Re: [asterisk-users] Echo Cancellation
On Tue, Aug 20, 2013 at 3:01 PM, Ghanshyam btcs.em...@gmail.com wrote: Shaun Ruffell sruffell at digium.com writes: On Thu, Jul 25, 2013 at 02:51:02AM -0700, bilal ghayyad wrote: Hello; If our Digium Telephony Card does not support echo cancellation like (1TDM410PLF or 1AEX410PLF), what is the best and simple way to overcome the echo? Regards Bilal Also, just FYI, those cards do support adding a hardware echocancelation module. But I would recommend trying the software solutions first. I am new to asterisk and pbxiaf. I have setup a VM and intend to use it for a number of android phones (using sipdroid) all on local wifi for a conference call. There is a lot of echo. I also added a GoogleVoice account, even on a standard phone call the remote party gets a late but large echo. As I understand, DAHDI works with the cards, seeing as my complete system is software based, how would I get rid of the echo. Thanks in advance. Ghanshyam -- _ -- 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 Unlike our friend Ghanshyam, i'm sorry for the hijack however does OSLEC only work with telephony cards or, will it also work on a purely Asterisk SIP environment? Thanks in Advance, 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] Echo Cancellation
Thanks Eric, I breezed through the documentation and got the impression that this was the case. Good luck on getting rid of that echo Bilal! N. -- _ -- 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] Am I being hacked?
They are sending requests from his own public ip huh? Trade secrets H, IPTaibles, Fail2Ban (as a preventative), there is something I am missing What the f is it called again? Oh yeah Pike!!! alwaysauthreject = yes I don't know about that However, using the mac address of the device as the `sipbuddies.name`, and having `sipbuddies.secret` other than `12345a` ;), I would say yes too. One of Asterisk's dirty little secrets is that it does not show the source IP when a device or hacker tries sending a call without registering. The rejection message in the logs do not show the IP of the attacker. Yes it sucks, yes it has been that way for many many years. Does not good if the address is spoofed as it seems is the case here. IPTables, class c filter rule buy yourself a burger or a slice... Be strong my legit brotherins!!! N. -- _ -- 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] Am I being hacked?
#!/bin/bash IPTABLES='/sbin/iptables' #Set interface values INTIF1='eth0' # Set Limits LIMIT=2/sec LOGLIMIT=5/min LIMITBURST=5 #flush rules and delete chains $IPTABLES -F $IPTABLES -X #echo -e- Dropping Forward Requests $IPTABLES -P FORWARD DROP #echo -e- Dropping Input Requests $IPTABLES -P INPUT DROP #echo -e- Dropping output requests $IPTABLES -P OUTPUT DROP #echo -e- Accepting input lo traffic $IPTABLES -A INPUT -i lo -j ACCEPT #echo -e- Accepting output lo traffic $IPTABLES -A OUTPUT -o lo -j ACCEPT #echo -e- Defined Chains $IPTABLES -N ICMP $IPTABLES -N TCP $IPTABLES -N UDP $IPTABLES -N LOGINPUT $IPTABLES -N LOGOUTPUT #echo -e- Accepting incoming SIP Traffic $IPTABLES -A UDP -p udp -m udp -s local /24 --sport 5060 -d asterisk server --dport 5060 -j ACCEPT $IPTABLES -A UDP -p udp -m udp -s time warner ip --sport 5060 -d asterisk server --dport 5060 -j ACCEPT # $IPTABLES -A UDP -p udp -m udp -s 0.0.0.0/0 --sport 5060 -d asterisk server --dport 5060 -j DROP #echo -e- Accepting outgoing SIP Traffic $IPTABLES -A UDP -p udp -m udp -s asterisk server --sport 5060 -d local /24 --dport 5060 -j ACCEPT $IPTABLES -A UDP -p udp -m udp -s asterisk server --sport 5060 -d time warner sip server--dport 5060 -j ACCEPT # $IPTABLES -A UDP -p udp -m udp -s asterisk server --sport 5060 -d 0.0.0.0/00 --dport 5060 -j DROP RTP Traffic *may* or *may* not come from the same server as the SIP messages. It also *may* or *may not* come from the server provider's net mask or an underline either way, until you have determined this: #echo -e- Accepting incomming RTP Traffic $IPTABLES -A UDP -p udp -m udp --dport 8000:65000 -j ACCEPT # $IPTABLES -A UDP -p udp -m udp -d asterisk server --dport 8000:65000 -j ACCEPT # $IPTABLES -A UDP -p udp -m udp -s local /24 -d asterisk server --dport 8000:65000 -j ACCEPT # $IPTABLES -A UDP -p udp -m udp -s time warner -d asterisk server --dport 8000:65000 -j ACCEPT # $IPTABLES -A UDP -p udp -m udp -s 0.0.0.0/0 -d asterisk server --dport 8000:65000 -j DROP #echo -e- Accepting outgoing RTP Traffic $IPTABLES -A UDP -p udp -m udp --sport 8000:65000 -j ACCEPT # $IPTABLES -A UDP -p udp -m udp -s asterisk server --sport 8000:65000 -j ACCEPT # $IPTABLES -A UDP -p udp -m udp -s asterisk server -d local /24 --dport 8000:65000 -j ACCEPT # $IPTABLES -A UDP -p udp -m udp -s asterisk server -d time warner --dport 8000:65000 -j ACCEPT # $IPTABLES -A UDP -p udp -m udp -s asterisk server -d 0.0.0.0/0 --dport 8000:65000 -j DROP #echo -e- Accepting input ICMP, TCP, and UDP traffic to open ports $IPTABLES -A INPUT -i $INTIF1 -p icmp -j ICMP $IPTABLES -A INPUT -i $INTIF1 -p tcp -j TCP $IPTABLES -A INPUT -i $INTIF1 -p udp -j UDP #echo -e- Accepting output ICMP, TCP, and UDP traffic to open ports $IPTABLES -A OUTPUT -o $INTIF1 -p icmp -j ICMP $IPTABLES -A OUTPUT -o $INTIF1 -p tcp -j TCP $IPTABLES -A OUTPUT -o $INTIF1 -p udp -j UDP #echo -e- Logging Dropped Input Traffic $IPTABLES -A LOGINPUT -i $INTIF1 -p icmp -m limit --limit $LOGLIMIT --limit-burst $LIMITBURST -j LOG --log-prefix ICMP LOGINPUTDROP: --log-tcp-options --log-i$ $IPTABLES -A LOGINPUT -i $INTIF1 -p tcp --tcp-flags FIN,SYN,RST,ACK SYN -m limit --limit $LOGLIMIT --limit-burst $LIMITBURST -j LOG --log-prefix TCP LOGINPUTDRO$ $IPTABLES -A LOGINPUT -i $INTIF1 -p udp -m limit --limit $LOGLIMIT --limit-burst $LIMITBURST -j LOG --log-prefix UDP LOGINPUTDROP: --log-tcp-options --log-ip-$ $IPTABLES -A LOGINPUT -i $INTIF1 -f -m limit --limit $LOGLIMIT --limit-burst $LIMITBURST -j LOG --log-prefix FRAGMENT LOGINPUTDROP: --log-tcp-options --log-ip$ $IPTABLES -A LOGINPUT -j DROP $IPTABLES -A INPUT -p icmp -i $INTIF1 -j LOGINPUT $IPTABLES -A INPUT -p tcp -i $INTIF1 -j LOGINPUT $IPTABLES -A INPUT -p udp -i $INTIF1 -j LOGINPUT #echo -e- Logging Dropped Output Traffic $IPTABLES -A LOGOUTPUT -o $INTIF1 -p icmp -m limit --limit $LOGLIMIT --limit-burst $LIMITBURST -j LOG --log-prefix ICMP LOGOUTPUTDROP: --log-tcp-options --log$ $IPTABLES -A LOGOUTPUT -o $INTIF1 -p tcp --tcp-flags FIN,SYN,RST,ACK SYN -m limit --limit $LOGLIMIT --limit-burst $LIMITBURST -j LOG --log-prefix TCP LOGOUTPUTD$ $IPTABLES -A LOGOUTPUT -o $INTIF1 -p udp -m limit --limit $LOGLIMIT --limit-burst $LIMITBURST -j LOG --log-prefix UDP LOGOUTPUTDROP: --log-tcp-options --log-i$ $IPTABLES -A LOGOUTPUT -o $INTIF1 -f -m limit --limit $LOGLIMIT --limit-burst $LIMITBURST -j LOG --log-prefix FRAGMENT LOGOUTPUTDROP: --log-tcp-options --log-$ $IPTABLES -A LOGOUTPUT -j DROP $IPTABLES -A OUTPUT -p icmp -o $INTIF1 -j LOGOUTPUT $IPTABLES -A OUTPUT -p tcp -o $INTIF1 -j LOGOUTPUT $IPTABLES -A OUTPUT -p udp -o $INTIF1 -j LOGOUTPUT #echo -e- Rejecting input TCP and UDP traffic to closed ports $IPTABLES -A INPUT -i $INTIF1 -p tcp -j REJECT --reject-with tcp-rst $IPTABLES -A INPUT -i $INTIF1 -p udp -j REJECT --reject-with icmp-port-unreachable #echo -e- Rejecting output
Re: [asterisk-users] G729 Passthrough How To
Anyone? :) N. On 8/13/13, Nick Khamis sym...@gmail.com wrote: Hello Everyone, We are currently experiencing some higher load on our servers, and since signaling comes into our servers on G729, we would like to implement G729 pass-through. A few questions arise, do we need to convert all the recording to the codec, and what about voicemail? We are also using A2Billing (hope I am not violating any thread rules), and would like to convert all that recording to G729 as well. Any help is greatly appreciated. Kind Regards, Nick from Toronto. -- _ -- 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] G729 Passthrough How To
Hey!!! Eric thank you so much for your response. Could you guys please direct us in achieving as much as possible. For example: * What linux command can we use to convert all recording to G729 * Which files do we need to convert and there locations * For *testing* how do we make sure Asterisk NEVER EVER transcodes. Do we still need the G729 codec installed on the asterisk machine if we manage to implement pass-through that would suffice our needs. Kind Regards, 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] G729 Passthrough How To
I forgot to mention that all our equipment (phones etc..) are using G729, and this is for internal use over the net. The problem, concurrent calls, and bad bandwidth at some locations... N. -- _ -- 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] G729 Passthrough How To
Hello Ashgar, Thank you so much for your response. As removing A2B is not an option we would first like to begin by converting all audio files (Asterisk, VM, A2B prompts etc...) to G729 to minimize unneeded trascoding. Linux commands and the list of recording would be a great help. Sorry, not new to VoIP but new to Asterisk :). N. -- _ -- 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] G729 Passthrough How To
Hey Eric, I do have the codec installed, and I remember hearing about the CLI command to convert. Is there a recent how-to of blog already discussing this somewhere? N. On 8/14/13, Nick Khamis sym...@gmail.com wrote: I wanted to mention that I do not mind posting the converted files on this list for future individuals, given that I am not doing anything illegal... N. -- _ -- 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] G729 Passthrough How To
I wanted to mention that I do not mind posting the converted files on this list for future individuals, given that I am not doing anything illegal... N. -- _ -- 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] G729 Passthrough How To
Not really no... And how do I make sure Asterisk always generates prompts and VM recordings in G729 from now on. This is also hard to find information.. N. On 8/14/13, Eric Wieling ewiel...@nyigc.com wrote: I have no idea, though Google might. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis Sent: Wednesday, August 14, 2013 11:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] G729 Passthrough How To Hey Eric, I do have the codec installed, and I remember hearing about the CLI command to convert. Is there a recent how-to of blog already discussing this somewhere? N. On 8/14/13, Nick Khamis sym...@gmail.com wrote: I wanted to mention that I do not mind posting the converted files on this list for future individuals, given that I am not doing anything illegal... N. -- _ -- 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
[asterisk-users] G729 Passthrough How To
Hello Everyone, We are currently experiencing some higher load on our servers, and since signaling comes into our servers on G729, we would like to implement G729 pass-through. A few questions arise, do we need to convert all the recording to the codec, and what about voicemail? We are also using A2Billing (hope I am not violating any thread rules), and would like to convert all that recording to G729 as well. Any help is greatly appreciated. Kind Regards, Nick from Toronto. -- _ -- 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] PCI Passthrough of T1 cards
Asterisk does fine in a virtual instance. The key is finding hardware that would support more than just virtualization (i.e., SR-IOV) Not sure if such a card exist. -- _ -- 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] SIP Trunking Mantra (Origination)
Any other experts out there? -- _ -- 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] SIP Trunking Mantra (Origination)
