Re: [asterisk-users] Location
I thought the internet was entirely driven by the negative energy of its users. Monsters Inc was based on the Internet, wasn't it? Steve On 11/07/2009 08:31 AM, Thomas Perron wrote: I am trying to find others in my area. Have a sense of enjoyment instead of a negative attitude. On Fri, Nov 6, 2009 at 7:17 PM, Jai Rangi jpra...@gmail.com mailto:jpra...@gmail.com wrote: What kind of question is this. People are all over world. -Jai On Fri, Nov 6, 2009 at 4:02 PM, Thomas Perron thomas.per...@gmail.com mailto:thomas.per...@gmail.com wrote: Where is everyone located? I am in Washington DC. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Location
Italy Milan 2009/11/7 Thomas Perron thomas.per...@gmail.com Where is everyone located? I am in Washington DC. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Giancarlo Lombardo ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Difference between 'core show channels' and 'sip show channels' ??
vps*CLI iax2 show channels Channel Peer UsernameID (Lo/Rem) Seq (Tx/Rx) Lag Jitter JitBuf Format 0 active IAX channels vps*CLI core show channels Channel Location State Application(Data) 0 active channels 0 active calls vps*CLI sip show channels Peer User/ANRCall ID Seq (Tx/Rx) Format Hold Last Message ip_peer(None) 58139462bde 00101/20006 0x0 (nothing)No Rx: REGISTER 1 active SIP channel Core show channels shows 0 active channels. Sip show channels shows 1 active channel. I find it odd to have 1 active channels... Mostly there are 2, 4, 6 channels as a SIP-conversation consists of 2 channels : user1 asterisk user2 And then some minutes further : vps*CLI core show channels Channel Location State Application(Data) 0 active channels 0 active calls vps*CLI sip show channels Peer User/ANRCall ID Seq (Tx/Rx) Format Hold Last Message ip_peer 0a27b9c7d 37df36b96ba 04717/0 0x0 (nothing) No ip_peer (None) 24ddc4be2b2 00101/00119 0x0 (nothing)No Rx: REGISTER 2 active SIP channels Can someone enlighten this ? Jonas. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Location
Amsterdam, The Netherlands Ron giancarlo lombardo schreef: Italy Milan 2009/11/7 Thomas Perron thomas.per...@gmail.com mailto:thomas.per...@gmail.com Where is everyone located? I am in Washington DC. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Giancarlo Lombardo ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- NeoNova BV innovatieve internetoplossingen http://www.neonova.nl Science Park 140 1098 XG Amsterdam info: 020-5611300 servicedesk: 020-5611302 fax: 020-5611301 KvK Amsterdam 34151241 Op dit bericht is de volgende disclaimer van toepassing: http://www.neonova.nl/maildisclaimer ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Nov 7 TODAY Nov 22 - Join Global FreeSW GNU(Linux) HW Culture meeting via VOIP - BerkeleyTIP GlobalTIP - For Forwarding
CONTENTS: Meeting days/times Howto - Mark your calendar's dates;
Videos; Hot topics; Opportunities; Announcement Flyers; New webpages
=
Come join in with the Global Free SW HW Culture community at the
BerkeleyTIP/GlobalTIP meeting, via VOIP.
Two meetings this month:
Sat Nov 7, 12Noon - 3PM Pacific Time (=UTC-8)
Sun Nov 22, 12Noon - 3PM Pacific Time (=UTC-8)
Mark your calendars, 1st Sat, 3rd Sun every month.
{Note: 4th Sunday this November, to give 2 week spacing.}
Join online #berkeleytip on irc.freenode.net
we'll help you get your voip HW SW working:
http://sites.google.com/site/berkeleytip/remote-attendance
Or come to the FreeSpeech Cafe at UC Berkeley in person meeting.
Join the global mailing list
http://groups.google.com/group/BerkTIPGlobal
I hope to see you there. :)
= Talk Videos for November 2009:
Django Development - Richard Kiss, Eddy Mulyono, Glen Jarvis, Simeon
Franklin; BayPiggies
Python for scientific research, discussion with Guido van Rossum;
UCBSciPy
Netbooks - Michael Gorven, Dave Mackie, and Jonathan Carter; CLUG
Japan Linux Symposium Keynote, Linus Torvalds Jim Zemlin; Linux
Foundation
http://sites.google.com/site/berkeleytip/talk-videos
Download watch them before the meetings, discuss at the meetings.
