ideally I'd like DiDs in Otawa, Toronto, NY San Jose. Anyone out ther who can help me with a recommendation? Vonage seemed clueless when I called them. Broadvoice is good but no Canadian DIDs... Thanks, Hugh ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Fwd: Re: [Asterisk-Users] Asterisk for Call Center (missing reference)]
John Todd, Can you please answer that question or just give me your feedback about it? I'll be very thankfull to hear something from you! regards, Telles ---BeginMessage--- Hi, Does any body knows some thing about it? Thanks in advance. Telles Rodrigo P. Telles wrote: Hi Folks, I've been searching for an specific feature on asterisk and I found an e-mail from John Todd asking for the same thing. http://lists.digium.com/pipermail/asterisk-users/2004-May/045882.html To be able to listen (zapbarge, zapscan or chanspy) an specific channel and can talk to one side (the operator). That feature is very usefull in call centers in Brazil so if you want to use Asterisk as a Call Center PBX you have to support it. John Todd post it in May 2004 so perhaps now (Jan 2006) it's possible or there is another app (commercial?) that can support it. John: have you found a solution for your question? if so, please let me know! Thanks in advance, -- Rodrigo P. Telles [EMAIL PROTECTED] IT Manager Devel-IT - http://www.devel.it IVOZ # 1029 +55 14 3324-1200 Bestcom Group ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---End Message--- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Outbound Call SIP Results
David Brazier wrote:
We make outbound calls via our Asterisk (*1), then via SIP to a 3rd
party Asterisk (*2), which then routes to PSTN, again via SIP. If the
called number is invalid or out of service, *2 gets a 404 Not Found,
which seems appropriate. However, *2 then passes on a 403 Forbidden to
*1, which is not really the right response. *1 then returns a 486 Busy
Here, which also seems wrong, as it would generally mean the called
number was busy (engaged), I think.
Is it possible to vary this behaviour? The ${DIALSTATUS} variable
doesn't seem to be fine-grained enough to help.
Would it be possible to see som log files?
/O
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Re: [Asterisk-Users] Anyone know a good ITSP in Canada that supports *?
On Thursday 02 February 2006 07:39, hugolivude wrote: I'm looking for a new Internet Telephony Service Provider for my company in Canada to terminate calls from my Asterisk PBX. Ideally I'd like DiDs in Otawa, Toronto, NY San Jose. Anyone out ther who can help me with a recommendation? Unlimitel.ca. CAD$0.011/min for origination and on-net termination. Excellent, and I mean *excellent* customer service. Not affiliated, but a very happy customer. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: DeadAGI variables confusion
In article [EMAIL PROTECTED],
Dave Brooks [EMAIL PROTECTED] wrote:
Hi * users,
We're using calls to external scripts through AGI at various points
throughout our IVR system. We use these scripts to log certain events
and to make certain choices that I wasn't sure would be possible in
the dial plan. The problem comes with with the final call to our
script. We use this line:
exten = h,1,deadagi(log.php|{$service}|Hung up|${UNIQUEID})
I know there are some issues with getting variables through DeadAGI,
but I just wanted some clarification, because I haven't seen it
explained clearly. Certainly the value of UNIQUEID was being
successfully passed to log.php in earlier versions of * but isn't now
(I just installed version 1.2.4)
Any advice welcome. Even if it is telling me we've beein doing this
all wrong the whole time!
You should be able to refer to channel variables in the 'h' extension.
If it's broken, then that's a bug.
However, if your example is an exact copy from your dialplan, perhaps
the parser is getting confused, because {$service} should be ${service}
Cheers
Tony
--
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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Re: [Asterisk-Users] Anyone know a good ITSP in Canada that supports *?
iBell just announced termination only to CA for I believe $0.0039 a minute. --- Andrew Kohlsmith [EMAIL PROTECTED] wrote: On Thursday 02 February 2006 07:39, hugolivude wrote: I'm looking for a new Internet Telephony Service Provider for my company in Canada to terminate calls from my Asterisk PBX. Ideally I'd like DiDs in Otawa, Toronto, NY San Jose. Anyone out ther who can help me with a recommendation? Unlimitel.ca. CAD$0.011/min for origination and on-net termination. Excellent, and I mean *excellent* customer service. Not affiliated, but a very happy customer. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call completes but then drops?
Feb 1 22:13:37 VERBOSE[18623] logger.c: -- Zap/2-1 answered SIP/102-9fda Feb 1 22:13:37 DEBUG[18623] channel.c: Avoiding initial deadlock for 'SIP/102-9fda' Feb 1 22:13:43 DEBUG[18623] channel.c: Didn't get a frame from channel: SIP/102-9fda Feb 1 22:13:43 DEBUG[18623] channel.c: Bridge stops bridging channels SIP/102-9fda and Zap/2-1 Feb 1 22:13:43 VERBOSE[18623] logger.c: -- Hungup 'Zap/2-1' Feb 1 22:13:43 DEBUG[18623] app_dial.c: Exiting with DIALSTATUS=ANSWER. Can anyone explain why this call dropped? The person dialed a number, the call WAS completed and connected to the PSTN through a PRI, but they never heard audio and the call was disconnected by Asterisk. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk on laptop connected to POTS line
Anyone know of any equipment that I can use to connect a laptop running asterisk to a POTS line (RJ11) ? Regards, Dovid __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Regarding cdr_manager.conf
Hello, My question is.. How does cdr_manager work? Does it suppose to populate cdr-csv/Master.csv? What about the cdr table on the database? What is the event some people talk about? I have changed (and reloaded) my configuration of cdr_manager.conf to ; ; Asterisk Call Management CDR ; [general] enabled = yes and it doesn't seem to make any difference. After originate a call from the manager interface my Master.csv is empty, cdr in my database also empty and I don't get any new event apart of Newchannel or Hangup from the manager interface. An this is all I get from the Asterisk console: -- Reloading module 'cdr_manager.so' (Asterisk Call Manager CDR Backend) == Parsing '/etc/asterisk/cdr_manager.conf': Found So..? Kind Regards, Victor. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on laptop connected to POTS line
On Thu, Feb 02, 2006 at 05:20:01AM -0800, Dovid Bender wrote: Anyone know of any equipment that I can use to connect a laptop running asterisk to a POTS line (RJ11) ? A SIP ATA with an FXO port? (e.g. the Sipura 3000) -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk on laptop connected to POTS line
Sipura SPA-3000 will give you 1 fxs and 1 fxo so you can connect to a POTS line AND a analog phone at the same time with one small box. Makes a great demo system. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Dovid Bender Sent: Thursday, February 02, 2006 6:20 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk on laptop connected to POTS line Anyone know of any equipment that I can use to connect a laptop running asterisk to a POTS line (RJ11) ? Regards, Dovid ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk on laptop connected to POTS line
The Grandstream ATA (480 I think...) does this and usually costs less than the Sipura. It has 1 FXS and 1 FXO. -Jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Damon Estep Sent: Thursday, February 02, 2006 8:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk on laptop connected to POTS line Sipura SPA-3000 will give you 1 fxs and 1 fxo so you can connect to a POTS line AND a analog phone at the same time with one small box. Makes a great demo system. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Dovid Bender Sent: Thursday, February 02, 2006 6:20 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk on laptop connected to POTS line Anyone know of any equipment that I can use to connect a laptop running asterisk to a POTS line (RJ11) ? Regards, Dovid ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk on laptop connected to POTS line
Look at Xorcom's USB channel Bank. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dovid Bender Sent: Thursday, February 02, 2006 8:20 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk on laptop connected to POTS line Anyone know of any equipment that I can use to connect a laptop running asterisk to a POTS line (RJ11) ? Regards, Dovid __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE411P or TE406P
Matt wrote: You will need a minimum 3.4Ghz Dual xeon with 1G ECC DDR, and hardware voice processing capable E1/T1 card, such as the sangoma 104d quad pci card, in order to run 120 PSTN calls, 1000 calls is impossible for 1 server. Centos Linux should be fine. This has to be some of the poorest advice I have seen on this list... what is 'hardware voice processing capable'? We run 120 channels of TDM on single CPU servers all the time (no transcoding of course), and the amount of RAM is nearly irrelevant. Independent tests have shown there is no appreciable performance difference between the available quad-port T1 cards. The poster did not ask about handling '1000 calls' nor about Linux distributions. We sell supermicro based * solutions you can contact me off list. This entire response was clearly an advertisement for your products/services, and as such is inappropriate for this list. To the OP: The TE406P and TE411P are identical except for PCI bus interface voltage. Use whichever your server can accept, and if it can do both (which is rare), use the TE411P as in the future 5V slots will be harder to find. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] meetme and dtmf
Accursio Avona wrote:
Step 2: The IAX client make a second call executing again
Dial(ZAP/g1/${EXTEN})
an IVR answer this call and the IAX client have to send some
DTMF stil now everything works very well.
