Re: [asterisk-users] Block Spam Calls

On Tue, Dec 10, 2019 at 11:01 AM Alexander Perkins wrote: > Hi All. Does anybody know if Google/Android has an API I can sign up for > that will allow us to query the caller ID and find out if it is spam or a > robocaller? I don't think that there is a public (free) API. All robocall

Re: [asterisk-users] Digium IP Phones UNREACHABLE after registration

On Thu, Apr 12, 2018 at 11:43 AM, Antony Stone wrote: >> A few seconds after registration, the Digium phones will become >> UNREACHABLE. Right after that, the entire VoIP network (where the >> Digiums are located) will be also dropped - all other devices >>

[asterisk-users] Digium IP Phones UNREACHABLE after registration

I'm trying to solve a mystery for the last couple of days. I have a mix of D70, D50 and D40 behind NAT. Server is in a colocation, not a VPS. For several years, everything was working fine, no issues. A few days ago I started having problems at one particular site. NO CHANGES have been made to

Re: [asterisk-users] Dock-N-Talk with Asterisk, Anyone?

Jeng Yu wrote: I would like to hear if anyone out there in Asteriskland has used the Dock-N-Talk (DNT) box to connect cell phones to Asterisk box. The only problem I noticed is that after a random amount of time the box will lost contact/synch with the cell phone. I'm using DockNTalk for about

Re: [asterisk-users] Asterisk-1.2.22 DeadAGI Hangup

Arun Kumar wrote: I've upgraded my server to asterisk-1.2.22 from 1.2.10 after that my DeadAGI scripts are not working properly. Like after hangup I used to do some more work now its not working. Try, at your own risk, this:

Re: [asterisk-users] Best wifi IP phone for asterisk

Michelle Dupuis wrote: We're looking at a large wifi phone deployment, and we're looking for wifi phones that: 1. Are SIP compliant (Asterisk friendly) 2. Provision capable (ideally TFTP of own MAC address) 3. Industrial quality (no cheap plastic stuff). 4. Well documented (and none of

Re: [asterisk-users] GSM Cards for Asterisk (UK)

Matt Brown wrote: Does anyone have any experience with a GSM card, preferably Quad Span (4 GSM modules or higher) for use in the UK. I have seen the Junghanns* version but I am not keen on the limitation of having to use a BriStuffed version of Asterisk. I'm buying this one to test:

Re: [asterisk-users] Apple IPhone mobile is released in India?

Crazy Boy wrote: If IPhone is released in India, Can you tell me any Apple authorized showroom in Hyderabad (Andhrapradesh, India)? Oh gosh... another troll... Google IS your friend: http://www.google.com/search?q=apple+iphone ___ --Bandwidth and

Re: [asterisk-users] saydigits in another language

Julian Lyndon-Smith wrote: however, I get no errors, but still get the default Allison sounds for the digits. Anyone got any clues on what I'm doing wrong ? 1) Create a directory named your_country_iso_code (AR|MX|ES|ETC) [1] under the main sounds directory (/var/lib/asterisk/sounds/ ???);

Re: [asterisk-users] sip_header=value?

Rizwan Hisham wrote: is there anyway i can set SIP_HEADER(To) to the value i like? If voip-info is correct, you can read, but you can't change. http://www.voip-info.org/wiki/index.php?page=Asterisk+func+sip_header ___ --Bandwidth and Colocation

Re: [asterisk-users] cisco 7905

Khaled Chehab wrote: How to configure cisco 7905 with asterisk ,if you please can send me step by step configuration steps . This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to

[asterisk-users] Asterisk x Mera MVTS

I'm having trouble to send calls to a Mera MVTS softswitch (with SIPHIT) when the asterisk box has a dynamic IP address. If the Asterisk box has a fixed IP, everything is OK. Any ideas? I'm looking for a working sample of the sip.conf in this case... user.cfg (for MVTS) is also appreciated if

Re: [asterisk-users] Problem in using Two BRi Cards in Asterisk

Farooq Ahmed wrote: And any idea about the issue on card one... means why outgoing is not working. Not quite sure if Traverse Technology Netjet ISDN-s will really work. Last time I had to use a ISDN BRI I bought one with Cologne chipset and used bristuff. Worked like a charm...