Thanks to all the Speakers, Videographers, Groups! :)
[Record your local meeting! Put the video online, email me for
inclusion for next month. :) ]
= Hot topics: Ubuntu 9.10 - Problems? Fixes? Upgrade? Install?
Freeswitch VOIP server - setup for BTIP
Flyers outreach to UCBerkeley.
Outreach to other UC campuses next semester.
= Opportunities - Learn new, or increase your job skills, /or
volunteer help the community: Set up any of: a BTIP Mailing List, web
server/site, Freeswitch VOIP server, or Virtual Private Network SSL
= Announcement Flyers: Print Post them in your community. 4/5
available - Freedom, Karmic Koala, Free Culture, SciPy, OLPC. See bottom
of page: http://groups.google.com/group/BerkTIPGlobal
= New BTIP Webpages @ http://sites.google.com/site/berkeleytip/
UC Campus local groups; Free Hardware; System Administration;
Announcement Flyers; Opportunities
For Forwarding - You are invited to forward this announcement wherever
it would be appreciated.
___
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Re: [asterisk-users] Difference between 'core show channels' and 'sip show channels' ??
On Sat, 7 Nov 2009, jonas kellens wrote: vps*CLI iax2 show channels Channel Peer UsernameID (Lo/Rem) Seq (Tx/Rx) Lag Jitter JitBuf Format 0 active IAX channels vps*CLI core show channels Channel Location State Application(Data) 0 active channels 0 active calls vps*CLI sip show channels Peer User/ANRCall ID Seq (Tx/Rx) Format Hold Last Message ip_peer(None) 58139462bde 00101/20006 0x0 (nothing)No Rx: REGISTER 1 active SIP channel You caught an endpoint registering - that is a channel while the conversation to register is occurring. vps*CLI sip show channels Peer User/ANRCall ID Seq (Tx/Rx) Format Hold Last Message ip_peer 0a27b9c7d 37df36b96ba 04717/0 0x0 (nothing) No ip_peer (None) 24ddc4be2b2 00101/00119 0x0 (nothing)No Rx: REGISTER 2 active SIP channels One is a register again, not sure about the other. j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trouble registering Cisco 7942
On Sat, Nov 7, 2009 at 12:56 AM, Warren Selby wcse...@selbytech.com wrote: That typically means you've got an error in your phone specific config file, the SEP[MAC].cnf.xml. You need to login to the phone via ssh and use the log/log login. Once you've done that, look at the logs and see what line of the config is giving it grief. Once you know that, you'll know what's causing the Unprovisioned message. I set the username and password but am unable to log into the phone. I provided an updated config below. I am prompted for the username and password though. Secondly should I be using IP or hostnames for the proxy and processNodeName or does it not matter? Thanks device deviceProtocolSIP/deviceProtocol sshUserIdadmin/sshUserId sshPassword123/sshPassword devicePool callManagerGroup members member priority=0 callManager ports ethernetPhonePort2000/ethernetPhonePort sipPort5060/sipPort securedSipPort5061/securedSipPort /ports processNodeNameSIPSERVER/processNodeName /callManager /member /members /callManagerGroup /devicePool sipCallFeatures cnfJoinEnabledtrue/cnfJoinEnabled callForwardURIx--serviceuri-cfwdall/callForwardURI callPickupURIx-cisco-serviceuri-pickup/callPickupURI callPickupListURIx-cisco-serviceuri-opickup/callPickupListURI callPickupGroupURIx-cisco-serviceuri-gpickup/callPickupGroupURI meetMeServiceURIx-cisco-serviceuri-meetme/meetMeServiceURI abbreviatedDialURIx-cisco-serviceuri-abbrdial/abbreviatedDialURI rfc2543Holdfalse/rfc2543Hold callHoldRingback2/callHoldRingback localCfwdEnabletrue/localCfwdEnable semiAttendedTransfertrue/semiAttendedTransfer anonymousCallBlock2/anonymousCallBlock callerIdBlocking2/callerIdBlocking dndControl0/dndControl remoteCcEnabletrue/remoteCcEnable /sipCallFeatures natEnabledtrue/natEnabled natAddress172.16.2.1/natAddress phoneLabel102/phoneLabel sipLines line button=1 featureID9/featureID featureLabel102/featureLabel contact102/contact proxySIPSERVER/proxy port5060/port name102/name displayNameAtlas/displayName authName102/authName authPasswordPASS/authPassword sharedLinefalse/sharedLine /line /sipLines /device ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trouble registering Cisco 7942
