[Asterisk-Users] matching sip connection under sip.conf
Anybody know how to match under sip.conf and cisco 53xx ? It looks like due to dynamic port number, it is not able to authorize it. Here is what I get under debug Using latest request as basis request Sending to 216.236.160.15 : 5060 (non-NAT) Found no matching peer or user for '216.236.160.15:53182' It looks like the port number is changing and that is why the * can not recognize it. Is there way to get around to this? My sip.conf [216.236.160.15] type=friend username=696 fromuser=696 host=216.236.160.15 context=from-pstn Thanks ** C.M. Rahman Jr. IT Manager CCNP, MCSE SecuritySecure your self by securing your System CompTI Security Plus Certified CCS Internet http://www.ccsi.com 13706 Research Blvd. Suite 100 Austin, TX 78750 Tel: 512-257-2274 Ex: 115 \ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] g729 passthrough?
I got some advice from Josh Colp that has helped with some of my problem: it may have a little logic flaw in the way transcoding is supposed to be done, from the way your message is I would say you are getting hit by this. (Upgrading to latest CVS head will fix it) but one solution is to be the following in asterisk.conf in /etc/asterisk [options] transcode_via_sln = no Thatll cause it to bridge the two and not try to transcode through signed linear. Enjoy! Well that worked, after a fashion. Now AS LONG AS I ONLY USE G.729 ONLY things are fine. But the 841 does all kinds of codecs, and so I'd like it to use g.726 to talk to a provider that doesn't speak g.729. So I set the sip.conf for the phone to disallow=all; allow=g726,g729 and then try to connect to the g726-only server: Apr 25 00:58:42 NOTICE[5839]: channel.c:1833 set_format: Unable to find a path from g729 to g726 After playing with this for as long as I could stand to, it appears that IFF I am talking to a g729-only endpoint and I set the SIP phone to use g729 only, things are fine. Once I deviate from that (unfortunately restrictive) setup, I can't seem to do anything. In other words, if g729 is in the mix it seems to always choose it despite my preferences, and things get hosed. I'd love to hear from someone who has conquered this. Thanks. B. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] PA168 ip phone setup iax2 to LiveVoip
Hi Charl, Im sorry to tell you a disappointing comment about the PA168 IP Phone, I have here too such like that and it's a crap... It works in a while, sometimes it can even send eight digits ( I mean it fail after dialing around 4 to 5 digits). -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C W Nel Sent: Saturday, April 23, 2005 9:18 PM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] PA168 ip phone setup iax2 to LiveVoip Can anyone PLEASE help to get a pa168 ip phone connected to livevoip? If I set use service it does not work. If I unset it, it works for a while, then just busy tone. If I set and unset use service, it will again work for a while. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.10.2 - Release Date: 21/04/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.10.2 - Release Date: 4/21/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Quantum A800 (SIP) - Asterisk Config
Hi Basher, Currently im using my A800 Quintum registered in my Asterisk SIP server. For you to register your Quitum to Aterisk, define your asterisk in as proxy and registrar IP at SIP config at quintum (I can give you the sample config at private mail). Also setup a user account at sip_additionl.conf in your asterisk defining the username and password used and defined in the quantum. Here is the sample in the asterisk: [199] username=199 type=friend secret=199 qualify=no port=5060 pickupgroup= nat=never mailbox= host=xxx.xxx.xxx.xxx (IP add of your quintum) dtmfmode=rfc2833 disallow= context=from-internal canreinvite=no callgroup= callerid=Caller ID 199 allow= Shall you have more question, your free to ask. In return I want to ask also if have you tried to managed to register a D-Link DVG-1402S. I have here a demo unit but I can make it work... I am not sure what is missing.. anyone here in asterisk users-list? Regards, Jessie -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bashir Ullah - www.Lamsre.Com Sent: Monday, April 25, 2005 3:36 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Quantum A800 (SIP) - Asterisk Config Hi Is there any help for me to register my quantium A800 (SIP) with my Asterisk . Please help me what should me my Sip.conf now present i did [1234567] type=friend context=sip username= secret= nat=yes host=dynamic canreinvite=no defaultip=XXX.XXX.XXX.XXX disallow=all allow=g729 allow=gsm allow=g723.1 allow=ulaw and is there any special change need on quintum? Bashir ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.10.2 - Release Date: 4/21/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Playing mp3's while recording voicemail
Yeah the idea: its like a karaoke conversation between people via voicemail thats posted on a website as an audio thread under a creative commons licence. Base requirements: -record: pick up, and reply to recordings. --listening to music while recording --having that music mixed with the recording --background music specified by website --a web listing of audio threads So far we have built an * box played around with iax2 trunks, voicemail, IVRs, vmail.cgi and mysql integration. Out of the box these dont fit. Not sure if we need to adapt theses or find other ways of doing it. Need direction on what we need to do next. Going to play with http://www.voip-info.org/wiki-Asterisk+tips+wiki (audio wiki) and using monitor / mp3player next? Ideas please? This is a non-commercial 'arts' project as a registered charity working with youth music groups. (currently applying for funding as our dev skills are limited) Rafal Kaniewski Rafal#movingimagearts.com -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Race Vanderdecken Sent: 22 April 2005 17:09 To: 'Rod Bacon'; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Playing mp3's while recording voicemail Very Curious, As a developer and Big Idea person I would like to know more about this. I am kinda curious about the singing part. Is this karaoke via asterisk? -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.10.2 - Release Date: 21/04/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] No busy tone when dialing out over ISDN with Polycom 500 IP
Hi, what do i have to configure to get a busy tone when dialing out over ISDN channel with my Polycom 500 IP? Kib ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] NEED HELP PROGRAMING ASTERISK VoIP NETWORK
-Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Jean-Michel Hiver Envoyé : lundi 25 avril 2005 07:54 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [Asterisk-Users] NEED HELP PROGRAMING ASTERISK VoIP NETWORK Franz wrote: Please contact me Urgent... Hi Frantz, I can do custom programming. Here is some information about my company: http://ykoz.net/intl/ Let me know what you're after and I'll send you a preliminary quote. Cheers, Jean-Michel. -- Ykoz Un Max - La VoIP en pré-payé! Essayez gratuitement - 5 crédits offerts. --- http://ykoz.net/voip/max --- Hi Can you please do advertising for your company in Asterisk-Biz Aniway where are the legal notices and the RCS on you'r web site Seem's to be quit strange Thierry Wehr ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zap event On hook(1) handling problem
i am using X100P on RHEL4, all incoming calls doing well, during any outbound call from sip to pstn, it hangup right away when the remote side pick up the phone. i've been trying to trace out this problem for 2days. for the log snapshot below, DEBUG[2401]: Exception on 15, channel 1 DEBUG[2401]: Got event On hook(1) on channel 1 (index 0) the On hook event always happens when the remote user pick up the phone. that's mean when i doing outbound call and the remote user did not pick up the phone(that is the phone keep ringing) it won't drop off. to my understanding on hook should mean the remote side pick up the phone. this On hook event should be handled correctly by the hardware right? but then asterisk drop the connection right away. i have no problem running with the same hardware on centos 3(kernel 2.4) do you think it's related to any asterisk problem on kernel 2.6.9(RHEL4)? -vince DEBUG[2401]: Exception on 15, channel 1 DEBUG[2401]: Got event Hook Transition Complete(12) on channel 1 (index 0) DEBUG[2401]: Exception on 15, channel 1 DEBUG[2401]: Got event Dial Complete(9) on channel 1 (index 0) DEBUG[2401]: No echocancellation requested DEBUG[2401]: Dropping duplicate answer! VERBOSE[2401]: -- Zap/1-1 answered SIP/168-9dbb DEBUG[2401]: Ooh, format changed from unknown to ulaw DEBUG[2401]: Stopping retransmission on '[EMAIL PROTECTED]' of Response 10010: Found DEBUG[2401]: Exception on 15, channel 1 DEBUG[2401]: Got event On hook(1) on channel 1 (index 0) DEBUG[2401]: Didn't get a frame from channel: Zap/1-1 DEBUG[2401]: Bridge stops bridging channels SIP/168-9dbb and Zap/1-1 DEBUG[2401]: Hangup: channel: 1 index = 0, normal = 15, call wait = -1, thirdcall = -1 DEBUG[2401]: Set option TDD MODE, value: OFF(0) on Zap/1-1 DEBUG[2401]: Updated conferencing on 1, with 0 conference users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: How to prevent native bridging between SIP channels
Marc Storck [EMAIL PROTECTED] writes in reply to my question: add canreinvite=no to the sip user definition blocks for the SIP provider and for the SIP ATA. Regards, Unfortunately, I already have this parameter in the sip user definitons, as well as a t option in the Dial command, both of which, according to the article on SIP Media Path in the Asterisk-Wiki, should prevent Asterisk from trying to take itself out of the loop. But it still does :-( On the other hand, the same article says that Asterisk decides whether or not to take itself out of the media path depends on many variables -- I was hoping to get some information on some of the _other_ variables, in addition to the canreinvite=no and the transfer option to the Dial command. Regards, Wolf ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] signaling during a call
I am using Asterisk with SIP phones. is it possible to press a key during a conversation and get asterisk to do something? Like the # key, but I would like asterisk to take other actions instead of transfering. tulika _ Your [EMAIL PROTECTED] Spaces! http://www.msn.co.in/spaces Blogs, albums, music lists. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] [ANNOUNCEMENT] Amatix InstantPBX
Description Amatix will instantly transform your computer in a small PBX. You don't have to install any software, just plug the Amatix CD in your CD drive and let the computer boot from it. In few minutes you will get a running Linux system with a configured Asterisk PBX. Highlights * Amatix will automatically detect your telephony hardware and will configure the Asterisk PBX accordingly. You can use analog or ISDN trunks. * There are 4 preconfigured VoIP extensions using SIP. * Amatix is a LiveCD Debian-Linux distribution based on Morphix. It is running completely from CD and your disk will be not modified. * Amatix is customisable. You can modify the default settings and store your changes on the floppy disk or on the hard disk. The changes will be loaded on boot. Telephony hardware This version of the Amatix InstantPBX will try to configure the following card types: * Analog - X100P (using the zaptel driver and chan_zap) * ISDN - AVM Fritz! PCI (using the AVM's CAPI4Linux driver and chan_capi) - HFC-S (using the mISDN drivers and chan_misdn) More details about Amatix InstantPBX can be found at http://amatisoft.homelinux.com/amatix.html For questions please use http://amatisoft.homelinux.com/contact.html __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] i like my colors, thanks..
[EMAIL PROTECTED] wrote: On Thu, 2005-04-21 at 08:39 -0500, Matthew Boehm wrote: Using most recent CVS-HEAD and my terminal keeps changing colors. I'm using vt100 terminal emulation. How can I turn off asterisk's colors? Or at least turn off the black background. My normal terminal is white background, black font. But for some reason, asterisk is changing it to white font, black background. Add '-n' to your command line. 'asterisk -h' will print out a list of all of the command line switches that it supports. On the (admittedly relatively old version we're using) this will only work when I'm logging in via SSH. When working from the console -n doesn't work. -- Andreas SikkemaRits tele.com Van Vollenhovenstraat 33016 BE Rotterdam t: +31 (0)10 2245544 f: +31 (0)10 413 65 45 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asteriks without h/w
Hi Iam new to asteriks, i juz installed it on my system and also got hold of diax(on a windows client) to call the asteriks server,now before buying any diguim hardware i want to test asteriks by making both the computers talk. I dont have any kind of h/w now, i need help from u guys to make these two computers in the same n/w talk without any hardware Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can Asterisk do the following for me ?