At this point call is transfered to the previous conference room
and The IAX client reach the conference too.
Step 3 The Iax client heve to send some other DTMF to the IVR.
How is the IVR still involved if the call has been transferred into a
conference room?
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Re: [Asterisk-Users] Help with sip setup because can't receive calls!!!!!!
I am responding to my own problem because I found the answer finally which it may help others in future. I just had a break-through after 2 weeks struggling, finally I found the problem. the problem was in extensions.conf file. I misspelled "include" as "inculde" (finding a misspelled word in a long extensions.conf file is not so easy, trust me) but after checking "show dialplan" and going through it line by line, I found my sip sub-division is not there. abc def [EMAIL PROTECTED] wrote:not sure but this is the output from the pbx:sip show registryHost
Username Refresh State local_sip:5060 stargate3 105 Registered local_sip:5060 stargate2 105 Registered local_sip:5060 stargate1 105 Registered from sip phone I can any other phone (cisco with sccp or iax protoc
ol) but
I can't call any other sip phone, or receive phone calls. Facundo Ameal [EMAIL PROTECTED] wrote: are you sure your sip phone is registering ok?2006/2/1, abc def <[EMAIL PROTECTED]>: Thanks Facundo for instruction but it didn't work. there is nothing new in your suggestion compare to my conf files nevertheless I tried it but it didn't work. I can make call from my sip phone but can't receive any phone call. I am sure some one had had the same problem an d solved it. as always I appreciate for your suggestion, advice and/or correction to my config files. if you know how to solve this problem please give me some hint. thank you Facundo Ameal <[EMAIL PROTECTED]>wrote: i've tested it with this config files and i worked:
;
extensions.conf exten = 55,1,Dial(SIP/2271,20) sip.conf [2271] type=friend host=dynamic secret=sip allow=all qualify=200 nat=no Instead of 2271 you can put whatever you want. good luck. 2006/1/31, Facundo Ameal : Are you using a SIP Softphone or an ATA? 2006/1/31, Facundo Ameal : does it registers well? although i think you have to add "context=default" to the stargate1 section. ; try that and see what happens. 2006/1/31, abc def :Hi all, I am resending this message, so far no one has helped me with thisincoming call issue. there is no problem with outbound call but there is no
t;
inbound call to my sip phone. the only message I get when I call from pstnis "unable to create local channel for call forward to'Local/[EMAIL PROTECTED]' (case =0)". my configuration files are attachedbelow. any help would be greatly appreciated. many thanks in advance.ABC abc def wrote: there is no error message coming up on the pbx for in-bound calls (there isonly debugging messages for outbound calls). thanks in advance for any hint or suggestion.Ama I just post my configuration file here for sip phone:extensions.conf
-[globals][default]include = incominginclude = outgoinginclude = iaxinculde = sipinclude = sccp[sip]exten = 2171,1,Dial(SIP/stargate1,20);exten = 2171,1,Dial(SIP/2171,20)exten = 2171,2,Hangup gt; exten = 2172,1,Dial(SIP/stargate2,20);exten = 2172,1,Dial(SIP/2172,20)exten = 2172,2,Hangupexten = 2173,1,Dial(SIP/stargate3,20);exten = 2173,1,Dial(SIP/2173,20)exten = 2173,2,Hangup[sccp][skinny]&
gt;
[incoming]exten = ; _214943[5-9]6,1,Dial(SIP/stargate3)exten = _214943[5-9]6,2,Hangup[outgoing]exten = _,1,Dial(Zap/g1/${EXTEN})exten = _,2,Hangup-sip.conf-[general]context=default ; Default context for incoming callsg t;; Set this to your host name or domain namebindport=5060 ; UDP Port to bind to (SIP standard port is5060)bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds toall)srvlookup=yes
; Enable
DNS SRV lookups on outbound calls register = stargate1:[EMAIL PROTECTED]/2171register = stargate2:[EMAIL PROTECTED]/2172register = stargate3:[EMAIL PROTECTED]/2173;-- NAT SUPPORTnat=no ; Global NAT settings (Affects all peers andusers) [local_sip]type=friendhost=10.47.200.136context=default[stargate1] ;cisco 9760;[2171]; type=friendhost=dynamic ;10.47.200.140 ;dynamicdefaultip=10.47.200.140username=stargate1<
BR>
secret=xxxcallerid="21495071" 2171allow=allqualify=200nat=nodefaultip=10.47.200.140 [stargate2] ;Polycom 601;[2172]type=friendhost=dynamic ;10.47.200.141 ;dynamicdefaultip=10.47.200.141username=xxxsecret=2stargatecallerid="21495072" 2172allow=all g t; qualify=200nat=nodefaultip=10.47.200.141[stargate3] ;Aastra 480i;[2173]type=friendhost=dynamic ;10.47.200.137 ;dynamicdefaultip=10.47.200.137
t;
username=stargate3
[Asterisk-Users] RE: Outbound Call SIP Results
Here is part of the log from *1, showing a 403 received and 486 passed on (IP addresses, host names and telephone number changed):--- (10 headers 0 lines)- SIP read from ip.of.asterisk.2:5060:SIP/2.0 403 ForbiddenVia: SIP/2.0/UDP ip.of.asterisk.1:5060;branch=z9hG4bK596fad3eFrom: "1000" sip:[EMAIL PROTECTED];tag=as46f716d4To: sip:[EMAIL PROTECTED];tag=as0474d5e5Call-ID: [EMAIL PROTECTED]CSeq: 103 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFERContact: sip:[EMAIL PROTECTED]Content-Length: 0--- (10 headers 0 lines)---Transmitting (no NAT) to ip.of.asterisk.2:5060:ACK sip:[EMAIL PROTECTED] SIP/2.0Via: SIP/2.0/UDP ip.of.asterisk.1:5060;branch=z9hG4bK596fad3e;rportFrom: "1000" sip:[EMAIL PROTECTED];tag=as46f716d4To: sip:[EMAIL PROTECTED];tag=as0474d5e5Contact: sip:[EMAIL PROTECTED]Call-ID: [EMAIL PROTECTED]CSeq: 103 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0---Feb 1 14:48:12 WARNING[1436]: chan_sip.c:9532 handle_response_invite: Forbidden - wrong password on authentication for INVITE to '"1000" sip:[EMAIL PROTECTED];tag=as46f716d4' -- SIP/sip_channel-a404 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing Busy("SIP/vcs-7b2e", "") in new stackTransmitting (no NAT) to ip.of.client:5060:SIP/2.0 486 Busy HereVia: SIP/2.0/UDP ip.of.client:5060;received=ip.of.clientFrom: sip:[EMAIL PROTECTED];tag=d6e314d4-13c4-43e0ca2b-39df1197-4823To: sip:[EMAIL PROTECTED];tag=as2d609cdcCall-ID: d6e314d4-13c4-43e0ca2b-39df1197-29-1a214c0CSeq: 1 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYContact: sip:[EMAIL PROTECTED]Content-Length: 0X-Asterisk-HangupCause: Call Rejected
The dial plan is a basic:
exten == _0Z.,1,Dial(SIP/sip_channel/${EXTEN},30,j)exten == _0Z.,2,Congestionexten == _0Z.,102,Busy
David
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Re: [Asterisk-Users] Call completes but then drops?