I think your featureLabel definition is wrong. On the login issue, ssh to the ip of the phone and login first with the user/pass you defined in the file (admin/123), then at the second login prompt use log/log. That should get you the log files which will show you your error. Thanks, --Warren Selby On Nov 7, 2009, at 9:45 AM, Stephen Reese rsre...@gmail.com wrote: On Sat, Nov 7, 2009 at 12:56 AM, Warren Selby wcse...@selbytech.com wrote: That typically means you've got an error in your phone specific config file, the SEP[MAC].cnf.xml. You need to login to the phone via ssh and use the log/log login. Once you've done that, look at the logs and see what line of the config is giving it grief. Once you know that, you'll know what's causing the Unprovisioned message. I set the username and password but am unable to log into the phone. I provided an updated config below. I am prompted for the username and password though. Secondly should I be using IP or hostnames for the proxy and processNodeName or does it not matter? Thanks device deviceProtocolSIP/deviceProtocol sshUserIdadmin/sshUserId sshPassword123/sshPassword devicePool callManagerGroup members member priority=0 callManager ports ethernetPhonePort2000/ethernetPhonePort sipPort5060/sipPort securedSipPort5061/securedSipPort /ports processNodeNameSIPSERVER/processNodeName /callManager /member /members /callManagerGroup /devicePool sipCallFeatures cnfJoinEnabledtrue/cnfJoinEnabled callForwardURIx--serviceuri-cfwdall/callForwardURI callPickupURIx-cisco-serviceuri-pickup/callPickupURI callPickupListURIx-cisco-serviceuri-opickup/callPickupListURI callPickupGroupURIx-cisco-serviceuri-gpickup/callPickupGroupURI meetMeServiceURIx-cisco-serviceuri-meetme/meetMeServiceURI abbreviatedDialURIx-cisco-serviceuri-abbrdial/abbreviatedDialURI rfc2543Holdfalse/rfc2543Hold callHoldRingback2/callHoldRingback localCfwdEnabletrue/localCfwdEnable semiAttendedTransfertrue/semiAttendedTransfer anonymousCallBlock2/anonymousCallBlock callerIdBlocking2/callerIdBlocking dndControl0/dndControl remoteCcEnabletrue/remoteCcEnable /sipCallFeatures natEnabledtrue/natEnabled natAddress172.16.2.1/natAddress phoneLabel102/phoneLabel sipLines line button=1 featureID9/featureID featureLabel102/featureLabel contact102/contact proxySIPSERVER/proxy port5060/port name102/name displayNameAtlas/displayName authName102/authName authPasswordPASS/authPassword sharedLinefalse/sharedLine /line /sipLines /device ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] help in installing asterisk
hi all, i am installing asterisk. when i am compiling asterisk-1.4.26.3, i am getting errors of dependency. there are three tar files in asterisk.org (http://www.asterisk.org/downloads) can any one suggest me how to get rid of dependencies. n which asterisk version should i download and compile. thx Asterisk 1.6.1.9 Source Tarball Asterisk 1.6.0.17 Source Tarball Asterisk 1.4.26.3 Source Tarball ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Location
On Friday 06 November 2009 18:31:09 Thomas Perron wrote: I am trying to find others in my area. If that's what you're looking for, I'd suggest a service like meetup.com. Have a sense of enjoyment instead of a negative attitude. The problem is that there are literally thousands of people subscribed to this list, and your effort was, at best, misguided. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question about callerid?
