[asterisk-users] Test numbers Worldwide
Hi, We are searching for a pool of test numbers to call from Asterisk, record voice and test it with our non-intrusive voice quality testing software (NIQA). The problem is that we could find some test numbers, but our customer would like to have a global pool of test numbers, so that we can call them and test voice quality. Greatly appreciate any help! Thank you! Sevana Oy, Finland http://www.sevana.fi-- _ -- 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] E1 configuration
dear flaviormiranda thanks for your kind of help. I want to know this part( mfcr2 ) what does it mean? signalling=mfcr2 mfcr2_variant=br mfcr2_get_ani_first=no mfcr2_max_ani=20 mfcr2_max_dnis=4 mfcr2_category=national_subscriber mfcr2_logdir=span1 mfcr2_call_files=yes mfcr2_logging=all mfcr2_mfback_timeout=-1 mfcr2_metering_pulse_timeout=-1 mfcr2_allow_collect_calls=yes mfcr2_double_answer=no mfcr2_immediate_accept=yes mfcr2_forced_release=no mfcr2_charge_calls=yes If i want to install te121p on elastix what should i do? best regards On Wed, Oct 27, 2010 at 3:18 AM, Flavio Miranda flaviormira...@hotmail.comwrote: Hi, /etc/dahdi/system.conf Att,# Span 1: WCT1/0 Digium Wildcard TE110P T1/E1 Card 0 (MASTER) HDB3/ span=1,1,0,cas,hdb3 cas=1-15:1101 dchan=16 cas=17-31:1101 #echocanceller=mg2,1-15,17-31 /etc/asterisk/chan_dahdi.conf [trunkgroups] [channels] usecallerid=yes callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes signalling=mfcr2 mfcr2_variant=br mfcr2_get_ani_first=no mfcr2_max_ani=20 mfcr2_max_dnis=4 mfcr2_category=national_subscriber mfcr2_logdir=span1 mfcr2_call_files=yes mfcr2_logging=all mfcr2_mfback_timeout=-1 mfcr2_metering_pulse_timeout=-1 mfcr2_allow_collect_calls=yes mfcr2_double_answer=no mfcr2_immediate_accept=yes mfcr2_forced_release=no mfcr2_charge_calls=yes ;language=pt_BR context=Saida-de-ligacoes group=0 callgroup=0 pickupgroup=0 channel = 1-15,17-31 immediate=no #include dahdi-channels.conf /etc/asterisk/dahdi-channels.conf ; Span 1: WCT1/0 Digium Wildcard TE110P T1/E1 Card 0 (MASTER) HDB3/ group=0,11 context=Saida-de-ligacoes switchtype = national signalling = pri_cpe channel = 1-15,17-31 context = default group = 63 Flavio Roberto Miranda MSN:flaviormira...@hotmail.com msn%3aflaviormira...@hotmail.com Skype: flaviormiranda -- Date: Wed, 27 Oct 2010 00:15:01 +0330 From: seighal...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] E1 configuration dear please send these configurations. thanks On Tue, Oct 26, 2010 at 3:04 PM, Flavio Miranda flaviormira...@hotmail.com wrote: hi, So, I think it depend of what environment are you setting up your link . In my case, E1 R2 Digital Brazil standard (Variant=br), I needed to change dahdi-channels parameter,chan_dahdi.conf , system.conf as well. If you need I can send you such configuration. good look! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda -- Date: Tue, 26 Oct 2010 14:24:13 +0330 From: seighal...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] E1 configuration hi my friend would ou say what did you do for solving the problem? because i use a digium te121p and have many problems. thanks in advance On Mon, Oct 25, 2010 at 4:50 PM, Flavio Miranda flaviormira...@hotmail.com wrote: Sorry, thats right!! I the nest email I will post here what I did in order to sove my problem! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda -- Date: Sun, 24 Oct 2010 23:59:27 -0700 From: shakeel.abbas@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] E1 configuration although I don't need the solution personally But would like to request you that instead of posting forget it . if you post the solution to the problem it will be more helpful. In case some one else faces the same problem he can use your solution Good luck On Sun, Oct 24, 2010 at 7:10 PM, Flavio Miranda flaviormira...@hotmail.com wrote: Forget it !! After several attempts, I have solved !!! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda -- From: flaviormira...@hotmail.com To: asterisk-users@lists.digium.com Date: Sun, 24 Oct 2010 22:28:16 -0200 Subject: [asterisk-users] E1 configuration Hi all, Please, anybody that have some knowllege about E1 configuration could give some guidance about it? I trying to set an Asterisk with E1 CAS signalling and everything looks good, but when I try to go out with calls I receive the follow message: == Using SIP RTP CoS mark 5 -- Executing [21341...@local:1] Dial(SIP/4804-, DAHDI/g11/21341400,,t) in new stack == Everyone is busy/congested at this time (1:0/0/1) == Spawn extension (local, 21341400, 2) exited non-zero on 'SIP/4804-' The boad has succesfully installed: Digium Wildcard TE110P T1/E1 Card 0 OK 0 0 0 CAS HDB3 0 db (CSU)/0-133 feet (DSX-1) the channels are correct and mfcr2 too, but the calls dont go out. Thanks for any help.
