Re: [asterisk-users] Auto ban IP addresses
On Wednesday 02 January 2013, Frank wrote: Greetings all, I have been seeing a lot of [Jan 2 16:36:31] NOTICE[7519]: chan_sip.c:23149 handle_request_invite: Sending fake auth rejection for device 100sip:100@108.161.145.18;tag=2e921697 in my logs lately. Is there a way to automatically ban IP address from attackers within asterisk ? There is a more general-purpose way to block IP addresses from which unwanted traffic is coming: fail2ban. This scans various logfiles for failed login attempts, and can insert iptables rules to block the addresses whence they originate. On Ubuntu and Debian, just run $ sudo apt-get install fail2ban -- AJS Answers come *after* questions. -- _ -- 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] RES: Auto ban IP addresses
I am using fail2ban on all my asterisk server, but beware, fail2ban can be a dangerous software. The problem rely on the fact that SIP uses UDP, so it is possible to send messages with a forged source IP address. This way the bad guy out there can ban all your IP addresses. I say it is possible without having investigated in deep details what is really needed to do. Leandro 2013/1/3 Éder e...@openminds.com.br Howto fail2ban in asterisk http://www.voip-info.org/wiki/view/Fail2Ban+%28with+iptables%29+And+Asterisk -Mensagem original- De: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Em nome de Frank Enviada em: quarta-feira, 2 de janeiro de 2013 20:50 Para: Asterisk Users Mailing List - Non-Commercial Discussion Assunto: [asterisk-users] Auto ban IP addresses Greetings all, I have been seeing a lot of [Jan 2 16:36:31] NOTICE[7519]: chan_sip.c:23149 handle_request_invite: Sending fake auth rejection for device 100sip:100@108.161.145.18;tag=2e921697 in my logs lately. Is there a way to automatically ban IP address from attackers within asterisk ? Thank you -- _ -- 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] DAHDI: How to know since when it is used? How to shutdown after max time?
2013/1/3 bilal ghayyad bilmar...@yahoo.com Hi; How can I know the duration that the DAHDI channel is still used? I need to know its status and since when it is in this status, how? Also, is it possible to hangup the channel if it has been openned more than 90 minute? Other than using the timeout in the Dial command (because this I know it). What is happening with me that from time to time, I find some DAHDI channels are stayed connected (stuck) for long time. I know how to write the extensions.conf in a way to handle the hangup properly, also I send the incoming calls to the voicemail to be sure it is hanged up properly. One more thing, I set the rtptimeout in case there is any problem in the sip phone and the network .. But, still after sometime, I am surprised that some channels are stuck and stayed connected and then I have to reset it manually !! This is happening only in the analoge channels. What other than the rtptimeout, the hangup in the extensions.conf, the voicemail? Is there anything I have to take care for it that might cause this stuck and keeping the channel openned? By the way, for such cases, what should I place the value of the rtpkeepalive as currently it is 0? What other things I have to take care for it? Regards Bilal I checked on my PBX and I find no way to identify the duration of a call involving a DAHDI channel like it happens on SIP channels. I think the only way will be to assign a not so huge AbsoluteTimeout to each call. Leandro -- _ -- 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] RES: Auto ban IP addresses
On Thu, 2013-01-03 at 09:42 +0100, Leandro Dardini wrote: I am using fail2ban on all my asterisk server, but beware, fail2ban can be a dangerous software. The problem rely on the fact that SIP uses UDP, so it is possible to send messages with a forged source IP address. This way the bad guy out there can ban all your IP addresses. I say it is possible without having investigated in deep details what is really needed to do. The jail.conf in fail2ban allows for a whitelist of IPs that will never be banned -- Ishfaq Malik i...@pack-net.co.uk Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET NORTH, MANCHESTER SCIENCE PARK, MANCHESTER, M156SE COMPANY REG NO. 04920552 -- _ -- 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] User busy issue in A400P 4 FXO card
On Thursday 03 January 2013, Selva M wrote:
Hi,
I setup PBX with A400P 4 x FXo board. There are one analog line plugged
into port 1.
Internal extension cane make calls to PSTN without any issue.
When I make inbound call, caller get busy tone user busy' message right
away.
Asterisk log shows following log and internal extension (200) rings for
that call and hangup (log below).
I tested the system with some other service provider and it worked fine
for IB and OB calls.
i would like to get your feedback to resolve the issue and will
appreciate your feedback.
Thanks
Selva
Don't try to run before you can walk. First of all, simplify your dialplan
right down to the minimum. Have just this context for calls coming in from
the card:
[from-pstn]
s,1,NoOp(Incoming call from ${CALLERID(num)})
s,2,Dial(200)
s,3,Hangup()
; end of from-pstn context
This will write the caller's number to the console (step 1) and ring
extension 200 (step 2). Alter it if necessary. Do a reload, and call
yourself from your mobile (or get an assistant to call you) while watching
the console.
If it doesn't work, you have a DAHDI problem; most probably your country
settings are wrong (NB: in some countries, different telephone companies can
actually require different country settings!) Power your machine off and on
again after changing DAHDI settings, just to make absolutely sure that it
really is using the new ones.
Once you have that very simple dialplan working, then you can start
introducing extra bells and whistles; one at a time, and testing in between
each one and the next.
--
AJS
Answers come *after* questions.