On Nov 6, 2009, at 5:14 AM, John A. Sullivan III wrote: On Thu, 2009-11-05 at 20:57 -0800, Martin Joseph wrote: Hello again Asterisk people. I am running Asterisk 1.42 on an old PowerPC ibook. I have had this deployed for several years now, with pretty good results. Recently I added a callerid service to my landline (qwest). I am using the audiocodes MP114 2fxo/2fxs gateway, which is an outstanding piece of hardware once it's configured (lol). Anyhow, I can see that the gateway is passing caller id info to asterisk because the console will display something like: [Nov 4 13:01:19] NOTICE[32]: chan_sip.c:13362 handle_request_invite: Failed to authenticate user SEATTLE SCHOOLS sip:2062524...@89.89.89.253 ;tag=1c492497235 So the caller ID info is right there. However on my extensions (or softphones) the id shows as the extension # (ie 2003). Is there something I need to do to set the callerid? I can't seem to find this in the examples? Thanks in advance for helping with my (I am sure) stupid question... snip I'd like to understand this better myself as I know we don't have this right in our environment. I believe the reason you see that is because Asterisk is providing a B2BUA (I think it's called), i.e., your caller is not actually talking to your phone. Instead, your caller is talking to Asterisk on the inbound SIP ID (whatever that is) and then Asterisk is calling your phone from the extension in the dial plan. At least I think that's why the extension shows up in the callerID. OK, that makes sense. So since Asterisk is a back to to back user agent (ie the call is always going through it) then the Caller ID data isn't magically moved along... Still, the fact that it's showing up there in the console means there should be some way to grab it (the callerID data) and stuff into into the proper place for it to be passed along. I see that the callerid valiable can be set as per: http://www.voip-info.org/wiki/view/Setting+Callerid So that's nice, and the only question is how to I get the callerID info from where it show in the console as failed to authenticate? Either that, or I could reconfigure my audiocodes and my asterisk so that instead of incoming calls dialing my desired extension (ie 2020), asterisk could accept the calls from the domain of the audiocodes (ie it's IP address). Maybe that's how get the CID data. Don't really know, but suspect there are lots of people here who do? Thanks for any help in advance, Marty The identity can be overridden in sip.conf with the fromdomain and fromuser parameters. However, we found this introduced its own problems. I suppose we just need to build more sophisticated logic into our dialplan. The problem is, if we set the fromdomain/user, we now show correct sip sources when we make direct SIP calls and can return those calls from the phone's call history. However, it breaks all the internal dialing which wants to dial to the extension. If we remove fromdomain/user, the internal dialing works but public SIP calls now show the extension as the user rather than the user's public SIP ID. I'm sure as with most things in Asterisk, we can fix it if we just take the time to think through the programming logic. Hope this helps - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on a MiniITX board+Atom1.6 2gb+Sangoma USB?
On Fri, 6 Nov 2009 14:43:50 +, Veselin K wrote: Thank you Michael, Any advise on how to design my setup to avoid transcoding? Maybe: Incoming: PSTN -alaw- Asterisk -alaw- SIP Phone Outgoing: SIP Phone -alaw- Asterisk -alaw- IAX2 Provider Am I understanding this correctly? As long as the phone uses the same codec as the PSTN/IAX2 providers, then Asterisk should not need to transcode? Right. Keep it simple. Allow only alaw in your configs and use SIP phones set to prefer alaw. Michael Regards, Veselin K On Fri, Nov 06, 2009 at 06:43:51AM -0600, Michael Graves wrote: On Wed, 4 Nov 2009 16:44:02 +, vese...@campbell-lange.net wrote: Hello, does this sound as a good combination, mini-itx board with Atom dual core 1.6ghz 2G ram and a sangoma USB? For a setup with PSTN for incoming and IAX2(alaw/gsm) for outgoing calls. - Would you say its a good choice from a hardware perspective? - Roughly how many concurrent calls would one of these be able to handle? Probably as much as your bandwidth can handle. Check ont the voip wiki (http://www.voip-info.org) and use the search term dimensioning. You'll find lots of older references to systems running at 400 MHz - 1 GHz passing many calls as long as they don't transcode between codecs. I myself have a little FIT-PC2 that I'm starting to use for Asterisk. It's basically a netbook, like the hardware you describe, but tiny and very low power. Ideal for a small office or home office. Michael -- Michael Graves mgravesatmstvp.com http://www.mgraves.org o713-861-4005 c713-201-1262 sip:mgra...