On 6/25/13, Jai Rangi jpra...@didforsale.com wrote: Not a problem, I wanted to tell you the diff between PRI and sip trunking. I am sure there are lots of option we are just fine what ever works best for you. Back to subject we strongly believe that sip trunking is far better option than PRI and that's the way to go in future. Jai Hello Jai the benefits of SIP trunking is well noted. However, there will always be an underline interconnect that makes SIP trunking possible. What I am trying to say is that there will always be ISUP/ISDN trunk groups that we throw a TCP/IP stack on top, and offer clients with SIP trunking. We are looking for information on how to accomplish interconnect using ISUP trunks (i.e., SS7 interconnect). Kind Regards, 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
[asterisk-users] SIP Trunking Mantra (Origination)
Hello Everyone, We are currently having talks with various service providers, and trying to determine what the best way is to interconnect in order to have access to the PSTN network. As you know there are two ways of doing this: Traditional PRI: Have trunks grouped into a transport layer such as OC3/12. With DIDs attached to the group. As you many know, this approach would also require a POP near the CO of the exchange we want to service etc.. We could also have the service provider backhaul some of the NXX in areas we do not have a POP, to a location near by. SIP Trunking: SIP traffic coming through the end of transport layer such as OC3 or ethernet connection directly connected to the service provider, with DID that can come from anywhere. No need for a POP in Chicago, for example, when we are located in Kansas. The benefits of one over the other are known, and not the topic of this message. What we are trying to determine are: When talking market price, a virtual PRI/SIP Trunk interconnect costs about 500-550 per 24 channel virtual pri. This compared to a true ISDN/PRI which can costs between 200-500 dollars depending who you talk to. We also have to take into consideration the hardware needed for either setup i.e.: * Option 1: SIP Proxy * Option 2: media gatweays, multiplexers, media server Even though it was natural to talk about pricing, this is still not what we are interested in knowing. What we are interested in finding out is: * How are service providers that offer virtual pris interconnected with their suppliers? I would imagine that some (non-CLECS), are renting a connection from the LECs, and grouping PRI/ISDN trunks (option 2). And others (CLECS), have a A-Link/ISUP trunk interconnect to the CO. - Which brings up a second question. How does a PRI trunk group differ from an ISUP trunk. I don't know much about and ISUP trunks and would *really* appreciate having someone educate us on (i) the concept, (ii) what type of equipment would be needed, (iii) how it differs from ISDN trunk groups. (iv) is it only available for LECS I do have more questions, however for the sake of brevity will stop right here. And before anyone asks the it depends what you want to do, I will mention that we are trying to establish an interconnection that will sustain 2016 channels or 84 T1s, and 5000 DIDs. We are not trying to become a CLEC however, still feel that option 2 would be the better choice for reasons covered here, and some that are left implicit (i.e, quality, reliability of managing our own networks..). Your insights are 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] SIP Trunking Mantra (Origination)
Thank you mitul. N. -- _ -- 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] PCI Passthrough of T1 cards
Hello James, Thank you so much for your response. I should have chose my words carefully. PCI pass-through in terms of virtualization of devices and it's draw back are well know. I was leaning more towards near host performance virtualization using SR-IOV. This moves emphasis back to the production drivers of the interface card using virtual functions etc., and can provide near host performance. Rephrasing my question, are any of the T1 pci manufactures providing support for virtualization using SR-IOV and virutal functions? Kind Regards, 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] SIGTRAN Integration
Anyone? N. -- _ -- 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] PCI Passthrough of T1 cards
Anyone try this? I saw a post here: http://www.elastix.org/index.php/en/component/kunena/1-installation-issues/94041-setup-of-sangoma-a101-in-my-elastix.html But not sure if it's possible. What I am asking is if there are any T1 cards with virtual functions implemented in their drivers to allow pci-passthrough? Kind Regards, 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] SIGTRAN Integration
What about projects like YATE, DiaStar, and mobicents (even though I have no idea how to approach that project in terms of downloading etc..). Are there any mature SS7/SIGTRAN stacks? Kind Regards, 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
[asterisk-users] SIGTRAN Integration
Hello Everyone, I was wondering how SIGTRAN/SS7oIP can fit into our currently 100% SIP model. We are looking to interconnect with the PSTN world, and our supplier has given us a few options. We can either do this over traditional PRIs, A-Links or the SS7IP new. I am really interested in SIGTRAN, and was wondering how some of you have integrated it into your architecture. Can Asterisk handle SS70IP or do we have to put a yate or squire server at the end of that connection. Kind Regards, 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] SIGTRAN Integration
Hello Mitul, Thank you so much for your response. During the testing phase we would like to employ an open source solution, and wanted to know what people have had success with, given the different user part etc.. On a side note, anyone know of service providers offering SIGTRAN? Kind Regards, 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] ILEC Interconnect: Basic MUX: M13 vs DCS: VT1.5 vs DS3
Hello James, thank you so much for your response! On 6/14/13, James Cloos cl...@jhcloos.com wrote: If they will do atm over oc-n, perhaps that would work better. Yes they will do atm over oc-n only not sure if they will ring or spur it... Ie, a perm virt circ for SS7 and as-needed vc's for ulaw. I know you're a busy guy, can you please expand on this kindly :). Atm oc-n cards with linux sw support are widely available, according to goog. libss7 and and ast *might* need a bit of patching to work with it, but it shoudn't take too much. Will goog some more! Sip/rtp over private ptp ethernet is an option with at least some of the ILECs. They may call it virtual-pri or some such. Of course, if they are installing an actual sonet ring, and not just a spur, that can have built-in redundancy, depedning on physical routing. We are trying to position ourselves as facility based virtual PRI service provider here, and would like to put something of our own together vs. resell another LEC's red ribbon product. -JimC -- James Cloos cl...@jhcloos.com OpenPGP: 1024D/ED7DAEA6 PS I love your website!!! Kinid Regards, 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] ILEC Interconnect: Basic MUX: M13 vs DCS: VT1.5 vs DS3
Hello Brian, Thank you so much On 6/12/13, Brian LaVallee b.laval...@globaltank.jp wrote: Hi Nick, Going from DS1 to OC-n is a multi-step process. Typically requiring a hardware device to handle each MUX step. But you can find hardware that handles multiple MUX steps together. The connection is coming into our premise on the OC-n transport. The question now is should we have it multiplexed as DS1 or VT1.5s to the DS3s. What is common today, I think DS1 VT1.5s mappings are more flexible? VT1.5 is just a raw OC-n channel containing a single DS1. An M13 device converts between DS3 and DS1. Understood!!! A DACS (DCS or DXC) provides M13 conversion, sometimes even capable of extracting the raw VT1.5 signal directly to DS1. The ILEC transport option you choose really depends on the terminating interface. Do you want to connect with a DS3 or OC-n? The transport is coming in as OC-n. What I am trying to figure out are the advantages of mapping the STS-1 using DS1s or VT1.5s. No matter what hardware you choose, you will need to convert to single copper pairs (DS1/T1) to connect to your Asterisk boxes. So an M13 or DCS will be necessary to reach the DS1 level. The device you choose depends on budget and growth expectations. Typically a DCS is an expensive investment, handling hundreds of DS3's. An M13 device is typically a small unit that handles one or two DS3's. This is almost understood. Is an M13 device basically a MUX (In our case STS-1-DS1)? From there we would plug the signaling into the Quad Digiums as you mentioned (this is where I get more comfortable). Could you kindly post a link to an entry level DCS with OC-n interfacing and M13s being used today. That way I can see what functionality each provides and determine which better suits our need. I am guessing, but hate to presume: M13: Adtran MX2800 DCS: Mediant 3000, Metaswitch 0610 etc.. The advantage comes when you add the 29th DS1. With VT1.5 it's just adding a single channel, DS3 will require another whole DS3 to get an additional DS1. This is why we are going SONET. It's a new transport layer for me compared to DS3s, and want to make sure I can put everything together at the network level. Sincerely, Brian LaVallee Thank you kindly, Nick from Montreal. -- _ -- 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] ILEC Interconnect: Basic MUX: M13 vs DCS: VT1.5 vs DS3
On 6/12/13, Don Kelly d...@donkelly.biz wrote: Is there an OC-n to SIP solution that makes sense? --Don Hello Don, what will be coming out of the network discussed above would be SIP. Kind Regards, 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] ILEC Interconnect: Basic MUX: M13 vs DCS: VT1.5 vs DS3
Correction: I think VT1.5s mappings are more flexible? Sorry! N. -- _ -- 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] ILEC Interconnect: Basic MUX: M13 vs DCS: VT1.5 vs DS3
On 6/13/13, Eric Wieling ewiel...@nyigc.com wrote: Verizon (NE ILEC) has SIP handoff. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis Sent: Thursday, June 13, 2013 8:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ILEC Interconnect: Basic MUX: M13 vs DCS: VT1.5 vs DS3 Hello Eric, Thank you so much for your response. Is this an ISUP-IP interconnect (i.e., SS7IP), or are you referring to the traditional DID based VoIP. In either case, do you have a contact I can get a hold of. Kind Regards, 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] ILEC Interconnect: Basic MUX: M13 vs DCS: VT1.5 vs DS3
Hello Eric, Thank your for your reponse. We are discussing interconnects at a different level. We are more interested in SS7 or ISUP-IP SS7IP type interconnects. There are many people that offer DIDs channels etc. over the internet. Including us. N. -- _ -- 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] ILEC Interconnect
Hello Everyone, We are looking to interconnect with a local ILEC over an OC-n transport layer. They basically gave us two options in terms of mapping the SONET to the DS3: * VT1.5s mapping * DS1s mapping The second option is quite clear. We would MUX the connection, and plug the lines into qaud t1 cads etc... The tech mentioned that with the second option we would also need a DACS to convert back to M13 mapping. I was scared to tell him that I could not follow can someone explain that to me kindly :). I don't know much about VT1.5 mapping. Can someone kindly explain what the benefits or lack of are in choosing that option. Also what type of additional equipment we would need? In case I have overlooked something, can you gents please tell me what I will need in terms of hardware in both cases (minus routers and switches). What we are looking at is: CO | | | OC-n | v DS3 MUX | | v 21 Asterisk boxes with quad T1s Kind Regards, 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] Asterisk sends the INTERNAL IP address of my equipments to my SIP friends?!?