You know, that's exactly what I was looking for since the beginning! Unfortunately I only found one of these items for sale in the US and even then I'm not sure if it will be compatible with the European system! Maybe someone can enlighten me once and for all as far as the differences between North America/Europe in telephony. In any case, I already ordered 2 X100P cards which should be arriving in 1 week 1/2. This Asterisk software looks very promising and I might as well build a small Home Office PBX with different extensions! Another stupid question now: anyone knows who does the voices in all these nice systems ? Like, Welcome to Mycompany, for sales press 1, for support press 2 Thanks! Hello, that is even possible without MODEM hardware. It should work with a simple call forwarder/diverter. It connects to both line ends and works more or less like a analogue 2-port pbx with a fixed programmable forwarding number. Offered e.g. in Germany from AUERSWALD (A-BOX) http://www.auerswald.de/int/products/auerswald_box/box_intro.htm or at EBAY ...like here http://cgi.ebay.de/ws/eBayISAPI.dll?ViewItemitem=6386901484 No modems or VoIP equipment except the ATA is needed at all for this ... regards, Jürgen Hey guys, I am aware that Asterisk may be a bit overkill for what I need but I haven't found any other software. Here's what I need to do: I have 1 computer with 2 modems in it. Each modem (regular 56k )is plugged into a different phone line ( line A and line B ). Whenever a call comes on line A, a software application should automatically dial a fixed number on line B and form a connection between the two ends. In other words: call comes into modem, software dials a fixed number on second line, makes the connection and it works as if the caller dialed the end number. Why do I need this ? I currently use Vonage in an European country so that my North American friends can call me localy. The problem is that this North American phone number is only available at home and not when I'm outside, travelling, etc. Using call forwarding would require me to set up Vonage to forward calls to an international number and thus it will cost me extra! But, if I can manage to get the incoming Vonage call into a computer, then have the computer dial my local cell phone number and patch the incoming call I would have access to incoming North American calls everywhere and much cheaper too! Notice I only want this to happen one way, in the direction I described and not the other way around! So..does anyone know if Asterisk can do this, or another ( simpler ) software ? Also, would it work with regular 56k modems ? P.S. Only a voice call would come into Asterisk, no VoIP stuff and only voice should go out ( transit the system ) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NEED HELP PROGRAMING ASTERISK VoIP NETWORK
Hi Can you please do advertising for your company in Asterisk-Biz Sorry, I did use the 'reply to sender' functionality but this mailing list is utterly broken because it replaces the reply-to with the list address. I was just replying to the poster. I am very sorry for the inconvenience. Aniway where are the legal notices and the RCS on you'r web site Seem's to be quit strange Hey that's a good idea, I'll stick them on the /contact/ page. Thanks. Best Regards, Jean-Michel. -- Ykoz Un Max - La VoIP en pré-payé! Essayez gratuitement - 5 crédits offerts. --- http://ykoz.net/voip/max --- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Can Asterisk do the following for me ?
In article [EMAIL PROTECTED], [EMAIL PROTECTED] wrote: Another stupid question now: anyone knows who does the voices in all these nice systems ? Like, Welcome to Mycompany, for sales press 1, for support press 2 Allison Smith. See http://www.digium.com/index.php?menu=thevoice and http://www.theivrvoice.com/ Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco's description of echo
Another common problem that causes echo in networks is not setting your loss plan correctly. You need to be sure that you aren't coming in too hot at any of your analog interfaces. In general you should see a signal between -20dbm and -12dbm when someone is talking on the line. If it is significantly hotter then you run the chance of having a larger reflected signal resulting in echo. I typically try padding down analog levels by 3dB at a time to see if echo is reduced. How do you measure the amplitude of a pstn line? As an audio engineer in a previous life, I would love to be able to send standard level tones down a pstn line and measure the amplitude at my end, then adjust the input gain accurately. Is there a way to do this? Chris Mason US Number: (646)722-0001 US Fax (815)301-9759 Skype: netconcepts ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] each 64K channel's ABCD bits for E100P Digium Cards.
Hi ! I am trying to configure a E100P card with Channel bank, but I am a bit confused witheach 64K channel's ABCD bits. The * will be connected to a PSTN switch with E1 Channel Bank lines. The E1 lines will be used for incoming calls as FXS channels. My problem now is where to find the CAS signalling ABCD bit table ?(TxABCD, RxABCD) Could I change the ABCD value? Thanks for your help. These are my system info: [zaptel.conf]# E100P cardspan=1,0,0,cas,hdb3,crc4fxoks=1-12unused=13-15unused=17-31## Wildcard X100P cardfxsks=32defaultzone=usloadzone=us [zapata.conf][channels]musiconhold=defaultsignalling=fxo_ksechocancel=yesechocancelwhenbridged=yesechotraining=400callerid=asreceivedcontext=defaultchannel = 1-12 language=ensignalling=fxs_kschannel = 32 Output of zttool:- Alarms Span OK Digium Wildcard E100P E1/PRA Card 0 OK Wildcard X101P Board 1 Current Alarms: No alarms. Sync Source: Internally clocked IRQ Misses: 0 Bipolar Viol: 0 Tx/Rx Levels: 0/ 0 Total/Conf/Act: 31/ 12/ 0 112333 1234567890123456789012345789012 TxA TxB TxC TxD RxA RxB RxC RxD - [EMAIL PROTECTED] /]# lsmodModule Size Used by Not taintedwcfxo 9376 0 (unused)wct1xxp 13184 0 (unused)zaptel 179168 0 [wcfxo wct1xxp] [EMAIL PROTECTED] /]# cat /proc/interrupts CPU0 0: 68392 XT-PIC timer 1: 4 XT-PIC keyboard 2: 0 XT-PIC cascade 3: 453361 XT-PIC wcfxo 5: 1048 XT-PIC eth0 8: 1 XT-PIC rtc 9: 0 XT-PIC usb-ohci, usb-ohci, ehci-hcd10: 415190 XT-PIC t1xxp12: 32 XT-PIC PS/2 Mouse14: 5257 XT-PIC ide015: 0 XT-PIC ide1NMI: 0ERR: 0 [EMAIL PROTECTED] /]# cat /proc/zaptel/1Span 1: WCT1/0 "Digium Wildcard E100P E1/PRA Card 0" HDB3//CRC4 1 WCT1/0/1 FXOKS 2 WCT1/0/2 FXOKS ... 11 WCT1/0/11 FXOKS 12 WCT1/0/12 FXOKS -- Frank Lin [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] UK (english) sound files
Title: Message Hi all, After many complaints (including car manufacturers saying the american prompts are unexceptable, EEEK) I started on a quest for real "English" asterisk prompts. The only one I have found is here http://www.g7ltt.com/VoIP/vmfiles.html And no nothing else on the WIKI looked helpful (e.g. only American voice actors etc) These prompts are actually alot better than the standard prompts, according to my customer. But unfortunately they arent perfect. For example all of the queue prompts are missing as well as a number of other prompts. Personally I like Allisons sultry tones telling me that shes doing her utmost to connect my call :-) Couple of questions: 1) Does anyone else have english prompts they can share / point me to? 2) Does Mark (the kind guy that made the above) post on this list and is there any possiblity of adding some of the most needed prompts? Failing that I will give him an email and see what the chances are. 3) Failing everything else would anyone be interested in sharing the cost and getting some professional (female?) recordings done for all of the standard asterisk prompts? Currently I'm facing the possiblity of having three different people talking to the caller before they are put through. Company recording warning,UK transfer message and then American queue announcements. :-S So this has suddenly become a fairly urgent matter. thanks in advance for any help / advice on this Alex P.S. Sorry for the double post Asterisk-UK listers but as soon as I sent I thought it would be best to post here as well since this is pretty urgent for me. Information contained in this e-mail and any attachments are intended for the use of the addressee only, and may contain confidential information of Ubiquity Software Corporation. All unauthorized use, disclosure or distribution is strictly prohibited. If you are not the addressee, please notify the sender immediately and destroy all copies of this email. Unless otherwise expressly agreed in a writing signed by an officer of Ubiquity Software Corporation, nothing in this communication shall be deemed to be legally binding. Thank-you ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] UK (english) sound files
Ooops damn Outlook to Hades. Forgot to format in plain text. If you have been offended by this please feel free to ignore this thread. If not then I have left the original message below (this isnt a top post I swear) Thanks again alex -Original Message- From: Alex Barnes Sent: 25 April 2005 11:25 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] UK (english) sound files Hi all, After many complaints (including car manufacturers saying the american prompts are unexceptable, EEEK) I started on a quest for real English asterisk prompts. The only one I have found is here http://www.g7ltt.com/VoIP/vmfiles.html And no nothing else on the WIKI looked helpful (e.g. only American voice actors etc) These prompts are actually a lot better than the standard prompts, according to my customer. But unfortunately they arent perfect. For example all of the queue prompts are missing as well as a number of other prompts. Personally I like Allisons sultry tones telling me that shes doing her utmost to connect my call :-) Couple of questions: 1) Does anyone else have english prompts they can share / point me to? 2) Does Mark (the kind guy that made the above) post on this list and is there any possiblity of adding some of the most needed prompts? Failing that I will give him an email and see what the chances are. 3) Failing everything else would anyone be interested in sharing the cost and getting some professional (female?) recordings done for all of the standard asterisk prompts? Currently I'm facing the possiblity of having three different people talking to the caller before they are put through. Company recording warning, UK transfer message and then American queue announcements. :-S So this has suddenly become a fairly urgent matter. thanks in advance for any help / advice on this Alex P.S. Sorry for the double post Asterisk-UK listers but as soon as I sent I thought it would be best to post here as well since this is pretty urgent for me. Information contained in this e-mail and any attachments are intended for the use of the addressee only, and may contain confidential information of Ubiquity Software Corporation. All unauthorized use, disclosure or distribution is strictly prohibited. If you are not the addressee, please notify the sender immediately and destroy all copies of this email. Unless otherwise expressly agreed in a writing signed by an officer of Ubiquity Software Corporation, nothing in this communication shall be deemed to be legally binding. Thank you. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can Asterisk do the following for me ?
On April 24, 2005 11:58 pm, Lee Howard wrote: Certainly I can understand that Digium doesn't stand to make much money selling X100Ps at $10 each, and I can certainly understand them choosing to not sell them. But, by the same token I cannot understand the community's interest in discouraging other folks from joining the community in the way that economically suits them best. It has absolutely nothing to do with what economically suits them best -- it has everything to do with the fact that when you buy a clone X100P you DO NOT KNOW what you're getting. The chipset may be the same but as you can clearly see from searching this very list, the hybrid circuitry (a crucial crucial part of the design) can be VERY different, and even if the hybrid's fine, there are subtle variations in the chipset that can bite you in the ass. If you're just starting out with Asterisk, buck up and buy what is known to work and what is supported by Digium so that if the excrement DOES hit the air-conditioning you at least know your hardware's not at fault and there's someone who will log on to your system to help you fix it. In fact, Digium doesn't even sell the X100P/X101P anymore because the TDM FXO module has a dynamic impedance hybrid (not automatic, you need to specify which telco standard you're wiring in to) and even a nice simple FIR filter you can tune to help eliminate echo and reduce noise. It's simply a better product. Once you know how things work feel free to buy whatever you want. You'll have the understanding to know where to start troubleshooting if things go wrong and you won't be flooding the list and IRC with various Waah, I gots echo, Waah, I can't gets me CID, Waah, Asterisk sucks messages. Unless you know what you're doing (or are personally working with someone who does), buying the Digium stuff *IS* the most economical route. You may get lucky but generally speaking you'll waste far more time and resources pissing about getting the clone card to work than you will if using something known to work. This is along the exact same lines as those who come in here and post I juxt heard abouts this Aestrix thing... whutz teh ABSOLUTELY BARE MINIMUM hardware I need to make this work?! Early optimization (monteary, hardware or even software) is teh suck. It was the fall of the Roman empire, and it'll be the fall of your Asterisk empire if you're not careful. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can Asterisk do the following for me ?