[asterisk-users] Asterisk Strange Problem while call received from customer On PRI.
HI group, this is very strange problem with me when i received a call from Germany i am able to receive call on my PRI line everything is fine User connected with IVRS and user trying to enter a extension number like *1660976 *call goes to users company extension starting with *16.* is this very strange with me on asterisk. how this possible even if i want to explain to user in technical terms. i don't know user is using which PBX system. i think there is one possibility which i think User entered a number but i do not receive anything and user will try to re-enter number again in this time user PBX will redirect call to extension with 16 let give your thoughts regarding this. regards Dhaval * * -- _ -- 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] Auto provisioning from public server
On 10/26/2010 06:30 PM, Andrew Latham wrote: snom phones can do http digest authentication... I think this digest authentication is for accessing the phone's web interface, not for contacting a provisioning server Jonas. -- _ -- 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] Auto provisioning from public server
On Tue, 2010-10-26 at 17:31 +0200, Jonas Kellens wrote: Hello, has anyone experience with auto provisioning IP-phones on different locations through a central public provisioning server ? You use http or https ? Is there a danger that one uses a different MAC-address in the provisioning link to obtain SIP username / password settings ? Kind regards, Jonas. The company we use for provisioning snom phones delete the un pass info from the server once it has been picked up for the first time. That way no one else can access it by spoofing the MAC address -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- 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] Auto provisioning from public server
On Tue, Oct 26, 2010 at 11:31 AM, Jonas Kellens jonas.kell...@telenet.bewrote: Hello, has anyone experience with auto provisioning IP-phones on different locations through a central public provisioning server ? You use http or https ? Is there a danger that one uses a different MAC-address in the provisioning link to obtain SIP username / password settings ? Kind regards, Jonas. Yes, there is a danger, especially with TFTP, but also with FTP to a lesser degreee. If someone guessed correctly, they could download the config file for another phone. Thanks, Steve T -- _ -- 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] Auto provisioning from public server
On Wed, Oct 27, 2010 at 4:04 AM, Ishfaq Malik i...@pack-net.co.uk wrote: On Tue, 2010-10-26 at 17:31 +0200, Jonas Kellens wrote: Hello, has anyone experience with auto provisioning IP-phones on different locations through a central public provisioning server ? You use http or https ? Is there a danger that one uses a different MAC-address in the provisioning link to obtain SIP username / password settings ? Kind regards, Jonas. The company we use for provisioning snom phones delete the un pass info from the server once it has been picked up for the first time. That way no one else can access it by spoofing the MAC address -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 What company is that? I have seen companies that do this but have never felt very secure handing the keys to the castle over to a 3rd party service. It seems like a good idea, but I have trust issues, especially when you top off your prepaid service with $15k a week. Thanks, Steve T -- _ -- 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] Auto provisioning from public server
On Wed, 2010-10-27 at 04:10 -0400, Steve Totaro wrote: On Wed, Oct 27, 2010 at 4:04 AM, Ishfaq Malik i...@pack-net.co.uk wrote: On Tue, 2010-10-26 at 17:31 +0200, Jonas Kellens wrote: Hello, has anyone experience with auto provisioning IP-phones on different locations through a central public provisioning server ? You use http or https ? Is there a danger that one uses a different MAC-address in the provisioning link to obtain SIP username / password settings ? Kind regards, Jonas. The company we use for provisioning snom phones delete the un pass info from the server once it has been picked up for the first time. That way no one else can access it by spoofing the MAC address -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 What company is that? I have seen companies that do this but have never felt very secure handing the keys to the castle over to a 3rd party service. It seems like a good idea, but I have trust issues, especially when you top off your prepaid service with $15k a week. Thanks, Steve T It's our hardware supplier, the provisioning server is a free service if you purchase the hardware from them. I totally understand your point but there's always got to be some trust at some point whether it be in your suppliers or even your employees or co workers They are a UK based company called Provu, I'm pretty sure they are active on this list too -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- 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] Auto provisioning from public server