Matt wrote: Can anyone explain why this call dropped? The person dialed a number, the call WAS completed and connected to the PSTN through a PRI, but they never heard audio and the call was disconnected by Asterisk. Very difficult to guess without any information about your system. If you are using Asterisk 1.2.2, this is a known problem. If not, we'll need a lot more information to be able to even try to help you. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TE411P or TE406P
It sucks when a lead developer of Asterisk has to be the moderator. People please, use some common sense. KPF has ENOUGH work on his plate. I find it embarrassing that HE is the one policing the list. Alex Original and follow-up posts snipped for bandwidth sake. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin P. Fleming Sent: Thursday, February 02, 2006 9:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] TE411P or TE406P ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: DeadAGI variables confusion
On 02/02/06, Tony Mountifield [EMAIL PROTECTED] wrote:
You should be able to refer to channel variables in the 'h' extension.
If it's broken, then that's a bug.
However, if your example is an exact copy from your dialplan, perhaps
the parser is getting confused, because {$service} should be ${service}
Thank you for your idiocy detection skills! That was indeed the
problem. Although a warning from the parser would have helped.
Grumble, grumble.
Cheers,
DaveB
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Re: [Asterisk-Users] Outbound Caller ID number on E1
Thanks Steve We replaced an old Siemens that used to produce the outgoing numbers correctly, so we must have the 3rd option. The extensions are all set up to be in that range to provide DID. When the SIP extensions dial out will they provide the extension number which gets prefixed with what? Do I need to send the complete number, 3 digit area code + 4 digit extension to the Telko? Does the zapata.conf add the prefix? How can I check what callerid number is being passed to the Telko? Garth Steve Underwood wrote: Garth van Sittert wrote: Hi All I am having a problem setting the outbound callerid number on a PRI E1 in South Africa. The outbound number keeps on appearing as the main PRI number. How does it work between Asterisk and the Telko? More importantly how do I get it working? Kind Regards Garth Telcos usually arrange outgoing CLI in one of 3 ways: - a free for all - you can put what you like as your CLI, and no checks are made - a rigid arrangement - no matter what you give as the CLI, the telco will replace it with a fixed value before passing the message on - a constrained arrangement - if you give a CLI within the range that is valid for you, it will be passed on. If you give something which is not allocated to you, the telco wil replace it with a fixed value. Sounds like you do not have the first arrangement. You might have the third, though. It could be you just aren't specifying your number correctly - either the digits themselves or the TON/NPI pair might not be right. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Garth van Sittert BSc (Physics Computer Science) - Mobile: +27 (0)83 791 6662 Email: [EMAIL PROTECTED] Phone: 08600 BITCO Web:www.bitco.co.za ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] (newby) Is PING a good indicator of latency?
For a newbie, it's a start, but you are absolutely right. The work we are doing with RTCP support will help in this, measuring quality per call. Those of you that wants to test, please go to the bug tracker for What qualify stats are good for is watching over a very long period and comparing several providers. Sure, it's flawed, but three months of data of who was unreachable how many times is very interesting. I'm trying to keep records over *years* now of both local and overseas providers. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] POTS lines vs. using T1 to connect phone services?? HELP
Need help...I need to install a card to terminate 7 lines...I have not order the phone lines yet...I can either do analog lines 1FBs or order a fractional T1...any suggestions on what hardware would be easier to install and configure...also if I went with a T1...do I need an external CSU/DSU or anything or does it just plug into the T1 card...thanks.. Kevin J. Steil Steil Technologies ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Contents of Asterisk-Users digest...
@lists.digium.com asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060202/9d75b053/attachment-0001.htm -- Message: 4 Date: Thu, 02 Feb 2006 20:29:40 +0800 From: Steve Underwood [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Outbound Caller ID number on E1 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed Garth van Sittert wrote: Hi All I am having a problem setting the outbound callerid number on a PRI E1 in South Africa. The outbound number keeps on appearing as the main PRI number. How does it work between Asterisk and the Telko? More importantly how do I get it working? Kind Regards Garth Telcos usually arrange outgoing CLI in one of 3 ways: - a free for all - you can put what you like as your CLI, and no checks are made - a rigid arrangement - no matter what you give as the CLI, the telco will replace it with a fixed value before passing the message on - a constrained arrangement - if you give a CLI within the range that is valid for you, it will be passed on. If you give something which is not allocated to you, the telco wil replace it with a fixed value. Sounds like you do not have the first arrangement. You might have the third, though. It could be you just aren't specifying your number correctly - either the digits themselves or the TON/NPI pair might not be right. Steve -- Message: 5 Date: Thu, 2 Feb 2006 23:30:50 +1100 From: Mark van Kerkwyk [EMAIL PROTECTED] Subject: [Asterisk-Users] Delaying media stream by short period after 183 is sent To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Hi, anyone know of a way that I can delay the RTP stream a little bit once the 183 is sent, I just want to delay it by around 100ms or so for some troubleshooting. Also, I always see a RTP packet before the 183 is sent for each call, it is just a single packet, is something wrong here as my firewall won't open a connection entry until 183 is processed. Mark -- next part -- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060202/595f514c/attachment-0001.htm -- Message: 6 Date: Thu, 2 Feb 2006 07:41:21 -0500 From: Alexander Lopez [EMAIL PROTECTED] Subject: RE: RE: [Asterisk-Users] Blocked Callerid To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii YES Asterisk will support this. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Thursday, February 02, 2006 1:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: RE: [Asterisk-Users] Blocked Callerid I think they have a 1-800 number so you might be right. But the important question is still - will Asterisk support this? PaulH Alexander Lopez [EMAIL PROTECTED] wrote: They are using ANI instead of CallerID. If they have an 800 number thya have the right to know who is calling them because they are paying for the call. the *ANI*DNIS* format is known as Feature Grooup D. Alex From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joe Pukepail Sent: Wednesday, February 01, 2006 3:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Blocked Callerid Do they have an 800 number? If so perhaps their 800 number provider is doing it via DTMF. Search around on the internet, I believe the standard format for the DTMF is *CALLERID*CALLEDNUMBER* (or perhaps reversed). On 2/1/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: I have been discussing an asterisk solution with a company that has a custom written dialogic based solution. The issue is that their dialogic solution can read callerid from incoming calls, even if the callerid is blocked. I have read before that Asterisk can do this, and they want me to make sure that their new system will be able to do this. A quick poke around inside the zaptel source code was unproductive... Any ideas? PaulH
Re: [Asterisk-Users] meetme and dtmf
Kevin P. Fleming wrote:
Accursio Avona wrote:
Step 2: The IAX client make a second call executing again
Dial(ZAP/g1/${EXTEN})
an IVR answer this call and the IAX client have to send some
DTMF stil now everything works very well.