You mean the SDP payload? You kind of need that c= is used for RTP transmission. o= always confuses me so I will just say it's important at well. You can put a proxy in the middle and do topology hiding I guess however, that is beyond the scope of this list? Kind Regards, Nick. On 6/12/13, Mickael MONSIEUR mickael.monsi...@gmail.com wrote: Good morning, or Good afternoon! It depends :-) I have a standard Asterisk configuration: SIP friends (phones)-Asterisk-SIP gateway to PSTN converter 80.236.215.61 109.69.217.6internal IP ( 10.4.0.10/255.255.255.0) When analyzing traffic on a SIP friend/phone I see this: INVITE sip:@80.236.215.61:64946;ob SIP/2.0 Via: SIP/2.0/UDP 109.69.217.6:5060;branch=z9hG4bK52d50250;rport Max-Forwards: 70 From: sip:@109.69.217.6;tag=as15b47581 To: test sip:@109.69.217.6;tag=kp1VwHD80rA9MVdBjTF4jyFIaCkrJcjh Contact: sip:x@109.69.217.6 Call-ID: MSMhw2bsheHWAQgHlae3O7yKQ2P9EcsM CSeq: 102 INVITE User-Agent: Asterisk Require: timer Session-Expires: 1800;refresher=uas Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 217 v=0 o=root 664087974 664087976 IN IP4 10.4.0.10 s=Asterisk c=IN IP4 10.4.0.10 t=0 0 m=audio 8652 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv My equipement IP 10.4.0.10 is visible to the user, why? Thank you, Mickael -- _ -- 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] OC3/STM-1 Line Card
Anyone? -- _ -- 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] OC3/STM-1 Line Card
Thank you so much for your responses!!! With this route we would have to manage so many * boxes with T1s, not to mention, the hit we would take on the MUX. Any decent DS/T3 cards out there? N. -- _ -- 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] OC3/STM-1 Line Card
Hello Everyone, Anyone know of a way of bypassing the 90K audiocodes mediant 3000 equipped for STM-1 interface using line cards and a linux box :). Kind Regards, 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
[asterisk-users] Implementing G729 Passthrough - VM recordings, maybe even a2billing
We would like implement G729 passthrough for our calls and get rid of the encoding overhead, and a little confused as to how to do this, and some unanswered questions. Do we need the open source G729? If so, do we still need the patent license. Not so much of an issue, just checking. Finally, a recent howto of how to enforce pci passthrough and disable encoding would be greatly appreciated. Oh, and there is also the issue with VM recordings, message etc.. On a slightly unrelated, we are using a2billing on some of our machines, and I think that we would have to convert that media from slink to G729 if not done so already... Thanks in Advance, 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
[asterisk-users] Fiber or regular DSL Supported Gateways/PRI
Hello Everyone, I am looking to getting converged with the local ILEC here in Canada (Bell or Telus), and was wondering if I can get some more information about typical setups. DIDs and channel offerings from third party clecs does not fit our business model and that's why we are looking to purchasing phone numbers directly from the ILEC, and manging our own PRIs for a convergence of sorts. My questions are: * Do we need a T1/3 of DS3 connection setup, or can we function of regular IPX, DSL, Fiber... * If we can function on a regular fiber or DSL connection, which media gateways provide a WAN interface for such a connection, or will we need a ethernet to T1 adapter to plug our connection to the appliance (e.g., audiocodes or sangoma). Finally, if you have recently setup such an interconnection, some details on what to expect, options and so on would be greatly appreciated. Kind Regards, 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] Fiber or regular DSL Supported Gateways/PRI
Anyone? -- _ -- 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 on Solaris
Hello Doug, A quick sift through http://www.mail-archive.com/search?l=asterisk-users%40lists.digium.comq=solaris+10, yielded many unanswered questions, questions with returning questions etc... There was even an email that had the same subject line. Surely, the creator of that email could take a second and say don't do it, or the SolarisVoIP project is a flop Surely a lot has changed since this question was raised in 2006. I'm not asking how to get my zaptel running on Solaris, but rather, if there are any performance that can be gained using a Solaris+Asterisk setup for SIP. Also, what are some recent experiences... Kind Regards, 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] Asterisk on Solaris
Bump On 5/23/13, Nick Khamis sym...@gmail.com wrote: Hello Everyone, I have bumped into the thralling penguin page on linux vs solaris for asterisk. Does the benchmark still hold with the newer versions of kernels? Curious to know of your thoughts. Also, they mentioned running it on Sun Fire x2100, but no benchmarks were given for that. Can increased performance be accomplished simply by changing to Solaris or OpenSolaris? Kind Regards, 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
[asterisk-users] Asterisk on Solaris
Hello Everyone, I have bumped into the thralling penguin page on linux vs solaris for asterisk. Does the benchmark still hold with the newer versions of kernels? Curious to know of your thoughts. Also, they mentioned running it on Sun Fire x2100, but no benchmarks were given for that. Can increased performance be accomplished simply by changing to Solaris or OpenSolaris? Kind Regards, 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] Tier 1 Service Providers (ATT, Level 3)
Hello Roel, Thank you so much for your response. We currently employ a number of similar companies. Given our increasing traffic we are really looking towards the incumbents for various reasons. The purpose of my post is in the hopes that someone watching will let us know how to setup interconnection agreements. We are also looking at XO communications and Verizon. Kind Regards, 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] Tier 1 Service Providers (ATT, Level 3)
On 5/10/13, Nick Khamis sym...@gmail.com wrote: Anyone here using Level 3 or ATT wholesale sip terminations services? I would like to know on any minimums they would require? Also, an idea of how competitive the rates are. I am not asking to disclose your custom rate deck, just a what to expect. Finally, if you guys can PM me contact info to someone from the wholesale department, I would really appreciate it. Kind Regards, Nick. Anyone? -- _ -- 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] Tier 1 Service Providers (ATT, Level 3)
Anyone here using Level 3 or ATT wholesale sip terminations services? I would like to know on any minimums they would require? Also, an idea of how competitive the rates are. I am not asking to disclose your custom rate deck, just a what to expect. Finally, if you guys can PM me contact info to someone from the wholesale department, I would really appreciate it. Kind Regards, 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] ISP trunk session ID?
Sorry to chime in here, is it possible to change the Server: Asterisk , s=Asterisk, and o= within sip.conf? What are the directives exactly please? Thanks in Advance, Nick. On 5/10/13, Asghar Mohammad asghar...@gmail.com wrote: hi, you can try to change sip user agent and sdp session s , owner in sip config same as your phone,s (modem). asterisk by default send user agent = asterisk version , s= asterisk , o= asterisk. some providers are not happy if they see asterisk word :) On Sat, May 11, 2013 at 12:27 AM, Sergej Petrovsky sergej5...@yandex.comwrote: Hi folks, What I trying to do here is exactly this: http://www.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/I_sect14_tt599.html My provider given me a Huawei modem which have 2 phone jacks on it, but instead of using it I rather redirect my POTS number to my PBX. I ran into couple of bumps on the road but now it's half-working. I extracted the SIP user, pass, server info from the modem and even managed to put my PBX into the same VLAN they use, on the exact same IP address like the modem but there is 1 problem: It seems this modem also sends some session ID to the ISP's sip server, something what Asterisk doesn't by default. So if I do this: 1, Let the modem register at the sip service (the phone number can be called and ringing out) 2, Disconnect the modem 3, Let the PBX connect to the SIP server 4, PBX accepts the calls 5, About 5-10 minutes later it stops doing it, when I call the number it shows busy (beep, beep, beep), no matter if I restart Asterisk or not it won't work anymore just if I do the same trick again I'm sure the remote SIP server breaks the voip channel or something, it does NOT drop me out tho, my PBX can register any time without problem but no packets will ever come forward me anymore. It's kind of hard to solve this from 1 side. There must be some solution for this. Please help! Thank You, Sergej -- _ -- 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] caller_id vs cid_number
Are these both caller id presentation related? If not, which on is currently being used. Finally, is there a latest sip_peers table structure to use with 1.8, without the obvious hacks, deprecations. and redundancies? Thanks in Advance, 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
[asterisk-users] Network based transcoding
Hello Everyone, We are looking for solutions where the transcoding is abstracted away from our * box (i.e., to the network layer) using some carrier grade gateway, or router. The reason for my post is to know about solutions people have used in the past, and how it fits into their overall architecture. Our transcoding needs consists mainly of u/alaw - g729, and gsm would also be good General details on how this would work, and concurrent capacities are greatly appreciated. Thanks in Advance, 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] Network based transcoding
Hello Gentlemen, Thank you so much for your response, we have adopted transcoding cards in our old system, and they do have some limitations, especially when it comes to concurrent calls. We were looking more into the lines of a scalable multi server router like a cisco 3745. And loading it with maximum number of packet voice DSP modules. A gateway like the ones mentioned would also work fine. Anything but an SBC, since we handle that on our own, and it would just be redundant. What I am kind of unclear about, is how a network appliance would fit into our architecture when it comes to processing the RTP streams before passing it on to our servers for example Please excuse my noob question ;) N. On 4/12/13, jg webaccou...@jgoettgens.de wrote: Did you already look at transcoding cards? E.g.: http://www.sangoma.com/media-processing/voice-transcoding-boards/ They also have separate boxes (http://www.sangoma.com/products/netborder-transcoding-appliance/). Personally, I prefer to have everything in a single box if there aren't too many parallel calls. jg -- _ -- 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] Network based transcoding