On April 25, 2005 12:25 am, Kerry Garrison wrote: What year is this? 2005 right? Doesn't everyone on the planet know that you get what you pay for these days? If you want to experiment with Asterisk there is nothing wrong with using clone X100P cards at $6.95 a pop. If you No there is something very wrong with experimenting with Asterisk with a $7 clone card. When it doesn't work the lists get flamed, Asterisk gets blamed, and the experimenter leaves with a bad taste in his mouth about the whole VOIP process. If you're new to Asterisk, use Digium hardware. Once you understand what's going on, buy whatever cheapass shit you can find, at least you'll KNOW that the system does work with the right hardware. fork over some cash for a quality piece of equipment. If you are really diving into Asterisk, you would probably want to get the developer's kit just so you are working with equipment that you will most likely be using in a production environment. For us, our demo systems and backup systems run clone cards but our production systems all use Digium cards. You've got it completely and utterly backward. Until you know what you're doing you have no idea whether the problem is with the card, with Asterisk, with your system or with your configuration. By using known good cards you eliminate two of those potential sources, *AND* you get Digium's technical support department to help with the rest. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Large Asterisk Setup (~500 Concurrent Calls + Scalability)
On Thu, 2005-04-21 at 17:52 -0400, Matt Roth wrote: Daniel, You're correct that if we instructed the Monitor command to mix the files the mixing would occur on the master server. I looked at the documentation and source (res_monitor.c) of the Monitor command to confirm that the default behavior is to NOT mix the files. The options argument must contain the character 'm' for the mixing to occur. We will be executing the Monitor command WITHOUT mixing, then running a periodic process on the Digital Recording Client to mix the files and compress the result to an MP3. I think you can define the 'soxmix' command, which in your case, might simply send some small signal to your 'Digital Recording Client' with the filenames it should process. Otherwise, it is difficult for it to 'know' when the files are finished with. Other option is to have it listen to the manager interface, then when it sees a monitor start, it watches for either an end monitor/end of call, and then processes the files. Just my 0.02c worth... Good luck with your project. Regards, Adam ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] UK (english) sound files
Alex I too am on the hunt for the same. I am hoping that my good friend with the recording studio and his lovely wife will be able to perform this. My only issue at the moment is getting the scripts that was worked to, failing that, next weekend I am spending hours writing down what alison says :) David On 4/25/05, Alex Barnes [EMAIL PROTECTED] wrote: Ooops dan Outlook to Hades. Forgot to format in plain text. If you have been offended by this please feel free to ignore this thread. If not then I have left the original message below (this isnt a top post I swear) Thanks again alex -Original Message- From: Alex Barnes Sent: 25 April 2005 11:25 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] UK (english) sound files Hi all, After many complaints (including car manufacturers saying the american prompts are unexceptable, EEEK) I started on a quest for real English asterisk prompts. The only one I have found is here http://www.g7ltt.com/VoIP/vmfiles.html And no nothing else on the WIKI looked helpful (e.g. only American voice actors etc) These prompts are actually a lot better than the standard prompts, according to my customer. But unfortunately they arent perfect. For example all of the queue prompts are missing as well as a number of other prompts. Personally I like Allisons sultry tones telling me that shes doing her utmost to connect my call :-) Couple of questions: 1) Does anyone else have english prompts they can share / point me to? 2) Does Mark (the kind guy that made the above) post on this list and is there any possiblity of adding some of the most needed prompts? Failing that I will give him an email and see what the chances are. 3) Failing everything else would anyone be interested in sharing the cost and getting some professional (female?) recordings done for all of the standard asterisk prompts? Currently I'm facing the possiblity of having three different people talking to the caller before they are put through. Company recording warning, UK transfer message and then American queue announcements. :-S So this has suddenly become a fairly urgent matter. thanks in advance for any help / advice on this Alex P.S. Sorry for the double post Asterisk-UK listers but as soon as I sent I thought it would be best to post here as well since this is pretty urgent for me. Information contained in this e-mail and any attachments are intended for the use of the addressee only, and may contain confidential information of Ubiquity Software Corporation. All unauthorized use, disclosure or distribution is strictly prohibited. If you are not the addressee, please notify the sender immediately and destroy all copies of this email. Unless otherwise expressly agreed in a writing signed by an officer of Ubiquity Software Corporation, nothing in this communication shall be deemed to be legally binding. Thank you. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P delayed ring on incoming calls?
Joseph Gutowski [EMAIL PROTECTED] wrote: [...] I wasn't suggesting Asterisk should magically be able to pick up the call before it rings at all, just that if my old roommate could manage to dive across the room and pick up half way through the first ring 99% of the time, surely a computer could do it (if it wasn't waiting for caller ID or distinctive ring determination). This tangential thread was referring to the UK. Your roommate would have to be very alert and fit to be able to realise the phone is ringing and answer it within 400ms :) With a 20Hz ringing current, that's just eight cycles of AC per ring. A piece of kit that tries to detect ringing without just leaving it to finish ringing is likely to suffer from false positives. And the 1 ring wasn't constant -- sometimes it's one ring, sometimes it's 3 -- with no apparent reason (test server with nothing to do except answer one X100P and play an IVR menu). Well, the X100P (or at least a clone) is an unreliable piece of junk anyway, so this doesn't surprise me. Mine has now taken to randomly answering the line even when there's no inbound call, and has now been relegated to being just a Zaptel timing source. -- PGP key ID E85DC776 - finger [EMAIL PROTECTED] for full key Please contribute to the beer fund and a tidier house: http://search.ebay.co.uk/_W0QQfgtpZ1QQfrppZ25QQsassZpndc ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_capi: no dialstatus, no causes, no branches
On 4/22/05, Stefan Gofferje [EMAIL PROTECTED] wrote:
Hi folks,
I'm using a Fritz!PCI with chan_capi 0.3.5.
I found that chan_capi neither seems to signal Busy or Congestion to
callers from ISDN nor does it seem to set HANGUPCAUSE, CAUSECODE or
DIALSTATUS if an outgoing call fails. There is also no branch to n+101
if the called party is busy.
Are there any known solutions how to get this working?
A simple one is to change your dial string from
Dial(CAPI/MSN:${EXTEN}) to Dial(CAPI/MSN:b${EXTEN}) This will provide
busy tone from the Carrier
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[Asterisk-Users] What small PC can take 8 FXS + 8 FXO cards
Hi, I would like to setup a 8 FXS + 8 FXO interface using the X100P Cards. The problem now comes in the PCI ports. Is there any PC that can handle 16 ports? What is most optimal solution? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What small PC can take 8 FXS + 8 FXO cards
If i'm understanding this correctly, you shouldn't need 16 ports. If you buy 2 TDM400P cards, and load them up with 8 FXS (4 on each card) then buy 2 TDM400P cards, and load them up with 8 FXO (4 on each card) This should reduce your PCI count down to a more manageable 4 cards In total your shopping list would be 4 TDM400P PCI cards 8 FXS Daughter cards 8 FXO daughter cards hope that helps David On 4/25/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi, I would like to setup a 8 FXS + 8 FXO interface using the X100P Cards. The problem now comes in the PCI ports. Is there any PC that can handle 16 ports? What is most optimal solution? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Best of the best of IP Phones
The Polycoms also include a power supply and SIP firmware, which the Ciscos do not. Overall I just think the Polycoms are a better value. Cisco SIP firmware does not support subcribe/notify method (busy line/extension status). Polycom does support it. I have cisco phones right now and I'm planning to buy Polycom. Sergio ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Trouble with call parking/transfer
Make sure you have canreinvite=no in your sip peers definition, and/or that you pass 't' or 'T', to the Dial statement. Julian J. M. On 4/25/05, Tim Pushor [EMAIL PROTECTED] wrote: Hi all, I am still unable to initiate a call transfer with the keypresses defined in features.conf in a couple month old version of asterisk from CVS HEAD. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] UK (english) sound files
[EMAIL PROTECTED] wrote: Alex I too am on the hunt for the same. I am hoping that my good friend with the recording studio and his lovely wife will be able to perform this. My only issue at the moment is getting the scripts that was worked to, failing that, next weekend I am spending hours writing down what alison says :) Anyone interested producing (replicating) current USA voice sounds by getting a quote and then sharing the total cost. Interested parties please send your acknowledgment to senad at bicomsystems.com Ta Senad ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What small PC can take 8 FXS + 8 FXO cards
On April 25, 2005 07:39 am, [EMAIL PROTECTED] wrote: I would like to setup a 8 FXS + 8 FXO interface using the X100P Cards. The problem now comes in the PCI ports. Is there any PC that can handle 16 ports? What is most optimal solution? The most optimal solution would be a TE110P + a channel bank. The TE110P is about US$500 and a channel bank with 8FXS and 8FXO (with option to expand to 8 more ports) will run probably US$700-1000 on ebay. There is 1 PCI card in your computer and a piece of external equipment (the channel bank). You could go with 4 TDM400Ps to get the same number of ports but you will undoubtedly have trouble with sharing IRQs and the interrupt overhead is going to eat you alive. Channel banks are great; the better ones (Adit600) can do far-end disconnect supervision and I think pretty much all of them do dynamic impedance adjustment, meaning they're FAR less prone to echo. Just about anyone's FXS modules work, but be careful with FXO modules on channel banks. Access Bank I and IIs do *NOT* do far-end disconnect, meaning if someone on the other side hangs up, Asterisk won't be able to tell. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What small PC can take 8 FXS + 8 FXO cards
Thanks very much for this info Andrew. Selon Andrew Kohlsmith [EMAIL PROTECTED]: On April 25, 2005 07:39 am, [EMAIL PROTECTED] wrote: I would like to setup a 8 FXS + 8 FXO interface using the X100P Cards. The problem now comes in the PCI ports. Is there any PC that can handle 16 ports? What is most optimal solution? The most optimal solution would be a TE110P + a channel bank. The TE110P is about US$500 and a channel bank with 8FXS and 8FXO (with option to expand to 8 more ports) will run probably US$700-1000 on ebay. There is 1 PCI card in your computer and a piece of external equipment (the channel bank). You could go with 4 TDM400Ps to get the same number of ports but you will undoubtedly have trouble with sharing IRQs and the interrupt overhead is going to eat you alive. Channel banks are great; the better ones (Adit600) can do far-end disconnect supervision and I think pretty much all of them do dynamic impedance adjustment, meaning they're FAR less prone to echo. Just about anyone's FXS modules work, but be careful with FXO modules on channel banks. Access Bank I and IIs do *NOT* do far-end disconnect, meaning if someone on the other side hangs up, Asterisk won't be able to tell. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] UK (english) sound files
Interestingly enough I'm looking to do the same for a Canadian English version... does anyone to collaborate on this one? Ian Pattison, Senior Analyst Technology Associates Inc. Tel: 905-459-2100 ext. 204 Mobile: 416-568-6548 E-mail: [EMAIL PROTECTED] WWW: http://www.technologyassociates.ca [EMAIL PROTECTED] 25/04/2005 06:24 Hi all, After many complaints (including car manufacturers saying the american prompts are unexceptable, EEEK) I started on a quest for real English asterisk prompts. The only one I have found is here http://www.g7ltt.com/VoIP/vmfiles.html http://www.g7ltt.com/VoIP/vmfiles.html And no nothing else on the WIKI looked helpful (e.g. only American voice actors etc) These prompts are actually a lot better than the standard prompts, according to my customer. But unfortunately they arent perfect. For example all of the queue prompts are missing as well as a number of other prompts. Personally I like Allisons sultry tones telling me that shes doing her utmost to connect my call :-) Couple of questions: 1) Does anyone else have english prompts they can share / point me to? 2) Does Mark (the kind guy that made the above) post on this list and is there any possiblity of adding some of the most needed prompts? Failing that I will give him an email and see what the chances are. 3) Failing everything else would anyone be interested in sharing the cost and getting some professional (female?) recordings done for all of the standard asterisk prompts? Currently I'm facing the possiblity of having three different people talking to the caller before they are put through. Company recording warning, UK transfer message and then American queue announcements. :-S So this has suddenly become a fairly urgent matter. thanks in advance for any help / advice on this Alex P.S. Sorry for the double post Asterisk-UK listers but as soon as I sent I thought it would be best to post here as well since this is pretty urgent for me. Information contained in this e-mail and any attachments are intended for the use of the addressee only, and may contain confidential information of Ubiquity Software Corporation. All unauthorized use, disclosure or distribution is strictly prohibited. If you are not the addressee, please notify the sender immediately and destroy all copies of this email. Unless otherwise expressly agreed in a writing signed by an officer of Ubiquity Software Corporation, nothing in this communication shall be deemed to be legally binding. Thank you. BEGIN:VCARD VERSION:2.1 FN:Ian Pattison EMAIL;WORK;PREF:[EMAIL PROTECTED] TEL;WORK:416-657-2464 ext. 204 N:Pattison;Ian TITLE:Senior Analyst ADR;INTL;WORK;PARCEL;POSTAL:;;9052 Creditview Rd.;Brampton;Ontario;L6V 1A1;Canada LABEL;INTL;WORK;PARCEL;POSTAL;ENCODING=QUOTED-PRINTABLE:Ian Pattison=0A= 9052 Creditview Rd.=0A= Brampton, Ontario L6V 1A1=0A= Canada LABEL;DOM;WORK;PARCEL;POSTAL;ENCODING=QUOTED-PRINTABLE:Ian Pattison=0A= 9052 Creditview Rd.=0A= Brampton, Ontario L6V 1A1 TEL;CELL:416-568-6548 TEL;PREF:416-657-2464 ext. 204 TEL;WORK:905-459-2100 ext. 204 ORG:Technology Associates Inc. END:VCARD ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco's description of echo