@mstvp.onsip.com skype mjgraves Twitter mjgraves ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Graves mgravesatmstvp.com http://www.mgraves.org o713-861-4005 c713-201-1262 sip:mgra...@mstvp.onsip.com skype mjgraves Twitter mjgraves ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [DAHDI 2.2.0.2] failed on channel 1: No such device or address
Hello No matter if I use the entry-level OpenVox A400 with a single FXO module or the most-often-garbage card from ww.x100p.com, I get the following error after just adding this card to Mini-ITX with a single PCI slot (OS = CentOS 5.4): # lspci -v 03:00.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device b100:0003 Flags: bus master, medium devsel, latency 64, IRQ 12 I/O ports at a000 [size=256] Memory at e200 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 # cat /proc/interrupts CPU0 CPU1 0: 409671 0IO-APIC-edge timer 1: 2 0IO-APIC-edge i8042 7: 0 0IO-APIC-edge parport0 8: 3 0IO-APIC-edge rtc 9: 0 0 IO-APIC-level acpi 14: 7380 0IO-APIC-edge ide0 15: 3406 0IO-APIC-edge ide1 169:122 0 IO-APIC-level uhci_hcd:usb5, HDA Intel 201: 0 0 IO-APIC-level ehci_hcd:usb1, uhci_hcd:usb2 209: 0 0 IO-APIC-level uhci_hcd:usb3 217: 0 0 IO-APIC-level uhci_hcd:usb4 225:333 0 PCI-MSI eth0 NMI: 0 0 LOC: 412563 419484 ERR: 0 MIS: 0 # cat /etc/dahdi/system.conf loadzone=fr defaultzone=fr fxsks=1 # dahdi_cfg -vv DAHDI Tools Version - 2.2.0 DAHDI Version: 2.2.0.2 Echo Canceller(s): Configuration --- Channel map: Channel 01: FXS Kewlstart (Default) (Echo Canceler: none) (Slaves: 01) 1 channels to configure. DAHDI_CHANCONFIG failed on channel 1: No such device or address (6) If someone's already seen this error message, any idea what the cause is, and what I could try to solve it? Thank you for any tip. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.6.1 + Cisco AS5300 + Fax T38 ?
Hi I have finished the installation of my VoIP basic configuration ... Actually: - All calls from my E1 are received by a Cisco AS5300 and sent to my Asterisk (in G711 by SIP). - All user are connected by SIP to the Asterisk - All calls from User are sent by asterisk to the Cisco AS5300 Now, i want see if i can supply T38 Fax Gateway I am search to: - Cisco Receive all Fax, two poss: he detect automatiqueley a Fax, or based on phone number. - Sent the fax in T38 to Asterisk for Fax Routing - Based on the number and extensions, Asterisk sent the T38 call to the T38 terminaison (other server) .. Anyone know if it's possible ? and if yes, what is the configuration process (on the cisco and on the Asterisk Gateway) Thanks Jérôme ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Help with concurrent VoIP calls
Hi. I'm having trouble figuring out why I'm not able to make many concurrent VoIP calls on my system. I'm not aiming for a huge number, because I have purposely bought a low powered system, but I would think that I could get more. Here are the details: I have a small-form-factor Asterisk server with an Intel Atom 230 CPU (1.6 GHz, 533 MHz FSB) and 512 MB DDR2 533. It is running Ubuntu Server 9.04 with the default Debian package manager installation of Asterisk. (version 1.4) Here is what is going on: I'm making outgoing calls (with .call files) via SIP (using Vitelity's service, if anyone wants to know) with about 55.0 ms latency between my Bellsouth DSL connection their servers. I'm using GSM-format prompts with GSM encoding (disallow=all, allow=gsm in sip.conf) and I'm able to make about 7 concurrent calls. I have a very fast internet connection, so there is still plenty of bandwidth, and the top command shows that Asterisk is only at about 5% CPU and 10% RAM. Even with only 7 calls, a landline phone will skip occcasionally, but cell phones have perfect quality. I don't think that 7 calls is very many, I'll be happy if I can get 10 good-sounding calls. Can anyone give suggestions? (If this has been hashed out elsewhere, I'm happy with a link to more information!) Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6.1 + Cisco AS5300 + Fax T38 ?
