Re: RE : [Asterisk-Users] CDRTool
I have 4.5.4 CDRTool version. I patched my cdr_addon_mysql like this: cd ../asterisk-addons - Add a line into asterisk-addons/Makefile reading: CFLAGS+=-DMYSQL_LOGUNIQUEID - edit cdr_addon_mysql.c and replace the line reading AST_MUTEX_DEFINE_STATIC(mysql_lock); with static ast_mutex_t mysql_lock = AST_MUTEX_INITIALIZER; - change the asterisk table name from cdr to asterisk_cdr in cdr_addon_mysql.c chmod 644 cdr_addon_mysql.so cp cdr_addon_mysql.so /usr/lib/asterisk/modules/ restart Asterisk But when I make , I've got error like this: cdr_addon_mysql.c:61: error: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared here (not in a function) make: *** [cdr_addon_mysql.o] Error 1 rm app_saycountpl.o I had a similar problem and so ignored that patching suggestion. In my testing so far it doesn't seem to have caused a problem. You could post to the cdrtool-users list at freelist.org Cameron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CallerID is not displaying for my incoming calls
Hi,As you said, I have tested. But, still callerid is not displaying. On the console, Asterisk is giving below error: *CLI Aug 14 15:09:58 ERROR[27056]: callerid.c:276 callerid_feed: fsk_serie made mylen 0 (-8) Aug 14 15:09:58 WARNING[27056]: chan_zap.c:6087 ss_thread: CallerID feed failed: Success Aug 14 15:09:58 WARNING[27056]: chan_zap.c:6131 ss_thread: CallerID returned with error on channel 'Zap/1-1'Please tell me the solution. Looking forward to your response. Thanks Regards,Chandra.Ira [EMAIL PROTECTED] wrote: At 02:14 AM 8/14/2006, you wrote:We have installed Asterisk with Digium 04B card (4 FXO ports). Now, I have connected my PSTN line directly to first port. I am making outgoing calls and receiving incoming calls successfully through my Asterisk. The problem is: When I am receiving a call from outside (PSTN), I am not getting the callerid number and getting callerid as "Asterisk" in my softphones (XLite). Here I am giving my configuration files.[incoming]exten = s,1,wait(2)exten = s,n,Answerexten = s,n,SetMusicOnHold(default)exten = s,n,DigitTimeout,5exten = s,n,ResponseTimeout,10exten = s,n,Background(/tmp/virg2)exten = s,n,Goto(s,1)include = leaderWhat I have to do to display the PSTN caller number on my softphones? Please tell me. Looking forward to your response. Thank you.When I had this problem, adding a wait() in front of the answer cured the problem. I have the same TDM04 card and we get callerid no problem now.Ira ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail Beta.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OPENSER / SER and Asterisk
Absolutely. The SER/OpenSER documentation is terrible, and if you post to the OpenSER mailing list, you get very cryptic replies. ___ Whilst I would agree with you regarding SER, the documentation of OpenSER is far better. Documentation of Asterisk Realtime on the other hand. Now *that's* terrible. Cameron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: New Device
On 2006-08-15 13:10:05 -0700, Dovid Bender [EMAIL PROTECTED] said: This is a multi-part message in MIME format. I have spoken to some one who is interested in investing into building = equipment for asterisk. I am looking to find out what products that the = asterisk community would like to see be built. This can be products that = already exists but lack certain functionality as well as things that = arent out there but you would want to see it. Thanks. Dovid I think a good quality single port FXO (with or without and FXS) that is external (ie ethernet) would be a very successful product at the right price point. Basically all the products that pretend to this are garbage in my experience. Although the SPA3000 does seem to work well for some people, it's echo issues are legendary. Just my thought. Marty ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OPENSER / SER and Asterisk
Hi all Onsip.org is the best option for startup and openser has many more option integrating with Voice mail with Astrisks openser.org have lot of documentation Ram On 8/16/06, kjcsb [EMAIL PROTECTED] wrote: Absolutely. The SER/OpenSER documentation is terrible, and if you post tothe OpenSER mailing list, you get very cryptic replies. ___Whilst I would agree with you regarding SER, the documentation of OpenSER isfar better.Documentation of Asterisk Realtime on the other hand. Now *that's* terrible. Cameron___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CallerID is not displaying for my incoming calls