--
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Re: [asterisk-users] Asterisk for Razberry Pi
Wow! Thanks so much for all the information. I now have a lot to look over. Bob R On 01/02/2013 10:03 AM, Tzafrir Cohen wrote: On Wed, Jan 02, 2013 at 09:55:44AM -0500, Robert Rawlinson wrote: Has anyone ported Asterisk to the Razzberry Pi? If so could you point me to info on doing so? apt-get install asterisk -- _ -- 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] new user help required to build voice recorder with asterisk
I don't think this should be an issue, but we have seen a lot of sites going live and discovering too late that they had recording problems. Maybe you won't need to implement an external recorder, but it's better to plan in advance, not when you are in production! :) l. 2013/1/2 Leandro Dardini ldard...@gmail.com I don't know how many I/O can be achieved on a modern hardware, but I don't think 60 concurrent calls will be a problem. 60 calls are just 4 Mbit/s of data. However can be a good idea to start loading a server and be prepared to share the load on another server. Leandro -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.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] Auto ban IP addresses
On Wednesday, January 2, 2013, Frank wrote: Is there a way to automatically ban IP address from attackers within asterisk ? As others have mentioned, fail2ban does a good job. However, it may not be enough as these attacks sometimes come from older versions of the SipVicious hacking tool that keep trying even after they cease getting a response -- i.e. the attack continues even after fail2ban has jailed the host, which eats into your bandwidth and can cause denial of service in extreme cases. FWIW, I suffered one such attack last year after my router died and the temporary replacement couldn't selectively block or forward UDP 5060 based on WAN IP address. The attack continued for over eight days and consumed over a gigabyte a day of my bandwidth for the first three of those days -- until I'd replaced the temporary router and taken proactive measures. An initial LART to the attacking host's owner and their provider achieved little. I ended up installing SipVicious to a virtual machine to which I router all SIP requests from the attacker. On the VM I set up svcrash to automatically crash the attacking script each time it received a SIP request. This cut the attack down to one request every couple of seconds. In the end, I suggested to the owner of the attacking host that it might be a good idea for them to remove Python unless it was actually needed and in any case to remove from that machine all instances of svwar.py and svcrack.py together with the remainder of the SipVicious suite. The attack stopped shortly after. I suspect that any system that responds to all SIP requests is likely to attract such attacks. My solution is to silently drop SIP traffic from all but my SIP providers, which means that attackers perceive that my Asterisk box doesn't exist. This is not ideal as it also prevents legitimate direct SIP calls and reinvites, but IMO better that than having bandwidth I pay for by the gigabyte consumed by brute force attacks. -- Geoff -- _ -- 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] RES: Auto ban IP addresses
Interesting... -Mensagem original- De: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Em nome de Geoff Lane Enviada em: quinta-feira, 3 de janeiro de 2013 10:06 Para: Asterisk Users Mailing List - Non-Commercial Discussion Assunto: Re: [asterisk-users] Auto ban IP addresses On Wednesday, January 2, 2013, Frank wrote: Is there a way to automatically ban IP address from attackers within asterisk ? As others have mentioned, fail2ban does a good job. However, it may not be enough as these attacks sometimes come from older versions of the SipVicious hacking tool that keep trying even after they cease getting a response -- i.e. the attack continues even after fail2ban has jailed the host, which eats into your bandwidth and can cause denial of service in extreme cases. FWIW, I suffered one such attack last year after my router died and the temporary replacement couldn't selectively block or forward UDP 5060 based on WAN IP address. The attack continued for over eight days and consumed over a gigabyte a day of my bandwidth for the first three of those days -- until I'd replaced the temporary router and taken proactive measures. An initial LART to the attacking host's owner and their provider achieved little. I ended up installing SipVicious to a virtual machine to which I router all SIP requests from the attacker. On the VM I set up svcrash to automatically crash the attacking script each time it received a SIP request. This cut the attack down to one request every couple of seconds. In the end, I suggested to the owner of the attacking host that it might be a good idea for them to remove Python unless it was actually needed and in any case to remove from that machine all instances of svwar.py and svcrack.py together with the remainder of the SipVicious suite. The attack stopped shortly after. I suspect that any system that responds to all SIP requests is likely to attract such attacks. My solution is to silently drop SIP traffic from all but my SIP providers, which means that attackers perceive that my Asterisk box doesn't exist. This is not ideal as it also prevents legitimate direct SIP calls and reinvites, but IMO better that than having bandwidth I pay for by the gigabyte consumed by brute force attacks. -- Geoff -- _ -- 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
[asterisk-users] Verizon SIP trunking Field Trial