On Sat, Nov 7, 2009 at 2:18 PM, Phibee Network Operation Center n...@phibee.net wrote: I am search to: - Cisco Receive all Fax, two poss: he detect automatiqueley a Fax, or based on phone number. - Sent the fax in T38 to Asterisk for Fax Routing - Based on the number and extensions, Asterisk sent the T38 call to the T38 terminaison (other server) .. Anyone know if it's possible ? and if yes, what is the configuration process (on the cisco and on the Asterisk Gateway) I don't know how to do automatic fax detection on Cisco. It is probably possible. Sending T.38 to asterisk is supported by Cisco voice IOS and by asterisk, with the caveat that you should read the release notes for whatever IOS you have chosen. Pretty much every voice IOS release notes I've ever read has pointed out conditions where T.38 has known issues. Doing routing based on number dialed definitely works, and you tell Cisco to send those calls over T.38 with G.711 failover if T.38 doesn't negotiate properly. If you google Cisco T.38 dialplan you should see some examples. If you search the asterisk-users archives you'll see dial-peer samples involving T.38 faxing between Cisco and asterisk. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help with concurrent VoIP calls
On Sat, Nov 7, 2009 at 2:45 PM, John Timms johngti...@gmail.com wrote: I have a small-form-factor Asterisk server with an Intel Atom 230 CPU (1.6 GHz, 533 MHz FSB) and 512 MB DDR2 533. It is running Ubuntu Server 9.04 with the default Debian package manager installation of Asterisk. (version 1.4) What kind of NIC are you using and what's the network config? ie Bellsouth - router - switch - you Are you NAT'd? Where are your endpoints connected? (locally, outside?) I have a very fast internet connection, so there is still plenty of bandwidth what is the specs for fast? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help with concurrent VoIP calls
John Timms wrote: Hi. I'm having trouble figuring out why I'm not able to make many concurrent VoIP calls on my system. I'm not aiming for a huge number, because I have purposely bought a low powered system, but I would think that I could get more. Here are the details: I have a small-form-factor Asterisk server with an Intel Atom 230 CPU (1.6 GHz, 533 MHz FSB) and 512 MB DDR2 533. It is running Ubuntu Server 9.04 with the default Debian package manager installation of Asterisk. (version 1.4) Most of my installations are Soekris net5501's with 512MB ram and a 500mhz Geode LX processor. Unless Ubuntu is running a ton of extra junk in the background, that processor should be more than adequate. Here is what is going on: I'm making outgoing calls (with .call files) via SIP (using Vitelity's service, if anyone wants to know) with about 55.0 ms latency between my Bellsouth DSL connection their servers. I'm using GSM-format prompts with GSM encoding (disallow=all, allow=gsm in sip.conf) and I'm able to make about 7 concurrent calls. I have a very fast internet connection, so there is still plenty of bandwidth, and the top command shows that Asterisk is only at about 5% CPU and 10% RAM. Even with only 7 calls, a landline phone will skip occcasionally, but cell phones have perfect quality. Your only connection to the PSTN is via SIP right? Then this is likely coincidental that 'landline' calls are different than 'cell phone' calls. The ONLY possibility is that the problem is with your SIP termination provider, but even that is unlikely. As Fred pointed out your DSL connection is likely the cause. Do you have any traffic shaping on the network? If not, you really should have a firewall that's capable of prioritizing voice traffic over bulk data traffic. What is the actual down and up speed of your DSL connection? I don't think that 7 calls is very many, I'll be happy if I can get 10 good-sounding calls. Can anyone give suggestions? (If this has been hashed out elsewhere, I'm happy with a link to more information!) Use this calculator to see how much bandwidth 10 concurrent calls will take. http://www.asteriskguru.com/tools/bandwidth_calculator.php Darrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help with concurrent VoIP calls