Hi,As you said, I have changed my configurations. But, callerid is not displaying. What I have to do? Please tell me.ThanksRegards,Chandra.Rich Adamson [EMAIL PROTECTED] wrote: Crazy Boy wrote: Hi Friends, We have installed Asterisk with Digium 04B card (4 FXO ports). Now, I have connected my PSTN line directly to first port. I am making outgoing calls and receiving incoming calls successfully through my Asterisk. The problem is: When I am receiving a call from outside (PSTN), I am not getting the callerid number and getting callerid as "Asterisk" in my softphones (XLite). Here I am giving my configuration files. zaptel.conf file contents: loadzone = us defaultzone=us fxsks=1-4 zapata.conf file contents: [channels] context=incoming signalling=fxs_ks busydetect=1 busycount=7 relaxdtmf=yes callwaiting=yes callwaitingcallerid=yes threewaycalling=yes cancallforward=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 callerid=asreceived language=en usecallerid=yes hidecallerid=no echocancel=yes transfer=yes immediate=no group=1 callgroup=9 pickupgroup=9 channel = 1The above entries appear to be reasonable and correct. If you have not properly set rxgain and txgain, it "could" impact callerid. If those gains are too high/low, asterisk will not properly read the callerid data when sent to you. extensions.conf file contents: [incoming] exten = s,1,Answer exten = s,2,SetMusicOnHold(default) exten = s,3,DigitTimeout,5 exten = s,4,ResponseTimeout,10 exten = s,5,Background(/tmp/virg2) exten = s,6,Goto(s,1) include = leader Got event 18 (Ring Begin)... Aug 14 14:11:58 WARNING[26744]: pbx.c:5869 pbx_builtin_dtimeout: DigitTimeout is deprecated, please use Set(TIMEOUT(digit)=timeout) instead. Aug 14 14:11:58 WARNING[26744]: pbx.c:5845 pbx_builtin_rtimeout: ResponseTimeout is deprecated, please use Set(TIMEOUT(response)=timeout) instead.The above two WARNING statements are telling you that either you are copying those exten= statements from someone's old config files, or, you haven't read the asterisk documentation. The message is telling you that your statement "exten = s,3,DigitTimeout,5" should be replaced with the Set(TIMEOUT(digit)=timeout) syntax. Your statements are still executing properly today, but the next time you upgrade asterisk code, they are likely to fail due to the old syntax not being supported.Try 'show function TIMEOUT' from your CLI and read it. What I have to do to display the PSTN caller number on my softphones? Please tell me. Looking forward to your response. Thank you.___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] capi (divas4linux) bearer setting
hi, we've got a Diva Server BRI-2M PCI, SN:3485 card in our asterisk server. we use the latest divas4linux-melware-3.0.g-106.628.1-1 driver for it. the card is connected to a Bosch Integral33 PBX. the two system connected with an S0 line in order the two pbx be able to call eachother. when we call from the bosch to asterisk everything is working properly. but when we call from the a x-ten soft phone client through asterisk to the bosch the it's not working. which means the asterisk pass the call to the bosch, bosch receive but don't ring the given number. after we debug the capi layer with bosch experts from bosch we found the while the bosch call asterisk it request SPEECH time bearer, but when asterisk call bosch it set bearer to MULTIUSE. i found it in ./divactrl/common/dbg_tapi.c LINE_BEARER_MODE__SPEECH, LINE_BEARER_MODE__MULTIUSE. so probably the problem is thet we (x-ten, asterisk, a divas4linux do not set the bearer to proper value. is this the real reason? how can i set the bearer to speech in divas4linux or in capi or in asterisk's capi or ...? thank you for your help in advance. yours. -- Levente Si vis pacem para bellum! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX unstable with large number of calls?
Hi Curt,That probably suggests that with SIP they're handing off the RTP to their upstream provider and just dealing with the signalling which is very low overhead. With IAX they have to transport both unless they're interconnecting upstream by IAX and can transfer. In my experience the load is about the same for IAX or SIP+RTP and limited [on a single box] by the specification of the box itself. Whilst I risk being shot down, I'd be wary of any provider who isn't themselves handling the RTP for a multitude of quality reasons (and just because that is what you're paying them for), as well as one who quotes capacities in single box terms. SimonOn 8/15/06, Curt Shaffer [EMAIL PROTECTED] wrote: I was just talking with an unnamed provider and the guy told me that they recommend their users not to use IAX because it is unstable at 50 concurrent calls and unusable at 100 or more calls. Now I have personally worked on an asterisk box that was pushing more than 50 and there were no problems. Anyone else out there have any data either for or against this suggestion? Thanks Curt ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RTP Stream not set up correct at outgoing call
Dear astreisk-users mailing list subscribers, with my asterisk under debian (Version: 1:1.0.9.dfsg-5) i have the problem, that the RTP stream is not set up correct with an outgoing call. Incoming calls are working with no problems. The problem is, that the RTP stream is initiated from IP A and my asterisk or my router/with firewall (ISA 2004) replies to the port the RTP stream comes from, but to the ip-adress it is talking SIP, not the IP A, from where the RTP stream is set up. For a better understaning please see http://www.ontae.net/tmp/graph_outgoing-call.JPG; or http://www.ontae.net/tmp/graph_outgoing-call.txt;. More infos can be found in the trace i took at my router/firewall at the external interface (myIP): http://www.ontae.net/tmp/outgoing-call.trace.sip.txt;. Please have a look at my problem and give me a feedback. Thanks in advance, ontae ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can budgetone 101 display name part of cid?