On 10/27/2010 10:06 AM, Steve Totaro wrote: On Tue, Oct 26, 2010 at 11:31 AM, Jonas Kellens jonas.kell...@telenet.be mailto:jonas.kell...@telenet.be wrote: Hello, has anyone experience with auto provisioning IP-phones on different locations through a central public provisioning server ? You use http or https ? Is there a danger that one uses a different MAC-address in the provisioning link to obtain SIP username / password settings ? Kind regards, Jonas. Yes, there is a danger, especially with TFTP, but also with FTP to a lesser degreee. If someone guessed correctly, they could download the config file for another phone. Thanks, Steve T If I find a way to implement it... https would be safer ? Or is the only safe way to work with certificates that are loaded on the IP-phone ?! Jonas. -- _ -- 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] Mobile Phones and Asterisk
Hi, On 10/27/2010 05:55 AM, GBR Icasiano, Ryan A. wrote: anyone??? regards, RYAN ICASIANO Hi, I changed my sip.conf and added call-limit. At first I thought it works ok, since i tried calling a cellphone that is currently busy(phone answers 1st softphone, then another softphone calls the same number, it now returns INUSE). But then, i tried calling a different number while the first phone is busy, but it returns INUSE. It seems that the status being returned was from the peer itself(both phones uses the same peer) and not from the device(mobile phone) which i believe is more logical. I also tried using DIALSTATUS(which of course you need to DIAL first), but then I only hear a busy tone and the dialstatus will return a noanswer. Do I have to configure it first in order to capture the busy status of a device? Have you done something similar to this? I'm using ver. 1.6. Thanks in advance. I'm not sure I understand your setup. Are you using SIP for trunking, or for extensions? Are you calling a normal mobile phone, or a SIP client on a mobile phone? Sebastian regards, RYAN ICASIANO From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of GBR Icasiano, Ryan A. [raicasi...@globalbridgeresources.com] Sent: Tuesday, October 26, 2010 10:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Mobile Phones and Asterisk Hi, Is the dev_state can also be used to track a mobile phone's status via SIP? I tried it on several phones(nokia, samsung) but it returns NOANSWER but i can hear a beep beep beep sound indicating that it is currently busy. regards, RYAN ICASIANO __ From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Sebastian [s...@open-t.co.uk] Sent: Tuesday, October 26, 2010 7:50 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Mobile Phones and Asterisk On 10/26/2010 12:30 PM, ayodele abejide wrote: Hello Jonathan, The solution would work only if the ISP has one public address, but in my solution they have a pool of public address, any other possible solution? With dynamic dns, you either install a piece of software on your server (dynamic dns client) or you use the facility provided by your router (some firewall/router/access point combo's have them). This software updates automatically the record with dyndns every time your IP address changes. Sebastian ABEJIDE, Ayodele A. (CCNA) +2348039269311 From: ayodeleabej...@hotmail.com To: asterisk-users@lists.digium.com Date: Tue, 26 Oct 2010 11:01:09 + Subject: Re: [asterisk-users] Mobile Phones and Asterisk thanks i would check it up ABEJIDE, Ayodele A. (CCNA) +2348039269311 Date: Tue, 26 Oct 2010 12:52:30 +0200 From: jonathan@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Mobile Phones and Asterisk Try http://www.dyndns.com/ that should solve your problem with dynamic IPs. Regards, Jonathan On Tue, Oct 26, 2010 at 12:40 PM, ayodele abejide ayodeleabej...@hotmail.commailto:ayodeleabej...@hotmail.com wrote: Dear Asterisk-Users, I have this Asterisk Box I run in my house, I need to terminate and originate remote calls through the box via internet (SIP), the problem is in Nigeria most ISPs would not provide you with Public Addresses, all they provide is dynamic Natted addresses which change each time one connects, I have thought of all possible solutions and cannot come up with one, can anyone please help. Thanks in anticipation ABEJIDE, Ayodele A. (CCNA) +2348039269311 -- _ -- 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 -- Personal webpage - www.jonbaraq.euhttp://www.jonbaraq.eu -- _ -- 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