Another common problem that causes echo in networks is not setting your loss plan correctly. You need to be sure that you aren't coming in too hot at any of your analog interfaces. In general you should see a signal between -20dbm and -12dbm when someone is talking on the line. If it is significantly hotter then you run the chance of having a larger reflected signal resulting in echo. I typically try padding down analog levels by 3dB at a time to see if echo is reduced. How do you measure the amplitude of a pstn line? As an audio engineer in a previous life, I would love to be able to send standard level tones down a pstn line and measure the amplitude at my end, then adjust the input gain accurately. Is there a way to do this? One way is to buy a relatively inexpensive analog transmission test set ($400 US). Most have a tone generator and level meter built in. You didn't mention which country you're located in, but ensure whatever test set you purchase, that it supports the line impedance in use by your telco. The inexpensive test sets won't function with digital circuits, however by using something like a cisco ata186 (with a known rx tx loss), one can use the analog test set to measure almost anything going on with asterisk and the pstn. Most US telco's have a milliwatt generator and quiet termination box attached to some local telephone number. Many of the US telco's encourage their installation people to use the test sets to access those resources for new pstn line installations to verify end to end functions. Its usually fairly easy to obtain the telephone numbers assigned to those boxes and use them to measure pstn loss, noise, etc. The Triplett Model 4 is one model. There are many others. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] UK (english) sound files
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Mon, 25 Apr 2005, David John Walsh wrote: Alex I too am on the hunt for the same. I am hoping that my good friend with the recording studio and his lovely wife will be able to perform this. My only issue at the moment is getting the scripts that was worked to, failing that, next weekend I am spending hours writing down what alison says :) Take a look at http://www.voip-info.org/wiki-Asterisk+sound+files and http://www.voip-info.org/wiki-Asterisk+sound+files These scripts seem to be cover all the standard and additional sound files. - -- Ron Wellsted http://www.wellsted.org.uk [EMAIL PROTECTED] FWD: 519961 N 52.567623, W 2.137621 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (GNU/Linux) iQEVAwUBQmzm+0tP/KMNOfRbAQK8PwgAqRxXp2flCXqTKeavdHbMswURHquzZjYh DyJeou3WCXsNeTthH7lAi+J8xLQEwjlOva+vW+cUvlEqAzCetGoDLEtsC+HBwCfr /8AXPXnKfbiMNtaHeedv45t6Ydv8tTdHeEEG3l19tgoKzrgxxet0seSXV6iqcpfZ n2ukYyKkX7rnHqonlN4/2h7d/MVSiKFRZXULL3Ha7pVhPYLavbKgBIc3/t5/0TGg Pi/C9ls38nPCnGbNai5BH0bMdl9Xf4JBPUFWfo3C+ko/Y0I+AcPbTjYkuyF2qPr4 iVEciB7kUOfeGiXWE7U9RaFtzgO7Yp9Y+4gfDGe3yuFPoALDV/BDog== =iaJr -END PGP SIGNATURE- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] UK (english) sound files
-Original Message- From: Ron Wellsted [mailto:[EMAIL PROTECTED] Sent: 25 April 2005 13:48 To: David John Walsh; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] UK (english) sound files -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Mon, 25 Apr 2005, David John Walsh wrote: Alex I too am on the hunt for the same. I am hoping that my good friend with the recording studio and his lovely wife will be able to perform this. My only issue at the moment is getting the scripts that was worked to, failing that, next weekend I am spending hours writing down what alison says :) Take a look at http://www.voip-info.org/wiki- Asterisk+sound+files and http://www.voip-info.org/wiki-Asterisk+sound+files These scripts seem to be cover all the standard and additional sound files. - -- Ron Wellsted http://www.wellsted.org.uk [EMAIL PROTECTED] FWD: 519961 N 52.567623, W 2.137621 This link seems to have a bigger list, tho will check if the file names are still correct tonight. http://www.voip-info.org/wiki-Asterisk+sound+files+additional Information contained in this e-mail and any attachments are intended for the use of the addressee only, and may contain confidential information of Ubiquity Software Corporation. All unauthorized use, disclosure or distribution is strictly prohibited. If you are not the addressee, please notify the sender immediately and destroy all copies of this email. Unless otherwise expressly agreed in a writing signed by an officer of Ubiquity Software Corporation, nothing in this communication shall be deemed to be legally binding. Thank you. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] UK (english) sound files
Ian you do realise that alison is actually canadian :) (well as far as I know she is) On 4/25/05, Ian Pattison [EMAIL PROTECTED] wrote: Interestingly enough I'm looking to do the same for a Canadian English version... does anyone to collaborate on this one? Ian Pattison, Senior Analyst Technology Associates Inc. Tel: 905-459-2100 ext. 204 Mobile: 416-568-6548 E-mail: [EMAIL PROTECTED] WWW: http://www.technologyassociates.ca [EMAIL PROTECTED] 25/04/2005 06:24 Hi all, After many complaints (including car manufacturers saying the american prompts are unexceptable, EEEK) I started on a quest for real English asterisk prompts. The only one I have found is here http://www.g7ltt.com/VoIP/vmfiles.html http://www.g7ltt.com/VoIP/vmfiles.html And no nothing else on the WIKI looked helpful (e.g. only American voice actors etc) These prompts are actually a lot better than the standard prompts, according to my customer. But unfortunately they arent perfect. For example all of the queue prompts are missing as well as a number of other prompts. Personally I like Allisons sultry tones telling me that shes doing her utmost to connect my call :-) Couple of questions: 1) Does anyone else have english prompts they can share / point me to? 2) Does Mark (the kind guy that made the above) post on this list and is there any possiblity of adding some of the most needed prompts? Failing that I will give him an email and see what the chances are. 3) Failing everything else would anyone be interested in sharing the cost and getting some professional (female?) recordings done for all of the standard asterisk prompts? Currently I'm facing the possiblity of having three different people talking to the caller before they are put through. Company recording warning, UK transfer message and then American queue announcements. :-S So this has suddenly become a fairly urgent matter. thanks in advance for any help / advice on this Alex P.S. Sorry for the double post Asterisk-UK listers but as soon as I sent I thought it would be best to post here as well since this is pretty urgent for me. Information contained in this e-mail and any attachments are intended for the use of the addressee only, and may contain confidential information of Ubiquity Software Corporation. All unauthorized use, disclosure or distribution is strictly prohibited. If you are not the addressee, please notify the sender immediately and destroy all copies of this email. Unless otherwise expressly agreed in a writing signed by an officer of Ubiquity Software Corporation, nothing in this communication shall be deemed to be legally binding. Thank you. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] need resources to include iax softphone functionality in vb6 app
Hi there I am looking for an open-source softphone / control for windows that I can use in a VB 6 application that will be for commercial use. I also need support for GSM, ulaw / alaw and possibly ilbc / speex. I have found a couple of possibilities, but none of them quite suit my needs: IaxPhone - http://www.sokol-associates.com/IaxPhone.htm - but the problem is that the open source version only supports GSM. IaxClientOcx http://www.angelfire.com/falcon/babarnazmi/iaxclient/iaxclientocx.htm - this active x control can only be used privately not commercially. Does anyone know if there is anything out there which would suit my requirements. Many thanks Steven ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] What small PC can take 8 FXS + 8 FXO cards
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Andrew Kohlsmith Sent: Monday, April 25, 2005 8:04 AM snip Channel banks are great; the better ones (Adit600) can do far-end disconnect supervision and I think pretty much all of them do dynamic impedance adjustment, meaning they're FAR less prone to echo. Just about anyone's FXS modules work, but be careful with FXO modules on channel banks. Access Bank I and IIs do *NOT* do far-end disconnect, meaning if someone on the other side hangs up, Asterisk won't be able to tell. Andrew, What configuration do you need to do to the Adit in order to get it to recognize FXO side disconnect? I have tried a number of different settings and can never get it to pass through to *. I am configured with 2 FXS cards and 1 FXO. My FXO card is running SW ver. 1.12 and my Mainboard is at ver. 7.0.3. I am using POTS lines from SBC. My lines are getting a loop drop as I have done some testing with Voicetronix OpenSwitch cards and they do see the disconnect. Any suggestions would be appreciated. Karl Putz ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Recording via monitor
I haven't played with it yet but from the info I read I understand that I can specify to record conversations on any extensions with the Monitor command. Is there any interface for replaying these recordings? I'm thinking of something like the CDR analysis package asterisk-stats with a link to the sound file for that CDR record. Has anyone worked with anything I should look at? Chris Mason ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco's description of echo
Rich Adamson wrote: One way is to buy a relatively inexpensive analog transmission test set ($400 US). Most have a tone generator and level meter built in. You didn't mention which country you're located in, but ensure whatever test set you purchase, that it supports the line impedance in use by your telco. The inexpensive test sets won't function with digital circuits, however by using something like a cisco ata186 (with a known rx tx loss), one can use the analog test set to measure almost anything going on with asterisk and the pstn. Most US telco's have a milliwatt generator and quiet termination box attached to some local telephone number. Many of the US telco's encourage their installation people to use the test sets to access those resources for new pstn line installations to verify end to end functions. Its usually fairly easy to obtain the telephone numbers assigned to those boxes and use them to measure pstn loss, noise, etc. The Triplett Model 4 is one model. There are many others. What's wrong with a little software for * to do all of the above and far more without cost or inconvenience? Regards, Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] UK (english) sound files
[EMAIL PROTECTED] wrote: Ian you do realise that alison is actually canadian :) yeah.. she is as far as I know as well... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco ATA 186
I'm nothing understand now. I have Cisco ATA 186 with one analog phone and
the following problem:
The next config works just fine:
sip.conf:
[150]
type=friend
port=5060
context=officepbx-outgoing
qualify=yes
secret=password
user=150
username=150
fromuser=150
defaultip=XXX.XXX.XXX.XXX
host=dynamic
dtmfmode=rfc2833
canreinvite=no
disallow=all
allow=ulaw
allow=g729
allow=alaw
extensions.conf:
TEST = SIP/[EMAIL PROTECTED]
agent_150 = ${TEST}
exten = 150,1,Macro(callfullext,${TEST},,,30,N)
But if I rename 150 to Cisco, by example, I get the following error
message:
NOTICE[1174440880]: chan_sip.c:7519 handle_request: Registration from
'sip:[EMAIL PROTECTED];user=phone' failed for 'XXX.XXX.XXX.XXX'
sip.conf:
[Cisco]
...
extensions.conf:
TEST = SIP/[EMAIL PROTECTED]
...
Cisco configuration:
UID0: 150
PWD0: password
UID1: 0
UseLoginID: 0
What is going on?
--
Serge Matveev
Relcom Corp., St.Petersburg
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Re: [Asterisk-Users] What small PC can take 8 FXS + 8 FXO cards
On April 25, 2005 09:05 am, Karl H. Putz wrote: What configuration do you need to do to the Adit in order to get it to recognize FXO side disconnect? I have tried a number of different settings and can never get it to pass through to *. It just worked for me, nothing unusual or funny. You tell the Adit600 to use LSCPD on its FXO interfaces, and you tell zapata.conf to use fxs_ks signaling; it Just Worked. :-) I'd be happy to assist further if needed. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco's description of echo
The Triplett Model 4 is one model. There are many others. Thanks, I found something on Ebay for $120 - great advice, I appreciate it. Chris Mason www.anguillaguide.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco's description of echo
On April 25, 2005 09:09 am, Steve Underwood wrote: What's wrong with a little software for * to do all of the above and far more without cost or inconvenience? Do we not already have this with ztmonitor and app_milliwatt.so? That's what I used, at least on Zap interfaces. If you need to adjust the gains on your SIP gear, get the Zap stuff aligned then just monitor the SIP stuff though the Zap channel and adjust accordingly. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asttapi and identapop pro
Hi folks, I've been trying to get Identapop Pro to work properly and am having no success for inbound Caller ID and Name. I've upgraded to the recent release of asttapi. Calling from Microsoft Outlook contacts works great. However on any inbound call Identapop Pro reports the following: 23:25:56:527 Asterisk Ready 23:26:44:466 CallState=OFFERING 23:26:44:506 RING 0 23:26:44:717 Examining: LINECALLINFO=0 Anyone run across this with a fix? Evidently asttapi is not sending the CID info to Identapop that it needs. Thanks. Tavis Patterson Network Engineer TAZ Networks Networking Done Right Voice: 517.579.0578 Email: [EMAIL PROTECTED] Web: www.taznetworks.com Blog: www.taznetworks.com/rss/webblog.html ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] i like my colors, thanks..
