Re: [asterisk-users] Remote Unix Connection
Nope, Its a totally normal self-built Asterisk. Dan Zeeshan Zakaria zisha...@gmail.com wrote: Do you use FreePBX by any chance? Zeeshan A Zakaria -- www.ilovetovoip.comhttp://www.ilovetovoip.com On 2010-10-16 6:38 PM, Dan Journo d...@keshercommunications.commailto:d...@keshercommunications.com wrote: Serious answer: Looks like a process running asterisk -r. Do you have any sort of AGI, cron j... Thanks for lightning my day! Is there any way to debug this because as far as i'm aware, there's nothing running that command, (except for me) -- _ -- Bandwidth and Colocati... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Remote Unix Connection
I think I've seen this where I am trying to start another instance of asterisk using safe_asterisk, when I already have an instance running Julian On 16 October 2010 22:36, Dan Journo d...@keshercommunications.com wrote: Hi, Does anyone know where this is suddenly coming from? -- Remote UNIX connection -- Remote UNIX connection disconnected -- Remote UNIX connection -- Remote UNIX connection disconnected -- Remote UNIX connection -- Remote UNIX connection disconnected Thanks Dan p.s. sorry about the last post. hit the mouse by mistake and it sent the email. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Follow Ode To Politics by HB Tasker at http://twitter.com/HBTasker -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Detect incoming fax on PSTN and route to fax machine on DADHI extension?
mù:l;:kj,nb hgyuè 2010/10/16 Frank Tarczynski ft...@mindspring.com I'm running an AsteriskNow V1.7.1 with both a PSTN connection and fax machine. Both are connected to a DAHDI board. I'd like to route incoming PSTN fax calls to the extension of the fax machine and process non-fax calls through different dialplan.logic. What's the best way to go about doing this? I've looked into Fax for Asterisk, bit I'm not sure that I want it or NVFax detection. Any pointers to share? Hi, The trouble with using the same DID for both voice and fax calls is with the way fax priority seems to function, user experience is a bit rough: - user phone is ringing - user answers and hear a brief tone while fax detection happens - then user is hearing an hangup tone (while the fax is received in another channel) What is missing here is the possibility to playback a message to end user instead of playing a tone. If you find a way to work around this, please, do not hesitate to share. Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Remote Unix Connection
Some service is definitely connecting to your asterisk using AMI. Such services use username/password described in manager.conf. Usually its is some monitoring service. Although the message says 'remote UNIX connection' but it can be very well something from localhost. I would suggest to use tcpdump to find out the IP of this service. AMI uses TCP port 5038. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-10-17 3:37 AM, Dan Journo d...@keshercommunications.com wrote: Nope, Its a totally normal self-built Asterisk. Dan Zeeshan Zakaria zisha...@gmail.com wrote: Do you use FreePBX by any chance? Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-10-16 6:38 ... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fraud advice (Also advice on using ipbanning)
- Original Message - When we designed our systems on asterisk we designed it to me multi-tenant. Se we use customer prefixes on all extensions. This allows us to have multiple customers using the same extension pools. It also reduces the hack foot print as hackers must know the prefix for a customer to try and brute force things. All passwords use 8+ characters with alfa/numeric and special characters. As I see it Asterisk does very good keeping out the hackers if you use a solid design in your peer and dialplans. At the least put an alpha character post or pre other wise you are just asking for it. Use your head you can be smarter then they are. We are looking into ipban as well. If any one has an example of ipban I would love to see how best to implement it. In a 4 year period we have not had a breach but we do get about 10 to 15 hack attempts a week. We have blocking scripts that block ip's at the primary firewall but I would like to trigger the ipban at each switch level. Could I also use the ipban method to trigger the audo updates to our primary firewalls? Any advice is appreciated. Bryant You could also use OSSEC http://www.ossec.net and a custom decoder and rule: decoder name=local-asterisk-denied prematchNOTICE[\d+] \S+: Registration from /prematch regex offset=after_prematch^\S+ failed for '(\d+.\d+.\d+.\d+)'/regex ordersrcip/order /decoder rule id=110005 level=5 decoded_aslocal-asterisk-denied/decoded_as descriptionAsterisk Potentially Under Attack/description /rule rule id=110006 level=10 frequency=5 timeframe=10 if_matched_sid110005/if_matched_sid same_source_ip / descriptionAsterisk Under Brute Force Attack/description /rule -- Thanks, Phil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Remote Unix Connection