Re: [asterisk-users] Fax Degium channel License
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[asterisk-users] phoneprov
Hi List, Can anyone please tell me how to use the phoneprov.conf to provision my client's atas. I read the file but dont know how to actually use it. -- Best Regards Rizwan Qureshi -- _ -- 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] Auto provisioning from public server
Hi, On Tue, Oct 26, 2010 at 05:31:00PM +0200, Jonas Kellens wrote: Hello, has anyone experience with auto provisioning IP-phones on different locations through a central public provisioning server ? You use http or https ? What is it exactly that you want to guarantee? Authenticating the client? The server? Avoiding any leak of data to some eavesdropper? Is there a danger that one uses a different MAC-address in the provisioning link to obtain SIP username / password settings ? On a LAN it wouls be quite difficult to forge the MAC without it getting detected. But in your case, the MAC is merely an arbitrary ID of the client. It can probably serve as a useful unique ID. See the above question regarding authentication. I also guess you should not use TFTP. Unless you have some spare time at boot. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- 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] Asterisk died without any message, segfault
Hi! We've experienced asterisk has gone without any message, it wasn't any segfault, anything in asterisk messages log that says about shutting down. Asterisk process has just diapered. Has anybody got similar problem? Asterisk is version 1.4.29-1 from digium repository. -- _ -- 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] phoneprov
You can read some here http://www.asterisk.org/astdocs/node272.html or here http://etel.wiki.oreilly.com/wiki/index.php/Dynamic_Phone_Provisioning_with_res_phoneprov_and_TFTP There will be more on this topic in the coming months... ~ Andrew lathama Latham lath...@gmail.com * Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software * Learn more about Linux http://en.wikipedia.org/wiki/Linux * Learn more about Tux http://en.wikipedia.org/wiki/Tux On Wed, Oct 27, 2010 at 7:39 AM, Rizwan Hisham rizwanhas...@gmail.com wrote: Hi List, Can anyone please tell me how to use the phoneprov.conf to provision my client's atas. I read the file but dont know how to actually use it. -- Best Regards Rizwan Qureshi -- _ -- 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] Auto provisioning from public server
Jonas A quick look at the snom wiki will tell you that I am right... On 10/26/2010 06:30 PM, Andrew Latham wrote: snom phones can do http digest authentication... I think this digest authentication is for accessing the phone's web interface, not for contacting a provisioning server Jonas. -- _ -- 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] Asterisk died without any message, segfault
On 27/10/2010 12:59, Krzysztof Urbaniak wrote: Hi! We've experienced asterisk has gone without any message, it wasn't any segfault, anything in asterisk messages log that says about shutting down. How do you launch asterisk ? did you try without 'safe_asterisk' or anything like it, just 'asterisk -cvvv' within a 'screen' for example ? Has anybody got similar problem? Have you searched the bugs repository ? Asterisk is version 1.4.29-1 from digium repository. there is a few new releases for 1.4.x, it is mostly bug fixes. I would suggest you try the latest one and if it still dies build it with debugging options -- _ -- 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] Dial Plan Conf
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nile Kaledon Sent: Wednesday, October 27, 2010 4:15 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Dial Plan Conf Jigar, You should use Read() instead of Background() component. See attached Visual Dialplan file. Nile Finally got VDP to show me this dialplan. A Gotoif will satisfy rest of OP's request. -- _ -- 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] Dial Plan Conf
On Wed, 27 Oct 2010, Nile Kaledon wrote: You should use Read() instead of Background() component. We conf file weenies call them applications. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] Dial Plan Conf