Andreas Sikkema wrote: [EMAIL PROTECTED] wrote: On Thu, 2005-04-21 at 08:39 -0500, Matthew Boehm wrote: Using most recent CVS-HEAD and my terminal keeps changing colors. I'm using vt100 terminal emulation. How can I turn off asterisk's colors? Or at least turn off the black background. My normal terminal is white background, black font. But for some reason, asterisk is changing it to white font, black background. Add '-n' to your command line. 'asterisk -h' will print out a list of all of the command line switches that it supports. -n doesn't do anything. (CVS-HEAD 4/24/05) -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Basic telephony hardware questions
Hi, I am in the process of setting up an Asterisk-based PBX at work. I get the concept of how Asterisk works pretty decently. I am more confused about the proliferation of TLAs like FXO, FXS, TDP, SIP, After some intense reading I have come to some understanding of the hardware I need to set things up. I am doing this in India and am getting a friend of mine bring the cards with him when he comes here in a couple of weeks. I don't know when my next trip to a place of abundant Asterisk-capable hardware will be. So I cannot afford making the wrong decisions. I need to live with my choice for atleast the next 4-6 months. My Setup Our office currently has 3 (to expand into 4 in the next few months) incoming PSTN lines. All these lines have RJ11 terminations. There are a couple of hundred employees working in the office. But not all of them need or have a phone on their desktops. My Requirements --- I need about 10-12 POTS phones to be connected to the PBX. When their extension is dialled, they need to ring. I need about 3 IP-phones to connect to the PBX over Ethernet. There will be some 50 users who will use soft phones on their desktops to connect to the PBX to make and receive calls. I also need IVRS for incoming calls and voicemail for all the extensions. Based on all of the above: * The cheapest option for me to get started seems to be 4 Digium PCI cards on a box running Asterisk. Will a setup with an Asterisk box with 4 Digium cards work? * I have identified 'TDM04B - 4-port FXO bundle' as the card I need to connect to the incoming PSTN lines. Is this identification correct? Also, will the incoming RJ11 terminations connect to this card? Or do I need something else? * I have identified 'TDM40B - 4-port FXS bundle' as the card I need to connect my in-office POTS phones. Is this identification correct? Also, will these cards enable the connection of RJ11 cables connecting to the POTS phones? * What do I need to connect to my local 100-base-T LAN to the PBX? I want desktops on the LAN to be be able to run soft phones and connect to the PBX. Do I need any other card? * Similarly, do I need anything extra on the Asterisk box to connect the IP phones? * What are TE410 and TE405 cards used for? Any help will be appreciated. Thaths -- Lisa: Why are you dedicating your life to blasphemy? Homer: Don't worry, sweetheart. If I'm wrong, I'll recant on my deathbed. Slacker Without Bordershttp://openscroll.org/ Key fingerprint = 8A 84 2E 67 10 9A 64 03 24 38 B6 AB 1B 6E 8C E4 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco's description of echo
One way is to buy a relatively inexpensive analog transmission test set ($400 US). Most have a tone generator and level meter built in. You didn't mention which country you're located in, but ensure whatever test set you purchase, that it supports the line impedance in use by your telco. The inexpensive test sets won't function with digital circuits, however by using something like a cisco ata186 (with a known rx tx loss), one can use the analog test set to measure almost anything going on with asterisk and the pstn. Most US telco's have a milliwatt generator and quiet termination box attached to some local telephone number. Many of the US telco's encourage their installation people to use the test sets to access those resources for new pstn line installations to verify end to end functions. Its usually fairly easy to obtain the telephone numbers assigned to those boxes and use them to measure pstn loss, noise, etc. The Triplett Model 4 is one model. There are many others. What's wrong with a little software for * to do all of the above and far more without cost or inconvenience? Absolutely nothing. Can you offer up something that is accurate and usable? (Might want to read through the mile long comments in bug 2023 though, since that problem has never been addressed.) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco's description of echo
The Triplett Model 4 is one model. There are many others. Thanks, I found something on Ebay for $120 - great advice, I appreciate it. I seen that one too. Be careful... it doesn't imply the meter is in working order, nor any warranty, etc. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Trouble with call parking/transfer
Thank you! Did you change the default transfer key? Doesn't the sipura 'eat' the #'s? yes, at least I know it should work (as I suspected). Thanks again, Tim David John Walsh wrote: call parking and transfer works great for me, on a variety of devices noteably: sipura 2000 / 3000 xten x-lite snom 360 using the keys set in feature.conf I guess even thou its bad news for you, it shows it works. On 4/25/05, Tim Pushor [EMAIL PROTECTED] wrote: Hi all, I am still unable to initiate a call transfer with the keypresses defined in features.conf in a couple month old version of asterisk from CVS HEAD. Before I go ripping things apart, I was really wondering if this is by design, or should it work on all my devices? I have an iaxy, phones hanging off fxs ports on a pair of tdm400p's, a sipura 841, a sipura 3000, and a pair of sipura 2000's and a Polycom IP 500. It only works on the phones hanging off the tdm400p. Should this work on all phones? Does anyone have it working on non digium FXS phones? Thanks, Tim ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] using * for Internet call waiting
You need a V92 capable modem for your client and a V92 capable access server for you. The feature is called modem on hold, it lets you pick up a call without loosing your internet connection, and resume the dialup session after hangup. The only feature you need for your telco is call waiting. It does not need forward on busy. Regards, That's one way of doing it. The other is call forward busy and how most of the existing services do it. Gary ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Repost: Dialing problem - Cisco 7290 to anything
Hi All,
Still having problems :-(
I have an Asterisk 1-0-7 setup on Debian 3.1 (Sparc)
I have severel SIP phones that call between each other and can chat no
probs. I can even call from the SIP phones to the sccp 7920 no
probs
However when I call from the 7290 to any SIP phone it just doesn't recognise
that the other person has answered the SIP phone, it just carries on making
the 'ringing' noise. When I hit hangup, the display of the 7290 changes to
onhook state but I can still hear the ringing
Any Ideas?
here are some copies of my config..
sccp.conf
[general]
keepalive = 5
context = home
dateFormat = D-M-Y ; M-D-Y in any order (5 chars max)
bindaddr = 192.122.122.22;
port = 2000; listen on port 2000 (Skinny, default)
[SEP000D282E89AA]
description = Walnuts Wireless
type = 7920
context = home
tzoffset = 0
autologin = wireless
[wireless]
id = 2210
context = home
callwaiting = 1
mailbox = 2210
callerid= Wireless, 2210
extensions.conf
[globals]
PHONES10=SCCP/wireless
PHONES10VM=wireless
[home]
exten = 2210,1,SetCalledParty(wireless 2000)
exten = 2000,2,Dial(SCCP/wireless)
exten = 2210,3,Macro(vmessage,${PHONES10VM})
exten = 2210,4,Hangup
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Re: [Asterisk-Users] sm bounty validate length of e164/e212 number for all countries
On Sun, 2005-04-24 at 11:15 -0700, Thomas Miller wrote: For example, Australia phone numbers can be either 6 or 7 digits, while USA phone numbers are always 10 digits. No, they aren't Most 'local' phone numbers are 8 digits, long distance (ie, including area code) they are 10 digits. That will also cover mobile phone numbers. Then, you get to deal with 4 digits numbers (1223/information/etc) and 6 digit numbers 13 (destination routed based on callers location (by exchange)). You could also add 000, which is 3 digits, and I am sure there are other rather unique numbers which don't really fit into any of that... I just can't recall them off the top of my head... Anyway, this doesn't help you with your request, but... it isn't as easy as you might think/home :) Regards, Adam -- -- Adam Goryachev Website Managers Ph: +61 2 9345 4395[EMAIL PROTECTED] Fax: +61 2 9345 4396www.websitemanagers.com.au ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Call Recording via monitor
It really depends on how you want the whole thing to function. After the call is done you could simply use system to email the recording as an attachment or you could use a php page to list all of the recordings. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Mason (Lists) Sent: Monday, April 25, 2005 7:08 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Call Recording via monitor I haven't played with it yet but from the info I read I understand that I can specify to record conversations on any extensions with the Monitor command. Is there any interface for replaying these recordings? I'm thinking of something like the CDR analysis package asterisk-stats with a link to the sound file for that CDR record. Has anyone worked with anything I should look at? Chris Mason ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM card periodic buzz
The box isn't doing anything else at all. It just started this problem recently, the only change I can correlate it to is moving to Asterisk 1.0.7. I'm pretty much not able to use the TDM card anymore now. I am thinking of just offering it upon ebay; there doesn't seem to be anything I can do about it. :-( (Sorry for the delay in responding; personal things are interfering with my time!) -Trent On Apr 13, 2005, at 5:01 PM, Andrew Kohlsmith wrote: With it occurring, log in and type zttest and let it run for a minute and tell us the accuracy min/max/avg. On April 18, 2005 10:17 am, Trent Tuggle wrote: Opened pseudo zap interface, measuring accuracy... --- Results after 109 passes --- Best: 100.00 -- Worst: 99.987793 What exactly does zttest test? On Apr 18, 2005, at 10:28 AM, Andrew Kohlsmith wrote: That's not terribly bad; Were you able to tell if the buzz occurrs when the timing drops down below 100%? What else is this box doing? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Call Recording via monitor
Message: 13 Date: Mon, 25 Apr 2005 09:08:15 -0400 From: Chris Mason (Lists) [EMAIL PROTECTED] Subject: [Asterisk-Users] Call Recording via monitor To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii I haven't played with it yet but from the info I read I understand that I can specify to record conversations on any extensions with the Monitor command. Is there any interface for replaying these recordings? I'm thinking of something like the CDR analysis package asterisk-stats with a link to the sound file for that CDR record. Has anyone worked with anything I should look at? Chris Mason MONITOR is great, we put in infront of all incoming / outgoing calls apart from calls to DQ services. From what I can tell it allocates a unique ID to the call file which is then written to disk. In terms of 'pulling back' the call then I have a CDR viewing tool from Andrews and Arnold (aa.gg) that enables me to find the call and the download the file to my local hard drive for playback based on selecting the unique ID. Unfortunately the otherwise excellent Areski stat tool doesn't seem to include the unique ID function and thus I can't pull a file back directly from that tool What I'd really like is the abillity to have an interface that a) Allows me to move important calls out from the standard archive into a 'special folder' on disk b) Allows me to tag the call with all the CDR information, as well as further info such as customer, reason for the call etc. This is so that 'important' call recordings don't get lost in the 000's of files you get when running Monitor 12x6. Anyone fancy some development activity? Greg Greg Eaton Product Development Director Intelicoms Mobile: 07957 144997 Phone: 0870 2403020 Fax: 0870 0501180 Website:www.intelicoms.co.uk Communications EBS Ltd trading as Intelicoms Registered in England. Company No 4745757 KBC House, 42-50 Hersham Road Walton-On-Thames, Surrey, KT12 1RZ The information contained in this message or any of its attachments may be confidential and is intended for the exclusive use of addressee(s). Any disclosure, reproduction, distribution or other dissemination or use of this communication is strictly prohibited without the express permission of the sender. This e-mail and any response may be monitored by Intelicoms. If you have received this e-mail in error, please notify Intelicoms on 08702 40 30 20 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BYOD provider other than broadvoice
Dan Perik wrote: Michael Lyszczek wrote: I have broadvoice and they suck lately. Can you elaborate? - Dan Yes, please elaborate. Do you mean to say they didn't suck previously but now they do suck? I can't imagine them staying in business much longer if that is truly the case. There are some bad providers out there who at least improve slightly over time. Going in the opposite direction is just too radical in this industry. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM card periodic buzz
On April 25, 2005 10:45 am, Trent Tuggle wrote: The box isn't doing anything else at all. It just started this problem recently, the only change I can correlate it to is moving to Asterisk 1.0.7. So go back to the version that was working... I'm trying to help you figure out what happened... maybe there's a bug in 1.0.7? basically change one thing at a time ... get it working then find out what made it break. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] astrecipes v2.0
Hello, if anyone is interested, there is a new wiki about Asterisk recipes, i.e. step-by-step descriptions on how to perform something with your * box. This is quite different from most * sites around, that are either questions-and-answers forums or are dedicated to documenting a feature. The point of AstRecipes is how to implement something. See http://www.oinko.net/astrecipes All content is licenced as creative commons, so if you got a recipe to spere, feel free to post it. Thanks l. -- Assum est, versa et manduca. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OH323 incoming audio stutter
Hi Michael and Tony I have the same problem here and I have been able to check that this problem can be solved disabling VAD in h323 destination routers, I think this is a common problem with h323 and oh323 modules users and for me has become a nightmare because my service provider can no longer disable VAD support independently for my connection... I will appreciate if you can include it in some future releases. regards rafael risco Millicom Peru On 4/22/05, Michael Manousos [EMAIL PROTECTED] wrote: Hi Tony, Can you get an ethereal trace of the RTP packets on both directions? A short analysis of those streams (from within the ethereal tools) would help us find the problem. Michael. Tony Mountifield wrote: I'm using asterisk-oh323-0.6.5 with the Janus patch 4 versions of pwlib (v1.6.6.3) and openh323 (v1.13.5.3), and using it to connect to my provider's switch. The effect that I am seeing is that a call starts off fine, but suddenly after a few minutes the audio coming into Asterisk via OH323 gets very broken up to the point of being about 90% silence with occasional brief snippets of audio getting through. When this happens, the audio going out from Asterisk to the other end is still fine, with no disturbances. I have observed this both when using SIP for the local leg of the call and when using IAX. I'm not really sure where to look to diagnose this, not whether it is likely to be an Asterisk problem or something in the switch. Any advice would be appreciated! Cheers Tony ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- rrgv ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] BYOD provider other than broadvoice