Sorry for the missing info. Our current architecture is as such: NAT - SIP/RTP Proxy - *(n) Our concurrent sessions usually peak at between 700-800 channels. On average about 450. I will of course look at the documentation to better understand how a transcoding appliance would fit in our architecture, and thank you so much for the links!!! But generally speaking, does the appliance process/transocde the RTP stream and then forward it to our SIP/RTPProxy? Is it really that easy? :) Again, I will breeze through the documentation to get the detailed how it works info I am looking for. Thanks Kindly, N. On 4/12/13, jg webaccou...@jgoettgens.de wrote: What do you mean with servers? A simple proxy, or a B2BUA (aka Asterisk)? Depending on the basic configuration the server might or might not have to deal with some or all of the RTP-streams. The already mentioned company Sangoma usually has good documentation about their products (see http://www.sangoma.com/products/d500-400-2000-sessions/ or http://www.sangoma.com/products/d150-30-400-sessions/) and examples on how to use them. jg -- _ -- 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] Network based transcoding
For anyone else that may be interested in the future, I found a detailed depiction here: http://wiki.sangoma.com/ntg-theory-of-operation Thanks again, N. On 4/12/13, Nick Khamis sym...@gmail.com wrote: Sorry for the missing info. Our current architecture is as such: NAT - SIP/RTP Proxy - *(n) Our concurrent sessions usually peak at between 700-800 channels. On average about 450. I will of course look at the documentation to better understand how a transcoding appliance would fit in our architecture, and thank you so much for the links!!! But generally speaking, does the appliance process/transocde the RTP stream and then forward it to our SIP/RTPProxy? Is it really that easy? :) Again, I will breeze through the documentation to get the detailed how it works info I am looking for. Thanks Kindly, N. On 4/12/13, jg webaccou...@jgoettgens.de wrote: What do you mean with servers? A simple proxy, or a B2BUA (aka Asterisk)? Depending on the basic configuration the server might or might not have to deal with some or all of the RTP-streams. The already mentioned company Sangoma usually has good documentation about their products (see http://www.sangoma.com/products/d500-400-2000-sessions/ or http://www.sangoma.com/products/d150-30-400-sessions/) and examples on how to use them. jg -- _ -- 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] [OpenSIPS-Users] 404 When BYE initiated by external callee
On Tue, Apr 9, 2013 at 1:22 PM, Bogdan-Andrei Iancu bog...@opensips.orgwrote: ** Hi Nick, The BYE is not properly formed and rejected by script - in the 200 OK of the INVITE, you can see that your opensips is doing Record-Routing, but the BYE does not contain the corresponding Route hdr, so SIP routing is impossible. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developerhttp://www.opensips-solutions.com On 04/09/2013 08:05 PM, Nick Khamis wrote: Hello Everyone, I saw an earlier post about this issue: http://www.mail-archive.com/users@lists.opensips.org/msg23052.html And was wondering if there was anything we can do on our end to fix this problem? It seems that providers are not obligated to maintain RR? When the caller (internal) initiates the BYE everything is ok, but not the case when the callee (external) initiates the BYE. 192.168.2.5: OpenSIPS 192.168.2.10: Asterisk 70.10.163.44: Public IP 108.59.2.133: Service Provider U 2013/04/09 12:17:02.920454 192.168.2.10:5060 - 192.168.2.5:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.2.5;branch=z9hG4bKac2e.554c6e93.0;received=192.168.2.5;rport=5060. Via: SIP/2.0/UDP 192.168.2.11:5060 ;rport=5060;received=192.168.2.11;branch=z9hG4bK42f3f16e7BC15DF1. Record-Route: sip:192.168.2.5;lr;did=392.62562fb2. From: 1001 sip:1...@server.example.com;tag=FCA0BFC0-B585477D. To: sip:15178342...@server.example.com;user=phone;tag=as0a76fcde. Call-ID: 595ad334-f06e97fa-3bbc8137@192.168.2.11. CSeq: 1 INVITE. Server: Asterisk PBX UNKNOWN__and_probably_unsupported. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Contact: sip:15178342008@192.168.2.10:5060. Content-Type: application/sdp. Content-Length: 312. . v=0. o=root 1860889533 1860889534 IN IP4 192.168.2.10. s=Asterisk PBX UNKNOWN__and_probably_unsupported. c=IN IP4 192.168.2.10. t=0 0. m=audio 60646 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. ACC: transaction answered: timestamp=1365524222;method=INVITE;from_tag=FCA0BFC0-B585477D;to_tag=as0a76fcde;call_id= 595ad334-f06e97fa-3bbc8137@192.168.2.11;code=200;reason=OK U 2013/04/09 12:17:02.939608 192.168.2.5:5060 - 192.168.2.11:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.2.11:5060 ;rport=5060;received=192.168.2.11;branch=z9hG4bK42f3f16e7BC15DF1. Record-Route: sip:192.168.2.5;lr;did=392.62562fb2. From: 1001 sip:1...@server.example.com;tag=FCA0BFC0-B585477D. To: sip:15178342...@server.example.com;user=phone;tag=as0a76fcde. Call-ID: 595ad334-f06e97fa-3bbc8137@192.168.2.11. CSeq: 1 INVITE. Server: Asterisk PBX UNKNOWN__and_probably_unsupported. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Contact: sip:15178342008@192.168.2.10:5060. Content-Type: application/sdp. Content-Length: 329. . v=0. o=root 1860889533 1860889534 IN IP4 192.168.2.10. s=Asterisk PBX UNKNOWN__and_probably_unsupported. c=IN IP4 192.168.2.5. t=0 0. m=audio 31148 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. a=nortpproxy:yes. U 2013/04/09 12:17:06.988918 108.59.2.133:5060 - 192.168.2.5:5060 BYE sip:1001@70.10.163.44:5060 SIP/2.0. Max-Forwards: 64. To: 1001 sip:1001@70.10.163.44;tag=as4b40d9b4. From: sip:001110215178342...@sbc.voxbeam.com;tag=3574513019-870807. Reason: Q.850;cause=16;text=. Call-ID: 705605f129adbf5a38b5a0ff72de8f39@70.10.163.44:5060. CSeq: 2 BYE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Via: SIP/2.0/UDP 108.59.2.133;branch=z9hG4bK2deb.8bfd0b06.0. Contact: sip:callee@108.59.2.133;did=e9e.a6618961. Allow-Events: as-feature-event. Allow-Events: call-info. Allow-Events: presence. Allow-Events: line-seize. Allow-Events: dialog. Allow-Events: refer. Allow-Events: message-summary. Content-Length: 0. . Forcing RPORT: sip:001110215178342...@sbc.voxbeam.com U 2013/04/09 12:17:06.989421 192.168.2.5:5060 - 108.59.2.133:5060 SIP/2.0 404 Not here. To: 1001 sip:1001@70.10.163.44;tag=as4b40d9b4. From: sip:001110215178342...@sbc.voxbeam.com;tag=3574513019-870807. Call-ID: 705605f129adbf5a38b5a0ff72de8f39@70.10.163.44:5060. CSeq: 2 BYE. Via: SIP/2.0/UDP 108.59.2.133;received=108.59.2.133;rport=5060;branch=z9hG4bK2deb.8bfd0b06.0. Content-Length: 0. Or is asterisk the culprit? Looking at the forwarded INVITE (on the asterisk server), I see that the RR has been re-written, as opposed to appended when contacting the provider: U 2013/04/09 12:52:52.109611 192.168.2.10:5060 - 108.59.2.133:5060 INVITE sip:001110215178342...@sbc.voxbeam.com SIP/2.0. Via: SIP/2.0/UDP 70.10.163.44:5060;branch=z9hG4bK75a764b9;rport. Max-Forwards: 70. From: 1001 sip
[asterisk-users] Asterisk Peaking and 91 Calls And not a Dime More!
Hello Everyone, We are running some torcher tests on our * box using SIPP. The overall idea of the test is to contact asterisk and play a g729 encoded recording. On the asterisk side, we are initiating the echo app for the contacted extension, simulating a two way conversation. For some reason we cannot get past *91* calls on every test, with a lot of resources left: *top* top - 14:28:45 up 1 day, 1:45, 2 users, load average: 1.09, 0.80, 0.59 Tasks: 56 total, 1 running, 55 sleeping, 0 stopped, 0 zombie %Cpu(s): 7.6 us, 8.5 sy, 0.0 ni, 82.7 id, 0.0 wa, 0.0 hi, 1.2 si, 0.0 st KiB Mem: 3825108 total, 164480 used, 3660628 free,16324 buffers KiB Swap: 2097148 total,0 used, 2097148 free,97404 cached PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND 7229 root 20 0 70400 25m 5808 S 35.5 0.7 7:29.26 asterisk *iftop* Press H or ? for help 1.91Mb 3.81Mb5.72Mb 7.63Mb 9.54Mb └┴─┴───┴┴ test.example.com = 192.168.2.100 1.75Mb 1.75Mb 1.71Mb = 1.70Mb 1.70Mb 1.66Mb test.example.com = db.example.com 37.3Kb 37.3Kb 36.5Kb = 10.1Kb 10.1Kb 9.87Kb TX: cumm: 8.28MB peak: 1.79Mb rates: 1.79Mb 1.79Mb 1.74Mb RX: 7.93MB 1.72Mb 1.71Mb 1.71Mb 1.67Mb TOTAL: 16.2MB 3.51Mb 3.50Mb 3.50Mb 3.41Mb The SIPP Results -- Scenario Screen [1-9]: Change Screen -- Call-rate(length) Port Total-time Total-calls Remote-host 10.0(0 ms)/1.000s 50602089.21 s20802 192.168.2.10:5060(UDP) 0 new calls during 0.000 s period 0 ms scheduler resolution 0 calls (limit 100)Peak was 91 calls, after 9 s 0 Running, 332 Paused, 0 Woken up 0 dead call msg (discarded)0 out-of-call msg (discarded) 1 open sockets Messages Retrans Timeout Unexpected-Msg INVITE -- 20802 0 0 100 -- 20802 0 0 0 180 -- 0 0 0 0 200 -- E-RTD1 20802 0 0 0 ACK -- 20802 0 [ NOP ] Pause [ 8000ms] 20802 0 [ NOP ] Pause [ 1000ms] 20802 0 BYE -- 20802 0 0 200 -- 20802 0 0 0 -- Test Terminated - Statistics Screen --- [1-9]: Change Screen -- Start Time | 2013-04-09 14:08:07:797 1365530887.797642 Last Reset Time| 2013-04-09 14:42:57:025 1365532977.025339 Current Time | 2013-04-09 14:42:57:025 1365532977.025537 -+---+-- Counter Name | Periodic value| Cumulative value -+---+-- Elapsed Time | 00:00:00:000 | 00:34:49:227 Call Rate |0.000 cps |9.957 cps -+---+-- Incoming call created |0 |0 OutGoing call created |0 |20802 Total Call created | |20802 Current Call |0 | -+---+-- Successful call|0 |20802 Failed call|0 |0 -+---+-- Response Time 1| 00:00:00:000 | 00:00:00:003 Call Length| 00:00:00:000 | 00:00:09:010 -- Test Terminated Can we clear OS and * bottlenecks down into the different parts: OS - Simple commands such as ulimit etc... Asterisk - Startup directives that will increase whatever (i.e., allocated memory, -p value) before addressing hardware resources? 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
Re: [asterisk-users] [OpenSIPS-Users] 404 When BYE initiated by external callee