At this point call is transfered to the previous conference
room and The IAX client reach the conference too.
Step 3 The Iax client heve to send some other DTMF to the IVR.
How is the IVR still involved if the call has been transferred into a
conference room?
The IVR records the conversation between the other partecipant to the
conference and wait '#' to stop recording and a '1' to save the file.
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Re: [Asterisk-Users] Call completes but then drops?
Asterisk CVS HEAD built by root on a i686 running Linux on 2005-09-03 01:57:23 UTC It's not an on going issue, just once in a great while someone will make a call, and it goes through, rings for a moment, and then they hang up. I guess I'm more trying to figure out what Feb 1 22:13:43 DEBUG[18623] channel.c: Didn't get a frame from channel: SIP/102-9fda means. On 2/2/06, Kevin P. Fleming [EMAIL PROTECTED] wrote: Matt wrote: Can anyone explain why this call dropped? The person dialed a number, the call WAS completed and connected to the PSTN through a PRI, but they never heard audio and the call was disconnected by Asterisk. Very difficult to guess without any information about your system. If you are using Asterisk 1.2.2, this is a known problem. If not, we'll need a lot more information to be able to even try to help you. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] POTS lines vs. using T1 to connect phone services?? HELP
A fractional T1 is what I would suggest and it is easy to setup and configure. You should only need to plug in the T1 line directly into the T1 Card on the server. The provider will supply the equipment to terminate the line on your premises. On 2/2/06, Kevin Steil [EMAIL PROTECTED] wrote: Need help...I need to install a card to terminate 7 lines...I have not order the phone lines yet...I can either do analog lines 1FBs or order a fractional T1...any suggestions on what hardware would be easier to install and configure...also if I went with a T1...do I need an external CSU/DSU or anything or does it just plug into the T1 card...thanks.. Kevin J. Steil Steil Technologies ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Streaming MOH
Not tried 1.2.4 yet I'm using 1.2.3 and an old version of mpg123 You should be able to use any streaming mp3 that you can find on shoutcast for test. http://www.shoutcast.com Click one of the 'tune in buttons' to download a playlist (pls) file and open in your favorite text editor. Or let it open in your MP3 player and view the properties of the stream. I have several streaming servers here, if you need a test link or want to listen to live air traffic in Detroit Michigan, send me a personal email and I can give you a link for testing. I'd rather not post it here only to end up indexed by google in a few days ;-) Steve Hi, I'm having some problems getting this to work with Asterisk 1.2.4. Does it work for anyone? Does anyone have a site I can test this with? Regards Lee ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Outbound Caller ID number on E1
Hi,
I just had the same problem (see post Setting MSN for outgoing ISDN
calls).
It was very helpful to enable pri debug (pri debug span X). Just try
different values for pridialplan, prilocaldialplan. And also try to do
CallerPres right before the Dial command.
How do you set your CallerID or MSN? I just do:
exten = _X.,1,CallingPres(0)
exten = _X.,2,SetCIDNum(123456)
exten = _X.,3,Dial(Zap/g1/${EXTEN},60, T)
exten = _X.,4,Busy()
123456 is my number without area code. (prilocaldialplan=local).
Hope that helps,
Henry
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~/.signature to help me spread!
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Re: [Asterisk-Users] Regarding cdr_manager.conf
Hi Victor. in /etc/asterisk/modules.conf you MUST have autoload=yes, or better yet, just load what you need INCLUDING the modules cdr_manager.so, you can test if you have it by doing in the asterisk console show modules, if you dont have it, you can load it immediatly from the console doing load cdr_manager.so. If you have it, the new event is Cdr with the following fields: AccountCode Source Destination DestinationContext CallerID Channel DestinationChannel LastApplication LastData StartTime AnswerTime EndTime Duration BillableSeconds Disposition AMAFlags UniqueID UserField RegardsOn 2/2/06, Victor Alvarez [EMAIL PROTECTED] wrote: Hello, My question is.. How does cdr_manager work? Does it suppose to populatecdr-csv/Master.csv? What about the cdr table on the database? What is theevent some people talk about? I have changed (and reloaded) my configuration of cdr_manager.conf to ;; Asterisk Call Management CDR;[general]enabled = yes and it doesn't seem to make any difference. After originate a call from themanager interface my Master.csv is empty, cdr in my database also empty and I don't get any new event apart of Newchannel or Hangup from the managerinterface.An this is all I get from the Asterisk console: -- Reloading module 'cdr_manager.so' (Asterisk Call Manager CDR Backend) == Parsing '/etc/asterisk/cdr_manager.conf': FoundSo..?Kind Regards, Victor.___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on laptop connected to POTS line
On Thu, Feb 02, 2006 at 09:13:29AM -0500, Alexander Lopez wrote: Anyone know of any equipment that I can use to connect a laptop running asterisk to a POTS line (RJ11) ? Look at Xorcom's USB channel Bank. Which is a great product and you should all get one (and the fact that I'm a Xorcom employee has nothing to do with this recommendation), but sadly, still lacks FXO ports. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Streaming MOH
Is what I am looking for the Location of the stream ie (http://ipadd:8043) I Tried that. But I get no audio. Could you please post the url for a working stream but change the IP to 127.0.0.1 so that it doesn't get indexed, but I have an example??? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Gladden Sent: Thursday, February 02, 2006 10:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Streaming MOH Not tried 1.2.4 yet I'm using 1.2.3 and an old version of mpg123 You should be able to use any streaming mp3 that you can find on shoutcast for test. http://www.shoutcast.com Click one of the 'tune in buttons' to download a playlist (pls) file and open in your favorite text editor. Or let it open in your MP3 player and view the properties of the stream. I have several streaming servers here, if you need a test link or want to listen to live air traffic in Detroit Michigan, send me a personal email and I can give you a link for testing. I'd rather not post it here only to end up indexed by google in a few days ;-) Steve Hi, I'm having some problems getting this to work with Asterisk 1.2.4. Does it work for anyone? Does anyone have a site I can test this with? Regards Lee ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Rewind MusicOnHold?
Does anyone know how to rewind the music on hold? Thanks Dan Journo ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Anyone know a good ITSP in Canada that suppo rts *?