Re: [asterisk-users] Cisco 7970 with skinny on * 1.4.1

Richard Klingler wrote: Has any1 got their 7970 to work with * 1.4.x ? Why don't you use 7970 with SIP firmware? I'm running SIP 8.0.4SR2 without problems (Asterisk 1.2.16). Just remember that 7970 only will register if your Asterisk is at the same network - no NAT between them - check

Re: [asterisk-users] asterisk on debian

Josu Lazkano Lete wrote: I need to download the sources or just with apt-get install is enought??? apt-get is the easiest way, but won't give you the latest release. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing

Re: [asterisk-users] zttool always reports OK on TDM400P

Yuan LIU wrote: Just noticed that no matter what the line condition is, zttool always reports OK, so it's pretty useless. (In contrast, I'd get Red alert if I unplug the line connecting to an X100P.) This is the normal behavior. Only X100P will report the real status.

Re: [asterisk-users] Cisco 7912

Matt Putnam wrote: anything useful any sugestions? Are they requesting anything via TFTP? Do you have the full tftp files ready? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update

Re: [asterisk-users] Cisco 7912

Tom Lynn wrote: Do they appear to have failed as a result of Daylight Savings time? DST for 7905/7912 are set inside the lddefault/gkdefault - or the individual config file (ldMAC / gkMAC), but can't be set in advance like 7940/7960. DST is not the reason here...

Re: [asterisk-users] Adding 4 more POTS lines

Jim Freeze wrote: I suppose that is my alternative - remove the 4FXO card and add an 8FXO card. But I'm not seeing the prices you list. The Digium TDM2402B is listed at $837.00. Am I missing something? Digium is releasing a new 8 FXO/FXS card TDM800P, based on the same expansion cards used

Re: [asterisk-users] International Provider

Carlos Rojas wrote: Anyone know a good carrier of voip for international calls? Please use asterisk-biz list http://lists.digium.com/mailman/listinfo/asterisk-biz ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list

Re: [asterisk-users] AdvancedVoIP Billing ?

Noc Phibee wrote: after 2 mounth of search, i don't have see a billing solution for my small business.. Not quite sure as I didn't research very much their product, but did you check Aradial? http://www.aradial.com/voip-billing-radius.html ___

Re: [asterisk-users] g729 codec help

programming dept wrote: What happens is that if we terminate calls to carriers who accept only the g729 codec we get a 503 service unavailable. are you sure that your carrier will accept g.729? Sometimes they don't accept under iax2 and do accept under sip... check your debug for more

Re: [asterisk-users] CDR - mysql with asterisk 1.2.12 not working

On Sun, Oct 08, 2006 at 10:04:51PM -0600, Joseph wrote: What am I missing? Maybe your /etc/mysql/my.cnf ? # Instead of skip-networking you can listen only on # localhost which is more compatible and is not less secure. # bind-address = 127.0.0.1 #skip-networking

Re: [asterisk-users] CDR - mysql with asterisk 1.2.12 not working

On Sun, Oct 08, 2006 at 10:39:26PM -0600, Joseph wrote: I have bind-address = 127.0.0.1 in my.cnf the cdr was working find with asterisk 1.0.1 just after upgrade something is not connecting. I don't know if asterisk will use the localhost or the network IP to connect. Just try to comment your

Re: [asterisk-users] Dropping extra frame of G.729 ?

Noc Phibee wrote: anyone know where i can solve this problems ? : 1) By doing a quick google search; 2) By reading previous posts regarding the same issue; 3) By disabling VAD (Voice activity detection) in your device. ___ --Bandwidth and Colocation

Re: [asterisk-users] Call center reports

Technical Support wrote: Can someone point me to call center reports available from Asterisk? http://queuemetrics.loway.it/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options

Re: [asterisk-users] G729 Replacement Codec - FREE or may ne cheaper than existing one.

PLEASE DON'T CROSS POST! Kannaiyan Natesan wrote: I heard of a news, that there is a replacement codec available for g729 and accept the g729 codec data for decoding. [...] If there is any royalty need to pay, is that cheaper than the existing g729 cost?. G729 is not royalty free.

Re: [asterisk-users] Sipura SPA3000

Michael Strelnikov wrote: 1. I want all incoming calls are redirected from SPA3000 to my asterisk server. 2. Asterisk then should direct this call to my SIP phones (including Sipura) 3. In case asterisk server is down I want that call be directed straight to the handset connected to the Sipura

[asterisk-users] Cisco 7960G SIP firmware 8.4

Cisco released last Aug 23 the latest SIP firmware for Cisco 7960G. Any info? SIP Flash Image for 7940/7960 IP Phone v8.4(0) - Non CallManager ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or

Re: [asterisk-users] Portuguese sound files available?