On Tuesday 15 August 2006 19:05, Doug Lytle wrote: Jessee J Holmes wrote: Doug, That is correct you can only display the number on the BudgetTone 101, 102, and 200. If you wish to display the name as well, you will need to upgrade to the GXP-2000 phone. I'm not, Guus is. Doug Ok, too bad it's not possible :( Thanks everybody who responded! Guus ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7960 password reset
Hi on my 7940 Phones here, this is the first Part of the Factory Reset Procedure after Step 3 and the Status Message you have to hit all Keys on the Number Pad (1 - 2 - 3 - 4 - #) and then answer the Question by hitting Number 2 Cu David Maxx Lobo schrieb: Fastest way (wipes everything out): 1. Power off the phone completely. 2. Hold down the # key, then power the phone on. 3. Continue holding the # key until the LCD gives you a status message. 4. Follow the prompts to do a full factory reset, which resets the password as well. --Maxx Ferguson, Michael wrote: G'Day List, I am trying, once again, to configure my 7960 to work with asterisk. Where abouts do I go to reset the password on the phone? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problems with outgoing calls on a TE410P
We make a call from a PBX through asterisk to ISDN (E1) Denmark. We use asterisk to record incoming and outgoing calls Sometimes we experience that a outgoing call doesnt get through. In the message.log we get WARNING[7594] app_dial.c: Unable to forward voice In the Master.csv we get source number but in the destination number we get a t and not the dialled number. Does Anyone have experience or ideas to solve this problem. Medvenlig hilsen Nils Rasmussen ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple registrations to the same asterisk server
Hi, I couldn't find in you sip.conf of your central server, the line context=clientes-sip, did u forget to past, or u r missing it, or i'm missunderstanding?That could be the problem! You MUST define the context for your ATA devices in central server, so * will look for this context in
extensions.conf, that's your dialplan.Hope it helps,Ps. Plse give me some feedbackOn 8/16/06, Juan Luis Moyano
[EMAIL PROTECTED] wrote:Marco Mouta escribió: Hi , Please post here your
extensions.conf in your central server only with that i can figured out or at least try to help u. Best regards, Marco Mouta On 8/15/06, * Juan Luis Moyano*
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi All, I have the following scenario: A central Asterisk server where
all the ATAs register themselves. This server runs Asterisk 1.2.5 and ATAs are SPA-2002. So far everything is OK. Now I have another location where I want to connect 4 analog phones. I thought setting up 2
SPA-2002 but as I already have a TDM400P card and I want to use it, I had configured asterisk 1.0.7 on the second machine. So far I can place calls from the second server to any extension on the central
server. But I cant get an ATA on the central server to reach an extension on the second server. Please help me solve this situation. Thanks in advance. Juan Luis Moyano
The configs are as follows: Central Server -- -sip.conf [40019] username=USER1 callerid=40019 type=friend
host=dynamic secret= mailbox=40019 accountcode=USER1 [40028] username=USER2 callerid=40028 type=friend host=dynamic
secret= mailbox=40028 accountcode=USER2 [4] username=USER3 callerid=4 type=friend host=dynamic secret=
mailbox=4 accountcode=USER3 [40023] username=USER4 callerid=40023 type=friend host=dynamic secret= mailbox=40023
accountcode=USER4 -extensions.conf[clientes-sip]exten = _4.,1,Macro(stdexten,SIP/${EXTEN},${EXTEN})[macro-stdexten]exten = s,1,Dial(${ARG1},30,Tr)exten = s,2,Voicemail(u${ARG2})
exten = s,3,Hangupexten = s,102,Voicemail(b${ARG2})exten = s,103,Hangup Second Server - -sip.conf register =
40019:[EMAIL PROTECTED]/40019 register = 40028:[EMAIL PROTECTED]/40028 register = 4:[EMAIL PROTECTED]/4 register = 40023:[EMAIL PROTECTED]/40023 [40019]
type=friend secret= username=40019 host=10.32.1.16 http://10.32.1.16 insecure=very
[4] type=friend secret= username=4 host=10.32.1.16 http://10.32.1.16
insecure=very [40028] type=friend secret= username=40028 host=10.32.1.16
http://10.32.1.16 insecure=very [40023] type=friend secret= username=40023 host= 10.32.1.16
http://10.32.1.16 insecure=very -extensions.conf [globals] USER1=Zap/2 USER2=Zap/3 USER3=Zap/4
USER4=Zap/5 [extensions] exten = 40019,1,Dial(${USER1}) exten = 40023,1,Dial(${USER2}) exten = 40028,1,Dial(${USER3}) exten = 4,1,Dial(${USER4})
[outbound] exten = _.,1,Dial(SIP/[EMAIL PROTECTED]) ___ --Bandwidth and Colocation provided by
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-- Com os melhores cumprimentos, Marco Mouta ___
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-- Com os melhores cumprimentos,Marco Mouta
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Re: Re[2]: [asterisk-users] Softphone for Windows Mobile 5?