Title: Message There are a number of them, try Comwave, Voxipor Wiztel. Depends on what you need we may also provide it... email me privately if you're interested. Some provide IAX, some only SIP, H323, MGCP... -Original Message-From: hugolivude [mailto:[EMAIL PROTECTED] Sent: Thursday, February 02, 2006 7:39 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] Anyone know a good ITSP in Canada that supports *?Hi,I'm looking for a new Internet Telephony Service Provider for my company in Canada to terminate calls from my Asterisk PBX. Ideally I'd like DiDs in Otawa, Toronto, NY San Jose. Anyone out ther who can help me with a recommendation?Vonage seemed clueless when I called them. Broadvoice is good but no Canadian DIDs...Thanks,Hugh DISCLAIMER: This email message may contain information that is confidential, privileged, and for communication only to its intended recipient or recipients. If you have received this message in error, please immediately notify the sender and delete it. courrier lectronique est confidentiel et protg. L'expditeur ne renonce pas aux droits et obligations qui s'y rapportent. Toute diffusion, utilisation ou copie de ce message ou des renseignements qu'il contient par une personne autre que le (les) destinataire(s) dsign(s) est interdite. Si vous recevez ce courrier lectronique par erreur, veuillez m'en aviser immdiatement, par retour de courrier lectronique ou par un autre moyen. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Rewind MusicOnHold?
Turn phone over and shake! :-) MOH plays in a loop, It has no player controls. You can't Pause, Stop, FF,or RW. (gasp!) You can try killing the mpg123 or whatever process you use for MOH. Asterisk should restart it if it needs to. (high-unrecomended) From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dan JournoSent: Thursday, February 02, 2006 10:27 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] Rewind MusicOnHold? Does anyone know how to rewind the music on hold? Thanks Dan Journo ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT O'Reilly Asterisk TFOT
Whilst it can be downloaded I find that a paper copy is easier to read. I bought it for that reason alone. I also find it's a usefull addition to my tool box. I can't always access the net whilst on site. If I get stuck doing something I can look it up in the book. Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com Dave Cotton wrote: I went to the Linux Solutions exhibition in Paris yesterday, visited the well stocked O'Reilly stand and saw a nice pile of Asterisk TFOT, 6 hours later there was only one left. It must say something, also it was the English version. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call completes but then drops?
Ugh.. NPF - It was user error not server error. On 2/2/06, Matt [EMAIL PROTECTED] wrote: Asterisk CVS HEAD built by root on a i686 running Linux on 2005-09-03 01:57:23 UTC It's not an on going issue, just once in a great while someone will make a call, and it goes through, rings for a moment, and then they hang up. I guess I'm more trying to figure out what Feb 1 22:13:43 DEBUG[18623] channel.c: Didn't get a frame from channel: SIP/102-9fda means. On 2/2/06, Kevin P. Fleming [EMAIL PROTECTED] wrote: Matt wrote: Can anyone explain why this call dropped? The person dialed a number, the call WAS completed and connected to the PSTN through a PRI, but they never heard audio and the call was disconnected by Asterisk. Very difficult to guess without any information about your system. If you are using Asterisk 1.2.2, this is a known problem. If not, we'll need a lot more information to be able to even try to help you. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] POTS lines vs. using T1 to connect phoneservices?? HELP
Thanks...just need to see what the cost is...compared to getting 6 lines.. -Original Message- From: Tom Vile [mailto:[EMAIL PROTECTED] Sent: Thursday, February 02, 2006 9:58 AM To: Asterisk User List Subject: Re: [Asterisk-Users] POTS lines vs. using T1 to connect phoneservices?? HELP A fractional T1 is what I would suggest and it is easy to setup and configure. You should only need to plug in the T1 line directly into the T1 Card on the server. The provider will supply the equipment to terminate the line on your premises. On 2/2/06, Kevin Steil [EMAIL PROTECTED] wrote: Need help...I need to install a card to terminate 7 lines...I have not order the phone lines yet...I can either do analog lines 1FBs or order a fractional T1...any suggestions on what hardware would be easier to install and configure...also if I went with a T1...do I need an external CSU/DSU or anything or does it just plug into the T1 card...thanks.. Kevin J. Steil Steil Technologies ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Streaming MOH
I've got it working now but the playback through the handset is sloow. I can tell it's music but you couldn't sing along to it... Still maybe it's about the right speed for a hangover. Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Gladden Sent: 02 February 2006 15:01 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Streaming MOH Not tried 1.2.4 yet I'm using 1.2.3 and an old version of mpg123 You should be able to use any streaming mp3 that you can find on shoutcast for test. http://www.shoutcast.com Click one of the 'tune in buttons' to download a playlist (pls) file and open in your favorite text editor. Or let it open in your MP3 player and view the properties of the stream. I have several streaming servers here, if you need a test link or want to listen to live air traffic in Detroit Michigan, send me a personal email and I can give you a link for testing. I'd rather not post it here only to end up indexed by google in a few days ;-) Steve Hi, I'm having some problems getting this to work with Asterisk 1.2.4. Does it work for anyone? Does anyone have a site I can test this with? Regards Lee ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk on laptop connected to POTS line
Original Message Subject: Re: [Asterisk-Users] Asterisk on laptop connected to POTS line From: Tzafrir Cohen [EMAIL PROTECTED] Date: Thu, February 02, 2006 9:15 am To: asterisk-users@lists.digium.com On Thu, Feb 02, 2006 at 09:13:29AM -0500, Alexander Lopez wrote: Anyone know of any equipment that I can use to connect a laptop running asterisk to a POTS line (RJ11) ? Look at Xorcom's USB channel Bank. Which is a great product and you should all get one (and the fact that I'm a Xorcom employee has nothing to do with this recommendation), but sadly, still lacks FXO ports. If Xorcom could offer something similar with 2-4 FXOs I'd just have to buy at least one. Heck of an idea for a product, a quad FXO adapter interfaced to Asterisk via local USB port. Wow! Michael Graves Sr Product Specialist Pixel Power Inc [EMAIL PROTECTED] o(713) 861-4005 o(800) 905-6412 f(713) 864-8668 c(713) 201-1262 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] limit sip sessions
hi, is there a way to limit the sip session per username?. i mean, if i have a sip session with asterisk using xxx as username, nobody can register with that username until my session is terminated. thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Rewind MusicOnHold?