Some service is definitely connecting to your asterisk using AMI. Such services use username/password described in manager.conf. Usually its is some monitoring service. Although the message says 'remote UNIX connection' but it can be very well something from localhost. I would suggest to use tcpdump to find out the IP of this service. AMI uses TCP port 5038. I ran the following command and waited for the cli to show the remote unix connection message a few times. [r...@sip2 ~]# tcpdump port 5038 -w tcpdump.log -s0 tcpdump: listening on eth0, link-type EN10MB (Ethernet), capture size 65535 bytes The result was 0 packets captured 0 packets received by filter 0 packets dropped by kernel Therefore, it seems like nothing is connecting to the AMI? Also, in manager.conf enabled=no Any other ideas? Is this a bug? Thanks Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Create channel bank with TDMoE
Hi, I can use TDMoE for trunking of Asterisks. it is no problem. In use of TDMoE for channel bank, my problem is: All of channels in trunk are OFFHOOK! Can you help me? On 10/14/10, Luis Antonio Prata Barbosa luispratalis...@gmail.com wrote: Hi Karim, Here you find a basic example for configuration: http://www.voip-info.org/wiki/index.php?page=Asterisk+TDMoE First step is to connect two asterisk using this basic configuration. did you get that ? There are other possibilites when using two asterisks like connecting them using IAX trunking. Luis A P Barbosa 2010/10/14 Karim Davoodi karimdavo...@gmail.com Tanks I want to create a channel bank with TDMoE. I have not to buy a product. Best, Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Remote Unix Connection
Do you have freepbx anywhere it always tries to connect -- via a socket I think and it usually uses the manager, so if you disable the manager it will break things. Also take the port stanza off of the tcpdump and you will soon see what is connecting. You will get other stuff, but this will tell you. Dan Journo d...@keshercommunications.com wrote: Some service is definitely connecting to your asterisk using AMI. Such services use username/password described in manager.conf. Usually its is some monitoring service. Although the message says 'remote UNIX connection' but it can be very well something from localhost. I would suggest to use tcpdump to find out the IP of this service. AMI uses TCP port 5038. I ran the following command and waited for the cli to show the remote unix connection message a few times. [r...@sip2 ~]# tcpdump port 5038 -w tcpdump.log -s0 tcpdump: listening on eth0, link-type EN10MB (Ethernet), capture size 65535 bytes The result was 0 packets captured 0 packets received by filter 0 packets dropped by kernel Therefore, it seems like nothing is connecting to the AMI? Also, in manager.conf enabled=no Any other ideas? Is this a bug? Thanks Dan Alternatives: -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Good Day! 2010.10.17.21.4.4
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Re: [asterisk-users] Remote Unix Connection
On Sun, Oct 17, 2010 at 6:55 AM, cov...@ccs.covici.com wrote: Do you have freepbx anywhere it always tries to connect -- via a socket I think and it usually uses the manager, so if you disable the manager it will break things. Also take the port stanza off of the tcpdump and you will soon see what is connecting. You will get other stuff, but this will tell you. Dan Journo d...@keshercommunications.com wrote: Some service is definitely connecting to your asterisk using AMI. Such services use username/password described in manager.conf. Usually its is some monitoring service. Although the message says 'remote UNIX connection' but it can be very well something from localhost. I would suggest to use tcpdump to find out the IP of this service. AMI uses TCP port 5038. I ran the following command and waited for the cli to show the remote unix connection message a few times. [r...@sip2 ~]# tcpdump port 5038 -w tcpdump.log -s0 tcpdump: listening on eth0, link-type EN10MB (Ethernet), capture size 65535 bytes The result was 0 packets captured 0 packets received by filter 0 packets dropped by kernel Therefore, it seems like nothing is connecting to the AMI? Also, in manager.conf enabled=no Any other ideas? Is this a bug? Thanks Dan Alternatives: -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Sounds like either FreePBX or some other script using astmanproxy or just the AMI in general. Another possible cause is a script (or terminal) constantly accessing asterisk -r or rasterisk (+ any other arguments) to either run an Asterisk CLI command, or to just watch' the console output. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Remote Unix Connection