Our experience with BroadVoice over the past two months: Pros Good voice quality Zero downtime (not counting our ISP going down several times) Solid connections Low ping times Cons Would be nice if they supported more codecs (nothing new there) Takes on average of 45 minutes to talk to tech support (nothing new there) Overall, if you have everything working, it stays working and works well. If you are having problems than you need to have lots of patience with them. They are trying to improve and have even asked to use my using broadvoice with asterisk article on http://geekgazette.com to help their customers (which I gave them permission to use). -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Sent: Monday, April 25, 2005 8:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] BYOD provider other than broadvoice Dan Perik wrote: Michael Lyszczek wrote: I have broadvoice and they suck lately. Can you elaborate? - Dan Yes, please elaborate. Do you mean to say they didn't suck previously but now they do suck? I can't imagine them staying in business much longer if that is truly the case. There are some bad providers out there who at least improve slightly over time. Going in the opposite direction is just too radical in this industry. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] QoS Help and survey
Hi - We've been using IAX forwards between sites for a little while now (with centralized VM). For the most part, it is fine, but I have some very minor, yet persistent QoS issues on calls over the IAX forwards. For most normal calls, there are very occasional minor glitches, just an infrequent popping sound. It is something most of my users don't really care about, although it is a minor annoyance for some of them. Strangely, the problem is significantly more noticeable on voicemail and directory calls, and it is not limited to just pops. I also get large drops and strange metallic sounding echos and repeated sounds (voicema-ma-ma-ma-ma-mail - a la Max Headroom, for those that remember that). The issue isn't horrible, but it is a little weird and annoying. It generally only happens when network traffic between the sites is heavy. So, my survey question is - Is this normal? Should I expect to be able to get PSTN quality calls over these IAX forwards, or are some audio glitches just part of the package? I use a commercial VoIP service at home, and I don't have any of these issues, so I'm guessing it must be something in my network or setup. Our setup: - CVS HEAD from about a month ago on all machines (problem was also there with CVS HEAD as far back as 11/04) - Late Model Dell Servers 1600SC and SC420's - Cisco Routers - 1751 and 1721's (using Low Latency Queueing, matched to UDP 4569) - T1's - 10/100 Switches - 10/100 hubs at one site (is this a problem for anyone?) - SIP phones (Polycom IP300, IP500 and IP600, Snom 190) Thanks, Noah ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: astrecipes v2.0
lenz [EMAIL PROTECTED] writes: Hello, if anyone is interested, there is a new wiki about Asterisk recipes, i.e. step-by-step descriptions on how to perform something with your * box. This is quite different from most * sites around, that are either questions-and-answers forums or are dedicated to documenting a feature. The point of AstRecipes is how to implement something. See http://www.oinko.net/astrecipes All content is licenced as creative commons, so if you got a recipe to spere, feel free to post it. Thanks l. Good idea, but don't we have already the Wiki tips/hints, editable by anybody ? I understand people like to contribute, which is great. But spreading the info all over the web instead of centralizing it might be not so a great. Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: OH323 incoming audio stutter
In article [EMAIL PROTECTED], Rafael J. Risco G.V. [EMAIL PROTECTED] wrote: Hi Michael and Tony I have the same problem here and I have been able to check that this problem can be solved disabling VAD in h323 destination routers, I think this is a common problem with h323 and oh323 modules users and for me has become a nightmare because my service provider can no longer disable VAD support independently for my connection... I will appreciate if you can include it in some future releases. Rafael - thank you for your comments! I had suspected it was something to do with VAD. When conducting some tests with a colleague, we noticed that the audio coming into Asterisk via H.323 had silence suppression on it, and by experimenting with varying the level of background noise we could hear the squelch cut in and out. I don't know why it works for a while and then fails, unless there is something in chan_oh323 or openh323 that builds up and then reaches a threshold. I have demonstrated that two Asterisk boxes talking OH323 to each other do not experience the problem. Michael - does H.323 allow a peer to negotiate with the other end to disable VAD? If so, is there a way in chan_oh323 to invoke that option? I haven't had the chance yet to capture any RTP traces. Later this week if it is still an issue with new boxes being installed. regards rafael risco Millicom Peru On 4/22/05, Michael Manousos [EMAIL PROTECTED] wrote: Hi Tony, Can you get an ethereal trace of the RTP packets on both directions? A short analysis of those streams (from within the ethereal tools) would help us find the problem. Michael. Tony Mountifield wrote: I'm using asterisk-oh323-0.6.5 with the Janus patch 4 versions of pwlib (v1.6.6.3) and openh323 (v1.13.5.3), and using it to connect to my provider's switch. The effect that I am seeing is that a call starts off fine, but suddenly after a few minutes the audio coming into Asterisk via OH323 gets very broken up to the point of being about 90% silence with occasional brief snippets of audio getting through. When this happens, the audio going out from Asterisk to the other end is still fine, with no disturbances. I have observed this both when using SIP for the local leg of the call and when using IAX. I'm not really sure where to look to diagnose this, not whether it is likely to be an Asterisk problem or something in the switch. Any advice would be appreciated! Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: astrecipes v2.0
In data Mon, 25 Apr 2005 17:33:14 +0200, Bruno Hertz [EMAIL PROTECTED] ha scritto: Good idea, but don't we have already the Wiki tips/hints, editable by anybody ? I understand people like to contribute, which is great. But spreading the info all over the web instead of centralizing it might be not so a great. Regards, Bruno. Hello Bruno, I see your point; but I think that AstRecipes should be a different level from what I have currently found with *: it is more like a Linux HOW-TO container than a forum or a documentation project. I think one can find quite a wealth of information on * around, but hey, there is not much when you just want to know how to start with something with a minimal fuss. Whether centralization per se is a good idea or not I am not sure, I believe it is more an issue of ranking and rating content and making it available than having everything on a single server (and this could be the case, because all content is creative commons). Thanks l. -- Assum est, versa et manduca. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Alternatives to SpanDSP??
Hello all. I'm trying to see if anyone knows of an alternative solution, commercial or non-commercial, to SpanDSP. I'm specifically looking for another software-based, DSP fax that doesn't require me to add a tie up a bunch of extensions on my PBX. Has anyone ever seen such an animal, or gotten such it to play nice with Asterisk? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: astrecipes v2.0
In article [EMAIL PROTECTED], lenz [EMAIL PROTECTED] wrote: Hello, if anyone is interested, there is a new wiki about Asterisk recipes, i.e. step-by-step descriptions on how to perform something with your * box. This is quite different from most * sites around, that are either questions-and-answers forums or are dedicated to documenting a feature. The point of AstRecipes is how to implement something. See http://www.oinko.net/astrecipes All content is licenced as creative commons, so if you got a recipe to spere, feel free to post it. I've just looked at your Asterisk-OH323 recipe, and wanted to point out that with Asterisk 1.0.x the correct version of asterisk-oh323 is 0.6.5. Version 0.7.1 is only for use with CVS HEAD. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: astrecipes v2.0
In data Mon, 25 Apr 2005 16:12:08 + (UTC), Tony Mountifield [EMAIL PROTECTED] ha scritto: In article [EMAIL PROTECTED], lenz [EMAIL PROTECTED] wrote: Hello, if anyone is interested, there is a new wiki about Asterisk recipes, i.e. step-by-step descriptions on how to perform something with your * box. This is quite different from most * sites around, that are either questions-and-answers forums or are dedicated to documenting a feature. The point of AstRecipes is how to implement something. See http://www.oinko.net/astrecipes All content is licenced as creative commons, so if you got a recipe to spere, feel free to post it. I've just looked at your Asterisk-OH323 recipe, and wanted to point out that with Asterisk 1.0.x the correct version of asterisk-oh323 is 0.6.5. Version 0.7.1 is only for use with CVS HEAD. Cheers Tony Thanks, I fixed it. See http://www.oinko.net/astrecipes/index.php?n=40 If you notice other bugs or problems, please let me know. l. -- Assum est, versa et manduca. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Realtime voicemail
I have the latest Asterisk CVS as of 4/21/05 and a Fedora FC3 machine. I also set up the system to use Realtime for the voicemail mailboxes. I am successfully using Realtime for extensions and sip clients on this machine, but as yet, cannot get the voicemail system to recognize the mailboxes as defined in the MySQL database. The other tables, Sip and Extensions are part of the same database and they are accessed correctly. When the voicemail system does a MySQL query, the debug output shows that the correct mailbox is requested, but the context in the query is default, not the context that should be active at the moment, in my case analog-phones. Of course, if I define the extension in the voicemail.conf file, it works perfectly for the same context. I must be doing something wrong, but I cannot see what. Any help would be greatly appreciated. Ed Horton ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Grandstream ATA 286 problems
Anobody had any problem with GS ata 286? The past few days Ive been having some problem with it, while making a call or during a call, I suddely hear a low noise like a car engine starting and then the ata dies, as if it got stuck or frozen. Anybody had these problems? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: astrecipes v2.0
your queue recipie, does that monitor record from when the agent answers or the music on hold prior to taking the call? thanks On 4/25/05, lenz [EMAIL PROTECTED] wrote: In data Mon, 25 Apr 2005 16:12:08 + (UTC), Tony Mountifield [EMAIL PROTECTED] ha scritto: In article [EMAIL PROTECTED], lenz [EMAIL PROTECTED] wrote: Hello, if anyone is interested, there is a new wiki about Asterisk recipes, i.e. step-by-step descriptions on how to perform something with your * box. This is quite different from most * sites around, that are either questions-and-answers forums or are dedicated to documenting a feature. The point of AstRecipes is how to implement something. See http://www.oinko.net/astrecipes All content is licenced as creative commons, so if you got a recipe to spere, feel free to post it. I've just looked at your Asterisk-OH323 recipe, and wanted to point out that with Asterisk 1.0.x the correct version of asterisk-oh323 is 0.6.5. Version 0.7.1 is only for use with CVS HEAD. Cheers Tony Thanks, I fixed it. See http://www.oinko.net/astrecipes/index.php?n=40 If you notice other bugs or problems, please let me know. l. -- Assum est, versa et manduca. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Static and echo on PRI
Michael Welter wrote: Mark Johnson wrote: I tested and I do in fact get from 40-50% system util every 5 seconds or so. After removing the wctdm module, the system util drops to 0 and stays there. I have not loaded the wcfxs and wcfxo modules because I could never get them to work right. I instead load the wctdm and it has seemed to work fine. I only need to make the fx port to the paging system work and the others can stay idle. What modules and order so you suggest. Here is what I load in this order: wct4xxp wctdm Do you still have the static on the PRI without the TDM modules? I finally got to test... Removing the tdm module makes no difference in the static. I still hear it for any incoming sound. Removing it does, however clean up the CPU usage but quite a bit. One odd thing was with the tdm module removed, it seemed to introduce a little delay in the conversation. I also tried to recompile the zaptel drivers with the aggresive cancellation. This seems to made a HUGE improvement to my echo problem. Any ideas? Mark ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Large Asterisk Setup (~500 Concurrent Calls + Scalability)
Interesting. Can anyone out there tell me how many concurrent Monitors an Asterisk box can handle under my scenario (see below)? 1) Monitor commands are executed on the Asterisk server. 2) Audio packets are saved to files on a remote machine via mounted drive. 3) All handling of the audio files (mixing, compression, etc.) is done on the remote machine. If you could point out the bottlenecks and how to circumvent them, that would be appreciated as well. It seems that scaling an Asterisk setup is no trivial task. Thank you, Matthew Roth http://www.voip-info.org/tiki-index.php?page=Running%20Asterisk%20on%20Debian Daniel Salama wrote: The way aheeva handles this is by integrating call recording capabilities in their proprietary softphone. After the call is ended, the softphone uses their proprietary technology to transfer the audio file back to the server. It's a neat solution, but not scalable over WANs since the audio streams congest the WAN during busy periods. - Daniel On Apr 22, 2005, at 9:10 PM, Brian Roy wrote: On 4/21/05, Matt Roth [EMAIL PROTECTED] wrote: Daniel, I would be interested to hear if anyone knows of a method to completely offload the Monitor command from the master server. It is the missing piece of the puzzle to optimizing the digital recording process. You might want to talk to the folks at aheeva. www.aheeva.com They built their platform around * very much like you are. They offloaded quite a bit (including recording calls) to other boxes. Now, they built their solution around a much less stable Asterisk build, but they have some great experiences. I talked to them quite a bit last your at Astricon. I do remember them saying that after about 60 concurrent monitor's the * box would get unstable. Looks like you have some good research going you just need a little more proof of concept. -Brian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Random SIP Phone Problem
I got the same problem with 04/19/05 CVS version. I am using Grandstream phones. I also noticed that when this happens, an already hung-up call was still shown as bridged between a SIP phone and a Zap channel. On 4/18/05, Shaun Tierney [EMAIL PROTECTED] wrote: I am currently running CVS-HEAD-04/15/05-13:15:00 and I have an issue that just recently cropped up. I upgraded to this version of Asterisk last Friday and now twice in the last two hours, all of my Aastra SIP phones lose service suddenly. Network connectivity is still there between the phones and the PBX, and I have restart Asterisk to fix the issue. Would it be worth my time to move to the latest CVS Asterisk release even though it has only been three days since I installed the version in operation? Or would I be better off going with a previous CVS release to fix the problem? I can't use the stable release because I use macro arguments in the dial command. Here are the error messages that seem to show up for the duration of the problem. Apr 18 13:43:50 NOTICE[16997] chan_sip.c: Peer 'brettb' is now UNREACHABLE! Last qualify: 1045 Apr 18 13:44:03 VERBOSE[16997] logger.c: Don't know what to do if second ROSE component is of type 0x6 Apr 18 13:44:07 NOTICE[16997] app_queue.c: Added interface 'SIP/brettb' to queue 'psc' Apr 18 13:44:14 NOTICE[16997] chan_sip.c: Peer 'brettb' is now REACHABLE! (76ms / 2000ms) Regards, Shaun ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Realtime voicemail
Edwin Horton wrote: I have the latest Asterisk CVS as of 4/21/05 and a Fedora FC3 machine. I also set up the system to use Realtime for the voicemail mailboxes. I am successfully using Realtime for extensions and sip clients on this machine, but as yet, cannot get the voicemail system to recognize the mailboxes as defined in the MySQL database. The other tables, Sip and Extensions are part of the same database and they are accessed correctly. When the voicemail system does a MySQL query, the debug output shows that the correct mailbox is requested, but the context in the query is default, not the context that should be active at the moment, in my case analog-phones. Of course, if I define the extension in the voicemail.conf file, it works perfectly for the same context. I must be doing something wrong, but I cannot see what. Any help would be greatly appreciated. Ed Horton Send your extensions.conf section relative to this VM call. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Alternatives to SpanDSP??