I thought someone was going to say that. Does anyone know a way to do the following:- 1) Answer incoming call 2) Begin dialing an extension 3) While extension is ringing play a welcome message to the caller 4) Then play MoH until the extension is answered 5) Connect the incoming and outgoing when the extension is answered. Thanks Dan On 02/02/06, Alexander Lopez [EMAIL PROTECTED] wrote: Turn phone over and shake! :-) MOH plays in a loop, It has no player controls. You can't Pause, Stop, FF,or RW. (gasp!) You can try killing the mpg123 or whatever process you use for MOH. Asterisk should restart it if it needs to. (high-unrecomended) From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Dan Journo Sent: Thursday, February 02, 2006 10:27 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] Rewind MusicOnHold? Does anyone know how to rewind the music on hold? Thanks Dan Journo___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Callerid Name
Anyone know why zaptel would ignore a facility message from an ISDN PRI. I am trying to get Callerid name to work. The carrier says it on and I see it in the pri debug but asterisk never gets it. Any help would be appreciated. Thanks John Bittner Simlab.net Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 572/0x23C) (Terminator) Message type: ALERTING (1) [1e 02 81 88]I Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Progress Description: Inband information or appropriate pattern now available. (8) ] -- SIP/69.60.198.130-5119 is ringing Protocol Discriminator: Q.931 (8) len=36 Call Ref: len= 2 (reference 572/0x23C) (Originator) Message type: FACILITY (98) [1c 1d 9f 8b 01 00 a1 17 02 01 01 02 01 00 80 0f 42 55 4c 4b 41 4e 27 53 2c 48 45 41 4c 54 48] Facility (len=31, codeset=0) [ 0x9f, 0x8b, 0x01, 0x00, 0xa1, 0x17, 0x02, 0x01, 0x01, 0x02, 0x01, 0x00, 0x80, 0x0f, 'BULKAN', 0x27, 'S', 0x2c, 'HEALTH' ] -- Processing IE 28 (cs0, Facility) Handle Q.932 ROSE Invoke component ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] return code from AGI
Hello friends, Asterisk applications like Dial and other commands return codes. When AGI script is executed, it returns -1 on hangup and 0 on non hangup exit. How do I check these return codes from the extensions.conf . I want to check these return codes and control the dialplan. Please help me how do I track this. Thanks all for reading this mail. With warm regards. Vivek J. Joshi. [EMAIL PROTECTED] Trikon electronics Pvt. Ltd. --Sweat saves blood, blood saves lives, and brains saves both. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Euro-ISDN
[EMAIL PROTECTED] is believed to have said: [setup tool] Sorry, I cannot answer that one. I don't know enough about these cards and their drivers. Armin, thanks alot. One has to do some research and experimentation on his own every now and then; and see if there is anything interesting that might even end up in the wiki... ;-) So in the end there a lot of reasons to go for a 'better' card. Yes, a lot reasons. But actually, it depends on what you need and what you want to do. You are right.. I was asking about ISDN cards to see if there was any simple integration path for some more expensive units (where one might use those with ISDN interfaces first). But now I'll have to fiddle with the newly arrived GSM-gateway first. And this is a nice, little analog unit. Thank you very much, Aldo ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Euro-ISDN
[EMAIL PROTECTED] is believed to have said: With BRIstuff you get to use ztcfg, etc. Cannot say anything about mISDN, CAPI... Francesco, thank you; this is important to know Aldo ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Directed Call Pickup
Hi All I am having problems with Directed Call Pickup in Asterisk 1.2.1 If extension 100 is ringing, a user at another extension is supposed to be able to dial *8100 and pickup the call to 100. It isn't working for me and I cannot figure out why. I have in features.conf: pickupexten = *8 Kind Regards Garth -- Garth van Sittert BSc (Physics Computer Science) - Mobile: +27 (0)83 791 6662 Email: [EMAIL PROTECTED] Phone: 08600 BITCO Web:www.bitco.co.za ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Callerid Name
Look at http://bugs.digium.com/view.php?id=1192 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Bittner Sent: Thursday, February 02, 2006 11:27 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Callerid Name Anyone know why zaptel would ignore a facility message from an ISDN PRI. I am trying to get Callerid name to work. The carrier says it on and I see it in the pri debug but asterisk never gets it. Any help would be appreciated. Thanks John Bittner Simlab.net Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 572/0x23C) (Terminator) Message type: ALERTING (1) [1e 02 81 88]I Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Progress Description: Inband information or appropriate pattern now available. (8) ] -- SIP/69.60.198.130-5119 is ringing Protocol Discriminator: Q.931 (8) len=36 Call Ref: len= 2 (reference 572/0x23C) (Originator) Message type: FACILITY (98) [1c 1d 9f 8b 01 00 a1 17 02 01 01 02 01 00 80 0f 42 55 4c 4b 41 4e 27 53 2c 48 45 41 4c 54 48] Facility (len=31, codeset=0) [ 0x9f, 0x8b, 0x01, 0x00, 0xa1, 0x17, 0x02, 0x01, 0x01, 0x02, 0x01, 0x00, 0x80, 0x0f, 'BULKAN', 0x27, 'S', 0x2c, 'HEALTH' ] -- Processing IE 28 (cs0, Facility) Handle Q.932 ROSE Invoke component ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Rewind MusicOnHold?
Hi Dan Have a look at setting up queues. Kind Regards Garth Dan Journo wrote: I thought someone was going to say that. Does anyone know a way to do the following:- 1) Answer incoming call 2) Begin dialing an extension 3) While extension is ringing play a welcome message to the caller 4) Then play MoH until the extension is answered 5) Connect the incoming and outgoing when the extension is answered. Thanks Dan On 02/02/06, *Alexander Lopez* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Turn phone over and shake! :-) MOH plays in a loop, It has no player controls. You can't Pause, Stop, FF, or RW. (gasp!) You can try killing the mpg123 or whatever process you use for MOH. Asterisk should restart it if it needs to. (high-unrecomended) *From:* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]] *On Behalf Of *Dan Journo *Sent:* Thursday, February 02, 2006 10:27 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [Asterisk-Users] Rewind MusicOnHold? Does anyone know how to rewind the music on hold? Thanks Dan Journo ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/ -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Garth van Sittert BSc (Physics Computer Science) - Mobile: +27 (0)83 791 6662 Email: [EMAIL PROTECTED] Phone: 08600 BITCO Web:www.bitco.co.za ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Rewind MusicOnHold?
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dan JournoSent: Thursday, February 02, 2006 11:18 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Rewind MusicOnHold? I thought someone was going to say that. Does anyone know a way to do the following:- 1) Answer incoming call exten = s,1,Answer() 2) Begin dialing an extension 3) While extension is ringing play a welcome message to the caller Here you got a problem. What do you do if callee picks up too fast. so I would exten = s,2.PlayBack(mesage-to-caller0 exten = s,3,Dial(SIP/123||m) 4) Then play MoH until the extension is answered 5) Connect the incoming and outgoing when the extension is answered. Thanks Dan On 02/02/06, Alexander Lopez [EMAIL PROTECTED] wrote: Turn phone over and shake! :-) MOH plays in a loop, It has no player controls. You can't Pause, Stop, FF,or RW. (gasp!) You can try killing the mpg123 or whatever process you use for MOH. Asterisk should restart it if it needs to. (high-unrecomended) From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Dan JournoSent: Thursday, February 02, 2006 10:27 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] Rewind MusicOnHold? Does anyone know how to rewind the music on hold? Thanks Dan Journo___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Directed Call Pickup
On Thursday 02 Feb 2006 16:46, Garth van Sittert wrote: Hi All I am having problems with Directed Call Pickup in Asterisk 1.2.1 If extension 100 is ringing, a user at another extension is supposed to be able to dial *8100 and pickup the call to 100. It isn't working for me and I cannot figure out why. I have in features.conf: pickupexten = *8 At the CLI, show features should tell you if it is configured. If so, you need to tell us what happens on the console. If not, then you are liable to get asked my car does not work, does anyone know why?. B -- http://www.mailtrap.org.uk/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Rewind MusicOnHold?