- Original Message - Sounds like either FreePBX or some other script using astmanproxy or just the AMI in general. Another possible cause is a script (or terminal) constantly accessing asterisk -r or rasterisk (+ any other arguments) to either run an Asterisk CLI command, or to just watch' the console output. It's asterisk -r or asterisk -rx. The message that says remote unix connection means a remote connection to Asterisk over the UNIX domain socket, which is what the remote console uses. AMI connections have a different message associated with them. -- Russell Bryant Digium, Inc. | Engineering Manager, Open Source Software 445 Jan Davis Drive NW -Huntsville, AL 35806 - USA jabber: rbry...@digium.com-=-skype: russell-bryant www.digium.com -=- www.asterisk.org -=- blogs.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Remote Unix Connection
SSH?? JN Dan Journo wrote: Hi, Does anyone know where this is suddenly coming from? -- Remote UNIX connection -- Remote UNIX connection disconnected -- Remote UNIX connection -- Remote UNIX connection disconnected -- Remote UNIX connection -- Remote UNIX connection disconnected Thanks Dan p.s. sorry about the last post. hit the mouse by mistake and it sent the email. -- Dog is my Co-pilot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DMTF Mode
I'm really struggling with this DTMF issue. In order to test it, I've tried a few different providers and DTMF RFC2833 does work with any of them, even though a few of them insist that it is. Is this a bug with 1.4.36? Has anyone else experienced this problem? The Asterisk CLI is showing the DTMF signals properly, the tones just don't get repeated to the opposite end of the call properly. Any help would be greatly appreciated! Thanks Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Detect incoming fax on PSTN and route to fax machine on DADHI extension?
On Sun, Oct 17, 2010 at 3:59 AM, Olivier oza_4...@yahoo.fr wrote:
mù:l;:kj,nb hgyuè
2010/10/16 Frank Tarczynski ft...@mindspring.com
I'm running an AsteriskNow V1.7.1 with both a PSTN connection and fax
machine. Both are connected to a DAHDI board. I'd like to route
incoming PSTN fax calls to the extension of the fax machine and process
non-fax calls through different dialplan.logic.
What's the best way to go about doing this? I've looked into Fax for
Asterisk, bit I'm not sure that I want it or NVFax detection.
Any pointers to share?
Hi,
The trouble with using the same DID for both voice and fax calls is with the
way fax priority seems to function, user experience is a bit rough:
- user phone is ringing
- user answers and hear a brief tone while fax detection happens
- then user is hearing an hangup tone (while the fax is received in another
channel)
What is missing here is the possibility to playback a message to end user
instead of playing a tone.
If you find a way to work around this, please, do not hesitate to share.
Regards
extensions.conf:
[incoming]
exten = fax,1,NoOp(Fax Detected) ;; the fax line
exten = fax,n,GoTo(incoming-fax,s,1)
exten = fax,n,Hangup();; the fax machine
exten =s,1,Answer()
exten =s,n,Wait(6)
exten =s,n,Dial(${House_Phones},60)
You may still get one or two rings, so don't run to answer :-)
See:
https://issues.asterisk.org/view.php?id=17064
sean
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Re: [asterisk-users] DMTF Mode
Whats payload used for in rfc2833? I'm wondering if that is incompatible. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo Sent: 17 October 2010 14:54 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DMTF Mode I'm really struggling with this DTMF issue. In order to test it, I've tried a few different providers and DTMF RFC2833 does work with any of them, even though a few of them insist that it is. Is this a bug with 1.4.36? Has anyone else experienced this problem? The Asterisk CLI is showing the DTMF signals properly, the tones just don't get repeated to the opposite end of the call properly. Any help would be greatly appreciated! Thanks Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DMTF Mode