I wasn't aware that SpanDSP tied up a bunch of extensions. Jeremy Melanson wrote: I'm trying to see if anyone knows of an alternative solution, commercial or non-commercial, to SpanDSP. I'm specifically looking for another software-based, DSP fax that doesn't require me to add a tie up a bunch of extensions on my PBX. Has anyone ever seen such an animal, or gotten such it to play nice with Asterisk? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Broadvoice Down?
Is anyone else having difficulty with their Broadvoice service? When I dial my number right now it rings either fast busy or tells me it cannot complete the call. I can make outgoing calls from my system through broadvoice however. Seems their inbound trunks hit capacity? Am I alone in this? -Max -- Max Clark max [at] clarksys.com http://www.clarksys.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: astrecipes v2.0
Naturally from when the agent and the user start talking to each other. l. In data Mon, 25 Apr 2005 17:45:00 +0100, David John Walsh [EMAIL PROTECTED] ha scritto: your queue recipie, does that monitor record from when the agent answers or the music on hold prior to taking the call? thanks On 4/25/05, lenz [EMAIL PROTECTED] wrote: In data Mon, 25 Apr 2005 16:12:08 + (UTC), Tony Mountifield [EMAIL PROTECTED] ha scritto: In article [EMAIL PROTECTED], lenz [EMAIL PROTECTED] wrote: Hello, if anyone is interested, there is a new wiki about Asterisk recipes, i.e. step-by-step descriptions on how to perform something with your * box. This is quite different from most * sites around, that are either questions-and-answers forums or are dedicated to documenting a feature. The point of AstRecipes is how to implement something. See http://www.oinko.net/astrecipes All content is licenced as creative commons, so if you got a recipe to spere, feel free to post it. I've just looked at your Asterisk-OH323 recipe, and wanted to point out that with Asterisk 1.0.x the correct version of asterisk-oh323 is 0.6.5. Version 0.7.1 is only for use with CVS HEAD. Cheers Tony Thanks, I fixed it. See http://www.oinko.net/astrecipes/index.php?n=40 If you notice other bugs or problems, please let me know. l. -- Assum est, versa et manduca. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Assum est, versa et manduca. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Trouble with call parking/transfer
Tim Pushor wrote: I am still unable to initiate a call transfer with the keypresses defined in features.conf in a couple month old version of asterisk from CVS HEAD. Before I go ripping things apart, I was really wondering if this is by design, or should it work on all my devices? I have an iaxy, phones hanging off fxs ports on a pair of tdm400p's, a sipura 841, a sipura 3000, and a pair of sipura 2000's and a Polycom IP 500. It only works on the phones hanging off the tdm400p. Should this work on all phones? Does anyone have it working on non digium FXS phones? Sounds to me like you have a DTMF problem. Does other DTMF work from your non-Zap devices? Not dialing, I mean like an IVR or VoicemailMain. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Broadvoice Down?
I am having the same broadvoice issue at the moment. jerry Is anyone else having difficulty with their Broadvoice service? When I dial my number right now it rings either fast busy or tells me it cannot complete the call. I can make outgoing calls from my system through broadvoice however. Seems their inbound trunks hit capacity? Am I alone in this? -Max -- Max Clark max [at] clarksys.com http://www.clarksys.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] chan capi: Long incomingmsn line in capi.conf?
I modified the source code as I have 10 msn numbers here at home, I will try to make a diff of the changes. Peter -Original Message- From: Stefan Helbing [mailto:[EMAIL PROTECTED] Sent: 22 April 2005 16:40 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] chan capi: Long incomingmsn line in capi.conf? Hello, the incomingmsn line in chan_capi's capi.conf is limited to 80 characters (AST_MAX_EXTENSION default value). My problem: I have to include several MSNs but NOT all. The interface is a 30 channel PRI card with a number area of 600 numbers, splitted in different functions. Some numbers are used for fax, some for PPP, some for telephony. (Example: 1234567xx is used for fax, 1234568xx is used for ppp, 1234569xx is used for telephony) When I set incomingmsn to * it's fine for asterisk - it gets all calls - but PPP and fax are not working anymore because they don't get any calls. In Germany I have to take the whole number without the leading zero of the area prefix. So every MSN has a length of 10 characters. This limits the count of usable MSN to 7 (7*10 + 6 commas = 76 chars). I tried out to use a wildcard in the string (using the example above: 1234569*) but this doesn't work. Any idea (except modifying the source code)? Thank you! :-) Best regards Stefan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ __ This email has been scanned for all viruses by the Star Internet Virus Screen. The service is provided in partnership with MessageLabs, the email security company. For more information on a higher level of virus protection visit www.star.net.uk __ __ This e-mail has been scanned for all viruses by Star. The service is powered by MessageLabs. For more information on a proactive anti-virus service working around the clock, around the globe, visit: http://www.star.net.uk ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No busy tone when dialing out over ISDN with Polycom 500 IP
Kib Eki wrote: Hi, what do i have to configure to get a busy tone when dialing out over ISDN channel with my Polycom 500 IP? Try priindication = inband in /etc/asterisk/zapata.con ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] UK (english) sound files
Does Mark post on this list? Hmm. Let me think about this one ;-} I was trying to get a movement going to make sound files in other English language variants but it all seemed to die off. I have not found the time to complete the Southern England Male prompts but if you send me a list of the ones you specifically need I could do them for you particularly if there was a fee involved ;-} Let me know off list. Mark Alex Barnes wrote: Hi all, After many complaints (including car manufacturers saying the american prompts are unexceptable, EEEK) I started on a quest for real English asterisk prompts. The only one I have found is here http://www.g7ltt.com/VoIP/vmfiles.html And no nothing else on the WIKI looked helpful (e.g. only American voice actors etc) These prompts are actually a lot better than the standard prompts, according to my customer. But unfortunately they arent perfect. For example all of the queue prompts are missing as well as a number of other prompts. Personally I like Allisons sultry tones telling me that shes doing her utmost to connect my call :-) Couple of questions: 1) Does anyone else have english prompts they can share / point me to? 2) Does Mark (the kind guy that made the above) post on this list and is there any possiblity of adding some of the most needed prompts? Failing that I will give him an email and see what the chances are. 3) Failing everything else would anyone be interested in sharing the cost and getting some professional (female?) recordings done for all of the standard asterisk prompts? Currently I'm facing the possiblity of having three different people talking to the caller before they are put through. Company recording warning, UK transfer message and then American queue announcements. :-S So this has suddenly become a fairly urgent matter. thanks in advance for any help / advice on this Alex P.S. Sorry for the double post Asterisk-UK listers but as soon as I sent I thought it would be best to post here as well since this is pretty urgent for me. Information contained in this e-mail and any attachments are intended for the use of the addressee only, and may contain confidential information of Ubiquity Software Corporation. All unauthorized use, disclosure or distribution is strictly prohibited. If you are not the addressee, please notify the sender immediately and destroy all copies of this email. Unless otherwise expressly agreed in a writing signed by an officer of Ubiquity Software Corporation, nothing in this communication shall be deemed to be legally binding. Thank-you ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Broadvoice Down?
I seem to be down right now too. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Max Clark Sent: Monday, April 25, 2005 10:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Broadvoice Down? Is anyone else having difficulty with their Broadvoice service? When I dial my number right now it rings either fast busy or tells me it cannot complete the call. I can make outgoing calls from my system through broadvoice however. Seems their inbound trunks hit capacity? Am I alone in this? -Max -- Max Clark max [at] clarksys.com http://www.clarksys.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Why can't I hear audio?
Hi Everybody can someone tell me why I can hear audio? My call is to my proxie which is directing it to my Asterisk box. The Voice mail is playing but I think its playing to my proxie. the phone is on 198.31.185.246:63257 Here is from the sip debug. Thanks Sip read: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 208.41.254.119:5060; branch=z9hG4bKAClkQjtHCQANCLBn5V+1x2CxseR9SzOu84U_ Via: SIP/2.0/UDP 198.31.185.246:63257;branch=z9hG4bKf9f703805e460dc4 From: Shelcomm call forwarding test sip:[EMAIL PROTECTED];user=phone;tag=100c9f35ec6f09a2 To: sip:[EMAIL PROTECTED];user=phone Contact: sip:[EMAIL PROTECTED]:63257;user=phone Supported: replaces Proxy-Authorization: DIGEST username=[EMAIL PROTECTED], realm=sip.shelcomm.com, algorithm=MD5, uri=sip:[EMAIL PROTECTED];user=phone, qop=auth, nc=0001, cnonce=1a605453cf8a557d, nonce=62JsQimn5xTAMoKDHL+EbAkTRGg=, response=874d55e7960ad550b78bb1d8660faf69 Call-ID: [EMAIL PROTECTED] CSeq: 55676 INVITE User-Agent: Grandstream BT100 1.0.5.23 Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Content-Length: 338 Record-Route: sip:208.41.254.119;lr;hash=sipd-0-2-2 asterisk1*CLI v=0 o=6262769011 8000 8001 IN IP4 198.31.185.246 s=SIP Call c=IN IP4 198.31.185.246 t=0 0 m=audio 63268 RTP/AVP 0 4 9 15 2 18 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:4 G723/8000 a=rtpmap:9 G722/16000 a=rtpmap:15 G728/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 16 headers, 15 lines Using latest request as basis request Sending to 208.41.254.119 : 5060 (non-NAT) Found no matching peer or user for '208.41.254.119:5060' Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 9 Found RTP audio format 15 Found RTP audio format 2 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 198.31.185.246:63268 Found description format PCMU Found description format G723 Found description format G722 Found description format G728 Found description format G726-32 Found description format G729 Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x115 (g723|ulaw|g726|g729)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) Looking for 9009 in from-sip-external list_route: hop: sip:208.41.254.119;lr;hash=sipd-0-2-2 list_route: hop: sip:[EMAIL PROTECTED]:63257;user=phone Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 208.41.254.119:5060; branch=z9hG4bKAClkQjtHCQANCLBn5V+1x2CxseR9SzOu84U_ Via: SIP/2.0/UDP 198.31.185.246:63257;branch=z9hG4bKf9f703805e460dc4 From: Shelcomm call forwarding test sip:[EMAIL PROTECTED];user=phone;tag=100c9f35ec6f09a2 To: sip:[EMAIL PROTECTED];user=phone;tag=as59b09f62 Call-ID: [EMAIL PROTECTED] CSeq: 55676 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 208.41.254.119:5060 -- Executing VoiceMail(SIP/208.41.254.119-089aef50, 9009) in new stack We're at 208.41.254.125 port 13630 Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 208.41.254.119:5060; branch=z9hG4bKAClkQjtHCQANCLBn5V+1x2CxseR9SzOu84U_ Via: SIP/2.0/UDP 198.31.185.246:63257;branch=z9hG4bKf9f703805e460dc4 Record-Route: sip:208.41.254.119;lr;hash=sipd-0-2-2 From: Shelcomm call forwarding test sip:[EMAIL PROTECTED];user=phone;tag=100c9f35ec6f09a2 To: sip:[EMAIL PROTECTED];user=phone;tag=as59b09f62 Call-ID: [EMAIL PROTECTED] CSeq: 55676 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Type: application/sdp Content-Length: 242 v=0 o=root 2330 2330 IN IP4 208.41.254.125 s=session c=IN IP4 208.41.254.125 t=0 0 m=audio 13630 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - to 208.41.254.119:5060 -- Playing 'vm-intro' (language 'en') asterisk1*CLI Sip read: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 208.41.254.119:5060; branch=z9hG4bKAClkQjtHCQAOCCEtoOY6oOebox7ZBwoRRiY_ Via: SIP/2.0/UDP 198.31.185.246:63257;branch=z9hG4bK2e698f7c1c332d46 From: Shelcomm call forwarding test sip:[EMAIL PROTECTED];user=phone;tag=100c9f35ec6f09a2 To: sip:[EMAIL PROTECTED];user=phone;tag=as59b09f62 Contact: sip:[EMAIL PROTECTED]:63257;user=phone Proxy-Authorization: DIGEST username=[EMAIL PROTECTED], realm=sip.shelcomm.com, algorithm=MD5, uri=sip:[EMAIL PROTECTED], qop=auth, nc=0002, cnonce=b85d4240018f156a, nonce=62JsQimn5xTAMoKDHL+EbAkTRGg=, response=4030f97656e76c9bffecee6942efbfcc Call-ID: [EMAIL PROTECTED] CSeq: 55676 ACK User-Agent: Grandstream BT100 1.0.5.23 Max-Forwards: 69 Allow:
Re: [Asterisk-Users] Trouble with call parking/transfer
Yes, no problem. Seems like asterisk is not 'breaking in'. I have it in the media path and dtmf all works properly. The sipura devices are a mismatch of codecs, but the 841 is g729 and the iaxy is of course ulaw. I am running the IPP g729 codec and I did wonder if that was the issue, but I have at least one sipura port ulaw, and the iaxy. IVR (both my IVR's and remote (such as the telephone banking places) work fine). I'll track down why it isn't working, I just wanted to know if it was supposed to work or not. Thanks, Tim Eric Wieling aka ManxPower wrote: Tim Pushor wrote: I am still unable to initiate a call transfer with the keypresses defined in features.conf in a couple month old version of asterisk from CVS HEAD. Before I go ripping things apart, I was really wondering if this is by design, or should it work on all my devices? I have an iaxy, phones hanging off fxs ports on a pair of tdm400p's, a sipura 841, a sipura 3000, and a pair of sipura 2000's and a Polycom IP 500. It only works on the phones hanging off the tdm400p. Should this work on all phones? Does anyone have it working on non digium FXS phones? Sounds to me like you have a DTMF problem. Does other DTMF work from your non-Zap devices? Not dialing, I mean like an IVR or VoicemailMain. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Broadvoice Down?