http://www.electronicscience.com/ has a good IAX2 softphone called ESC SoftphoneOn 8/16/06, David Thomas [EMAIL PROTECTED] wrote:Sorry, poor reply. Yes I use it on WM5, and have not seen any problems. I admit I don'tuse it a lot, but it does seem to work fine.regards,DaveOn 8/15/06, David Thomas [EMAIL PROTECTED] wrote: Yes, use it on WM5. Dave On 8/15/06, Christian [EMAIL PROTECTED] wrote: Hello, Many thanks, but it seems only to be available for Windows Mobile 2003. Will it work on WM5? Many thanks, ChristianOn 2006-08-15 at 14:00 David Thomas wrote: Try SJphone, it works for me. http://www.sjlabs.com/sjp.htmlThe latency is a little too much over my EVDO cannection though. :) It does work great over wifi. regards, DaveOn 8/15/06, Christian [EMAIL PROTECTED] wrote: Hi all, Does anyone know a Softphone for Windows mobile 5? Want to connect to my Asterisk when I am away. Many thanks, Christian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: New Device
- Original Message - From: Martin Joseph [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, August 16, 2006 7:50 AM Subject: [asterisk-users] Re: New Device On 2006-08-15 13:10:05 -0700, Dovid Bender [EMAIL PROTECTED] said: This is a multi-part message in MIME format. I have spoken to some one who is interested in investing into building = equipment for asterisk. I am looking to find out what products that the = asterisk community would like to see be built. This can be products that = already exists but lack certain functionality as well as things that = arent out there but you would want to see it. Thanks. Dovid I think a good quality single port FXO (with or without and FXS) that is external (ie ethernet) would be a very successful product at the right price point. Basically all the products that pretend to this are garbage in my experience. Although the SPA3000 does seem to work well for some people, it's echo issues are legendary. Just my thought. Marty ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Yes a good quality FXO device would be great, as Marty said the SPA3000 is almost it.. Another device would be a GSM to SIP / IAX Harvey ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7960 password reset
If the phone already had the SIP image running. Check the SIPDefault.cnf file there may be a phone_password= string this is the phone's current password use it remember to change to number or uppercase if need be Ferguson, Michael wrote: Maxx, Thanks much for the feedback. I will check into it and follow up with your instructions. 'preciate it. Best wishes. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Maxx Lobo Sent: Tuesday, August 15, 2006 5:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7960 password reset What Cisco image is the phone running? If it is really old (lower than P0S030203) then yeah, this won't work. If you upgrade the phone to P0S30203 and from there on to P0S3-06-3-00, and then these instructions will work fine. This should be pretty straightforward using ATFTP and the Cisco images. In response to your other question, a factory reset TMK does not wipe out the SIP image. Just the settings. --Maxx Ferguson, Michael wrote: Maxx, That did not work. Any other ideas? Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Maxx Lobo Sent: Tuesday, August 15, 2006 4:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7960 password reset Fastest way (wipes everything out): 1. Power off the phone completely. 2. Hold down the # key, then power the phone on. 3. Continue holding the # key until the LCD gives you a status message. 4. Follow the prompts to do a full factory reset, which resets the password as well. --Maxx Ferguson, Michael wrote: G'Day List, I am trying, once again, to configure my 7960 to work with asterisk. Where abouts do I go to reset the password on the phone? Thanks - - -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] REQ: BATM gw-232 sip firmware
Hi, I would like to test them with asterisk and sip. Could anybody send me Telco systems BATM GW-232 sip firmware? Does anybody have experience with BATM products? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Support a malformed SIP INVITE
Hello all, We have a pathetic legacy PBX that produces the most terrible SIP INVITE packet. In the past we have found a phone that can hope and just used that. We now want to connect the legacy PBX to asterisk, and we're (well, I'm) having problems. This is the INVITE that's sent to the asterisk server (ip 192.168.0.240) - INVITE sip:PBX.400T-portal SIP/2.0 To: 01000:[EMAIL PROTECTED] From: :;tag=8af2812a Via: SIP/2.0/UDP 192.168.0.181;branch=z9hG4bK-d87543-24f99000ddafb002-1--d87543-;rport Call-ID: GSM-SIPCall-Number-1 CSeq: 1 INVITE Contact: PBX.400T-portalsip:192.168.0.181:5060 Max-Forwards: 70 Accept-Language: en Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE Content-Type: application/sdp Date: Sun, 16 Sep 2076 19:04:20 GMT upported: sip-cc, sip-cc-01, timer, replaces User-Agent: PBX.400T-portal Content-Length: 276 - Nasty, eh? At the moment asterisk just says it's not a SIP address and sends a 404. I have put the full trace of how asterisk respnds. Does anyone have any ideas on how to get asterisk to accept an INVITE like this? Thanks, Tom -- The full asterisk response to the INVITE: - INVITE sip:PBX.400T-portal SIP/2.0 To: 01000:[EMAIL PROTECTED] From: :;tag=8af2812a Via: SIP/2.0/UDP 192.168.0.181;branch=z9hG4bK-d87543-24f99000ddafb002-1--d87543-;rport Call-ID: GSM-SIPCall-Number-1 CSeq: 1 INVITE Contact: PBX.400T-portalsip:192.168.0.181:5060 Max-Forwards: 70 Accept-Language: en Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE Content-Type: application/sdp Date: Sun, 16 Sep 2076 19:04:20 GMT upported: sip-cc, sip-cc-01, timer, replaces User-Agent: PBX.400T-portal Content-Length: 276 v=0 o=gsm-sip-portal 1 1 IN IP4 192.168.0.181 s=gsm-sip-voice-call c=IN IP4 192.168.0.181 t=0 0 m=audio 8000 RTP/AVP 3 0 8 13 101 a=fmtp:101 0-16 a=rtpmap:3 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:13 CN/8000 a=rtpmap:101 telephone-event/8000 --- (15 headers 12 lines)--- Using INVITE request as basis request - GSM-SIPCall-Number-1 Sending to 192.168.0.181 : 5060 (NAT) Aug 16 11:58:58 NOTICE[4587]: chan_sip.c:7112 check_user_full: From address missing 'sip:', using it anyway Found peer '01000' Found RTP audio format 3 