On Sun, 2010-10-17 at 09:53 -0400, Dan Journo wrote: I'm really struggling with this DTMF issue. In order to test it, I've tried a few different providers and DTMF RFC2833 does work with any of them, even though a few of them insist that it is. Is this a bug with 1.4.36? Has anyone else experienced this problem? The Asterisk CLI is showing the DTMF signals properly, the tones just don't get repeated to the opposite end of the call properly. Any help would be greatly appreciated! Thanks Dan Have you tried relaxdtmf or rfcc2833compensate? relaxdtmf=yes rfc2833compensate=yes ---fred Fred Posner http://qxork.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DMTF Mode
Have you tried relaxdtmf or rfcc2833compensate? Just tried it, but it didnt make a difference. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DMTF Mode
Well, I connected my sip phone directly to the provider and totally skipped the asterisk server. DTMF rfc2833 worked fine! Looks like Asterisk is doing something that's preventing it from working. However, looking at the tcpdump from asterisk, it all looks fine. Any ideas? Thanks Dan -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo Sent: 17 October 2010 15:42 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DMTF Mode Have you tried relaxdtmf or rfcc2833compensate? Just tried it, but it didnt make a difference. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Remote Unix Connection
I took a look in the source -- it is definitely asterisk -r (or rasterisk) and not AMI. AMI logs something like Manager 'username' logged on from 127.0.0.1. Check the timing between calls and see if a pattern appears. If so, it is some sort of cron/scheduled job. If not, keep looking! -M -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Error with Connecting Two Asterisk BOX with IAX
Hello, I'm trying to conect two 1.6.2.13 Asterisk server with IAX. This is my configuration: Asterisk A: iax.conf register = coiax:pa...@69.164.207.166 [smiax] type=friend host=dynamic trunk=yes secret=pass2 context=phones deny=0.0.0.0/0.0.0.0 permit=69.164.207.166/255.255.255.255 qualify=yes Console: iax2 registry 69.164.207.166:4569 N coiax 69.164.197.105:456960 Registered iax2 peers smiax69.164.207.166 (D) 255.255.255.255 4569 (T) OK (3 ms) Asterisk B: register = smiax:pa...@69.164.197.105 [coiax] type=friend host=dynamic trunk=yes secret=pass1 context=phones deny=0.0.0.0/0.0.0.0 permit=69.164.197.105/255.255.255.255 qualify=yes Console iax2 registry 69.164.197.105:4569 N smiax 69.164.207.166:456960 Registered iax2 peers coiax69.164.197.105 (D) 255.255.255.255 4569 (T) OK (3 ms) When I try to call from Asterisk A to Asterisk B I receive this error Asterisk A WARNING[28759]: chan_iax2.c:10287 socket_process: Call rejected by 69.164.207.166: No authority found AsteriskB NOTICE[26563]: chan_iax2.c:10522 socket_process: Host 69.164.197.105 failed to authenticate as coiax What's wrong? Thank you in advance. Regards - Bakko -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Detect incoming fax on PSTN and route to fax machine on DADHI extension?
On Sat, 2010-10-16 at 17:36 -0400, Paul Belanger wrote: On Sat, Oct 16, 2010 at 4:59 PM, Frank Tarczynski ft...@mindspring.com wrote: Any pointers to share? chan_dahdi.conf faxdetect=incoming extensions.conf exten = fax,1,Dial(DAHDI/4) This is what I do and it works great, as long as what is on the DAHDI channel (4 in this example) is just a fax machine or computer with fax modem (my case). --Greg -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Create channel bank with TDMoE
Karim, Try these 2 configurations for FXS and FXO hardware: For FXS hardware Remote asterisk (channel bank): Configure FXS to immediate answer. use s,1,Dial( dahdi / TDMoE_channel# ) in FXS context use s,1,Dial( dahdi / FXS_channel#) in TDMoE_channel# context Configure TDMoE_channel# to use fxoks signaling. Main asterisk Configure TDMoE_channel# to use fxsks signaling. For FXO hardware Remote asterisk (channel bank): use s,1,Dial( dahdi / TDMoE_channel# ) in FXO context use s,1,Dial( dahdi / FXO_channel#) in TDMoE_channel# context Configure TDMoE_channel# to use fxsks signaling com immediate answer Main asterisk Configure TDMoE_channel# to use fxoks signaling. I didn't test, but I hope it helps. Luis A P Barbosa 2010/10/17 Karim Davoodi karimdavo...@gmail.com Hi, I can use TDMoE for trunking of Asterisks. it is no problem. In use of TDMoE for channel bank, my problem is: All of channels in trunk are OFFHOOK! Can you help me? On 10/14/10, Luis Antonio Prata Barbosa luispratalis...@gmail.com wrote: Hi Karim, Here you find a basic example for configuration: http://www.voip-info.org/wiki/index.php?page=Asterisk+TDMoE First step is to connect two asterisk using this basic configuration. did you get that ? There are other possibilites when using two asterisks like connecting them using IAX trunking. Luis A P Barbosa 2010/10/14 Karim Davoodi karimdavo...@gmail.com Tanks I want to create a channel bank with TDMoE. I have not to buy a product. Best, Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DMTF Mode