Same here. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Geis Sent: Monday, April 25, 2005 10:14 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Broadvoice Down? I am having the same broadvoice issue at the moment. jerry Is anyone else having difficulty with their Broadvoice service? When I dial my number right now it rings either fast busy or tells me it cannot complete the call. I can make outgoing calls from my system through broadvoice however. Seems their inbound trunks hit capacity? Am I alone in this? -Max -- Max Clark max [at] clarksys.com http://www.clarksys.com smime.p7s Description: S/MIME cryptographic signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Broadvoice Down?
Working fine for me.. going through: proxy.mia.broadvoice.com if that helps.. -- Regards, Sean Milheim iDREUS Corporation http://www.idreus.com On Mon, 2005-04-25 at 10:27 -0700, Kerry Garrison wrote: I seem to be down right now too. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Max Clark Sent: Monday, April 25, 2005 10:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Broadvoice Down? Is anyone else having difficulty with their Broadvoice service? When I dial my number right now it rings either fast busy or tells me it cannot complete the call. I can make outgoing calls from my system through broadvoice however. Seems their inbound trunks hit capacity? Am I alone in this? -Max -- Max Clark max [at] clarksys.com http://www.clarksys.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message was scanned for spam and viruses by BitDefender For more information please visit http://www.idreus.com/index.php?page=subproductproduct_id=41 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Alternatives to SpanDSP??
I guess I didn't word this right. It's not that SpanDSP ties up extensions, as it definitely doesn't. I was more referring to the standard hardware-based solutions out there that need to have a dedicated line for an incoming fax. I need the ability to send and receive faxes with a good amount of reliability, and would love to integrate it with Asterisk. I'm just not keen on needing to buy a bunch of Digium TDM cards just to support such a solution. Don't get me wrong, SpanDSP is great! I'm just looking for something a little more enterprise-ready. Jeremy On Mon, 2005-04-25 at 12:07 -0500, Eric Wieling aka ManxPower wrote: I wasn't aware that SpanDSP tied up a bunch of extensions. Jeremy Melanson wrote: I'm trying to see if anyone knows of an alternative solution, commercial or non-commercial, to SpanDSP. I'm specifically looking for another software-based, DSP fax that doesn't require me to add a tie up a bunch of extensions on my PBX. Has anyone ever seen such an animal, or gotten such it to play nice with Asterisk? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Broadvoice Down?
Max Clark wrote: Is anyone else having difficulty with their Broadvoice service? When I dial my number right now it rings either fast busy or tells me it cannot complete the call. I can make outgoing calls from my system through broadvoice however. Seems their inbound trunks hit capacity? Am I alone in this? -Max Nope, I get the same thing. I can dial out though through my asterisk machine, but not in from pstn. JD -- JD Austin Twin Geckos Technology Services LLC email: [EMAIL PROTECTED] http://www.twingeckos.com phone/fax: 480.422.1250 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Realtime voicemail
Realtime voicemail configuration assumes the Voicemail Context to be 'default' unless otherwise specified. This is not the same as the Extensions Context. Having said that, can you specify what the actual problem is? Can't get voicemail to pick up; MWI doesn't work; etc. Matthew Boehm ([EMAIL PROTECTED]) wrote: Edwin Horton wrote: I have the latest Asterisk CVS as of 4/21/05 and a Fedora FC3 machine. I also set up the system to use Realtime for the voicemail mailboxes. I am successfully using Realtime for extensions and sip clients on this machine, but as yet, cannot get the voicemail system to recognize the mailboxes as defined in the MySQL database. The other tables, Sip and Extensions are part of the same database and they are accessed correctly. When the voicemail system does a MySQL query, the debug output shows that the correct mailbox is requested, but the context in the query is default, not the context that should be active at the moment, in my case analog-phones. Of course, if I define the extension in the voicemail.conf file, it works perfectly for the same context. I must be doing something wrong, but I cannot see what. Any help would be greatly appreciated. Ed Horton Send your extensions.conf section relative to this VM call. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Joe Dennick [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAX help
If you look at your iax.conf lines as under, you will notice that the
two contexts are illegal as they both have same name:
[telx-nyc]
type=user
secret=telx-nyc
context=from-telx-nyc
disallow=all
allow=ulaw
; telx-nyc-asterisk - Outgoing
;
[telx-nyc]
type=peer
username=telx-NY17S ; our username
secret=telx-NY17S ; our password
host=192.168.11.30 ; host to connect to
;qualify=yes
;trunk=yes ; use trunking
Make that into one and change type to type=friend
This is a starting point. You have defined a macro also with similar
name and that is confusing. Why don't you name it differently as that
may also be one of the causes for the problem you are seeing.
Seshu
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael
DiMartino
Sent: Friday, April 22, 2005 8:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] IAX help
I am trying to send calls from (telx-NY17S) to (telx-nyc) via an IAX2
channel. However the call is being rejected on the (telx-nyc) server.
See error below copied from telx-nyc CLI
Apr 22 13:56:57 NOTICE[147465]: chan_iax2.c:5390 socket_read:
Rejected
connect attempt from 192.168.0.251
I have icluded the following conf files
1. extensions.conf (telx-nyc)
2. iax.conf (telx-nyc)
3. extensions.conf (telx-NY17S)
4. iax.conf (telx-NY17S)
1. extensions.conf (telx-nyc)
[general]
static=yes
writeprotect=yes
[globals]
EMERGENCY=0
LINEOUT=Zap/1 ; Line to use in emergency
#include extensions.conf.macro
#include extensions.conf.telx
[from-telx-atl]
include = internal
include = ext-external-from-atl
[from-telx-NY17S]
include = internal
include = ext-internal
[from-jnctn]
include = aa-main
exten = _NXXNXX,1,Goto(aa-main,s,1)
[from-swifttel]
exten = _NXXNXX,1,NoOp(Context is from-swifttel) exten =
_NXXNXX,2,Goto(aa-main,s,1)
[default]
include = aa-main
exten = _NXXNXX,1,NoOp(Context is default) exten =
_NXXNXX,2,Goto(aa-main,s,1)
[internal]
include = ext-local
include = ext-internal
[ext-local]
exten = 7000,1,Goto(aa-main,s,1)
[ext-internal]
exten = _2XXX,1,Dial(${TELX-MICS2}/${EXTEN:${TELX-MICS2-MSD}})
exten = _2XXX,2,Congestion
exten = _3XXX,1,Dial(${TELX-MICS1}/${EXTEN:${TELX-MICS1-MSD}})
exten = _3XXX,2,Congestion
exten = _71XX,1,Dial(IAX2/telx-NY17S/${EXTEN})
exten = _71XX,2,Congestion
exten = _7XXX,1,Dial(IAX2/telx-atl/${EXTEN})
exten = _7XXX,2,Congestion
2. iax.conf (telx-nyc)
[general]
allow=all
jitterbuffer=no
tos=lowdelay
bindaddr=0.0.0.0
; Registration to Junction Networks
;register = telx:[EMAIL PROTECTED]
; Guest sections for unauthenticated connection attempts.
[guest]
type=user
context=default
callerid=Guest IAX User
; from Junction Networks
[jnctn]
type=user
context=from-jnctn
auth=rsa
inkeys=jnctn
; telx-atl-asterisk - Incoming
[telx-atl] ; name remote end will use to connect
type=user ; they will send calls to us
secret=telx-atl ; their password
context=from-telx-atl ; context for calling in
disallow=all
allow=ulaw
; telx-atl-asterisk - Outgoing
[telx-atl]
type=peer
username=telx-nyc ; our username
secret=telx-nyc ; our password
host=192.168.22.7 ; host to connect to
;qualify=yes
;trunk=yes ; use trunking
; telx-NY17S - Incoming
[telx-NY17S]
type=peer
secret=telx-NY17S
context=from-telx-NY17S
disallow=all
allow=ulaw
; telx-NY17S - Outgoing
[telx-NY17S]
type=peer
username=telx-nyc ; our username
secret=telx-nyc ; our password
host=192.168.0.251 ; host to connect to
;qualify=yes
;trunk=yes ; use trunking
[stealth]
type=friend
host=dynamic
auth=md5
secret=telxvoip
context=from-jnctn
permit=206.252.192.70/255.255.255.255
3. Extensions.conf (telx-NY17S)
[general]
static=yes
writeprotect=yes
[bogon-calls]
exten = _.,1,Congestion
[from-sip]
exten = 7101,1,Dial(SIP/7101,20)
exten = 7101,2,Voicemail(u7101)
exten = 7101,102,Voicemail(b7101)
exten = 7101,103,Hangup
exten = 7102,1,Dial(SIP/7102,20)
exten = 7102,2,Voicemail(u7102)
exten = 7102,102,Voicemail(b7102)
exten = 7102,103,Hangup
;Extentions at telx-nyc
exten = _7XXX,1,Dial(IAX2/telx-nyc/${EXTEN})
exten = _7XXX,2,Congestion
exten = _2XXX,1,Dial(IAX2/telx-nyc/${EXTEN})
exten = _2XXX,2,Congestion
exten = _3XXX,1,Dial(IAX2/telx-nyc/${EXTEN})
exten = _3XXX,2,Congestion
exten = 7199,1,VoicemailMain(${CALLERIDNUM})
[from-telx-nyc]
exten = _7XXX,1,Dial(SIP/7101,20)
exten = _7XXX,2,Voicemail(u7101)
exten = _7XXX,102,Voicemail(b7101)
exten = _7XXX,103,Hangup
[macro-telx-nyc]
exten = s,1,Noop()
exten = s,2,Dial(IAX2/telx-nyc/${ARG1})
[outgoing]
;ingnorepat = 9
exten = _9NXXNXX,1,Noop()
exten = _9NXXNXX,2,Macro(telx-nyc,${EXTEN})
exten = _9NXXNXX,3,Playback(invalid) exten =
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