All, We are in the process of trying to setup our network to use Verizon's SIP trunking product. They say that since Asterisk is not on their certified list of approved devices, we need to go through a field trial to get it approved before allowing us to use their service. Where we are at is getting the design approved. We are trying to watch our budget at the same time. We have used other providers without any issues with our current setup but it seems that Verizon has their own standards when it comes to this and they don't seem very keen on linux and open source. Yet, they are willing to work with us and want to see the field trial succeed instead of being rejected from another group within Verizon who will have to approve the final design. Has anyone in the community had experience with Verizon and their SIP product? Were you able to get through the field trial successfully? What was the design that you used to get Asterisk certified with Verizon's network? Where I am at is that they want us to use an SBC. One engineer asked about Cisco Call Manager. I told them that basically if I can accomplish the same thing with a Linux box (routing box and sip proxy box) without having to spend money on SBCs or expensive Cisco gear, that is the route we would like to go. We are looking at the possibility of handling 140 concurrent calls... that is what they are designing on their end as well. So, I am asking the community for any input. I have read on here and seen on IRC that some in the community are successfully using Asterisk with Verizon SIP. Verizon was going to check and see if they have any notes about that and those particular setups. Can anyone help share any information or tidbits on how they were able to sucessfully work with Verizon? Thanks, -- Michael L. Young -- _ -- 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] Verizon SIP trunking Field Trial
On 3 Jan 2013, at 15:13, Michael L. Young wrote: So, I am asking the community for any input. I have read on here and seen on IRC that some in the community are successfully using Asterisk with Verizon SIP. Verizon was going to check and see if they have any notes about that and those particular setups. Can anyone help share any information or tidbits on how they were able to sucessfully work with Verizon? I *think* Verizon require IPSEC for the signalling, so it may be worth reading up on configuring IPSEC in Linux (or acquiring additional hardware) whilst you're looking at the Asterisk part. This could have just been for a specific product / contract or something, I don't recall the details exactly. S -- _ -- 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] Verizon SIP trunking Field Trial
- Original Message - From: Steven Howes steve-li...@geekinter.net I *think* Verizon require IPSEC for the signalling, so it may be worth reading up on configuring IPSEC in Linux (or acquiring additional hardware) whilst you're looking at the Asterisk part. This could have just been for a specific product / contract or something, I don't recall the details exactly. I should have probably stated that this is going to be going through an MPLS network being setup with Verizon. They may not be requiring that since it is within their network, not going over the internet. They have not said anything about the the need to secure the traffic coming from them or to them since the VoIP traffic will be on Verizon's network. Thanks for the heads up, though. I will keep that in mind. Michael -- _ -- 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] Verizon SIP trunking Field Trial
2013/1/3 Steven Howes steve-li...@geekinter.net On 3 Jan 2013, at 15:13, Michael L. Young wrote: So, I am asking the community for any input. I have read on here and seen on IRC that some in the community are successfully using Asterisk with Verizon SIP. Verizon was going to check and see if they have any notes about that and those particular setups. Can anyone help share any information or tidbits on how they were able to sucessfully work with Verizon? I *think* Verizon require IPSEC for the signalling, so it may be worth reading up on configuring IPSEC in Linux (or acquiring additional hardware) whilst you're looking at the Asterisk part. This could have just been for a specific product / contract or something, I don't recall the details exactly. S -- I have no direct experience with Verizon, but another big player asks for a long series of tests, like call and answer, call and don't answer, call and cancel. It took me two full days of work to accomplish all the tasks. For every call I have to dump the Call-ID, the date and the hours... So, don't be scared by the field test, it will be probably long and tedious, but doable. Leandro -- _ -- 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] Verizon SIP trunking Field Trial
It doesn't matter. They still require IPSEC VPN. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michael L. Young Sent: Thursday, January 03, 2013 10:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Verizon SIP trunking Field Trial - Original Message - From: Steven Howes steve-li...@geekinter.net I *think* Verizon require IPSEC for the signalling, so it may be worth reading up on configuring IPSEC in Linux (or acquiring additional hardware) whilst you're looking at the Asterisk part. This could have just been for a specific product / contract or something, I don't recall the details exactly. I should have probably stated that this is going to be going through an MPLS network being setup with Verizon. They may not be requiring that since it is within their network, not going over the internet. They have not said anything about the the need to secure the traffic coming from them or to them since the VoIP traffic will be on Verizon's network. Thanks for the heads up, though. I will keep that in mind. Michael -- _ -- 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
[asterisk-users] Build asterisk for VIA C3