Ricardo Carvalho wrote: I've been searching for sound files in Portuguese language to use in Asterisk for example for voicemail, but I couldn't find anything... Does anyone know where I could find them for download, if there is such thing already? Brazilian Portuguese only...

Re: [asterisk-users] OT: Changing Cisco tftp root directory

Julian Lyndon-Smith wrote: Is there any way of specifying a directory to load tftp files from instead of from the root tftp directory when booting a cisco 7960 phone ? SIPDefault.cnf: # TFTP Phone Specific Configuration File Directory tftp_cfg_dir: ./7960/ ; Example: ./sip_phone/

Re: [asterisk-users] Asterisk time not the same as unit time ?

Andre Courchesne - Consultant wrote: [EMAIL PROTECTED] tmp]# date Mon Aug 14 16:44:15 EDT 2006 The Linux command line time is connect, but not Asterisk... just guessing... not sure: date -u is showing what? ___ --Bandwidth and Colocation

Re: [asterisk-users] CDR Variable

Abdul wrote: Could any one tell me how i can change CDR variable value from extentions.conf file. for the example i would like to change the src field value different that caller phone on the first attempt of call? exten = blabla,1,Set(CDR(fieldname)=new_value) (for asterisk = 1.2)

Re: [asterisk-users] Inbound Calls SIP/2.0 404 Not Found

Mr. Jones wrote: I have 20 DIDs, some I want to send to a menu, most directly to an extension. sip debug is (really) your friend. It should give you the [context] where your DID is being send to and the 404 not found error also. A particular line to look for: Looking for

Re: [asterisk-users] Callback and Asterisks

Vic wrote: I am in immediate need of configuring an Asterix to act as wake up call system. Amazing: http://www.google.com/search?q=asterisk+wake+up+call+site%3Avoip-info.org ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users

Re: [asterisk-users] HFC-S Cards in the UK

Ron Wellsted wrote: I have been trying all the major distributors but they are all out of stock with no dates for new stock to be delivered. As you are in the UK, why not talking directly to Billion? Maybe they can help: http://www.billion.uk.com/contact.htm I'm also trying to find a new

Re: [asterisk-users] Japanese Sound Files

Nhadie wrote: Does anyone here have Japanese version of the asterisk sound files? http://www.google.com/search?q=japanese+sound+files+site%3Avoip-info.org ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To

Re: [asterisk-users] Setting CALLERID on a residential telco line

Andrea Spadaccini wrote: Is there any hope to change the caller-id on a BRI line? I guess you can do it within the range assigned to you. If you have 2 numbers, you can choose between these two numbers. Not tested, as I have only 1 number here (and still fighting with the zaphfc: empty HDLC

Re: [asterisk-users] Check call duration on active call in CLI?

voiplist wrote: Is there a command to check the call duration of an active call in the CLI? show channels verbose ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:

Re: [Asterisk-Users] ISDN

Mimmus wrote: Could some goodwill man summarize this topic for me before I engage myself in the rediscovery of warm water? Read a topic posted a few days ago: ISDN BRI NetJet You will find good advice there. If you need to buy a Cologne chipset card, check here:

Re: [Asterisk-Users] Packet8 and Asterisk, do they play nice?

Grady Neely wrote: Has anyone gotten Packet8 setup as a sip trunk for Asterisk? I have it here. With a TDM400. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:

[Asterisk-Users] ISDN BRI NetJet

I'm trying to use a Teles (netjet) ISDN BRI card with asterisk 1.2.9.1 Anyone was able to use this card with asterisk? I couldn't find much information about it. Any help? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users

Re: [Asterisk-Users] instalacion

samuel wrote: I am of Argentina, and I do not speak very well English, I cannot install asterisk in red hat 9. Don't send HTML messages to the list. Install [EMAIL PROTECTED] Please remember that [EMAIL PROTECTED] will erase all data on your HD.

Re: [Asterisk-Users] FreeBSD Digium g.729 codec seg faults on rev 30652

Kim Culhan wrote: Was running the Digium FreeBSD g.729 codec until recently when the latest Asterisk bits were obtained via svn: svn checkout http://svn.digium.com/svn/asterisk/trunk asterisk MAYBE it is the same problem: http://lists.digium.com/pipermail/asterisk-users/2006-April/147577.html

Re: [Asterisk-Users] Is NuFone Really Dead?