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 13 Found RTP audio format 101 Peer audio RTP is at port 192.168.0.181:8000 Found description format PCMA Found description format PCMU Found description format PCMA Found description format CN Found description format telephone-event Capabilities: us - 0x8 (alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x3 (telephone-event|CN), combined - 0x1 (telephone-event) Aug 16 11:58:58 WARNING[4587]: chan_sip.c:6650 get_destination: Huh? Not a SIP header (:)? Reliably Transmitting (no NAT) to 192.168.0.181:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.0.181;branch=z9hG4bK-d87543-24f99000ddafb002-1--d87543-;rport;received=192.168.0.181 From: :;tag=8af2812a To: 01000:[EMAIL PROTECTED];tag=as757d8c87 Call-ID: GSM-SIPCall-Number-1 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- Retransmitting #1 (no NAT) to 192.168.0.181:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.0.181;branch=z9hG4bK-d87543-24f99000ddafb002-1--d87543-;rport;received=192.168.0.181 From: :;tag=8af2812a To: 01000:[EMAIL PROTECTED];tag=as757d8c87 Call-ID: GSM-SIPCall-Number-1 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- Retransmitting #2 (no NAT) to 192.168.0.181:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.0.181;branch=z9hG4bK-d87543-24f99000ddafb002-1--d87543-;rport;received=192.168.0.181 From: :;tag=8af2812a To: 01000:[EMAIL PROTECTED];tag=as757d8c87 Call-ID: GSM-SIPCall-Number-1 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- Retransmitting #3 (no NAT) to 192.168.0.181:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.0.181;branch=z9hG4bK-d87543-24f99000ddafb002-1--d87543-;rport;received=192.168.0.181 From: :;tag=8af2812a To: 01000:[EMAIL PROTECTED];tag=as757d8c87 Call-ID: GSM-SIPCall-Number-1 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- Retransmitting #4 (no NAT) to 192.168.0.181:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.0.181;branch=z9hG4bK-d87543-24f99000ddafb002-1--d87543-;rport;received=192.168.0.181 From: :;tag=8af2812a To: 01000:[EMAIL PROTECTED];tag=as757d8c87 Call-ID: GSM-SIPCall-Number-1 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL,
RE: [asterisk-users] Cisco 7960 password reset
David and Barry, Thanks for the help. 'preciate it. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Barry Fawthrop Sent: Wednesday, August 16, 2006 6:54 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7960 password reset If the phone already had the SIP image running. Check the SIPDefault.cnf file there may be a phone_password= string this is the phone's current password use it remember to change to number or uppercase if need be Ferguson, Michael wrote: Maxx, Thanks much for the feedback. I will check into it and follow up with your instructions. 'preciate it. Best wishes. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Maxx Lobo Sent: Tuesday, August 15, 2006 5:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7960 password reset What Cisco image is the phone running? If it is really old (lower than P0S030203) then yeah, this won't work. If you upgrade the phone to P0S30203 and from there on to P0S3-06-3-00, and then these instructions will work fine. This should be pretty straightforward using ATFTP and the Cisco images. In response to your other question, a factory reset TMK does not wipe out the SIP image. Just the settings. --Maxx Ferguson, Michael wrote: Maxx, That did not work. Any other ideas? Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Maxx Lobo Sent: Tuesday, August 15, 2006 4:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7960 password reset Fastest way (wipes everything out): 1. Power off the phone completely. 2. Hold down the # key, then power the phone on. 3. Continue holding the # key until the LCD gives you a status message. 4. Follow the prompts to do a full factory reset, which resets the password as well. --Maxx Ferguson, Michael wrote: G'Day List, I am trying, once again, to configure my 7960 to work with asterisk. Where abouts do I go to reset the password on the phone? Thanks - - -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems with outgoing calls on a TE410P
I'm not a guru,but it could help if you post the dialled number as well as your extensions.conf.On 8/16/06, Nils Rasmussen [EMAIL PROTECTED] wrote: We make a call from a PBX through asterisk to ISDN (E1) Denmark. We use asterisk to record incoming and outgoing calls Sometimes we experience that a outgoing call doesn't get through. In the message.log we get WARNING[7594] app_dial.c: Unable to forward voice In the Master.csv we get source number but in the destination number we get a "t" and not the dialled number. Does Anyone have experience or ideas to solve this problem. Medvenlig hilsen Nils Rasmussen ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Com os melhores cumprimentos,Marco Mouta ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Extension for Incoming Call through Zap Channel
Hi, In my zapata.conf, I have the following lines signalling = fxs_ks context = fromfxs channel = 1 When there is an incoming Zap call at Zap channel 1, the context fromfxs is entered and the entry s extension in the context is executed. Would it be possible to jump to a particular extension in context fromfxs instead of the s extension ? for e.g. when there is an incoming Zap call at Zap channel 1, the 123 extension in the context fromfxs is executed ? Thank you. regards, Kwang Mien ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: modprobe wctdm fails in /etc/rc.local on FC5
rc.local: touch /var/lock/subsys/local setpci -v -s 00:1f.1 LATENCY_TIMER=4 setpci -v -s 02:0e.0 LATENCY_TIMER=4 setpci -v -s 0b:07.0 LATENCY_TIMER=4 setpci -v -s 0c:08.0 LATENCY_TIMER=4 setpci -v -s 10:0d.0 LATENCY_TIMER=0 setpci -v -s 06:02.0 LATENCY_TIMER=ff sleep 5 echo UnLoading wct4xxp rmmod -v wct4xxp rmmod -v zaptel sleep 3 echo Loading wct4xxp /sbin/modprobe -v zaptel sleep 5 /sbin/modprobe -v wct4xxp sleep 5 # ztcfg - #sleep 5 echo 1 /proc/irq/201/smp_affinity echo 1 /proc/irq/217/smp_affinity echo 0 /proc/irq/209/smp_affinity echo 1 /proc/irq/14/smp_affinity /usr/sbin/amportal start -- -- Steven http://www.glimasoutheast.org Robert La Ferla [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Can someone send me their modprobe.conf file? I think that may be the problem. A zaptel file is created during install in /etc/ modprobe.d but modprobe.conf must need to reference it... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom upgrade issue