Is it difficult to publish a build asterisk.deb compiled for VIA C3 architecture ? Instead of using the binary just for me. So any one trying to install it on C3 CPU will need just to do: aptitude install asterisk The one that is installed by default doesn't work for such a CPU Should I contact debian dev team for that? Thanks OLD messages --- Message: 6 Date: Mon, 31 Dec 2012 12:08:40 -0500 From: neo haux neo.h...@gmx.com Subject: Re: [asterisk-users] Compile asterisk11.1 for i586 VIA C3 CPU To: asterisk-users@lists.digium.com Message-ID: CAHtT-j+Z2aD2qSN4ir17ZD5=gf4ljta9ggikstcojiu5+x9...@mail.gmail.com Content-Type: text/plain; charset=ISO-8859-1 Thanks George it works now ! Message: 6 Date: Sun, 30 Dec 2012 17:18:45 -0700 From: George Joseph george.jos...@fairview5.com Subject: Re: [asterisk-users] Compile asterisk11.1 for i586 VIA C3 CPU To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: cahkv19d6j0jfs3lgyyxsarqckhk32mgn6ciih9v8uwz4fqt...@mail.gmail.com Content-Type: text/plain; charset=utf-8 Try this... In menuselect, uncheck BUILD_NATIVE under Compiler Flags and recompile. On Sun, Dec 30, 2012 at 4:44 PM, neo haux neo.h...@gmx.com wrote: Hi, I've compiled asterisk 11.1 for my MiniITX card with VIA C3 Samuel2 800MHz CPU. A small box to play with PBX at home. I get this error when I start asterisk: root@pbx01:/usr/src/asterisk-11.1.0# /etc/init.d/asterisk start Illegal instruction Starting Asterisk PBX: asteriskIllegal instruction I compiled it on debian 6.0.6 with this options: ./configure --build=i386-pc-linux-gnu --host=i386-pc-linux-gnu \ --with-iksemel=/usr/src/iksemel-1.4 --with-pri=/usr/src/libpri-1.4.14 \ --with-dahdi='/usr/src/dahdi-linux-complete-2.6.1+2.6.1' --libdir=/usr/lib I was able last days to compile asterisk 1.8 and it did work perfectly except with Gtalk, and this is why I have to compile v11.1 -- _ -- 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] Verizon SIP trunking Field Trial
On Thu, Jan 3, 2013 at 8:13 AM, Michael L. Young myo...@acsacc.com wrote: Where I am at is that they want us to use an SBC. One engineer asked about Cisco Call Manager. I told them that basically if I can accomplish the same thing with a Linux box (routing box and sip proxy box) without having to spend money on SBCs or expensive Cisco gear, that is the route we would like to go. We are looking at the possibility of handling 140 concurrent calls... that is what they are designing on their end as well. So, I am asking the community for any input. I have read on here and seen on IRC that some in the community are successfully using Asterisk with Verizon SIP. Verizon was going to check and see if they have any notes about that and those particular setups. Can anyone help share any information or tidbits on how they were able to sucessfully work with Verizon? It may be too late for this, but in working with another RBOC who didn't want to deal with Asterisk, I just asked what they do support, and modified the headers sent by Asterisk to claim that it was one of the devices on that list. Done. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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] DAHDI: How to know since when it is used? How to shutdown after max time?
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leandro Dardini Sent: Thursday, January 03, 2013 2:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DAHDI: How to know since when it is used? How to shutdown after max time? 2013/1/3 bilal ghayyad bilmar...@yahoo.com Hi; How can I know the duration that the DAHDI channel is still used? I need to know its status and since when it is in this status, how? Also, is it possible to hangup the channel if it has been openned more than 90 minute? Other than using the timeout in the Dial command (because this I know it). What is happening with me that from time to time, I find some DAHDI channels are stayed connected (stuck) for long time. I know how to write the extensions.conf in a way to handle the hangup properly, also I send the incoming calls to the voicemail to be sure it is hanged up properly. One more thing, I set the rtptimeout in case there is any problem in the sip phone and the network .. But, still after sometime, I am surprised that some channels are stuck and stayed connected and then I have to reset it manually !! This is happening only in the analoge channels. What other than the rtptimeout, the hangup in the extensions.conf, the voicemail? Is there anything I have to take care for it that might cause this stuck and keeping the channel openned? By the way, for such cases, what should I place the value of the rtpkeepalive as currently it is 0? What other things I have to take care for it? Regards Bilal I checked on my PBX and I find no way to identify the duration of a call involving a DAHDI channel like it happens on SIP channels. I think the only way will be to assign a not so huge AbsoluteTimeout to each call. My suggestion would be to either do a cron job that executes asterisk -rx core show channels verbose and kill anything with a duration over 90 minutes or do the same thing with an AMI task (cron optional here). -- _ -- 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] Verizon SIP trunking Field Trial
On 01/03/2013 09:56 AM, Carlos Alvarez wrote: On Thu, Jan 3, 2013 at 8:13 AM, Michael L. Young myo...@acsacc.com mailto:myo...@acsacc.com wrote: Where I am at is that they want us to use an SBC. One engineer asked about Cisco Call Manager. I told them that basically if I can accomplish the same thing with a Linux box (routing box and sip proxy box) without having to spend money on SBCs or expensive Cisco gear, that is the route we would like to go. We are looking at the possibility of handling 140 concurrent calls... that is what they are designing on their end as well. So, I am asking the community for any input. I have read on here and seen on IRC that some in the community are successfully using Asterisk with Verizon SIP. Verizon was going to check and see if they have any notes about that and those particular setups. Can anyone help share any information or tidbits on how they were able to sucessfully work with Verizon? It may be too late for this, but in working with another RBOC who didn't want to deal with Asterisk, I just asked what they do support, and modified the headers sent by Asterisk to claim that it was one of the devices on that list. Done. ROFL!! Well done. j -- _ -- 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 11.1.2 Now Available (Security Release)