On Tue, Apr 9, 2013 at 2:31 PM, Joshua Colp jc...@digium.com wrote: Nick Khamis wrote: Is our asterisk server not relaying the RR along with the INVITE? If so, can we configure the PBX to do so using one of it's variables? * Mailing list CC'ed in this email... Asterisk is not a SIP proxy, it does not forward or relay INVITEs. It is a back to back user agent. Each leg is individual. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA 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-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users Hey Joshua, It was a poor choice of words on my part. What I meant to say was whether the problem was due to our asterisk configuration re-writing the RR when initiating the INVITE to our SIP trunk provider. Not sure if you had looked at the SIP trace included in the original email? If not I can resend it. Thanks in Advance, 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] [OpenSIPS-Users] 404 When BYE initiated by external callee
On Tue, Apr 9, 2013 at 3:04 PM, Joshua Colp jc...@digium.com wrote: Nick Khamis wrote: Hey Joshua, It was a poor choice of words on my part. What I meant to say was whether the problem was due to our asterisk configuration re-writing the RR when initiating the INVITE to our SIP trunk provider. Not sure if you had looked at the SIP trace included in the original email? If not I can resend it. I saw, but my response stands. Asterisk does not rewrite anything. The outgoing leg to your SIP trunk is completely separate, it is not a forwarded/modified INVITE. With the information you have available I don't think Asterisk is the problem here. The traces also illustrate this, the BYE in the trace is from a completely different call than the other messages. (You can see by looking at the Call-ID). -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA 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-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users Hello Joshua, Thanks again for your response. I can understand how * does not rewrite anything. When you mention the difference in call id, are you referring to: UA - OpenSIPS - Asterisk (Internal) Call-ID: 595ad334-f06e97fa-3bbc8137@192.168.2.11. Asterisk (Internal) - SIP Trunk (External) Call-ID: 5a5fb47111cadd6146746c4446a1790c@70.10.163.44:5060. SIP Trunk (External) BYE - OpenSIPS (Internal) Call-ID: 705605f129adbf5a38b5a0ff72de8f39@70.10.163.44:5060. The call id was changed twice Could this be a two part problem? N. -- _ -- 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] [OpenSIPS-Users] 404 When BYE initiated by external callee
On Tue, Apr 9, 2013 at 3:22 PM, Joshua Colp jc...@digium.com wrote: Nick Khamis wrote: Hello Joshua, Thanks again for your response. I can understand how * does not rewrite anything. When you mention the difference in call id, are you referring to: UA - OpenSIPS - Asterisk (Internal) Call-ID: 595ad334-f06e97fa-3bbc8137@**192.168.2.11595ad334-f06e97fa-3bbc8137@192.168.2.11 mailto:595ad334-f06e97fa-**3bbc8137@192.168.2.11595ad334-f06e97fa-3bbc8137@192.168.2.11 . Asterisk (Internal) - SIP Trunk (External) Call-ID: 5a5fb47111cadd6146746c4446a179**0c@70.10.163.44:5060http://5a5fb47111cadd6146746c4446a1790c@70.10.163.44:5060 http://**5a5fb47111cadd6146746c4446a179**0c@70.10.163.44:5060/http://5a5fb47111cadd6146746c4446a1790c@70.10.163.44:5060/ . SIP Trunk (External) BYE - OpenSIPS (Internal) Call-ID: 705605f129adbf5a38b5a0ff72de8f**39@70.10.163.44:5060http://705605f129adbf5a38b5a0ff72de8f39@70.10.163.44:5060 http://**705605f129adbf5a38b5a0ff72de8f**39@70.10.163.44:5060/http://705605f129adbf5a38b5a0ff72de8f39@70.10.163.44:5060/ . The call id was changed twice Could this be a two part problem? Yes. Until you can isolate it more it's all just a guess but it still doesn't seem like a problem with Asterisk itself. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA 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-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users Hello Joshua, Thanks again for your response. I re-ran the test, following a trace on the same call: 192.168.2.11 - UA 192.168.2.5 - OpenSIPS Server 192.168.2.10 - Asterisk Server 108.59.2.133 - SIP Trunk U 2013/04/09 15:44:00.549096 192.168.2.11:5060 - 192.168.2.5:5060 INVITE sip:15178392...@proxy.example.com:5060;user=phone SIP/2.0. Call-ID: ccc1a3e7-bcfc28f1-ed2257c4@192.168.2.11. U 2013/04/09 15:43:24.325964 192.168.2.5:5060 - 192.168.2.10:5060 INVITE sip:1517839...@asterisk.example.com:5060;user=phone SIP/2.0. Call-ID: ccc1a3e7-bcfc28f1-ed2257c4@192.168.2.11. U 2013/04/09 15:43:24.349274 192.168.2.10:5060 - 192.168.2.5:5060 SIP/2.0 100 Trying. Call-ID: ccc1a3e7-bcfc28f1-ed2257c4@192.168.2.11. U 2013/04/09 15:43:24.396204 192.168.2.10:5060 - 108.59.2.133:5060 INVITE sip:001110215178392...@sbc.voxbeam.com SIP/2.0. Call-ID: 58f65c9822f75d5a3da2992c0047c069@70.12.128.44:5060. 2013/04/09 15:44:15.086928 108.59.2.133:5060 - 192.168.2.5:5060 BYE sip:1001@70.12.168.99:5060 SIP/2.0. Call-ID: 58f65c9822f75d5a3da2992c0047c069@70.12.128.44:5060. U 2013/04/09 15:44:15.087277 192.168.2.5:5060 - 108.59.2.133:5060 SIP/2.0 404 Not here. Call-ID: 58f65c9822f75d5a3da2992c0047c069@70.12.168.99:5060. As I see asterisk rewrites the callid unexpectedly when initiating the INVITE with the SIP trunk (trace packet 4). In the same trace packet 4, the Record-Route Record-Route: sip:192.168.2.5;lr;did=7ea.60b64711. has also been removed. I am sure this is a configuration issue on our part/end, and was wondering how others with proxy--asterisk integrations addressed the issue. We can: 1) Rule out the provider as the source of the problem when it comes to the changing of the callid 2) Relay the non loose route BYE from our proxy to asterisk, which has record of the new callid. Not sure if this is a safe idea, or will even work? What is interesting to mention is the Session Progress: U 2013/04/09 15:43:32.211016 108.59.2.133:5060 - 192.168.2.10:5060 SIP/2.0 183 Session Progress. Call-ID: 58f65c9822f75d5a3da2992c0047c069@70.12.128.44:5060. U 2013/04/09 15:43:32.214127 192.168.2.10:5060 - 192.168.2.5:5060 SIP/2.0 183 Session Progress. Call-ID: ccc1a3e7-bcfc28f1-ed2257c4@192.168.2.11. Asterisk has mapped the call with the two different ids together. Any 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] Asterisk Peaking and 91 Calls And not a Dime More!
That's just it! Nothing! It just does not pass the 91 mark. There are no failed calls during the test: Successful call|0 |20802 Failed call|0 |0 It's locked on 91 calls. I think I have a channel limit or call limit thing set somewhere by accident? N. On 4/9/13, Paul Belanger paul.belan...@polybeacon.com wrote: On 13-04-09 02:49 PM, Nick Khamis wrote: Hello Everyone, We are running some torcher tests on our * box using SIPP. The overall idea of the test is to contact asterisk and play a g729 encoded recording. On the asterisk side, we are initiating the echo app for the contacted extension, simulating a two way conversation. For some reason we cannot get past *91* calls on every test, with a lot of resources left: *top* top - 14:28:45 up 1 day, 1:45, 2 users, load average: 1.09, 0.80, 0.59 Tasks: 56 total, 1 running, 55 sleeping, 0 stopped, 0 zombie %Cpu(s): 7.6 us, 8.5 sy, 0.0 ni, 82.7 id, 0.0 wa, 0.0 hi, 1.2 si, 0.0 st KiB Mem: 3825108 total, 164480 used, 3660628 free,16324 buffers KiB Swap: 2097148 total,0 used, 2097148 free,97404 cached PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND 7229 root 20 0 70400 25m 5808 S 35.5 0.7 7:29.26 asterisk *iftop* Press H or ? for help 1.91Mb 3.81Mb5.72Mb 7.63Mb 9.54Mb └┴─┴───┴┴ test.example.com = 192.168.2.100 1.75Mb 1.75Mb 1.71Mb = 1.70Mb 1.70Mb 1.66Mb test.example.com = db.example.com 37.3Kb 37.3Kb 36.5Kb = 10.1Kb 10.1Kb 9.87Kb TX: cumm: 8.28MB peak: 1.79Mb rates: 1.79Mb 1.79Mb 1.74Mb RX: 7.93MB 1.72Mb 1.71Mb 1.71Mb 1.67Mb TOTAL: 16.2MB 3.51Mb 3.50Mb 3.50Mb 3.41Mb The SIPP Results -- Scenario Screen [1-9]: Change Screen -- Call-rate(length) Port Total-time Total-calls Remote-host 10.0(0 ms)/1.000s 50602089.21 s20802 192.168.2.10:5060(UDP) 0 new calls during 0.000 s period 0 ms scheduler resolution 0 calls (limit 100)Peak was 91 calls, after 9 s 0 Running, 332 Paused, 0 Woken up 0 dead call msg (discarded)0 out-of-call msg (discarded) 1 open sockets Messages Retrans Timeout Unexpected-Msg INVITE -- 20802 0 0 100 -- 20802 0 0 0 180 -- 0 0 0 0 200 -- E-RTD1 20802 0 0 0 ACK -- 20802 0 [ NOP ] Pause [ 8000ms] 20802 0 [ NOP ] Pause [ 1000ms] 20802 0 BYE -- 20802 0 0 200 -- 20802 0 0 0 -- Test Terminated - Statistics Screen --- [1-9]: Change Screen -- Start Time | 2013-04-09 14:08:07:797 1365530887.797642 Last Reset Time| 2013-04-09 14:42:57:025 1365532977.025339 Current Time | 2013-04-09 14:42:57:025 1365532977.025537 -+---+-- Counter Name | Periodic value| Cumulative value -+---+-- Elapsed Time | 00:00:00:000 | 00:34:49:227 Call Rate |0.000 cps |9.957 cps -+---+-- Incoming call created |0 |0 OutGoing call created |0 |20802 Total Call created | |20802 Current Call |0 | -+---+-- Successful call|0 |20802 Failed call|0 |0 -+---+-- Response Time 1| 00:00:00:000 | 00:00:00:003 Call Length| 00:00:00:000 | 00:00:09
Re: [asterisk-users] Asterisk Peaking and 91 Calls And not a Dime More!
Hello Marie, Increasing the rate got us up 2 folds, Thank you so much for your help. We have a clustered asterisk setup, and it seems like 200 concurrent calls at 70% cpu is how we can keep these machine humming comfortably. Kind Regards, Nick On 4/9/13, Marie Fischer ma...@vtl.ee wrote: On 09.04.2013, at 23:43, Nick Khamis sym...@gmail.com wrote: That's just it! Nothing! It just does not pass the 91 mark. There are no failed calls during the test: Successful call|0 |20802 Failed call|0 |0 It's locked on 91 calls. I think I have a channel limit or call limit thing set somewhere by accident? The SIPP Results -- Scenario Screen [1-9]: Change Screen -- Call-rate(length) Port Total-time Total-calls Remote-host 10.0(0 ms)/1.000s 50602089.21 s20802 192.168.2.10:5060(UDP) So you have total calls = 20802. Does this number grow over time? 0 new calls during 0.000 s period 0 ms scheduler resolution 0 calls (limit 100)Peak was 91 calls, after 9 s IIRC, peak shows maximum concurrent calls. What command line do you use to start SIPP? I see your call rate is 10 calls/sec and maximum calls set to 100. Have you tried experimenting with increasing the call rate (-r command line parameter)? How long is the recording you are playing or have you set a call length for SIPP (-d command line option) - that is, how long are your calls? SIPP generates just as many calls as specified - if you have 10 calls per sec, it's quite logical to have ~90 ongoing calls after 9 secs. If your recording is about 9 secs, then the first calls will end at that time and you will never have more than ~90 concurrent calls. -- marie -- _ -- 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] Asterisk Peaking and 91 Calls And not a Dime More!