If you've got a bellsouth dsl connection because of the way their system works even with doing qos on the link you can really only do about 8 calls before you start to run into problems with their setup. Tom -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Timms Sent: Saturday, November 07, 2009 2:45 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Help with concurrent VoIP calls Hi. I'm having trouble figuring out why I'm not able to make many concurrent VoIP calls on my system. I'm not aiming for a huge number, because I have purposely bought a low powered system, but I would think that I could get more. Here are the details: I have a small-form-factor Asterisk server with an Intel Atom 230 CPU (1.6 GHz, 533 MHz FSB) and 512 MB DDR2 533. It is running Ubuntu Server 9.04 with the default Debian package manager installation of Asterisk. (version 1.4) Here is what is going on: I'm making outgoing calls (with .call files) via SIP (using Vitelity's service, if anyone wants to know) with about 55.0 ms latency between my Bellsouth DSL connection their servers. I'm using GSM-format prompts with GSM encoding (disallow=all, allow=gsm in sip.conf) and I'm able to make about 7 concurrent calls. I have a very fast internet connection, so there is still plenty of bandwidth, and the top command shows that Asterisk is only at about 5% CPU and 10% RAM. Even with only 7 calls, a landline phone will skip occcasionally, but cell phones have perfect quality. I don't think that 7 calls is very many, I'll be happy if I can get 10 good-sounding calls. Can anyone give suggestions? (If this has been hashed out elsewhere, I'm happy with a link to more information!) Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help with concurrent VoIP calls
Hi Fred. The NIC chip is a Realtek RTL8101E, on the motherboard. Network is Bellsouth = modem/router = Asterisk Yes, I am using NAT (assuming you mean that the Asterisk server does not have its own public IP address) Endpoints are outside the network, just standard POTS phones. Vitelity is my SIP provider. By fast I mean the best Business DSL Bellsouth has to offer: Up to 6.0 Mbps downstream - Up to 512 Kbps upstream I've used iftop on my server while running calls, and I'm under 200 Kbps while my calls are running. John Timms On Sat, Nov 7, 2009 at 4:25 PM, Fred Posner f...@teamforrest.com wrote: On Sat, Nov 7, 2009 at 2:45 PM, John Timms johngti...@gmail.com wrote: I have a small-form-factor Asterisk server with an Intel Atom 230 CPU (1.6 GHz, 533 MHz FSB) and 512 MB DDR2 533. It is running Ubuntu Server 9.04 with the default Debian package manager installation of Asterisk. (version 1.4) What kind of NIC are you using and what's the network config? ie Bellsouth - router - switch - you Are you NAT'd? Where are your endpoints connected? (locally, outside?) I have a very fast internet connection, so there is still plenty of bandwidth what is the specs for fast? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Set DESTINATION CID for outbound calls
I am wondering if anyone knows of a way to do this, as it would be much more meaningful for our CDR reports. We use FreePBX under the Elastix distro. We are able to set the CALLER's CID on inbound calls by using the Asterisk Phonebook module in FreePBX, then configure the Inbound Route settings to use it for CID. I haven't seen anything like this to apply those same rules to the DESTINATION CID on outbound calls. Instead, the caller id just shows the 7-10 digit number that was dialed by the person who made the call. If anyone could tell me how I can do this, or point me in the direction, or tell me who else I should talk to, I'd really appreciate it. Take care. Thanks, - Doug Mortensen Network Consultant Impala Networks Inc CCNA, MCP, Security+ Linux+, Network+, A+ . www.impalanetworks.com http://www.impalanetworks.com P: (505) 327-7300 F: (505) 327-7545 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help with concurrent VoIP calls
On Sat, Nov 7, 2009 at 4:45 PM, John Timms johngti...@gmail.com wrote: Hi Fred. By fast I mean the best Business DSL Bellsouth has to offer: Up to 6.0 Mbps downstream - Up to 512 Kbps upstream If you're running the GSM codec, 7 calls will hit around 200 Kbps. If you're running ulaw, 7 calls will hit over your max. The calculator Darrick linked to is a great tool. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Text messaging
IVR question: Users dial my DID numbers and get connected to macros and other vectors that guide them to the appropriate context. Once connected to a specific context I would like to send a text message to their phone. Do I need a PERL script or is there something native in Asterisk 1.6 that can trigger a text to the endpoint? Thank you [default] ;include = stdexten include = big10-IVR include = cleveland-IVR exten = _1703XXX,1,Goto(big10-IVR,s,1) exten = _1517XXX,1,Goto(cleveland-IVR,s,1) [big10-IVR] exten = s,1,Answer() exten = s,n,Background(dir-welcome) ;exten = s,n,WaitExten(1) ;exten = s,n,Background(astcc-please-enter-your) ;exten = s,n,Background(zip-code) ;exten = s,n,Wait(7) exten = s,n,Background(washington-dc) ;exten = s,n,Authenticate(,a) ;exten = s,n,Background(pin-number-accepted) exten = s,n,Playback(queue-thankyou) exten = s,n,Background(ginger110109) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help with concurrent VoIP calls