The Asterisk Development Team has announced a security release for Asterisk 11, Asterisk 11.1.2. This release addresses the security vulnerabilities reported in AST-2012-014 and AST-2012-015, and replaces the previous version of Asterisk 11 released for these security vulnerabilities. The prior release left open a vulnerability in res_xmpp that exists only in Asterisk 11; as such, other versions of Asterisk were resolved correctly by the previous releases. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases The release of these versions resolve the following two issues: * Stack overflows that occur in some portions of Asterisk that manage a TCP connection. In SIP, this is exploitable via a remote unauthenticated session; in XMPP and HTTP connections, this is exploitable via remote authenticated sessions. The vulnerabilities in SIP and HTTP were corrected in a prior release of Asterisk; the vulnerability in XMPP is resolved in this release. * A denial of service vulnerability through exploitation of the device state cache. Anonymous calls had the capability to create devices in Asterisk that would never be disposed of. Handling the cachability of device states aggregated via XMPP is handled in this release. These issues and their resolutions are described in the security advisories. For more information about the details of these vulnerabilities, please read security advisories AST-2012-014 and AST-2012-015. For a full list of changes in the current release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.1.2 The security advisories are available at: * http://downloads.asterisk.org/pub/security/AST-2012-014.pdf * http://downloads.asterisk.org/pub/security/AST-2012-015.pdf Thank you for your continued support of Asterisk - and we apologize for having to do this twice! -- _ -- 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] Moving User Agent To Remote Location
Hello Everyone, Before getting into SIP and RTP traces, I wanted to clarify some of the sip.conf settings that may to some seem redundant or have a misconception with. I do apologize if this has been discussed time and time again as I would imagine. If anything, this email would make google search results that much stronger :). With the UA local to my network I had tested two way audio, and now with the phone outside of network, we have no way audio. Before discussing NAT (which is enabled on the peer), and port forwarding (which is setup on the remote location), I would like to make sure I fully understand all the sip.conf settings. We are using Asterisk realtime via sip_buddies, and the fields in question are: (Enclosed in brackets are an example value for the setting) * host (dynamic): No problem here. Just wanted to mention that it's set as such * nat (yes): No problem here either * defaultuser (1...@example.com): Does the @example.com have to point to the UA (i.e., (1003@ua-public-ip), or is it just a name type field? * fullcontact: What to put here for a UA that is running at a remote location with dynamic external IP? * ipaddr (ua-public-ip): I did try setting it to the public ip of the UA, but is that really practical? What if I don't know where the initial registration request is coming from? I am guessing host=dynamic takes care of that. * defaultip?? * dynamic: Should this be set to yes, or is host=dynamic sufficient? The phone registers fine, and terminates a call through our providers. Just no audio both ways, which would suggest something more that an RTP issue which should at least have one way outgoing audio. Things that have been attempted: * Port forwarding to the phone * Changing defaultuser to 1003@ua-public-ip. This made our OpenSIPS sip proxy through a fit. Things I will attempt today: Calling the UA extension from an extension here SIP trace Your help is greatly appreciated!!! Nick. -- _ -- 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] Moving User Agent To Remote Location
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis Sent: Thursday, January 03, 2013 11:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Moving User Agent To Remote Location Hello Everyone, Before getting into SIP and RTP traces, I wanted to clarify some of the sip.conf settings that may to some seem redundant or have a misconception with. I do apologize if this has been discussed time and time again as I would imagine. If anything, this email would make google search results that much stronger :). With the UA local to my network I had tested two way audio, and now with the phone outside of network, we have no way audio. Before discussing NAT (which is enabled on the peer), and port forwarding (which is setup on the remote location), I would like to make sure I fully understand all the sip.conf settings. We are using Asterisk realtime via sip_buddies, and the fields in question are: (Enclosed in brackets are an example value for the setting) * host (dynamic): No problem here. Just wanted to mention that it's set as such * nat (yes): No problem here either * defaultuser (1...@example.com): Does the @example.com have to point to the UA (i.e., (1003@ua-public-ip), or is it just a name type field? * fullcontact: What to put here for a UA that is running at a remote location with dynamic external IP? * ipaddr (ua-public-ip): I did try setting it to the public ip of the UA, but is that really practical? What if I don't know where the initial registration request is coming from? I am guessing host=dynamic takes care of that. * defaultip?? * dynamic: Should this be set to yes, or is host=dynamic sufficient? The phone registers fine, and terminates a call through our providers. Just no audio both ways, which would suggest something more that an RTP issue which should at least have one way outgoing audio. Things that have been attempted: * Port forwarding to the phone * Changing defaultuser to 1003@ua-public-ip. This made our OpenSIPS sip proxy through a fit. Things I will attempt today: Calling the UA extension from an extension here SIP trace Your help is greatly appreciated!!! Nick. I'm going to vote for the RTP issue. If you are establishing a call but have no audio, you are getting the 5060 port, but not the 1-2 range that RTP normally expects. A better practice is to allocate 4 ports per line you expect to use in rtp.conf (1-2 would allow 2500 lines; much more that most folks need and more holes to monitor). -- _ -- 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] DAHDI: How to know since when it is used? How to shutdown after max time?