Hello Steve, Thank you so much for your response, we are testing each working nodes separately. These are computers the size of your palm that run on an average of 10 watts. With more's law in mind, how far can we push? I just kept the load under 80% cpu consumptions? Can we increase exponentially till something starts clunking and pinging? What I am asking is what is the general rule of thumb when performing such tests? Thanks in Advance, Nick. On 4/9/13, Steve Edwards asterisk@sedwards.com wrote: On Tue, 9 Apr 2013, Nick Khamis wrote: We have a clustered asterisk setup, and it seems like 200 concurrent calls at 70% cpu is how we can keep these machine humming comfortably. 200 channels on a single 7 year old server running CentOS 4.x and Asterisk 1.2.x would not be exceptional. I would expect more from a cluster of an unknown number of more modern servers running more modern software. -- 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 -- _ -- 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] Dedicated LCR Solutions
Hello Everyone, Was wondering what some of you for stand alone LCR implementations. I am aware of the LCR module within asterisk and a2billing however, we are looking for a standalone self less coupled solution. Not sure if such thing exist. Kind of like CDR Tool but for LCR... Thanks in Advance, 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] Optimizing Asterisk Environment
Hello Guys, Thank you so much for your response. We reran the sipp test: ./sipp -sf uac_pcap.xml -s 1001 vancouver.example.com -l 250 -trace_err -mp 3 -d 1 The scenario is the standard contact asterisk play some rtp media. On the asterisk, the echo test was used for the extension. This simulating a two way audio test. With ulimit set ulimit -n 65535, and while the test was running: # top PIDPR NI VIRT RES SHR S %CPU %MEMTIME+ COMMAND 16056 20 0 67568 25m 5812 S36 0.7 0:36.90asterisk #iftop (nice tool by the way :) vancouver.test.com = 192.168.2.100 1.75Mb1.75Mb 1.75Mb = 1.69Mb 1.70Mb 1.70Mb # free -m total used free sharedbuffers cached Mem: 3735518 3217 0 30438 -/+ buffers/cache: 48 3686 Swap: 2047 0 2047 # uptime 10:55:09 up 2 days, 1:45, 1 user, load average: 0.44, 0.46, 0.23 #ifconfig UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1 RX packets:13222747 errors:0 dropped:0 overruns:1 frame:1 TX packets:62311814 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:1000 # vmstat procs ---memory-- ---swap-- -io -system-- cpu r b swpd free buff cache si sobibo in cs us sy id wa 0 0 0 3293204 31824 45042400 1 1 931 1 1 98 0 # dmesg | grep -i duplex [ 14.622293] e100 :00:02.0: eth3: NIC Link is Up 100 Mbps Full Duplex We are running this on a test server (x330) just to help us with the dimensioning process for now. The important results from SIPP: Call-rate(length) Port Total-time Total-calls Remote-host 10.0(1 ms)/1.000s 5060 654.01 s 6450 192.168.2.10:5060(UDP) 0 calls (limit 250)Peak was 91 calls, after 9 s Elapsed Time |00:10:54:030 Call Rate |9.862 cps Successful call |0 | 6450 Failed call|0 |0 Is it safe to say that our test router is a lemon? Not sure if that's the bottleneck at this moment. Since only 36% of CPU is being utilized, and only 0.7% of memory. Are there any setting I should double check to run asterisk in full capacity. Thank you so much for your help, Nick. On 3/24/13, Steve Edwards asterisk@sedwards.com wrote: On Sat, Mar 23, 2013 at 09:33:38AM -0400, Nick Khamis wrote: We are getting some rather poor results (relative) with our Asterisk setup. On Sun, 24 Mar 2013, Tzafrir Cohen wrote: Run the system in full capacity and provide us some data. For starters: free -m uptime vmstat ethtool - make sure the interfaces are set correctly - look for 'Speed: 1000Mb/s' and 'Duplex: Full' ifconfig - look at the error counters iftop - how many bits are you pushing in each direction I've got a 7 year old Xeon box with 2GB running Asterisk 1.2 that handles 300 channels just fine. I suspect a modern box with a modern Asterisk could do that in 'sleep mode.' -- 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 -- _ -- 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] Optimizing Asterisk Environment
On the network side, Joshua had mentioned some network based encoding solutions. We would be sold on this. What are some of the routers out there that provide this capability with descent throughput? We were considering Cisco 3745 with a NM-HDV2 which transcoding from G.711 to G.729 handles 96 sessions. Not sure if this was the best bang for our buck? N. On 3/25/13, Nick Khamis sym...@gmail.com wrote: Hello Guys, Thank you so much for your response. We reran the sipp test: ./sipp -sf uac_pcap.xml -s 1001 vancouver.example.com -l 250 -trace_err -mp 3 -d 1 The scenario is the standard contact asterisk play some rtp media. On the asterisk, the echo test was used for the extension. This simulating a two way audio test. With ulimit set ulimit -n 65535, and while the test was running: # top PIDPR NI VIRT RES SHR S %CPU %MEMTIME+ COMMAND 16056 20 0 67568 25m 5812 S36 0.7 0:36.90asterisk #iftop (nice tool by the way :) vancouver.test.com = 192.168.2.100 1.75Mb1.75Mb 1.75Mb = 1.69Mb 1.70Mb 1.70Mb # free -m total used free sharedbuffers cached Mem: 3735518 3217 0 30438 -/+ buffers/cache: 48 3686 Swap: 2047 0 2047 # uptime 10:55:09 up 2 days, 1:45, 1 user, load average: 0.44, 0.46, 0.23 #ifconfig UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1 RX packets:13222747 errors:0 dropped:0 overruns:1 frame:1 TX packets:62311814 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:1000 # vmstat procs ---memory-- ---swap-- -io -system-- cpu r b swpd free buff cache si sobibo in cs us sy id wa 0 0 0 3293204 31824 45042400 1 1 931 1 1 98 0 # dmesg | grep -i duplex [ 14.622293] e100 :00:02.0: eth3: NIC Link is Up 100 Mbps Full Duplex We are running this on a test server (x330) just to help us with the dimensioning process for now. The important results from SIPP: Call-rate(length) Port Total-time Total-calls Remote-host 10.0(1 ms)/1.000s 5060 654.01 s 6450 192.168.2.10:5060(UDP) 0 calls (limit 250)Peak was 91 calls, after 9 s Elapsed Time |00:10:54:030 Call Rate |9.862 cps Successful call |0 | 6450 Failed call|0 |0 Is it safe to say that our test router is a lemon? Not sure if that's the bottleneck at this moment. Since only 36% of CPU is being utilized, and only 0.7% of memory. Are there any setting I should double check to run asterisk in full capacity. Thank you so much for your help, Nick. On 3/24/13, Steve Edwards asterisk@sedwards.com wrote: On Sat, Mar 23, 2013 at 09:33:38AM -0400, Nick Khamis wrote: We are getting some rather poor results (relative) with our Asterisk setup. On Sun, 24 Mar 2013, Tzafrir Cohen wrote: Run the system in full capacity and provide us some data. For starters: free -m uptime vmstat ethtool - make sure the interfaces are set correctly - look for 'Speed: 1000Mb/s' and 'Duplex: Full' ifconfig - look at the error counters iftop - how many bits are you pushing in each direction I've got a 7 year old Xeon box with 2GB running Asterisk 1.2 that handles 300 channels just fine. I suspect a modern box with a modern Asterisk could do that in 'sleep mode.' -- 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 -- _ -- 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] Using type=friend a mistake?
Hello Everyone, Just looking to secure our * box, and stumbled on the following This advice may run counter to the majority of documentation, sample files and examples shown on the voip-info.org site and on Asterisk forums, but you’ll have to take my word for it – using “type=friend” is a big mistake! It will make your Asterisk server much more vulnerable because “type=friend” actually causes two objects to be created – a SIP peer and a SIP user. This gives the potential hacker two entrance doors into your PBX, one of which has comparatively weak security. The problem is that a “user” is allowed to connect from any remote IP address, not just the address specified in the host parameter. Even if you want to allow connections from any address, it is much better to use “host=dynamic” than to use “type=friend”., http://kb.smartvox.co.uk/asterisk/secure-asterisk-pbx-part-2/ Is this true? Before I update all my type to peer, what are some of the things we needs to keep in mind when using friend vs. peer from a security standpoint? Thanks in Advance, 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
[asterisk-users] Optimizing Asterisk Environment
Hello Everyone, We are getting some rather poor results (relative) with our Asterisk setup. Not sure if we are using the sipp correctly etc.. but nevertheless, is there any documentation that describes how we can get the most our of our Asterisk box. For example when we hit the too many file error, and fixing it using ulimit. Also, is there any way we can allocate sufficient memory to our Asterisk instance when starting the PBX. An up to date and in-depth tutorial that covers this would be great. A quick search yielded pretty motivating success stories, but no little to no description on how to achieve them. Kind Regards, 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] Optimizing Asterisk Environment
Oh no secret. Some things I do is increase the ulimit size. I was wondering if there was a way to increase allocated memory. I have been reading about a -p option but when I start asterisk using asterisk -p -10 it does not accept it but asterisk -p 10 works fine. Not sure if that was the intended new value. Also, I just want to mention I am not trying to break any records. Just would like to get a ~200 concurrent call stable environment using G729 out of our setup. Thanks in Advance, 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] Optimizing Asterisk Environment
Hello Gentlemen, Thank you so much for your responses. We have been working on a SIP/RTP Proxy+Asterisk in backed by MySQL for a few weeks. Everything is working nicely I am pleased to say. And will be making some donations for G729 licenses etc.. (it's the least we can do to support the cause). Speaking about transcoding cards. We are functioning 100% on SIP using u/alaw and eventually G729. Some typical observations being great performance when not using G729 :)... Is there any transcoding happening when using only G729 and no other codec? We tried disallow=all and allow=g729 and judging by the CPU load 260% there seems to be... I hope this is not a silly question, but if we force the DID reseller to send only G729 encoded media, our asterisk server only allows G729, and finally for termination most sip trunk providers have g729 in there list of supported codecs, would there still be transcoding happening on our * box? I hope this is not as silly question as I think To answer your question, we also tried with only ulaw and alaw and we seem to be stuck on exactly 101 peak. Is there a limit setting hidden in one of the *.conf files? We let sipp run for almost 3 hours on our box, from another local computer using the following command: extensions.conf exten = 1002,1,Answer exten = 1002,n,Goto(demo,s,1) exten = 1002,n,Hangup ./sipp -sn uac -d 1 -s 1002 test.example.com -l 200 -mp 5606: And we got the following results: http://pastebin.com/J0YCprCb At 9.4 cps 96963 calls were executed with 0 failed calls. Where is the concurrent call figure in this tool? Please forgive me still getting use to it :). In regards to hardware transcoding cards for SIP protocol. Please let us know of some digium solutions. Again, we would love to support the cause. Nick. On 3/23/13, Andrew Latham lath...@gmail.com wrote: On Sat, Mar 23, 2013 at 12:06 PM, Joshua Colp jc...@digium.com wrote: Nick Khamis wrote: Oh no secret. Some things I do is increase the ulimit size. I was wondering if there was a way to increase allocated memory. I have been reading about a -p option but when I start asterisk using asterisk -p -10 it does not accept it but asterisk -p 10 works fine. Not sure if that was the intended new value. Also, I just want to mention I am not trying to break any records. Just would like to get a ~200 concurrent call stable environment using G729 out of our setup. Are you transcoding? If so then that is where most of your CPU is going, and the only option to make it go further is to use a hardware transcoding solution. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org +1 on hardware card. There are various other tools, even a network based encoding solution. Offloading to hardware can show you how stable/strong your system might already be. -- ~ Andrew lathama Latham lath...@gmail.com http://lathama.net ~ -- _ -- 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] Optimizing Asterisk Environment
Hello guys, no we do not do any recording of any kind. It was my assumption that processing media in g729 requires some sort of transcoding on the box? N. -- _ -- 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] Self Contained Least Cost Routing Solution
Hello Everyone, We are aware of a2billing and it's LCR functionality. I just wanted to know what other solutions you may be using. Maybe a tool that is a self contained module (ie a2billinig - Asterisk - LCR Tool - Trunk). Is there such a tool? It should be open source as is all good software. Kind Regards, Nick Khamis -- _ -- 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] Allow/Disallow
Hello Everyone, I have disallow=all and allow=g729 set in sip.conf however, it seems that asterisk still thinks it support other codecs: Capabilities: us - 0x8008000e (gsm|ulaw|alaw|h263|testlaw). How can I disable gsm,ulaw,alaw. Thanks in Advance, 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] Allow/Disallow