Andy Jefferson wrote: Went to their site today. Site claims they are still in biz. What is the story? What really happened to Nufone anyway? I'm using them for almost 3 years now. They are having some problems with OLD DIDs and toll free numbers, but newly assigned are working fine. I

Re: [Asterisk-Users] Now that Nufone is dead...

Carlos Chavez wrote: Now that Nufone is dead, what are other providers of 800 numbers that work with Asterisk? Nufone is NOT dead. It is working and I just added more funds into my account. You may also consider Asterlink. I'm a new client there, their support is a little slow, sometimes

Re: [Asterisk-Users] Is NuFone Really Dead?

Steve Totaro wrote: In what way is their email server configured badly? Wrong DNS entries. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:

Re: [Asterisk-Users] FW: NuFone Update: DIDs

Matt wrote: Is there more to this story then we know? No secrets, but at least some information may be found here: http://www.nufone.net/press/ Latest update April 28. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing

Re: [Asterisk-Users] Codec G729 / x86_64 bits.

Jefferson Carvalho wrote: I always used a compiled version for a x86 system From [...] Someone could help me on this? Yes, the folks at Digium will be more than happy to help you. Visit http://www.digium.com/en/products/voice/g729codec.php and get a licensed codec.

Re: [Asterisk-Users] Cisco 7940/7960 SIP 8.2 Freely Downloadable

Alexander Burke wrote: Just in case anyone here hadn't noticed, Cisco is apparently making 7940/7960 SIP 8.2 firmware freely downloadable by anyone: 8.2 isn't broken? Any comments? http://lists.digium.com/pipermail/asterisk-users/2006-March/143501.html

Re: [Asterisk-Users] Re: Cisco 7960 International

Shaun wrote: Well looks like the phone is sending some data... I was unable to debug the problem however.. Looking for 9011905326471222 in default (domain 204.10.xxx.xxx) Do you have a pattern in the default context that will match 9011905326471222 ?

Re: [Asterisk-Users] Cisco 7960 International

Shaun wrote: I'm having a problem with my Cisco 7960 phones with the SIP image. When i try to dial a international number i keep getting a busy signal but i dont see anything on the asterisk console (-vc) like i do when i dial local or long distance numbers. sip debug peer

Re: [Asterisk-Users] faxing setup

Corne Labuschagne wrote: How do I setup faxing in asterisk http://tinyurl.com/qddpf ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:

Re: [Asterisk-Users] OT: XML Content Manager for Cisco 79XX Phones

Corey S. McFadden wrote: PHP/MySQL based content manager for the Cisco 79XX series IP Phones Any mailing list available for this project? I have some questions/updates about this project... ___ --Bandwidth and Colocation provided by Easynews.com --

Re: [Asterisk-Users] Asterisk FXO Panasonic PBX

Waldo Rubinstein wrote: I'm looking for a reliable 2 FXO-port gateway to connect a Panasonic PBX to Asterisk. Can anyone recommend a stable and reliable one? Use 2x Sipura SPA-3000 - and you will also get 2x FXS... Or use a Digium TDM02B (2x FXO).

Re: [Asterisk-Users] name that vendor...

[EMAIL PROTECTED] wrote: Well yeah, I had no intention of buying one, I was just wondering what the hell it actually was that the seller was trying to hide. Their supplier? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users

Re: [Asterisk-Users] Cisco 7912G through NAT, problems with tones detection.

Diego Mariano Velo wrote: Hi, i have a cisco 7912G with SIP firmware, its connect to the asterisk through nat. The only problems is in the voice mailasterisk not detect the tones, therefore i cant access to my voice mail extension. Check the DTMF settings...