The reason we want to do it this way, is that we'd like to start dialing at the beginning, so that when the message finishes playing, the caller has actually already waited 10 seconds, leaving 10 seconds before the call is answered. Thanks Dan On 02/02/06, Alexander Lopez [EMAIL PROTECTED] wrote: From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED]] On Behalf Of Dan JournoSent: Thursday, February 02, 2006 11:18 AM To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Rewind MusicOnHold? I thought someone was going to say that. Does anyone know a way to do the following:- 1) Answer incoming call exten = s,1,Answer() 2) Begin dialing an extension 3) While extension is ringing play a welcome message to the caller Here you got a problem. What do you do if callee picks up too fast. so I would exten = s,2.PlayBack(mesage-to-caller0 exten = s,3,Dial(SIP/123||m) 4) Then play MoH until the extension is answered 5) Connect the incoming and outgoing when the extension is answered. Thanks Dan On 02/02/06, Alexander Lopez [EMAIL PROTECTED] wrote: Turn phone over and shake! :-) MOH plays in a loop, It has no player controls. You can't Pause, Stop, FF,or RW. (gasp!) You can try killing the mpg123 or whatever process you use for MOH. Asterisk should restart it if it needs to. (high-unrecomended) From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Dan Journo Sent: Thursday, February 02, 2006 10:27 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] Rewind MusicOnHold? Does anyone know how to rewind the music on hold? Thanks Dan Journo___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk on laptop connected to POTS line
Hello All , On Thu, 2 Feb 2006, [EMAIL PROTECTED] wrote: Original Message Subject: Re: [Asterisk-Users] Asterisk on laptop connected to POTS line From: Tzafrir Cohen [EMAIL PROTECTED] Date: Thu, February 02, 2006 9:15 am To: asterisk-users@lists.digium.com On Thu, Feb 02, 2006 at 09:13:29AM -0500, Alexander Lopez wrote: Anyone know of any equipment that I can use to connect a laptop running asterisk to a POTS line (RJ11) ? Look at Xorcom's USB channel Bank. Which is a great product and you should all get one (and the fact that I'm a Xorcom employee has nothing to do with this recommendation), but sadly, still lacks FXO ports. If Xorcom could offer something similar with 2-4 FXOs I'd just have to buy at least one. Heck of an idea for a product, a quad FXO adapter interfaced to Asterisk via local USB port. Wow! If one could get this in 1-3 FXO 1-3FXS ports(*) in an apropriate combination ... Where the USER can select which combo s/he wants at home , Not by buying a hardwired device . Then that would be something to buy . (*) 1FXO 3FXS , 2FXO 2FXS , 3FXO 1FXS . My $.02 , JimL -- +--+ | James W. Laferriere | SystemTechniques | Give me VMS | | NetworkEngineer | 3542 Broken Yoke Dr. | Give me Linux | | [EMAIL PROTECTED] | Billings , MT. 59105 | only on AXP | | http://www.asteriskhelpdesk.com/cgi-bin/astlance/r.cgi?babydr | +--+ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Rewind MusicOnHold?
1) Answer incoming call exten = s,1,Answer() 2) Begin dialing an extension 3) While extension is ringing play a welcome message to the caller Here you got a problem. What do you do if callee picks up too fast. In my situation, the caller wont pickup too fast. The message is 10 seconds long, and the shortest time for the person to answer is around 20 seconds. It will never be less. Thanks Dan On 02/02/06, Alexander Lopez [EMAIL PROTECTED] wrote: From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED]] On Behalf Of Dan JournoSent: Thursday, February 02, 2006 11:18 AM To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Rewind MusicOnHold? I thought someone was going to say that. Does anyone know a way to do the following:- 1) Answer incoming call exten = s,1,Answer() 2) Begin dialing an extension 3) While extension is ringing play a welcome message to the caller Here you got a problem. What do you do if callee picks up too fast. so I would exten = s,2.PlayBack(mesage-to-caller0 exten = s,3,Dial(SIP/123||m) 4) Then play MoH until the extension is answered 5) Connect the incoming and outgoing when the extension is answered. Thanks Dan On 02/02/06, Alexander Lopez [EMAIL PROTECTED] wrote: Turn phone over and shake! :-) MOH plays in a loop, It has no player controls. You can't Pause, Stop, FF,or RW. (gasp!) You can try killing the mpg123 or whatever process you use for MOH. Asterisk should restart it if it needs to. (high-unrecomended) From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Dan Journo Sent: Thursday, February 02, 2006 10:27 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] Rewind MusicOnHold? Does anyone know how to rewind the music on hold? Thanks Dan Journo___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] POTS lines vs. using T1 to connectphoneservices?? HELP
Kevin, Are you in the US? If so then you've probably got several carriers to choose from. In my experience analog lines have a flat expense of $20-$25 per month. That equates to about $140-$175 per month in flat fees, plus you have usage on top of that. (Your experience may vary.) I am currently experimenting with a company out of NY called Digizip (www.digizip.com) that sold me a Qwest PRI for about $150/month flat fee plus usage in the neighborhood of $.015 per minute. (Month-to-month term, no contract!) A PRI like this is attractive because you have the capability of having 23 simultaneous conversations, plus you can do DID. One drawback is the inability to do a Centrex transfer (aka DID to DOD transfer or off net transfer) but that usually isn't a big deal. One other note: in the US it is considered a legal requirement to have a CSU on any T1 circuit. However, it is not technically necessary. Also, some terminating equipment has the CSU built right in - e.g. Cisco T1/CSU WIC for their routers. I'm running 12 different T1 circuits, each with a CSU. I like having the CSU for testing and monitoring line conditions. If you go with a PRI (or any other T1-style circuit) then it's just matter of getting the right card for your system. The Digium and Sangoma cards have fine reputations for use in production machines. The advanced (and more expensive) models have echo cancellation built in. I'm currently using a knock-off of the older Tormenta2 Zapata card but that's only for testing. In a production environment I'll mostly likely upgrade to a better card. HTH and sorry for the ramble! -MC -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin Steil Sent: Thursday, February 02, 2006 7:46 AM To: Asterisk User List Subject: RE: [Asterisk-Users] POTS lines vs. using T1 to connectphoneservices?? HELP Thanks...just need to see what the cost is...compared to getting 6 lines.. -Original Message- From: Tom Vile [mailto:[EMAIL PROTECTED] Sent: Thursday, February 02, 2006 9:58 AM To: Asterisk User List Subject: Re: [Asterisk-Users] POTS lines vs. using T1 to connect phoneservices?? HELP A fractional T1 is what I would suggest and it is easy to setup and configure. You should only need to plug in the T1 line directly into the T1 Card on the server. The provider will supply the equipment to terminate the line on your premises. On 2/2/06, Kevin Steil [EMAIL PROTECTED] wrote: Need help...I need to install a card to terminate 7 lines...I have not order the phone lines yet...I can either do analog lines 1FBs or order a fractional T1...any suggestions on what hardware would be easier to install and configure...also if I went with a T1...do I need an external CSU/DSU or anything or does it just plug into the T1 card...thanks.. Kevin J. Steil Steil Technologies ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call completes but then drops?
Matt wrote: I guess I'm more trying to figure out what Feb 1 22:13:43 DEBUG[18623] channel.c: Didn't get a frame from channel: SIP/102-9fda means. Pretty much what it says... the SIP endpoint dropped its end of the call and the Asterisk channel was hung up as a result. Given the sheer number of bugfixes that have been made since you got that code, I would suggest that you are wasting time trying to debug this without upgrading :-) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Rewind MusicOnHold?