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Wednesday, October 27, 2010 4:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dial Plan Conf On Wed, 27 Oct 2010, Nile Kaledon wrote: You should use Read() instead of Background() component. We conf file weenies call them applications. Is that like a Perl Weenie? -- _ -- 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] No media being sent in SIP call
Hi! I've turned off t.38 and all of the codecs except ulaw; I still have the same problems. SOMETIMES it works. Other times, the sniffer clearly shows that the media simply isn't being sent. NOTHING is being sent. Anything else I should check? Look at the firewalls and possible SIP ALGs that are between the devices. Check for UDP port forwarding settings, and check that the RTP ports that have been negotiated for the call are not conflicting with those of other devices/calls/port forwarding settings. Philipp -- _ -- 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] No media being sent in SIP call
There are NO ACL's in place, either at the network level, or application level. We have a public address, so as far as I know, there are no forwarding rules in place. On Wednesday 27 October 2010 4:04:16 pm Philipp von Klitzing wrote: Hi! I've turned off t.38 and all of the codecs except ulaw; I still have the same problems. SOMETIMES it works. Other times, the sniffer clearly shows that the media simply isn't being sent. NOTHING is being sent. Anything else I should check? Look at the firewalls and possible SIP ALGs that are between the devices. Check for UDP port forwarding settings, and check that the RTP ports that have been negotiated for the call are not conflicting with those of other devices/calls/port forwarding settings. Philipp -- Take care and have fun, Mike Diehl. -- _ -- 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] Astribank Configuration Issues
I have recently updated from Centos/*1.2 to Ubuntu Server and FreePBX 2.8.0.2. We have an Astribank with 4 T1 ports and 16 FXS ports. After updating, we had it working for a while with one NT PRI and one TE PRI and, in the process of trying to configure another PRI, I ran into a couple problems. (1) As my configuration changes didn't seem to affect the Astribank, I power-cycled it. I found that it doesn't reload firmware automatically when it's connected. I can force it to load, but am missing something to reload automatically. (2) I would appreciate a step-by-step suggestion of how I can make configuration changes that propagate properly to the Astribank. (3) I'd like to know if it's possible to determine what configuration has been loaded into the Astribank without visiting the site and looking at the lights. I've spent quite a bit of time Googling, but haven't come up with the right combination of stuff. Thanks for any help you can give, --Don Don Kelly PCF Corp People Come First 651 842-1000 888 Don Kell(y) 651 842-1001 fax -- _ -- 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] No media being sent in SIP call
Do you have canreinvite=yes anywhere? If yes, try setting it to no. Also pasting your sip.conf here would be helpful. Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com (beta) On 2010-10-27 6:16 PM, Mike Diehl mdi...@diehlnet.com wrote: There are NO ACL's in place, either at the network level, or application level. We have a public address, so as far as I know, there are no forwarding rules in place. On Wednesday 27 October 2010 4:04:16 pm Philipp von Klitzing wrote: Hi! I've turned off t Take care and have fun, Mike Diehl. -- ___... -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a... -- _ -- 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] Integrating Asterisk 1.8 with Google Talk and Google Voice
On Tuesday, October 26, 2010 01:16:29 pm Stephen Reese wrote: http://messinet.com/trac/wiki/AsteriskGVGateway (AGI script) Is your .agi and .git the same script? I do not have a git client on this host to see for myself. I keep the AGI in Git as a version control system. But, you can view the AGI source here: http://messinet.com/trac/browser/gv/gv.agi And at the very bottom of that page is a link to download it as an individual file here: http://messinet.com/trac/export/b3229dbba3e01c887b3bdf6b0e0d93e897bd8a59/gv/gv.agi This is not the same thing as what is in the Changelog. I am using Asterisk 1.6 with this AGI. -A -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. -- _ -- 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] Mobile Phones and Asterisk
Hi,
Thanks for your reply. I'm calling a normal phone using the DIAL cmd. Here is
my sample dial command:
exten =s,4,Dial(SIP/xxx${extensi...@media_gateway,10,t)
but when I use:
exten =s,5,NoOp(SIP/xxx${extensi...@media_gateway has state ${DIALSTATUS})
I hear a busy tone, after the 10 sec. timeout it returns NOANSWER, as defined
in my DIAL func.