Re: [Asterisk-Users] Database update after hangup

Hermann Wecke wrote: I'm having a little problem to update the database after a call was placed. I have several PSTN lines and I need to split the calls between them. [...] Any idea? Solution: write to the database BEFORE the dial command. Worked very well

Re: [Asterisk-Users] International Dialing Code

Innocent Evil wrote: I am trying to download a list of international dialing codes. Would anybody please post a link to get it Google IS your friend. Did you try? Google: international country code Wikipedia: http://en.wikipedia.org/wiki/List_of_country_calling_codes

[Asterisk-Users] Database update after hangup

I'm having a little problem to update the database after a call was placed. I have several PSTN lines and I need to split the calls between them. The approach I used didn't work: [sipphone] include = trunktest ; other rules here blah blah blah [trunktest] exten =

[Asterisk-Users] Clipcomm CG-410 and caller-id from PSTN

Does anyone know if Clipcomm CG-410 [1] is able to handle caller-id information from PSTN and send it to Asterisk? Any trick on asterisk side to handle it? I tried several configurations but none worked. TIA [1] http://tinyurl.com/c6k4f ___

Re: [Asterisk-Users] stop asterisk when Idle

[EMAIL PROTECTED] wrote: I need to reboot every day an asterisk box, but I would like to do that only when asterisk is not doing anything. I have no idea *why* do you need to reboot the machine every day. What I do is a full asterisk restart - removing the modules and reinstalling them. My

Re: [Asterisk-Users] Dial command in extensions

Edwin Lam wrote: is there anyway to make the dial command return and execute the next line in the dial plan after the channel hangs up? Try g: exten = 1234,1,dial(SIP/1234,,g) exten = 1234,2,do something g: When the called party hangs up, exit to execute more commands in the current

[Asterisk-Users] Sipura 3000 x special dialling pattern (pin code)

I need to place a call using a pin code. To access an external line, the host PBX (a Ericsson MD-110) will require that I dial *72*pincode#phone_number to complete any (trunk) call. When I send the number, my Sipura 3000 will reject the call with Forbidden - wrong password on authentication

Re: [Asterisk-Users] lost g729 lic

altus wrote: We installed a box a long time ago and they bought g729a licenses Now we want to upgrade and reinstall,whats going to happen with the codec,if I give the box the same ip as always will it work? The Digium g729 license is bonded to the MAC address of all the interfaces you have.

Re: [Asterisk-Users] QoS for improvements

Kumara Jayaweera wrote: I want to run VoIP in the same LAN (15 windows clients) which we use for surfing the Internet. Some magic words: QoS Asterisk HTB TC. Not easy to find good material over the internet, but Google may give you some ideas - how to use them is another problem, which you have

[Asterisk-Users] TFTP question

I'm trying to use TFTP to update the firmware on a Cisco ATA188 but I'm receiving this error: May 1 06:51:50 mail2 in.tftpd[11499]: connect from 192.168.2.2 May 1 06:51:50 mail2 tftpd[11500]: tftpd: trying to get file: ata01234567890a May 1 06:51:50 mail2 tftpd[11500]: tftpd: serving file

[Asterisk-Users] Sipura 2000 x dual g729 channels x other choices?

I found a thread [1] last month about the poor/crappy g729 quality on Sipura units. Anyone noticed an improvement or the quality is still poor? If the Sipura firmware/g729 offers no quality yet, who else is offering a dual channel g729 ATA? I heard about Uniden, but I have no reports about

Re: [Asterisk-Users] Cisco 7960 SIP 7.4

Chris Lee wrote: Has anyone else upgraded to 7.4 and found that the date time no longer appears on the phone? This problem was pointed at the SIPPhoneReleaseNotes7_4.pdf file. What I noticed is that when the phone lost the internet connection the date/time will no longer be present on the phone.

Re: [Asterisk-Users] Sipura 2000 x dual g729 channels x other choices?

William Suffill wrote: According to the small print in the bottom graphic: http://www.sipura.com/products/spa2100.htm The SPA 2100 would give u 2 ports + 2 RJ45 as well as 2 G729 When I was placing an online order, I found this: support for two concurrent calls using the G.729 codec (in a

Re: [Asterisk-Users] Reg Asterisk

Sys Admin wrote: couldnt agree with u more !! And, please, add another one to the list: PLEASE TRIM THE ^*[EMAIL PROTECTED] MESSAGE. TIA. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com

Re: [Asterisk-Users] FWD to Vonage not working?

Brian Dingman wrote: The FWD - Vonage interconnect has been down for some time now. Vonage claimed there was a secuity issue and pulled the plug. No word when/if it will ever be working again. So I'm guessing that FWD - Packet8 falls into the same problem? Not working here for a couple of

Re: [Asterisk-Users] Broadvoice alternatives

Vicky Shrestha wrote: I have tried a lot of things to make broadvoice work with asterisk , but I failed each time. I had some problems here, mainly because I was trying to use g729 and broadvoice will only accept g711. Other than that, configuration itself took about 10~15 minutes with some

Re: [Asterisk-Users] Cisco 7960 SIP boot takes 2 minutes?