The native MOH type will, unless set to random=yes, play the music files in the same order as they appear with an ls of the directory. (someone, anyone, back me up here?) I would place the greeting in the same MOH class as your actual music, and name the file of the greeting something less than the filename of the music file. Additionally, to avoid repeating the greeting, should the music file play all the way through before an answer, you may want to make additional copies of the music file, named something greater than the greeting file. Just a thought, never tried to do it myself. Sincerely, Brent A. Torrenga [EMAIL PROTECTED] Torrenga Engineering, Inc. 907 Ridge Road Munster, Indiana 46321-1771 219.836.8918x325 Voice 219.836.1138 Facsimile www.torrenga.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk at SCALE 4x
Hello, Asterisk will have strong presence at SCALE 4x, the 2006 Southern California Linux Expo next week. On the exhibit hall floor both Digium and SwitchVox will have booths demonstrating asterisk and related products. The event will be held on Feb 11th and 12th at the Los Angeles Airport Radisson. In addition to Asterisk focused sponsors, we will have 3 talks on the topic of Asterisk and open-source VoIP: * Mark Spencer (Digium) - IP Communication: Open for Business * David Mandelstam (Sangoma) - It's a whole new world -- open source at the PBX, ready for prime time * Tim Fritchel - Case Study Switching from Motorola to Asterisk Other speakers include: Hans Reiser, Chris Dibona, Andi Gutmans and more.. For further details see the conference website at: http://www.socallinuxexpo.org Those interested in attending the show can use the promo code AST06 to get 40% off full access passes. (http://www.socallinuxexpo.org/order/) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] meetme and dtmf
Accursio Avona wrote: The IVR records the conversation between the other partecipant to the conference and wait '#' to stop recording and a '1' to save the file. Then I really don't understand at all... this is not functionality that I would call an 'IVR'. Can you show us the portions of the Asterisk dialplans that are involved here? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Directed Call Pickup
Same problem for me. Direct call pickup doesn't work. Global pickup is OK. This is 'show features' output: show features Builtin Feature Default Current --- --- --- Pickup*8 *8 Blind Transfer# #1 Attended Transfer #2 One Touch Monitor *1 Disconnect Call * * Dynamic Feature Default Current --- --- --- (none) Call parking Parking extension : 700 Parking context : parkedcalls Parked call extensions: 701-720 'show modules' says that app_directed_pickup.so is loaded: app_directed_pickup.so Directed Call Pickup Application 0 Then I have also: show application Pickup asterisk1*CLI -= Info about application 'Pickup' =- [Synopsis] Directed Call Pickup [Description] Pickup([EMAIL PROTECTED]): This application can pickup any ringing channel that is calling the specified extension. If no context is specified, the current context will be used. Any help? Thanks Mimmus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ISDN Eicon Diva Server V-BRI
Dear all, I'm planning to buying Eicon Diva Server V-BRI for my asterisk server and run with chan_capi. Is anybody using that card ? Would appreciate any feedback. Bartosz ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sip - no peer or user found on incoming call
Hi list, I try to connect to a GW which have one domain eg sip.mydomain.com and have few IPs related to this domain. I register * to this domain with host=sip.mydomain.com and type=user. So DNS will decide on which IP of my domain I will register (or redirection on the GW side). If an incoming call arrive, I would guess that, as type=user, it will not try to match the IP from INVITE as I want to match on username. But this is not true, I always have in logs Found no matching peer or user for 'xxx.xxx.xxx.xxx:5060' and asterisk then try to find a MyUserName extension in the SIP default context. I tried to play with deny/permit without luck. The call is finishing properly _only_ when the IP which with my * is registred to the GW match this from the incoming call, and then doesn't matter if type=user or type=peer, which is normal according to http://www.voip-info.org/wiki/view/Asterisk+SIP+user+vs+peer. I'm running Asterisk SVN-trunk-r8643M built by root @ keewi on a i686 running Linux on 2006-01-25 14:50:51 UTC Here is relevant part of my sip.conf register = MyUserName:MySecret@sip.mydomain.com/MyUserName [IN-UserName] type=user username=MyUserName fromuser=MyUserName fromdomain=MyFromDomainName secret=MySecret context=incoming-GW ;deny=0.0.0.0/0.0.0.0 ;permit=xxx.xxx.xxx.xx0/32 ;permit=xxx.xxx.xxx.xx1/32 ;permit=xxx.xxx.xxx.xx2/32 ;permit=xxx.xxx.xxx.xx3/32 ;permit=xxx.xxx.xxx.xx4/32 ;permit=xxx.xxx.xxx.xx5/32 host=sip.mydomain.com ;insecure=invite,port ;very ;nat=yes ;canreinvite=no ;qualify=1000 disallow=all allow=g726 Thanks for any clue. -- Daniel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Default value for ASTERISK_VERSION_NUM
Leo Ann Boon wrote: /* * version.h * Automatically generated */ #define ASTERISK_VERSION 1.2.4 #define ASTERISK_VERSION_NUM 00 This was a bug in the Makefile; it has been corrected in Subversion and will part of the 1.2.5 release. Sorry for the inconvenience. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Events when the target answer
Hi Group, I am developing a application, this use Manager API to connect with Asterisk. But when I call to an external number (over a zap channel), I dont receive any event when the target answer, Who can help me?, Which event notify me that the phone call was answered? Thank you. Ezequiel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN Eicon Diva Server V-BRI
On Thu, 2 Feb 2006, Bartosz Jozwiak wrote: Dear all, I'm planning to buying Eicon Diva Server V-BRI for my asterisk server and run with chan_capi. Is anybody using that card ? Would appreciate any feedback. I have the non-V version of that card multiple times in use with perfect results. Do you need specific information? Armin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 5,000 concurrent calls system rollout question
Hi, Joash, thank you for your email. I was very relieved to hear that someone was already doing this. Can you please tell me more about your test? Why did you test it in a first place? For me, we need to come up with a system that needs to: 1. Handle 5,000 inbound SIP calls 2. offer IVR capability 3. Billing I thought that Asterisk would be up to the task, but, I am not sure as to: 1. How many servers should I consider? 4? 10? Obviously, we will be talking about probably core Xeon servers if this is what we need. 2. How hard would it be to implement? 3. How bad is g729 quality? 4. IVR : if the call is SIP, can we do prompts without transcoding? Any other suggestions that you might have would really be appreciated. Joash Herbrink [EMAIL PROTECTED] wrote: I have tested an asterisk server with over 5000 concurrent calls. The system setup was a P4 HT 3Ghz, 4 Gb RAM, and 1 gbps Ethernet connection on a cisco 3560 switch. This works, but puts some serious stresses on the system. Why don't u considered using g.729 codec, this will at least lower the bandwidth consumption significantly, and, you can overcome the CPU resource issue by just using a server grade multi CPU xeon server. I would never the less still connect the system via 2 ethernet connections, just for some redundancy, as mentioned before in this thread. Bandwidth should be about 24 kbps (half duplex) per call So, 5000 * 24 is roughly 120 mbps, so a gigabit Ethernet should do just fine. Joash -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dustin Wildes Sent: Wednesday, February 01, 2006 8:54 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] 5,000 concurrent calls system rollout question Dinesh Nair wrote: On 02/01/06 09:29 Damon Estep said the following: Ok, now lets go for 5000 of them. 160kbps*5000=80kbps or 800mbps - full duplex. Have you ever seen a NIC or switch that can run GigE full duplex at 80% utilization and not at least start to fall apart? additionally, 5000 simultaneous SIP calls at 20ms intervals will send, 5,000 * 50 * 2 = 500,000 packets per second (full duplex). not too many boxes can handle such packet load, in spite of the relatively small packet sizes. Why not bond multiple NICs together to do a load balance output? Would provide redundancy as well. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 5,000 concurrent calls system rollout question
Hi, several of your mentioned signant as a viable option. Has anyone ever used them? Are there any reviews for their products? Did they just put together a lot of Asterisks into a large scale PC? (I am still struggling with the concept) Thanks, Vic ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