I also tried getting the DEVICE_STATE
exten =s,3,NoOp(SIP/xxx${extensi...@media_gateway has state
${DEVICE_STATE(SIP/xxx${extensi...@media_gateway)})
and same thing happens as stated on the scenario below.
Thanks again!
regards,
RYAN ICASIANO
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Sebastian
[s...@open-t.co.uk]
Sent: Wednesday, October 27, 2010 5:00 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Mobile Phones and Asterisk
Hi,
On 10/27/2010 05:55 AM, GBR Icasiano, Ryan A. wrote:
anyone???
regards,
RYAN ICASIANO
Hi,
I changed my sip.conf and added call-limit. At first I thought it works ok,
since i tried calling a cellphone that is currently busy(phone answers 1st
softphone, then another softphone calls the same number, it now returns
INUSE). But then, i tried calling a different number while the first phone is
busy, but it returns INUSE. It seems that the status being returned was from
the peer itself(both phones uses the same peer) and not from the
device(mobile phone) which i believe is more logical.
I also tried using DIALSTATUS(which of course you need to DIAL first), but
then I only hear a busy tone and the dialstatus will return a noanswer. Do I
have to configure it first in order to capture the busy status of a device?
Have you done something similar to this?
I'm using ver. 1.6. Thanks in advance.
I'm not sure I understand your setup. Are you using SIP for trunking, or
for extensions? Are you calling a normal mobile phone, or a SIP client
on a mobile phone?
Sebastian
regards,
RYAN ICASIANO
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of GBR Icasiano, Ryan A.
[raicasi...@globalbridgeresources.com]
Sent: Tuesday, October 26, 2010 10:41 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Mobile Phones and Asterisk
Hi,
Is the dev_state can also be used to track a mobile phone's status via SIP?
I tried it on several phones(nokia, samsung) but it returns NOANSWER but i
can hear a beep beep beep sound indicating that it is currently busy.
regards,
RYAN ICASIANO
__
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Sebastian
[s...@open-t.co.uk]
Sent: Tuesday, October 26, 2010 7:50 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Mobile Phones and Asterisk
On 10/26/2010 12:30 PM, ayodele abejide wrote:
Hello Jonathan,
The solution would work only if the ISP has one public address, but in
my solution they have a pool of public address, any other possible solution?
With dynamic dns, you either install a piece of software on your server
(dynamic dns client) or you use the facility provided by your router
(some firewall/router/access point combo's have them). This software
updates automatically the record with dyndns every time your IP address
changes.
Sebastian
ABEJIDE, Ayodele A. (CCNA)
+2348039269311
From: ayodeleabej...@hotmail.com
To: asterisk-users@lists.digium.com
Date: Tue, 26 Oct 2010 11:01:09 +
Subject: Re: [asterisk-users] Mobile Phones and Asterisk
thanks i would check it up
ABEJIDE, Ayodele A. (CCNA)
+2348039269311
Date: Tue, 26 Oct 2010 12:52:30 +0200
From: jonathan@gmail.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Mobile Phones and Asterisk
Try http://www.dyndns.com/ that should solve your problem with dynamic IPs.
Regards,
Jonathan
On Tue, Oct 26, 2010 at 12:40 PM, ayodele abejide
ayodeleabej...@hotmail.commailto:ayodeleabej...@hotmail.com wrote:
Dear Asterisk-Users,
I have this Asterisk Box I run in my house, I need to terminate and
originate remote calls through the box via internet (SIP), the
problem is in Nigeria most ISPs would not provide you with Public
Addresses, all they provide is dynamic Natted addresses which change
each time one connects, I have thought of all possible solutions and
cannot come up with one, can anyone please help.