Hello Asghar, I fixed the issue after I realized that I was specifying allow before disallow. Sorry for the noise!!! Nick. On 3/21/13, Asghar Mohammad asghar...@gmail.com wrote: please post sip.conf. On Thu, Mar 21, 2013 at 8:01 PM, Nick Khamis sym...@gmail.com wrote: Hello Everyone, I have disallow=all and allow=g729 set in sip.conf however, it seems that asterisk still thinks it support other codecs: Capabilities: us - 0x8008000e (gsm|ulaw|alaw|h263|testlaw). How can I disable gsm,ulaw,alaw. Thanks in Advance, 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 -- _ -- 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] Register Free Opensips/Asterisk Integration
Hello Everyone, I have gone through a few really good tutorials from the OpenSIPS site, Asterisk resources etc.. The unanswered question (and final piece of our puzzle) is if it's possible to have a register free environment in an OpenSIPS/Asterisk integration. Most approaches have OpenSIPS relay the UA's REGISTER request to Asterisk which has host=dynamic set for the Friend/Peer and everything works as expected. Where I run into problems is in Inbound calls. When I try to call the extension from a DID I am receiving Unable to create channel of type 'SIP' (cause 20 - Unknown). And rightfully so! Reason being: SIP Show Peers Yields: Name/username HostDynForcerport ACL Port Status Realtime 1001/1001 192.168.2.5N 5060 UNREACHABLE Cached RT TTrunk/sip.exp.com 192.168.2.5N 5060 UNKNOWN Cached RT As for who will keep track of the UA location, the OpenSIPS `location` table has the correct info: select username,domain,contact,socket from location; +--+++--+ | username | domain | contact| socket | +--+++--+ | 1001 | sip.exp.com | sip:1001@192.168.2.11:5060 | udp:192.168.2.5:5060 | +--+++--+ OpenSIPS: sip.exp.com OpenSIPS: 192.168.2.5 Asterisk: 192.168.2.10 UA: 192.168.2.11 I have set `host=sip.exp.com' for the UA but the UA is still `UNREACHABLE` by asterisk As for the rest of the media related stuff, everything works perfectly. Outbound works fine. As you know, this only poses a problem with inbound calls to the UAs. 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
[asterisk-users] Malicious traffic comming from 37.75.210.90
Hello Osama, and Hisham, At 1330GMT there was some malicious activity coming from your network IP 37.75.210.90. Please act accordingly. Things that may be of use 972599779558 N. -- _ -- 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
Hello Ishfaq, and Isrlgb, The canreinvite value for UA friend entries are set to no, and for the OpenSIPS peer entry it's set to yes. I do have esternip and localnet cid set in sip.conf. I did not want to start a new email, but part of my problem right now is that OpenSIPS is in charge of performing the AUTH and REGISTER. This is fine for peers with static host definition, but not for the dynamic ones. Is it possible to have the fullcontact realtime info in sip_buddies populated upon initial INVITE? This is my first problem right now. After that will come RTP, and Codec issues... PS I have seen fullcontact info get populated with the correctly in the past, just can't get it to do it every time Thanks for your help!!! Nick. On 1/4/13, isr...@gmail.com isr...@gmail.com wrote: Did you set externip and localnet in your sip conf ? -- _ -- 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] 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
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
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
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] Deleting an inadvertent message
Hello Everyone, Is there any way we can delete the following message sent to asterisk ml, instead of the actual user please? I appologize for the inconvenience however, my personal info is in the email. http://markmail.org/message/gwhg4trnw4wei74k Thanks in Advance!! 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] asterisk distributions
Tom you're killing me with the me's please! -- _ -- 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] India Telecom regulations
How can we get thise license? Who do we have to pay. Nick. On Tue, Dec 20, 2011 at 9:52 AM, khalid touati khalidtou...@gmail.com wrote: Thank you Raj, I hope it will soon require no license as I heard there is a project to change this law, for now I believe I will recommend our office in India to go for license (to bridge to PSTN). Thanks once more for your help! 2011/12/19 Raj Mathur (राज माथुर) r...@linux-delhi.org On Tuesday 20 Dec 2011, khalid touati wrote: Thank you Raj, so with VOIP license calls can go beyond our pbx to PSTN (india), right, if so this what i needed to know to call Indian cellphone from US (or other countries) If your objective is to originate calls in the US (using whatever technology), route them over SIP and then terminate them to the PSTN in India, then yes: your Indian presence would need a VoIP licence. Similarly for the reverse: originate a call from Indian PSTN to your local office here and route it using VoIP to any destination (whether within India or abroad). A licence is required in that case too. In general, interconnection of two different entities by bridging Indian PSTN with any other technology requires a licence. If you're only doing VoIP-VoIP, or PSTN-PSTN, or bridging an Indian VoIP call to PSTN outside India then it's permitted in principle. This is why, e.g., Skype is permitted: it doesn't connect to the Indian PSTN at any stage. Once again, IANAL and TINLA. This is purely from my (mostly informed) understanding of the current laws. Regards, -- Raj -- Raj Mathur || r...@kandalaya.org || GPG: http://otheronepercent.blogspot.com || http://kandalaya.org || CC68 It is the mind that moves || http://schizoid.in || D17F -- _ -- 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 -- Khalid Touati Network Administrator at Endosoft, LLC CCNA -- _ -- 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] India Telecom regulations
SIP in India is illegal. Nick. On Mon, Dec 19, 2011 at 3:06 PM, khalid touati khalidtou...@gmail.com wrote: Hi All, Because I am pretty sure we have people in this DL from India, I was hoping to get the 100% accurate information, is it legal to make calls from any coutry to Indian mobile phones through an Asterisk server based in India? -- Khalid Touati Network Administrator -- _ -- 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] Struggling with Extensions in Realtime
Hello Everyone, Can someone please let me know what the correct way to deal with extensions for a particular user using asterisk reatime. For a user 1001, we would like to support: Local Calls: 123-456-7890 LD Calls: 1-123-456-7890 INT Calls: 011-64-03-123-456-7890 PBX EXT:1002 Do I need to insert multiple records for use 1001, each pointing to the different extensions in the extensions table (i.e., local-context, ld-context, int-context, and pbx-context)? Thanks in Advance, 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
[asterisk-users] Contexts and Extensions
Hello Everyone, For inbound, I am trying to specify a specific context. Everything works fine using the IP address, however with domain name it's not working at all. I tried changing the: Via: SIP/2.0/UDP test.com, and the Record-Route: sip:test.com;lr;did=a1a.4d23bae4 If I have a peer with the host, fromdomain, and outboundprxy set as the IP address the correct context is found context-from-test, but not using the domain name test.com. Asterisk still knows that the call is coming from IP address: chan_sip.c:22081 handle_request_invite: Call from '' (192.168.2.102:5060) to extension '1001' rejected because extension not found in context 'internal'. SIP Trace: --- SIP read from UDP:192.168.2.102:5060 --- INVITE sip:1...@test.com:5060 SIP/2.0 Record-Route: sip:test.com;lr;did=a1a.4d23bae4 Via: SIP/2.0/UDP test.com;branch=z9hG4bK9e83.1b0fcd74.0 Via: SIP/2.0/UDP 208.44.220.234:5060;received=208.44.220.234;branch=z9hG4bK6d6940f3;rport=5060 From: Mike Peer sip:16058293047@208.44.220.234;tag=as62765da7 To: sip:1001@170.12.90.130 Contact: sip:16058293047@208.44.220.234 Call-ID: 0f920dff6eefb6bd70b48d73676be593@208.44.220.234 CSeq: 102 INVITE User-Agent: DiDXsuPErTecSIP5 Max-Forwards: 69 Remote-Party-ID: Mike Peer sip:16058293047@208.44.220.234;privacy=off;screen=no Date: Fri, 16 Dec 2011 04:15:10 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 382 --- Reliably Transmitting (no NAT) to 192.168.2.102:5060 --- SIP/2.0 404 Not Found Via: SIP/2.0/UDP test.com;branch=z9hG4bK9e83.1b0fcd74.0;received=192.168.2.102 Via: SIP/2.0/UDP 208.44.220.234:5060;received=208.44.220.234;branch=z9hG4bK6d6940f3;rport=5060 From: Mike Peer sip:16058293047@208.44.220.234;tag=as62765da7 To: sip:1001@170.12.90.130;tag=as51f932b5 Call-ID: 0f920dff6eefb6bd70b48d73676be593@208.44.220.234 CSeq: 102 INVITE Server: Asterisk PBX UNKNOWN__and_probably_unsupported Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 I am using OpenSIPS and changed the following: advertised_address=test.com record_route_preset(test.com); Again, if I create a peer, and set the host, fromdomain, and outboundprxy as 192.168.2.102, and everything woks fine, but I would like to use the domain name example.com. Thanks in Advance, 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] Simple Generic IVR to get us up an running Quick
That's too funny! What are some tricks to make it sound professional. What I mean is, what are some of the typcial things people do to the recording, to make it sound kind-of robotic? I have no idea how to explain it. Maybe those of you that have done ivr recordings for corporations could share what tools and tricks you use to get that professional look? Thanks in Advance, Nick. On Tue, Dec 6, 2011 at 5:07 AM, Hans Witvliet aster...@a-domani.nl wrote: On Sat, 2011-12-03 at 15:36 -0500, Nick Khamis wrote: Hello Everyone, Are there any descent generic IVR recordings, that we can use to quickly get our PBX up and running? It will obviously not include the company name. It's easy enough to make your own recordings. Word of caution though. It might be advisable to ask somebody outside the company to record the phrases, Wonder why? At home i did it my self, and i still hear people stating that they have been talking at me, totaly unaware that it was just the voicemail anouncements. Peope just hear a voice, but seldom listen. And not just 90-old aunts, But people from helpdesks and even CEO's. Sometimes i wonder, if i should ask/test the callers I.Q. , And adapt the IVR's accordingly ;=) hw -- _ -- 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] Simple Generic IVR to get us up an running Quick
What I would like to know is. How could you have possibly known I have a 90 year old aunt?!?! Sorry for the Noise! Merry Christmas/Happy Holidays, Nick. On Tue, Dec 6, 2011 at 9:29 AM, Danny Nicholas da...@debsinc.com wrote: There are some Allison Smith Speaks blogs out there with good IVR hints. Some hints from my experience 1. The recommended volume adjustment for asterisk is -3 DB (that's -3 if you look at the wav in Audaciity). This will vary depending on your flavor of Asterisk and your input (SIP/DAHDI/etc). 2. Use a metronome - regular speech has a large variance in tempo. If you say leave your message after the tone normally you are more likely to get Nick's 90 year old aunt than leave-your-message-after-the-tone. 3. If you aren't going to buy a high quality microphone and software, you're just as well off recording using the normal record function. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis Sent: Tuesday, December 06, 2011 8:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Simple Generic IVR to get us up an running Quick That's too funny! What are some tricks to make it sound professional. What I mean is, what are some of the typcial things people do to the recording, to make it sound kind-of robotic? I have no idea how to explain it. Maybe those of you that have done ivr recordings for corporations could share what tools and tricks you use to get that professional look? Thanks in Advance, Nick. On Tue, Dec 6, 2011 at 5:07 AM, Hans Witvliet aster...@a-domani.nl wrote: On Sat, 2011-12-03 at 15:36 -0500, Nick Khamis wrote: Hello Everyone, Are there any descent generic IVR recordings, that we can use to quickly get our PBX up and running? It will obviously not include the company name. It's easy enough to make your own recordings. Word of caution though. It might be advisable to ask somebody outside the company to record the phrases, Wonder why? At home i did it my self, and i still hear people stating that they have been talking at me, totaly unaware that it was just the voicemail anouncements. Peope just hear a voice, but seldom listen. And not just 90-old aunts, But people from helpdesks and even CEO's. Sometimes i wonder, if i should ask/test the callers I.Q. , And adapt the IVR's accordingly ;=) hw -- _ -- 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 -- _ -- 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] Simple Generic IVR to get us up an running Quick
Hello Everyone, Are there any descent generic IVR recordings, that we can use to quickly get our PBX up and running? It will obviously not include the company name. A sexy female's voice always do well yeah? Cheers, Nicholas. -- _ -- 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] Can't get off Europe/Bucharest timezone