Tom wrote: What times are others seeing for the load when you reboot a phone? About the same here, but I don't care as I never reboot my phone (about once every month or two). ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com

Re: [Asterisk-Users] Cisco 7960 x Asterisk CVS-HEAD-03/13/05 - registration issues

Rich Adamson wrote: Looks like a couple of problems here. I don't believe the Cisco phone handles md5, so remove that line. As I told before, tried 3 different approaches: 1) password; md5; 2) password, no md5; 3) no password, no md5. Only the third one worked. Trying to give SOME security, I

Re: [Asterisk-Users] Cisco 7960 x Asterisk CVS-HEAD-03/13/05 - registration issues

C F wrote: how are you telling the cisco what the password is? TFTP? TFTP (SIPmacaddress.cnf) you will not see anything on * CLI unelss you do sip debug And after sip debug I saw (among other lines): [...] Retransmitting #5 (NAT): SIP/2.0 407 Proxy Authentication Required [...] SIP/2.0 401

[Asterisk-Users] Cisco 7960 x Asterisk CVS-HEAD-03/13/05 - registration issues

After fighting with a Unable to create/find channel [1] [2], I gave up on my previous installation and rebuild my asterisk from CVS-Head. I guess the Debian package available today is broken somewhere (after a previous broken release made with an old libpri package), but now I'm having another

Re: [Asterisk-Users] Where to get (cheap) VoIP

Christian faucher wrote: I read that, using a modem,I can use a standard phone line, and convert that as input for Asterisk PBX, right? Not that simple, not every modem, but yes. Also, where can I get VOIP phones? eBay ___ Asterisk-Users mailing list

Re: [Asterisk-Users] Cisco 7960 x g729 x Unable to create/find channel

Guy Decarpentrie wrote: Try to configure your Cisco type=friend in your sip.conf It is already type=friend [1234] type=friend username=1234 auth=md5 secret=supersecret deny=0.0.0.0/0.0.0.0 permit=my_ip/255.255.255.255 canreinvite=no reinvite=no host=dynamic dtmfmode=rfc2833 qualify=1800

[Asterisk-Users] Cisco 7960 x g729 x Unable to create/find channel

I'm trying to place a call from my Cisco 7960 and I'm receiving this error: Mar 1 06:19:44 NOTICE[3060]: chan_sip.c:7399 handle_request: Unable to create/find channel Mar 1 06:19:58 NOTICE[3060]: chan_sip.c:7399 handle_request: Unable to create/find channel I can't place calls, but I can

Re: [Asterisk-Users] Servidor SIP

Max wrote: Pessoal estou querendo montar um servidor SIP para fazer testes [...] wrong list. For Portuguese mailing list please subscribe to http://groups.yahoo.com/group/asteriskbr/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com

Re: [Asterisk-Users] Zaptel Red Alarm

Matthew Boehm wrote: Is there a way for asterisk to notify you of this? Send an email? Send a page? Call you? Nagios (I believe now is called NetSaint) can do this and much more. But you must have the power to configure it... after that, Nagios can send you an email, a pager, even call you and

Re: [Asterisk-Users] Anyone had a Cisco 7970 working with Asterisk?

Paul A Brown wrote: Anyone had a Cisco 7970 working with Asterisk? As 7970 uses SCCP, you can do it with asterisk. I did it with 7960. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Re: FAX

Olaf Klein wrote: Why not just kill yourself, fucking wannabe spammer? DIE DIE DIE This is *REALLY* offtopic, but Isamar is the founder of Brazilian AntiSPAM - http://antispam.org.br/ and later http://spambr.org/ Does it matter here? I don't think so, but calling he (or even me) a spammer is

Re: [Asterisk-Users] chan_sip.c:7296 handle_request: Unable to create/find channel

Roger Schreiter wrote: But when dialing a number, I get: Feb 2 09:44:45 NOTICE[20380]: chan_sip.c:7296 handle_request: Unable to create/find channel After I installed my Digium g729 license, I'm trying to place a call from my Cisco 7960 and I'm receiving the same error: Feb 19 09:47:06

Re: [Asterisk-Users] ATA's

Matthew Boehm wrote: [...] In the meantime, get a Sipura 2100, supports 2 729 calls and has both WAN/LAN ports. I was told that the Uniden DTA200 also supports 2 g729 calls. I'm buying one to test. Street price around US$ 90. Another one with dual g729 channels is MTA V102. Street price US$ 100.