Manually config to point to your boot server, which should have a good copy of the software and it should go get it. If not sniff the traffic in/out and see what it IS doing. I have had several firmware updates get interrupted in the past corrupting the image and this has always worked. On Aug 16, 2006, at 12:15 AM, Dovid Bender wrote: I believe 468* resets the phone but dosent return it to the orig. firmware. Also try to name the files with the phones mac id and see what happens. I am doing this with 1.6.6 and its working fine. - Original Message - From: Curt Shaffer To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Tuesday, August 15, 2006 10:07 PM Subject: [asterisk-users] Polycom upgrade issue OK, I may have done something stupid. I was trying to upgrade my Polycom to the newest firmware I could find (1.6.7). I am also trying to get provisioning working from a central server. I tired to reset with holding 468* down and it kept the settings the phone had on the phone. From what I understand the settings on the phone override all. So I went into reset it from the phone and choose to format the firmware. Now when I try to boot it I am getting the following in the *-boot.log 0527180621|cfg |4|00|Could not get all 512 bytes of the header. 0527181013|cfg |4|00|Could not get all 512 bytes of the header. 0527181014|app1 |6|00|Error application is not present. 0527181014|app1 |6|00|Uploading boot log, time is SAT MAY 27 18:10:14 2006 I tried to put the old firmware and configs back in the directory but I get the same thing. Any help out there? Thanks! Curt ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Page()
I receive the following error in the Asterisk console when I try to
execute the Page() application:
WARNING[24360]: pbx.c:1700 pbx_extention_helper: No application 'Page'
for extention (intercom, *, 1)
EXTENSIONS.CONF
[Default]
Exten = *80,1,Goto(intercom,s,1)
[intercom]
exten = s,1,Answer
exten = s,n,SIPAddHeader(Call-Info: answer-after=0)
exten = s,n,Playback(beep)
exten = s,n,Set(TIMEOUT(digit)=5)
exten = s,n,WaitExten(10)
;Page
exten = *,1,Page(SIP/2000x1)
;Intercom
exten = _,1,Dial(SIP/${EXTEN})
Any clues?
Dennis Clark
DENPRO
WRK 207.618.1998
CEL 443.415.0527
FAX 1.888.811.8809
[EMAIL PROTECTED]
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Re: [asterisk-users] New Device
Dovid Bender wrote: I have spoken to some one who is interested in investing into building equipment for asterisk. I am looking to find out what products that the asterisk community would like to see be built. This can be products that already exists but lack certain functionality as well as things that arent out there but you would want to see it. Thanks. I have been away for a while, but one thing that I always wanted was an IAX phone (not soft phone). There was a guy here a while back that was working on one and I think he got one ready but I never heard anymore about it. Andy ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New Device
IAX based outboard device with FXO and/or FXS that overcomes the many shortcomings of the IAXy Should also support pulse dial, server address expressed in other than an IP, and all the good features of the SPa 2000 and 3000 JMO John Novack Dovid Bender wrote: I have spoken to some one who is interested in investing into building equipment for asterisk. I am looking to find out what products that the asterisk community would like to see be built. This can be products that already exists but lack certain functionality as well as things that arent out there but you would want to see it. Thanks. Dovid ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Training - Boston, US and Malaga, Spain
Just a quick note that Edvina in cooperation with Digium is starting the fall season of trainings again. Coming trainings are: * Asterisk Bootcamp, Boston - next week! We still have a few seats available * Asterisk Beachcamp, Malaga, Spain A class in a beach hotel in beautiful Malaga on the Spanish south coast Both classes are bootcamp-level classes with dCAP oppurtunities. Visit our web site for more information or send e-mail to [EMAIL PROTECTED] [EMAIL PROTECTED] - Voice On The Net in Boston For those of you going to Von Boston there will be a series of Asterisk seminars at von, labelled [EMAIL PROTECTED] I will be covering the coming release, 1.4 as well as run developer meetings and a meeting for the Asterisk Video Task Force. See the von web site at http://www.von.com for more information. Regards, /Olle -- http://edvina.net/training ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [EMAIL PROTECTED] - Von Fall, Boston Sept 11-14
[EMAIL PROTECTED] - There will be a lot of Asterisk-related activities at Voice On the Net FALL - Von - in Boston. Apart from Digium booth (#819), there will be Asterisk presentations as well as developer meetings. For the [EMAIL PROTECTED] agenda, see http://www.pulver.com/asterisk/ - there will be additions coming up soon. Mark Spencer, the creator of Asterisk, will speak on Wednesday, September 13th: Industry Perspective: An Open Source Carol: The Ghost of Open Source; Past, Present and Future Wednesday, September 13, 2006, 4:45pm - 5:15pm As the creator of Asterisk, the industry's first open source telephony platform, Mark Spencer, president of Digium, will discuss the phenomenal growth and industry acceptance of open source telephony since last year's Fall VON. Companies (from the enterprise to the SMB) as well as carriers and developers have come to realize the benefits of open source solutions go far beyond cost savings. In fact, flexibility and competitive advantage are two of the main drivers behind moving to an open source solution. Taking a glimpse at the past, present and future of open source telephony, Mark will discuss the role this industry has and will play in the development of next generation VoIP services. For the full agenda of the conference and the exhibition, see http:// www.von.com/index.html I am looking forward to meeting you at Von - the premium VoIP and Asterisk conference Trade show! /Olle ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Page()
Hi there;
Did you load the respective module?