Hello Everyone, The timezone is set correctly on the OS America/Toronto: mv /etc/localtime /etc/localtime.bak cp /usr/share/zoneinfo/America/Toronto /etc/localtime I even tried adding the timezone setting to sip.conf: timezone=America/Toronto However. Asterisk wants to be in Bucharest? Thinking about it, I want to be in Bucharest! Cheers, 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] Can't get off Europe/Bucharest timezone
I'm so sorry, i'm so sorry, i'm so sorry! Good thing I did not have a chance yet to transfer it to mysql realtime. It was in extensions.conf. Thanks for Everything, Nick. On Thu, Dec 1, 2011 at 3:41 PM, Danny Nicholas da...@debsinc.com wrote: Assuming it's nothing quirky in some mysql or odbc, I would do - grep Europe /etc/asterisk/* -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis Sent: Thursday, December 01, 2011 2:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Can't get off Europe/Bucharest timezone Hello Everyone, The timezone is set correctly on the OS America/Toronto: mv /etc/localtime /etc/localtime.bak cp /usr/share/zoneinfo/America/Toronto /etc/localtime I even tried adding the timezone setting to sip.conf: timezone=America/Toronto However. Asterisk wants to be in Bucharest? Thinking about it, I want to be in Bucharest! Cheers, 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 -- _ -- 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] Does Asterisk alter the Headers of INVITE Message
You want to talk SIP, you need to talk SIP proxy. Hint: http://www.kamailio.org/w/ ;) Nick from Toronto. On Sun, Nov 27, 2011 at 5:19 PM, Alex Balashov abalas...@evaristesys.com wrote: On 11/27/2011 04:53 PM, Faraj Khasib wrote: I tried that with my SIP Cleint but the custom Header is not reaching the cleint ... Does the asketrisk delete that? Are you sure? Have you taken a packet capture to confirm? -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.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 -- _ -- 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] Polycom Phantom Ringing
Nick. On Fri, Nov 18, 2011 at 9:38 AM, Danny Nicholas da...@debsinc.com wrote: If your phones are being “hacked” you have a firewall problem. Your phones should only be registering to your local DHCP server and your Asterisk box. DHCP Server 192.X.X.X Asterisk Server 192.X.Y.Y Phone 192.X.Z.Z From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr Sent: Friday, November 18, 2011 8:34 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Polycom Phantom Ringing I have a Polycom Soundpoint IP335. There are no inbound routes set to the phones yet. However, the phones are getting phantom rings. What is the legitimacy of these calls? Is there something I need to block to stop it? I believe its people trying to hack the phones/phone system but I cannot find where I read that before. Thanks, --E -- _ -- 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 I don't think that's it. I have had the Polycom on my test site ring unexpectadly. It could be a bug in asterisk. Are you using 1.8? 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] Polycom Phantom Ringing
Sorry, in my last email I mentioned a bug in *, but I meant a bug in Poly's firmware. The phone we have in our test lab is an IP301 Nick On Fri, Nov 18, 2011 at 10:28 AM, eherr email.eherr9...@gmail.com wrote: They do. It shows up asterisk on the physical phone. Nothing in the raw cdr file on the server. --E -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Friday, November 18, 2011 10:16 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Polycom Phantom Ringing I have Polycom 501's and they keep a log of all calls. I would expect the 335's to have that capability as well. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr Sent: Friday, November 18, 2011 8:50 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Polycom Phantom Ringing Well this is a remote site. I am running 1.4.26 I have multiple polycoms that do not experience this. They are getting dhcp from their local router. I am wondering if it could either be a bug in the polycom firmware or something like a probe into the phone. I am pretty sure I read that phantom rings are sip calls to a phone where they are probing for extensions or something; cant remember. --E -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis Sent: Friday, November 18, 2011 9:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom Phantom Ringing Nick. On Fri, Nov 18, 2011 at 9:38 AM, Danny Nicholas da...@debsinc.com wrote: If your phones are being “hacked” you have a firewall problem. Your phones should only be registering to your local DHCP server and your Asterisk box. DHCP Server 192.X.X.X Asterisk Server 192.X.Y.Y Phone 192.X.Z.Z From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr Sent: Friday, November 18, 2011 8:34 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Polycom Phantom Ringing I have a Polycom Soundpoint IP335. There are no inbound routes set to the phones yet. However, the phones are getting phantom rings. What is the legitimacy of these calls? Is there something I need to block to stop it? I believe its people trying to hack the phones/phone system but I cannot find where I read that before. Thanks, --E -- _ -- 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 I don't think that's it. I have had the Polycom on my test site ring unexpectadly. It could be a bug in asterisk. Are you using 1.8? 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 -- _ -- 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New
Re: [asterisk-users] Becoming a CLEC
The ride is over before it even began A local ILEC here in Canada, is already offering Unlimited World service. And this on a Tier 1 network, not the crap we're use to doing business on. Choose a different angle before you get anymore grey hairs on that head... http://www.bell.ca/Home_phone/Products/Unlimited_World?INT=HP_hpldpg_BAN_flatfee_Mass_20101110_cb_on_en 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] Becoming a CLEC
Hahah! Yeah it does doesn't it? What do we do? How do we stay a float, It almost seems like the ILECs will drop their rates to a penny once the people in this, and Kamailio lists ;) actually put a dent in their underline. Nick On Mon, Nov 14, 2011 at 9:08 PM, Jeff LaCoursiere j...@sunfone.com wrote: On Mon, 2011-11-14 at 20:51 -0500, Nick Khamis wrote: The ride is over before it even began A local ILEC here in Canada, is already offering Unlimited World service. And this on a Tier 1 network, not the crap we're use to doing business on. Choose a different angle before you get anymore grey hairs on that head... http://www.bell.ca/Home_phone/Products/Unlimited_World?INT=HP_hpldpg_BAN_flatfee_Mass_20101110_cb_on_en The Unlimited service seems pretty limited to me. Vonage may even have more reach than this. 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 -- _ -- 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] Becoming a CLEC
Yeah! That is what I was thinking... Bringing Voice and Video under one umbrella, things like that... I actually come from a speech recognition and natural language processing background. Trying to build the voice network, and seeing how I can bring it all together. P.S. I started by getting acquainted with the proxies of course ;) Nick On Mon, Nov 14, 2011 at 9:42 PM, Alex Balashov abalas...@evaristesys.com wrote: Only through new, innovative applications. They will always deliver transport and dialtone cheaper than you. -- This message was painstakingly thumbed out on my mobile, so apologies for brevity, errors, and general sloppiness. Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ On Nov 14, 2011, at 9:15 PM, Nick Khamis sym...@gmail.com wrote: Hahah! Yeah it does doesn't it? What do we do? How do we stay a float, It almost seems like the ILECs will drop their rates to a penny once the people in this, and Kamailio lists ;) actually put a dent in their underline. Nick On Mon, Nov 14, 2011 at 9:08 PM, Jeff LaCoursiere j...@sunfone.com wrote: On Mon, 2011-11-14 at 20:51 -0500, Nick Khamis wrote: The ride is over before it even began A local ILEC here in Canada, is already offering Unlimited World service. And this on a Tier 1 network, not the crap we're use to doing business on. Choose a different angle before you get anymore grey hairs on that head... http://www.bell.ca/Home_phone/Products/Unlimited_World?INT=HP_hpldpg_BAN_flatfee_Mass_20101110_cb_on_en The Unlimited service seems pretty limited to me. Vonage may even have more reach than this. 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 -- _ -- 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] State of Asterisk+Virtualization+Timing
Hahah... I was waiting on the sideline for this question. Nick. On Wed, Nov 9, 2011 at 8:10 AM, Anton Kvashenkin anton.juga...@gmail.com wrote: Is anybody using pci-passthrough? 2011/11/9 Nick Khamis sym...@gmail.com Hans, Thank you so much for your response. We will be moving everything to VM soon. Cheers, Nick. On Tue, Nov 8, 2011 at 6:11 PM, Hans Witvliet aster...@a-domani.nl wrote: On Mon, 2011-11-07 at 11:45 -0500, Nick Khamis wrote: That sucks! What about KVM or XEN? Nick. No problems here with XEN. (Perhaps i should mention, that i use paravirtualsisation to get the best performance. Distro: mix of SLES11sp1 /open_11.4) hw -- _ -- 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 -- _ -- 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] State of Asterisk+Virtualization+Timing
Smart card? I think we should be leaning more towards the network devices? Cheers, Nick. On Wed, Nov 9, 2011 at 5:23 PM, Hans Witvliet aster...@a-domani.nl wrote: On Wed, 2011-11-09 at 16:10 +0300, Anton Kvashenkin wrote: Is anybody using pci-passthrough? Yes, though quite a while ago. About three years ago, i used pci-passthrough to give a dom-U access to a localy mounted smartcard. But i have a vague feeling that you are up to something else... I know that forwarding has been done for ethernet and even VGA-cards, the mere idea of forwarding a analogue or PRI card is quite something else: Timing is here essential.. hw -- _ -- 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] State of Asterisk+Virtualization+Timing
Hans, Thank you so much for your response. We will be moving everything to VM soon. Cheers, Nick. On Tue, Nov 8, 2011 at 6:11 PM, Hans Witvliet aster...@a-domani.nl wrote: On Mon, 2011-11-07 at 11:45 -0500, Nick Khamis wrote: That sucks! What about KVM or XEN? Nick. No problems here with XEN. (Perhaps i should mention, that i use paravirtualsisation to get the best performance. Distro: mix of SLES11sp1 /open_11.4) hw -- _ -- 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] State of Asterisk+Virtualization+Timing
Could you give a little more detail please? We have been running
asterisk on vmware for years as our test bed.
Nick.
On Mon, Nov 7, 2011 at 8:00 AM, Michelle Dupuis mdup...@ocg.ca wrote:
Although you say SIMPLE...not all virtualization hosts allow software
installation. On VMware the host has become an appliance you can't really
mess with...
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen
[tzafrir.co...@xorcom.com]
Sent: Monday, November 07, 2011 6:03 AM
To: Asterisk Users List
Subject: Re: [asterisk-users] State of Asterisk+Virtualization+Timing
On Sun, Nov 06, 2011 at 03:50:21PM +, Gordon Henderson wrote:
On Tue, 1 Nov 2011, Nic Colledge wrote:
Have you thought about using LXC rather than OpenVZ.
+1
There are a few references to allowing guest access to timing
hardware online.
Simples. Load up the dahdi modules in the host and all the
containers see it.
Be sure to also create /dev/dahdi/{ctl,timer,pseudo,channel} for the
container, as they're likely not be allowed to create device files.
--
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
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Re: [asterisk-users] State of Asterisk+Virtualization+Timing
That sucks! What about KVM or XEN? 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] State of Asterisk+Virtualization+Timing
Do you gents feel that KVM and XEN hog too much resources which in turn effects the functionality of Asterisk? I really like the idea of Asterisk as an appllicance, for reasons stated in this email. It just makes life all pretty and green. Cheers, 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] DID from Direct from Telco
Thank you guys for your response, One FXS port can only handle one call. A PRI T1 gateway can handle 23 call channels. A single T1 Data line with SIP can handle about 18 call channels running G711, 37 channels running g729 I just want to make sure that a T1 Gateway (capable of 23 call channels), plugged into an FXS port (capable of one call), is not a bottleneck. I.e., even though our network can handle upto 23 channels, we can only support 1 concurrent call becuase of the single FXS? What I am trying to figure out is what would I need to have the same capabilities as a company offering DIDs. Which mediant, and maybe a nice illustration? Thanks in Advance, 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] DID from Direct from Telco
I realized there was an error in my last post. I meant analog gateway plugged into and FXO port. DIDs must start somwhere. And I am under the impression that the telcos are the one that have control over that? Therefore, we would first need an analog gateway plugged into an FXO, before being able to go through the T1s and media servers? Your insight 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