Thanks in anticipation
ABEJIDE, Ayodele A. (CCNA)
+2348039269311
--
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-- Bandwidth and Colocation Provided
Re: [asterisk-users] Dial Plan Conf
On Wed, 27 Oct 2010, Nile Kaledon wrote: You should use Read() instead of Background() component. On Wed, 27 Oct 2010, Steve Edwards wrote: We conf file weenies call them applications. On Wed, 27 Oct 2010, Danny Nicholas wrote: Is that like a Perl Weenie? Yes, and you can proudly wear as many self-congratulatory labels as you wish simultaneously. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] Intermittent failure when placing calls - unable to create channel of type SIP
Hello community, I've been running Asterisk on an embedded device for about six months, and my operation has been largely trouble-free. I'm hoping I could get some help with a minor problem: Every week or three, my PBX gets stuck in a state where it can receive calls, but it becomes completely unable to originate outgoing calls until I do a sip reload. After doing the SIP reload, everything immediately begins works perfectly again and I can make outgoing calls until it gets stuck again several weeks later. I recently upgraded to Asterisk 1.6.2.13, although I was also running 1.6.2.1 for a long time with identical symptoms. My system is an embedded ar71xx running the OpenWRT distribution. When I attempt to place a call, after Asterisk has transmitted the 100 Trying message to the calling extension (an ATA), I see the following Unable to create channel of type SIP message in the log: [Oct 27 18:46:48] DEBUG[25028]: pbx.c:3696 pbx_extension_helper: Launching 'Set' [Oct 27 18:46:48] DEBUG[25028]: pbx.c:3696 pbx_extension_helper: Launching 'Dial' [Oct 27 18:46:48] DEBUG[25028]: chan_sip.c:23241 sip_request_call: Asked to create a SIP channel with formats: 0x4 (ulaw) [Oct 27 18:46:48] DEBUG[25028]: chan_sip.c:7381 sip_alloc: Allocating new SIP dialog for 2ccf324d10670f2c73f478b523f92...@10.15.1.1 - INVITE (With RTP) Really destroying SIP dialog '2ccf324d10670f2c73f478b523f92...@10.15.1.1' Method: INVITE [Oct 27 18:46:48] WARNING[25028]: app_dial.c:1750 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) [Oct 27 18:46:48] DEBUG[25028]: rtp.c:2148 ast_rtp_early_bridge: Channel 'unspecified' has no RTP, not doing anything [Oct 27 18:46:48] DEBUG[25028]: app_dial.c:2326 dial_exec_full: Exiting with DIALSTATUS=CHANUNAVAIL. [Oct 27 18:46:48] DEBUG[25028]: pbx.c:3696 pbx_extension_helper: Launching 'Hangup' [Oct 27 18:46:48] DEBUG[25028]: pbx.c:4322 __ast_pbx_run: Spawn extension (phones,15102857673,3) exited non-zero on 'SIP/101-000a' [Oct 27 18:46:48] DEBUG[25028]: channel.c:1715 ast_softhangup_nolock: Soft-Hanging up channel 'SIP/101-000a' The calling extension then receives a 480 Temporarily Unavailable and a fast busy. Doing a sip show peers appears normal. When I do a detailed sip show mypeername, the one anomalous thing I see is that that the Addr-IP setting is listed as (Unspecified). * Name : voipms Secret : Set [...] ToHost : dallas.voip.ms Addr-IP : (Unspecified) Port 5060 Defaddr-IP : 0.0.0.0 Port 5060 Prim.Transp. : UDP Allowed.Trsp : UDP [...] In contrast, after I do a sip reload, outbound calls start working again and the sip show output looks identical except for showing the correct address under Addr-IP: * Name : voipms Secret : Set [...] ToHost : dallas.voip.ms Addr-IP : 74.54.54.178 Port 5060 Defaddr-IP : 0.0.0.0 Port 5060 Prim.Transp. : UDP Allowed.Trsp : UDP [...] Does anyone know how/where/why Asterisk could lose the IP address of the peer? One thing potentially related is that, in the previous registration to the peer (two minutes prior to my failed call), we do the usual REGISTER/Unauthorized+Nonce/REGISTER+Response/OK business. Immediately after that, we get a NOTIFY from the remote, which Asterisk responds to with a 489 Bad Event: NOTIFY sip:s...@my.ip.add.ress:6010 SIP/2.0 Via: SIP/2.0/UDP 74.54.54.178:5060;branch=z9hG4bK008e70db;rport From: Unknown sip:unkn...@74.54.54.178 sip%3aunkn...@74.54.54.178 ;tag=as5c60da37 To: sip:s...@my.ip.add.ress:6010 Contact: sip:unkn...@74.54.54.178 sip%3aunkn...@74.54.54.178 Call-ID: 266322e108872eab12fb307772a4a...