On Wed, Jan 2, 2013 at 5:39 PM, bilal ghayyad bilmar...@yahoo.com wrote: Hi; How can I know the duration that the DAHDI channel is still used? I need to know its status and since when it is in this status, how? Also, is it possible to hangup the channel if it has been openned more than 90 minute? Other than using the timeout in the Dial command (because this I know it). What is happening with me that from time to time, I find some DAHDI channels are stayed connected (stuck) for long time. I know how to write the extensions.conf in a way to handle the hangup properly, also I send the incoming calls to the voicemail to be sure it is hanged up properly. One more thing, I set the rtptimeout in case there is any problem in the sip phone and the network .. But, still after sometime, I am surprised that some channels are stuck and stayed connected and then I have to reset it manually !! This is happening only in the analoge channels. What other than the rtptimeout, the hangup in the extensions.conf, the voicemail? Is there anything I have to take care for it that might cause this stuck and keeping the channel openned? What kind of trunks you are using? Analog? Digital (E1, T1)? Saludos/Regards -- Ing. Gerardo Barajas Puente Proyectos Especiales/Preventa | www.neocenter.com T:+52 (55) 8590-9000 x 7003 -- _ -- 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] Moving User Agent To Remote Location
Oh that's so smart!!! So, if I did not misunderstand you, for this one call, have: rtpstart=10004 rtpend=1008 Just for testing purposes, and deduce my way from there? Right now I am trying to call the phone from my softphone. That being said, I currently I am not able to reach the remote extension from my location here. I think this is the root of the problem here: -- Executing [1003@context-from-toronto:1] Dial(SIP/OpenSIPS-0009, SIP/1003, 20) in new stack Really destroying SIP dialog '06775f8653ff88b47cfa9ec123abdd89@127.0.0.1:0' Method: INVITE [Dec 12 15:35:54] WARNING[1736]: app_dial.c:2198 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [1003@context-from-toronto:2] Wait(SIP/OpenSIPS-0009, 1) in new stack -- Executing [1003@context-from-toronto:3] Answer(SIP/OpenSIPS-0009, ) in new stack Audio is at 5060 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP It's actually not able to create the SIP channel between the two UA? I will try taking opensips out of the picture and work outwards... N. On 1/3/13, Danny Nicholas da...@debsinc.com wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis Sent: Thursday, January 03, 2013 11:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Moving User Agent To Remote Location Hello Everyone, Before getting into SIP and RTP traces, I wanted to clarify some of the sip.conf settings that may to some seem redundant or have a misconception with. I do apologize if this has been discussed time and time again as I would imagine. If anything, this email would make google search results that much stronger :). With the UA local to my network I had tested two way audio, and now with the phone outside of network, we have no way audio. Before discussing NAT (which is enabled on the peer), and port forwarding (which is setup on the remote location), I would like to make sure I fully understand all the sip.conf settings. We are using Asterisk realtime via sip_buddies, and the fields in question are: (Enclosed in brackets are an example value for the setting) * host (dynamic): No problem here. Just wanted to mention that it's set as such * nat (yes): No problem here either * defaultuser (1...@example.com): Does the @example.com have to point to the UA (i.e., (1003@ua-public-ip), or is it just a name type field? * fullcontact: What to put here for a UA that is running at a remote location with dynamic external IP? * ipaddr (ua-public-ip): I did try setting it to the public ip of the UA, but is that really practical? What if I don't know where the initial registration request is coming from? I am guessing host=dynamic takes care of that. * defaultip?? * dynamic: Should this be set to yes, or is host=dynamic sufficient? The phone registers fine, and terminates a call through our providers. Just no audio both ways, which would suggest something more that an RTP issue which should at least have one way outgoing audio. Things that have been attempted: * Port forwarding to the phone * Changing defaultuser to 1003@ua-public-ip. This made our OpenSIPS sip proxy through a fit. Things I will attempt today: Calling the UA extension from an extension here SIP trace Your help is greatly appreciated!!! Nick. I'm going to vote for the RTP issue. If you are establishing a call but have no audio, you are getting the 5060 port, but not the 1-2 range that RTP normally expects. A better practice is to allocate 4 ports per line you expect to use in rtp.conf (1-2 would allow 2500 lines; much more that most folks need and more holes to monitor). -- _ -- 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] Moving User Agent To Remote Location
Am 03.01.2013 21:21, schrieb Nick Khamis: Oh that's so smart!!! So, if I did not misunderstand you, for this one call, have: rtpstart=10004 rtpend=1008 do you mean 1_000_8 ? Markus -- _ -- 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] Moving User Agent To Remote Location
On 01/03/2013 02:23 PM, Markus Weiler wrote: Am 03.01.2013 21:21, schrieb Nick Khamis: Oh that's so smart!!! So, if I did not misunderstand you, for this one call, have: rtpstart=10004 rtpend=1008 do you mean 1_000_8 ? Markus I think he means 10007. -- _ -- 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] Moving User Agent To Remote Location
On Thu, Jan 3, 2013 at 2:21 PM, Nick Khamis sym...@gmail.com wrote: [Dec 12 15:35:54] WARNING[1736]: app_dial.c:2198 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) Can you check that the registration is happening correctly? Try `sip show peers` or `sip show peer [destination phone]`. Usually cause 20 means the peer isn't registered. -- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248 -- _ -- 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 ban IP addresses