Re: [Asterisk-Users] Debian way of compiling zaptel kernel modules

Tzafrir Cohen wrote: BTW: did I mention that we have binary packages for standard Debian Sarge kernels in our apt source? zaptel is the only package that never worked for me from apt-get. I need to download, compile and install the kernel (specially because the original debian install is pre

[Asterisk-Users] Help with dial command and h, H and g parameters

I'm trying to find some live examples on how to use the h, H and g parameters on the dial command (http://www.voip-info.org/wiki-Asterisk+cmd+dial) Any ideas? I was testing with the code below but after pressing * nothing happens (only after a long pause the goodye file was played) [testset]

Re: [Asterisk-Users] Cisco 7960 Beating a Dead Horse

Dave Green wrote: Following a top posted thread is a pain. not trimming the useless part of a reply is another pain... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE

Re: [Asterisk-Users] Regex in number dialed

Brian West wrote: exten = _8001133[12345789]XX.,1,Dial(SIP/france-gateway,60,tr) or exten = _8001133[1-57-9]XX.,1,Dial(SIP/france-gateway,60,tr) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com

[Asterisk-Users] Softphone x G729 x IAX

Is there any winblows softphone available offering g729 *and* IAX? I couldn't find any http://www.voip-info.org/wiki-VOIP+Phones The best choice should be dIAX, but it is only GSM. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com

Re: IAX long distance... Re: [Asterisk-Users] Asterisk for home office

Michael Graves wrote: [...] Although there have been a few (very few) times when I've notcied a brief pause after dialing and found that it had in fact dialed out on the last possible option. [...] The problem of your approach is that if you are out of credit with the first provider, your call

Re: [Asterisk-Users] Asterisk + Satellite connection

Eric Wieling aka ManxPower wrote: What company are you using for your service? Intelsat. But I'm not using it point-to-point as I'm not the primary contractor of this channel - I'm buying internet access. ___ Asterisk-Users mailing list [EMAIL

[Asterisk-Users] sipgate x asterisk: problems to receive PSTN calls?

I noticed that I'm no longer able to receive calls from PSTN to my SipGate DID number. I changed the sip.conf and extension.conf as per http://www.voip-info.org/tiki-index.php?page=Sipgate but the problem remains... However, I can receive calls from another sipgate user. The problem is only

Re: [Asterisk-Users] Asterisk + Satellite connection

Federico Gonzalez wrote: I have an Asterisk with one local Cisco ATA and one remote Cisco ATA connected to the Asterisk, the remore connection is a satellite link with an 900ms delay. This is the same delay I have here. Never less than 900, sometimes over 1500 ms. Check

Re: [Asterisk-Users] Question on IXAy (IAXy actually)

nkb wrote: So, do I still need to have an Asterisk server connected to my IAXy even after I've made provision for it? You can only connect IAXy to an asterisk server. Yours or from a VoIP provider. Like, can I just carry this IAXy around(after provision) and just plug into any broadband

Re: [Asterisk-Users] Playing reveived message WAV file

Joseph wrote: After somebody records a message asterisk notifies me and encloses the WAV file. Though I'm not sure if this is a WAV format. I can not play it. How to play received message? Did you try to use Windows Merdia Player? In other hand, if you are receiving a .GSM file, you can use the

Re: [Asterisk-Users] Question on IXAy

nkb wrote: I was wondering if I could use IXAy to forward my call via the internet to my destination, something of similar function to SIPURA 3000? The IAXy is similar to the Sipura 1000 or 2000, or the Cisco ATA 18x... You can use it to connect to a VoIP server with the IAX2 protocol (instead

Re: [Asterisk-Users] Fw: Gift for Mark Spencer

Steve Kann wrote: [...] I've gotten 270 already: [...] I've got only 1. But... what is the main issue now? Is this topic just another (endless) troll or someone is trying to get some config help for *? ___ Asterisk-Users mailing list [EMAIL PROTECTED]

Re: [Asterisk-Users] ATA186 V2.15.ms

Damon Estep wrote: [...] Contains a link you need for firmware. Correct URL is http://www.voip-info.org/wiki-Asterisk+phone+cisco+ATA18x URL:http://www.voip-info.org/wiki-Asterisk+phone+cisco+ATA18x ___ Asterisk-Users mailing list [EMAIL PROTECTED]

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