@74.54.54.178 CSeq: 102 NOTIFY User-Agent: VoIPMS SERAST Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 92 Messages-Waiting: no Message-Account: sip:aster...@74.54.54.178 sip%3aaster...@74.54.54.178 Voice-Message: 0/0 (0/0) SIP/2.0 489 Bad event Via: SIP/2.0/UDP 74.54.54.178:5060 ;branch=z9hG4bK008e70db;received=74.54.54.178;rport=5060 From: Unknown sip:unkn...@74.54.54.178 sip%3aunkn...@74.54.54.178 ;tag=as5c60da37 To: sip:s...@my.ip.add.ress:6010;tag=as4b162a1c Call-ID: 266322e108872eab12fb307772a4a...@74.54.54.178 CSeq: 102 NOTIFY Server: Asterisk PBX 1.6.2.13 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Immediately after that exchange, I see the following curious set of messages: [Oct 27 18:44:58] DEBUG[1676]: chan_sip.c:3608 __sip_xmit: Trying to put 'SIP/2.0 489' onto UDP socket destined for 74.54.54.178:5060 [Oct 27 18:44:58] DEBUG[1676]: chan_sip.c:22105 handle_request_do: Invalid SIP message - rejected , no callid, len 541 Could those messages be related to my problem? I see this 489 Bad Event issue may be related to https://issues.asterisk.org/view.php?id=17379, but it's unclear if this can somehow cause the SIP remote peer address to get lost, as opposed to just being potentially bad
Re: [asterisk-users] [asterisk-biz] D-Link Wifi Phones
Can they be used from any unsecured access point (eg they have a browser to enter in a password etc) or can you only use them from home AP's etc. Cheers, Dean -Original Message- From: asterisk-biz-boun...@lists.digium.com [mailto:asterisk-biz- boun...@lists.digium.com] On Behalf Of Mike White Sent: Wednesday, 27 October 2010 8:37 PM To: Commercial and Business-Oriented Asterisk Discussion; Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-biz] D-Link Wifi Phones Hello, I have about 100 unopened D-Link DPH-540 Wifi Phones that are new in the box. I am unloading these for $32 Each - Buy one or 100 :) I'll also entertain offers for bulk orders. http://short.e4strategies.com/dph540 Kind regards, Mike White .e4 http://8774e4voip.com PS - I also have many new Aastra 6739i and Polycom IP335 phones that are available at substantial discounts. Reseller? Contact me... -- __ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz -- _ -- 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] what interface for ISDN-10/20/30?
Hello all,I'm working with one of our offices (that is moving soon) and they're being offered ISDN-10/20/30 services from their TELCO. I'm wondering what kind of interface card I will need (I prefer using Digium's cards). Are the TE121/122/ or TE212/220 series cards compatible with this kind of service? Seems like the service would look like a PRI interface, but I'm not sure. The office is in Singapore.ThanksCassius Smith -- _ -- 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] what interface for ISDN-10/20/30?
On 10/27/2010 09:21 PM, Cassius Smith wrote: I'm working with one of our offices (that is moving soon) and they're being offered ISDN-10/20/30 services from their TELCO. I'm wondering what kind of interface card I will need (I prefer using Digium's cards). Are the TE121/122/ or TE212/220 series cards compatible with this kind of service? Seems like the service would look like a PRI interface, but I'm not sure. The office is in Singapore. Yes, you are right. That's an E1 circuit, configured with 10, 20 or 30 active B-channels. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.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] Asterisk 1.8 IAX Registration
On Tue, Oct 26, 2010 at 8:26 PM, Paul Belanger paul.belan...@polybeacon.com wrote: I'm going to try and look at this during Astricon :) Ok, just uploaded a new patch on https://issues.asterisk.org/view.php?id=18202 Let me know if it worked. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) | Blog: http://blog.polybeacon.com | Twitter: http://twitter.com/pabelanger -- _ -- 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] ss7_channel or ss7lib
Hi all, Are there anyone use ss7_lib or ss7_channel in production ?. What about its quality and reliablity ?. Can an Asterisk servce with ss7_lib or ss7_channel can processs 480 conccurent call (8 E1 line) ? Many thanks, Giang -- _ -- 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