I have been seeing a lot of [Jan 2 16:36:31] NOTICE[7519]: chan_sip.c:23149 handle_request_invite: Sending fake auth rejection for device 100sip:100@108.161.145.18;tag=2e921697 in my logs lately. Is there a way to automatically ban IP address from attackers within asterisk ? You may want to check out this presentation form the last Astricon, it may be relevant: http://www.astricon.net/2012/videos/Automated-Hacker-Mitigation.html Cheers. JR -- JR Richardson Engineering for the Masses -- _ -- 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] Moving User Agent To Remote Location
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jason Parker Sent: Thursday, January 03, 2013 2:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Moving User Agent To Remote Location On 01/03/2013 02:23 PM, Markus Weiler wrote: Am 03.01.2013 21:21, schrieb Nick Khamis: Oh that's so smart!!! So, if I did not misunderstand you, for this one call, have: rtpstart=10004 rtpend=1008 The rtpend should be 10008 and rtpstart should be 10005. A SIP call in Asterisk originates on 5060 (5061 for TLS) and then spawns two RTP channels for audio. AFAIK the odd channel is send and the even channel is receive (smarter folks than me like Tzafir can give you the specifics; this was covered at least twice in 2012 threads). If you open 5060 on your NAT/firewall, but open no RTP channels, you will establish a call with no sound. -- _ -- 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] Moving User Agent To Remote Location
Oooops yes of course 10004-10007!! Simple math does not come easy anymore... Anyhow, I singled out Opensips and I have two way audio form UA(local) - UA(remote) but not from UA - Siptrunk. That being said maybe a small diagram of the architecture. Please don't laugh!!! :) I know having a block of static IPs would make like easier however UA (Remote) - Router (Remote) - Internet - Router (Local) - OpenSIPS+RTPProxy - Asterisk Port forwarding (Remote): 5060, and 1-5 to UA Port Forwarding (Local): 5060. and 1-5 to OpenSIPS) No Audio Port Forwarding (Local): 5060. and 1-5 directly to Asterisk Two Way Audio Cheers Guys! Nick On 1/3/13, Danny Nicholas da...@debsinc.com wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jason Parker Sent: Thursday, January 03, 2013 2:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Moving User Agent To Remote Location On 01/03/2013 02:23 PM, Markus Weiler wrote: Am 03.01.2013 21:21, schrieb Nick Khamis: Oh that's so smart!!! So, if I did not misunderstand you, for this one call, have: rtpstart=10004 rtpend=1008 The rtpend should be 10008 and rtpstart should be 10005. A SIP call in Asterisk originates on 5060 (5061 for TLS) and then spawns two RTP channels for audio. AFAIK the odd channel is send and the even channel is receive (smarter folks than me like Tzafir can give you the specifics; this was covered at least twice in 2012 threads). If you open 5060 on your NAT/firewall, but open no RTP channels, you will establish a call with no sound. -- _ -- 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] Moving User Agent To Remote Location
Just for grins, run netstat -anp on the call using just Asterisk and then again with OpenSIPS in the mix. It sounds like OpenSIPS or your RTPproxy is block the audio channels. -- _ -- 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] Moving User Agent To Remote Location
To Answer Some of You Questions: Please not that I replace the true domain wtih example, and the true ip for the remote UA with public-ip. Nothing against no one here, just don't know who else would read this email in the future!!! PS: The public IP of the remote UA is correct. SIP Show Peers: Name/username HostDyn Forcerport ACL Port Status Realtime 1002/1002@toronto.example. 192.168.2.13 N5060 UNKNOWNCached RT 1003/1003@toronto.example. -public-ip- D N5060 OK (86 ms) Cached RT Peers look registered correctly. This has now become a sip proxy issue :S. Thank you so much for your time guys!!! N. On 1/3/13, Nick Khamis sym...@gmail.com wrote: Oooops yes of course 10004-10007!! Simple math does not come easy anymore... Anyhow, I singled out Opensips and I have two way audio form UA(local) - UA(remote) but not from UA - Siptrunk. That being said maybe a small diagram of the architecture. Please don't laugh!!! :) I know having a block of static IPs would make like easier however UA (Remote) - Router (Remote) - Internet - Router (Local) - OpenSIPS+RTPProxy - Asterisk Port forwarding (Remote): 5060, and 1-5 to UA Port Forwarding (Local): 5060. and 1-5 to OpenSIPS) No Audio Port Forwarding (Local): 5060. and 1-5 directly to Asterisk Two Way Audio Cheers Guys! Nick On 1/3/13, Danny Nicholas da...@debsinc.com wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jason Parker Sent: Thursday, January 03, 2013 2:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Moving User Agent To Remote Location On 01/03/2013 02:23 PM, Markus Weiler wrote: Am 03.01.2013 21:21, schrieb Nick Khamis: Oh that's so smart!!! So, if I did not misunderstand you, for this one call, have: rtpstart=10004 rtpend=1008 The rtpend should be 10008 and rtpstart should be 10005. A SIP call in Asterisk originates on 5060 (5061 for TLS) and then spawns two RTP channels for audio. AFAIK the odd channel is send and the even channel is receive (smarter folks than me like Tzafir can give you the specifics; this was covered at least twice in 2012 threads). If you open 5060 on your NAT/firewall, but open no RTP channels, you will establish a call with no sound. -- _ -- 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
[asterisk-users] faxdetect on/off on the fly?
Hello, We want the ability to choose from an AGI script whether or not to enable faxdetect for calls over SIP or DAHDI. Is this possible, or can anyone suggest a workaround? Thanks for any advice. -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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] faxdetect on/off on the fly?
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Cunningham Sent: Thursday, January 03, 2013 3:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] faxdetect on/off on the fly? Hello, We want the ability to choose from an AGI script whether or not to enable faxdetect for calls over SIP or DAHDI. Is this possible, or can anyone suggest a workaround? Thanks for any advice. You should be able to call the AGI and set a dialplan variable and use Gotoif to do/not do faxdetect. Reading the .sample files for 11.0 it seems that normally these are configured until restart/reload but with a little testing, the default should be overrideable. -- _ -- 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] faxdetect on/off on the fly?