Dan, I have to say, with 1.6.2.13 I am having the same or similar problems. I switched the affected customers to inband, haven't had time to delve too deep into the problem. what I did see though, is that rfc2833 are indeed being sent, but not recognized by two of the sip providers we use. Ron Op 17-10-10 17:20, Dan Journo schreef: Well, I connected my sip phone directly to the provider and totally skipped the asterisk server. DTMF rfc2833 worked fine! Looks like Asterisk is doing something that's preventing it from working. However, looking at the tcpdump from asterisk, it all looks fine. Any ideas? Thanks Dan -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo Sent: 17 October 2010 15:42 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DMTF Mode Have you tried relaxdtmf or rfcc2833compensate? Just tried it, but it didnt make a difference. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Meetme
Hi , Is it possible to have two meetme room in asterisk 1.6 which each one have a different language? I mean, one room the annoucement is in Portuguese an another in english? Today I can change over the sip.conf and it is valid for all room. regards! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Meetme
Hi Flavio, try with this funtion before the line with the english meetme application Set(CHANNEL(language)=en) and Set(CHANNEL(language)=pr) before the line with the portugues meetme application Regards - Bakko-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Meetme
hi Bakko, thanks! Acctualy, I had tried this but still don´t works! [conference]exten = 1001,3,MeetMe(1001,ipdM)exten = 1001,4,Set(CHANNEL(language)=pt_BR)exten = 1001,5,Playback(pt_BR/vm-goodbye)exten = 1001,6,Hangup this is my config! What´s wrong? thanks again! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda From: asannu...@gmail.com To: asterisk-users@lists.digium.com Date: Sun, 17 Oct 2010 16:36:34 -0500 Subject: Re: [asterisk-users] Meetme Hi Flavio, try with this funtion before the line with the english meetme application Set(CHANNEL(language)=en) and Set(CHANNEL(language)=pr) before the line with the portugues meetme application Regards - Bakko -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Meetme
Hi Flavio is: [conference] exten = 1001,3,Set(CHANNEL(language)=pt_BR) exten = 1001,4,MeetMe(1001,ipdM) exten = 1001,5,Playback(vm-goodbye) exten = 1001,6,Hangup Regards - Bakko-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Meetme
It works!!! Thanks a lot! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda From: asannu...@gmail.com To: asterisk-users@lists.digium.com Date: Sun, 17 Oct 2010 17:56:37 -0500 Subject: Re: [asterisk-users] Meetme Hi Flavio is: [conference] exten = 1001,3,Set(CHANNEL(language)=pt_BR) exten = 1001,4,MeetMe(1001,ipdM) exten = 1001,5,Playback(vm-goodbye) exten = 1001,6,Hangup Regards - Bakko -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DMTF Mode
On Sun, Oct 17, 2010 at 4:16 PM, Ron Arts ron.a...@neonova.nl wrote: I have to say, with 1.6.2.13 I am having the same or similar problems. I switched the affected customers to inband, haven't had time to delve too deep into the problem. I recommend reviewing [1] and look for possible regressions. There have been some update to RFC2833 over the last few months. [1] http://svnview.digium.com/svn/asterisk/branches/1.6.2/main/rtp.c?view=log -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX2 works one direction, but not the other...
I'm having trouble getting an IAX2 connection between a couple of servers. Ican make calls from server B to server A, but when I call from Server A to serverB, I get "No authority found".On ServerA I am running Asterisk 1.6.2.9On ServerB I'm running 1.6.2.13Any hints for me? The registrations in both directions seem to work fine when I do an iax2 reload from the CLI.config file snips shown below.ThanksCassius Smith=On server B, I have the following:[general]register = serverB:longsecretpasswo...@servera_ip[serverA]type=friendhost=dynamicauth=md5secret=longsecretpassword1context=no911[serverB]type=friendhost=dynamicauth=md5secret=longsecretpassword2context=no911On server A, I have the following:[general]register = serverA:longsecretpasswo...@serverb_ip[serverB]type=friendhost=dynamicauth=md5secret=longsecretpassword2context=no911[cary]type=friendhost=dynamicauth=md5secret=longsecretpassword1context=no911 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] app_swift for Asterisk 1.8
Just thought I'd let everyone know I've got a new beta version of app_swift up for Asterisk 1.8 on http://forge.asterisk.org. - Darren -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