By fast I mean the best Business DSL Bellsouth has to offer: Up to 6.0 Mbps downstream - Up to 512 Kbps upstream That almost sounds like an invitation to check out what business service your cableco offers. One thing to be aware of with DSL and cable modems is that there can be various ill effects as your line gets closer to its rated capacity; do not expect that you'll get a reliable 512Kbps upstream. VoIP is sensitive to loss, latency, and jitter. You may be able, for example, to only get 384Kbps reliably out of the link (before packet loss/jitter/etc wreck its suitability for VoIP). That's a good time to look seriously at a gateway package like pfSense that can prioritize certain classes of traffic while also limiting overall bandwidth. As an example, we noticed on the local business cable offering (2Mbps up) Shaped PL min avg max stddev 2.2M3 6.4 251 557 176 2.1M1 7.8 350 584 134 2.0M3 6.4 271 535 132 1.9M1 7 254 527 131 1.8M0 6 79 339 90 1.75M 0 5.9 14 92 11 1.7M0 5.4 13 77 10 1.65M 0 4.9 11 69 7 1.6M0 5.4 13 55 9 1.5M0 5.3 11 59 7 1.4M0 5 11 57 7 1.3M0 4.9 11 54 6 1.2M0 4.9 11 52 7 1.1M0 4.8 14 53 11 The max starts trending up after 1.6M (helps to graph it) and pretty much everything goes to hell after 1.75M. ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] how to check version of asterisk
hi all, i had installed asterisk under /etc. now i want to know by command which version of asterisk i had installed. how to know the version plz tell me. thx___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to check version of asterisk
asterisk -V Connecting to the CLI (asterisk -r) will produce a banner that will also tell you the version. aster...@opensourcesolution.in wrote: hi all, i had installed asterisk under /etc. now i want to know by command which version of asterisk i had installed. how to know the version plz tell me. thx ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to check version of asterisk
On Sun, Nov 08, 2009 at 06:20:46AM +, aster...@opensourcesolution.in wrote: hi all, i had installed asterisk under /etc. now i want to know by command which version of asterisk i had installed. how to know the version plz tell me. asterisk -V -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help with concurrent VoIP calls
How are you connecting your land line phones since this is where you have the problem? Also, I would not expect very many calls at the same time with that setup if each call takes 50K you can't get exactly the maximum anyway, maybe 80% of maximum. Hope this helps. Tom Moore tommym2...@gmail.com wrote: If you've got a bellsouth dsl connection because of the way their system works even with doing qos on the link you can really only do about 8 calls before you start to run into problems with their setup. Tom -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Timms Sent: Saturday, November 07, 2009 2:45 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Help with concurrent VoIP calls Hi. I'm having trouble figuring out why I'm not able to make many concurrent VoIP calls on my system. I'm not aiming for a huge number, because I have purposely bought a low powered system, but I would think that I could get more. Here are the details: I have a small-form-factor Asterisk server with an Intel Atom 230 CPU (1.6 GHz, 533 MHz FSB) and 512 MB DDR2 533. It is running Ubuntu Server 9.04 with the default Debian package manager installation of Asterisk. (version 1.4) Here is what is going on: I'm making outgoing calls (with .call files) via SIP (using Vitelity's service, if anyone wants to know) with about 55.0 ms latency between my Bellsouth DSL connection their servers. I'm using GSM-format prompts with GSM encoding (disallow=all, allow=gsm in sip.conf) and I'm able to make about 7 concurrent calls. I have a very fast internet connection, so there is still plenty of bandwidth, and the top command shows that Asterisk is only at about 5% CPU and 10% RAM. Even with only 7 calls, a landline phone will skip occcasionally, but cell phones have perfect quality. I don't think that 7 calls is very many, I'll be happy if I can get 10 good-sounding calls. Can anyone give suggestions? (If this has been hashed out elsewhere, I'm happy with a link to more information!) Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