Regards;
LK
On 8/16/06, Dennis P. Clark [EMAIL PROTECTED] wrote:
I receive the following error in the Asterisk console when I try to
execute the Page() application:
WARNING[24360]: pbx.c:1700 pbx_extention_helper: No application 'Page'
for extention (intercom, *, 1)
EXTENSIONS.CONF
[Default]
Exten = *80,1,Goto(intercom,s,1)
[intercom]
exten = s,1,Answer
exten = s,n,SIPAddHeader(Call-Info: answer-after=0)
exten = s,n,Playback(beep)
exten = s,n,Set(TIMEOUT(digit)=5)
exten = s,n,WaitExten(10)
;Page
exten = *,1,Page(SIP/2000x1)
;Intercom
exten = _,1,Dial(SIP/${EXTEN})
Any clues?
Dennis Clark
DENPRO
WRK 207.618.1998
CEL 443.415.0527
FAX 1.888.811.8809
[EMAIL PROTECTED]
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RE: [asterisk-users] Page()
What is the module I should be loading and how do I load it?
Dennis Clark
DENPRO
WRK 207.618.1998
CEL 443.415.0527
FAX 1.888.811.8809
[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Leonardo
Kamache (Gmail)
Sent: Wednesday, August 16, 2006 10:17 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Page()
Hi there;
Did you load the respective module?
Regards;
LK
On 8/16/06, Dennis P. Clark [EMAIL PROTECTED] wrote:
I receive the following error in the Asterisk console when I try to
execute the Page() application:
WARNING[24360]: pbx.c:1700 pbx_extention_helper: No application 'Page'
for extention (intercom, *, 1)
EXTENSIONS.CONF
[Default]
Exten = *80,1,Goto(intercom,s,1)
[intercom]
exten = s,1,Answer
exten = s,n,SIPAddHeader(Call-Info: answer-after=0) exten =
s,n,Playback(beep) exten = s,n,Set(TIMEOUT(digit)=5) exten =
s,n,WaitExten(10)
;Page
exten = *,1,Page(SIP/2000x1)
;Intercom
exten = _,1,Dial(SIP/${EXTEN})
Any clues?
Dennis Clark
DENPRO
WRK 207.618.1998
CEL 443.415.0527
FAX 1.888.811.8809
[EMAIL PROTECTED]
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RE: [Asterisk-Users] OPENSER / SER and Asterisk
*lol* The cryptic replies have been exactly my problem as well! -Original Message- From: kjcsb [mailto:[EMAIL PROTECTED] Sent: Wednesday, August 16, 2006 12:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] OPENSER / SER and Asterisk Absolutely. The SER/OpenSER documentation is terrible, and if you post to the OpenSER mailing list, you get very cryptic replies. ___ Whilst I would agree with you regarding SER, the documentation of OpenSER is far better. Documentation of Asterisk Realtime on the other hand. Now *that's* terrible. Cameron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] OPENSER / SER and Asterisk
-Original Message- From: kjcsb [mailto:[EMAIL PROTECTED] Sent: Wednesday, August 16, 2006 12:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] OPENSER / SER and Asterisk Absolutely. The SER/OpenSER documentation is terrible, and if you post to the OpenSER mailing list, you get very cryptic replies. ___ Whilst I would agree with you regarding SER, the documentation of OpenSER is far better. Documentation of Asterisk Realtime on the other hand. Now *that's* terrible. *lol* It's funny because it's so true! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Server Hardware
Hello I am curious as to what hardware folks are using successfully from HP or DELL. I will likely be running just a quad span T1 card with the system. I appreciate your input. Thanks, Dave ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Polycom upgrade issue
How did you find out about 468*??? It's sure as poop not documented in the Polycom Admin Guide anywhere. -Original Message-From: Dovid Bender [mailto:[EMAIL PROTECTED]Sent: Tuesday, August 15, 2006 11:16 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [asterisk-users] Polycom upgrade issue I believe 468* resets the phone but dosent return it to the orig. firmware. Also try to name the files with the phones mac id and see what happens. I am doing this with 1.6.6 and its working fine. - Original Message - From: Curt Shaffer To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Tuesday, August 15, 2006 10:07 PM Subject: [asterisk-users] Polycom upgrade issue OK, I may have done something stupid. I was trying to upgrade my Polycom to the newest firmware I could find (1.6.7). I am also trying to get provisioning working from a central server. I tired to reset with holding 468* down and it kept the settings the phone had on the phone. From what I understand the settings on the phone override all. So I went into reset it from the phone and choose to format the firmware. Now when I try to boot it I am getting the following in the *-boot.log 0527180621|cfg |4|00|Could not get all 512 bytes of the header. 0527181013|cfg |4|00|Could not get all 512 bytes of the header. 0527181014|app1 |6|00|Error application is not present. 0527181014|app1 |6|00|Uploading boot log, time is SAT MAY 27 18:10:14 2006 I tried to put the old firmware and configs back in the directory but I get the same thing. Any help out there? Thanks! Curt ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Page()
I just got done implementing this on a Realtime system and it works
flawlessly. You need to create a macro named page that you call from
the dialplan. Please refer to the wiki for more details:
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+page
Good luck!
Joe
Dennis P. Clark wrote:
What is the module I should be loading and how do I load it?