On Thu, 3 Jan 2013, David Cunningham wrote: We want the ability to choose from an AGI script whether or not to enable faxdetect for calls over SIP or DAHDI. What's the 'use case?' You're going to call in and execute an AGI that will enable faxdetect for future calls to this channel or other channels? Should the 'change' survive an Asterisk restart or an OS reboot? -- 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] Verizon SIP trunking Field Trial
On 03/01/2013 11:04 AM, Jeff LaCoursiere wrote: On 01/03/2013 09:56 AM, Carlos Alvarez wrote: On Thu, Jan 3, 2013 at 8:13 AM, Michael L. Young myo...@acsacc.com mailto:myo...@acsacc.com wrote: Where I am at is that they want us to use an SBC. One engineer asked about Cisco Call Manager. I told them that basically if I can accomplish the same thing with a Linux box (routing box and sip proxy box) without having to spend money on SBCs or expensive Cisco gear, that is the route we would like to go. We are looking at the possibility of handling 140 concurrent calls... that is what they are designing on their end as well. So, I am asking the community for any input. I have read on here and seen on IRC that some in the community are successfully using Asterisk with Verizon SIP. Verizon was going to check and see if they have any notes about that and those particular setups. Can anyone help share any information or tidbits on how they were able to sucessfully work with Verizon? It may be too late for this, but in working with another RBOC who didn't want to deal with Asterisk, I just asked what they do support, and modified the headers sent by Asterisk to claim that it was one of the devices on that list. Done. ROFL!! Well done. j -- _ -- 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 +1 -- Ron Wheeler President Artifact Software Inc email: rwhee...@artifact-software.com skype: ronaldmwheeler phone: 866-970-2435, ext 102 -- _ -- 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] Verizon SIP trunking Field Trial
Michael L. Young wrote: I should have probably stated that this is going to be going through an MPLS network being setup with Verizon. They may not be requiring that since it is within their network, not going over the internet. They have not said anything about the the need to secure the traffic coming from them or to them since the VoIP traffic will be on Verizon's network. Michael, Your email documents the same experience we had years ago. It was strange reading it and I was shocked that nothing has changed in that much time. Asterisk will work with Verizon's IP trunking product, but they're trying to make you jump through some old hoops first. We were using Verizon IP trunks over an MPLS network in 2008. At the time, they did not require IPSEC for signaling. However, they did want us to install an SBC and actually provided us with an AudioCodes nCite 1000 at their cost. It just acted as a proxy, so it didn't affect interoperability with Verizon's IP trunks and I wouldn't buy one only to satisfy them. We were quite happy with the service, so I'd encourage you to go ahead with the field trial without putting an SBC in place. Remember that you will be paying them, so they should be working to fit your design and if they reject you for some arcane reason then you are better off with another provider anyway. Don't hesitate to let them know that you know you're jumping through the same hoops that have been in place since 2008 and you'd appreciate it if they would streamline the process to save time and money. Tell them that Asterisk should already be on their certified list of approved devices because they've been running field trials and production setups on it for years. Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer -- _ -- 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] User busy issue in A400P 4 FXO card
Hi,
I tried the option and got following message.
PBX1*CLI
-- Starting simple switch on 'DAHDI/1-1'
== Starting DAHDI/1-1 at from-pstn,s,1 failed so falling back to exten 's'
== Starting DAHDI/1-1 at from-pstn,s,1 still failed so falling back to
context 'default'
-- Executing [s@default:1] Playback(DAHDI/1-1, vm-goodbye) in new
stack
-- DAHDI/1-1 Playing 'vm-goodbye.gsm' (language 'en')
-- Executing [s@default:2] Macro(DAHDI/1-1, hangupcall) in new stack
-- Executing [s@macro-hangupcall:1] GotoIf(DAHDI/1-1, 1?skiprg) in
new stack
-- Goto (macro-hangupcall,s,4)
-- Executing [s@macro-hangupcall:4] GotoIf(DAHDI/1-1, 1?skipblkvm)
in new stack
-- Goto (macro-hangupcall,s,7)
-- Executing [s@macro-hangupcall:7] GotoIf(DAHDI/1-1, 1?theend) in
new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] Hangup(DAHDI/1-1, ) in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on
'DAHDI/1-1' in macro 'hangupcall'
== Spawn extension (default, s, 2) exited non-zero on 'DAHDI/1-1'
-- Hungup 'DAHDI/1-1'
When i was looking at asterisk console, Caller phone (mobile phone) shows
that User busy message with busy tone.
Starting simple switch on 'DAHDI/1-1' shows after that phone get busy
tone.
The server is located in Canada and I use US as country.
Let me know your feedback.
Thanks,
Selva
On Thu, Jan 3, 2013 at 4:44 AM, A J Stiles asterisk_l...@earthshod.co.ukwrote:
[from-pstn]
s,1,NoOp(Incoming call from ${CALLERID(num)})
s,2,Dial(200)
s,3,Hangup()
; end of from-pstn context
--
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Re: [asterisk-users] faxdetect on/off on the fly?
Hi Steve, We have all calls going to an AGI, which decides where the number will get routed to, and if fax detection should be enabled for this call. The choice should only apply to the current call. Thanks very much. On 3 January 2013 17:46, Steve Edwards asterisk@sedwards.com wrote: On Thu, 3 Jan 2013, David Cunningham wrote: We want the ability to choose from an AGI script whether or not to enable faxdetect for calls over SIP or DAHDI. What's the 'use case?' You're going to call in and execute an AGI that will enable faxdetect for future calls to this channel or other channels? Should the 'change' survive an Asterisk restart or an OS reboot? -- 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-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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