Dennis Clark
DENPRO
WRK 207.618.1998
CEL 443.415.0527
FAX 1.888.811.8809
[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Leonardo
Kamache (Gmail)
Sent: Wednesday, August 16, 2006 10:17 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Page()
Hi there;
Did you load the respective module?
Regards;
LK
On 8/16/06, Dennis P. Clark [EMAIL PROTECTED] wrote:
I receive the following error in the Asterisk console when I try to
execute the Page() application:
WARNING[24360]: pbx.c:1700 pbx_extention_helper: No application 'Page'
for extention (intercom, *, 1)
EXTENSIONS.CONF
[Default]
Exten = *80,1,Goto(intercom,s,1)
[intercom]
exten = s,1,Answer
exten = s,n,SIPAddHeader(Call-Info: answer-after=0) exten =
s,n,Playback(beep) exten = s,n,Set(TIMEOUT(digit)=5) exten =
s,n,WaitExten(10)
;Page
exten = *,1,Page(SIP/2000x1)
;Intercom
exten = _,1,Dial(SIP/${EXTEN})
Any clues?
Dennis Clark
DENPRO
WRK 207.618.1998
CEL 443.415.0527
FAX 1.888.811.8809
[EMAIL PROTECTED]
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Re: [asterisk-users] Page()
Dennis P. Clark wrote: I receive the following error in the Asterisk console when I try to execute the Page() application: WARNING[24360]: pbx.c:1700 pbx_extention_helper: No application 'Page' for extention (intercom, *, 1) What version of Asterisk are you running? Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re[2]: [asterisk-users] Softphone for Windows Mobile 5?
Anyone tried any softphone on Cingular 8125 or HTC Wizard? On 8/16/06, Rajeev Natarajan [EMAIL PROTECTED] wrote: http://www.electronicscience.com/ has a good IAX2 softphone called ESC Softphone On 8/16/06, David Thomas [EMAIL PROTECTED] wrote: Sorry, poor reply. Yes I use it on WM5, and have not seen any problems. I admit I don't use it a lot, but it does seem to work fine.regards,DaveOn 8/15/06, David Thomas [EMAIL PROTECTED] wrote: Yes, use it on WM5. Dave On 8/15/06, Christian [EMAIL PROTECTED] wrote: Hello, Many thanks, but it seems only to be available for Windows Mobile 2003. Will it work on WM5? Many thanks, Christian On 2006-08-15 at 14:00 David Thomas wrote: Try SJphone, it works for me. http://www.sjlabs.com/sjp.htmlThe latency is a little too much over my EVDO cannection though. :) It does work great over wifi. regards, DaveOn 8/15/06, Christian [EMAIL PROTECTED] wrote: Hi all, Does anyone know a Softphone for Windows mobile 5? Want to connect to my Asterisk when I am away. Many thanks, Christian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Server Hardware
I've had an HP Proliant DL-360 G2 (Pentium IV - 3 GHz with 1 GB memory) in production for a couple of years with 20 Cisco phones and a single T-1. The load is generally nill. I've never seen the load over 1.0, its usually more like 0.1. It is also recording (in WAV format) all inbound calls for several of the extensions. I believe you will find that most of the modern server equipment you purchase will be more than sufficient to handle your needs. However, if you are serious about using 4 T-1s, I would look to use two servers (for redundancy) and have the VoIP phones dual-register to both servers. To make the configurations easy, you could use Realtime in a MySQL database either on a cluster or on another server. The whole point is to eleminate single points of failure. Good luck and have fun! Joe David Sampson wrote: Hello – I am curious as to what hardware folks are using successfully from HP or DELL. I will likely be running just a quad span T1 card with the system. I appreciate your input. Thanks, Dave ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] 1.2.10 - g726 Issues
Yeah, that's exactly the problem that I am having here (also switched to g729 and gsm). However, Teliax has told me that the g726 issue is with the * 1.2.10 release and as a result not an issue with their service. I'm not entirely convinced, but since we also use g726 for some of our internal phones we must support it and if it's broken in 1.2.10 then I won't upgrade. What version of * are you runing? Thanks, Cullin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Tuesday, August 15, 2006 3:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 1.2.10 - g726 Issues Cullin J. Wible wrote: I have hard that 1.2.10 has issues with voice quality through g726. Can anyone provide any feedback or point me in the right direction about the current status of this problem? Been using g726 between multiple * systems for some time and the quality has been very good. Recently, however, all calls via teliax.com using g726 have had very poor quality. Switching back to gsm for them cleared up the iax audio nicely. Not sure if teliax changed something or what, but had been working fine for several months. R. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Server Hardware
On Wed, 2006-08-16 at 10:57 -0400, David Sampson wrote: Hello – I am curious as to what hardware folks are using successfully from HP or DELL. I will likely be running just a quad span T1 card with the system. HP DL380 G4, 4GB mem, 2x 146GB U320 in RAID1, dual hotswap PS HP DL360 G4, 2GB mem, 2x 146GB U320 in RAID1, dual hotswap PS Some Dell models may have issues. Check the Digium website for compatibility (and perhaps the list archives). Both HP boxes work fine with 2 or 4 port E1 cards (hyperthreading is turned off). Regards, Patrick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Page()
1.2.10 Dennis Clark DENPRO WRK 207.618.1998 CEL 443.415.0527 FAX 1.888.811.8809 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle Sent: Wednesday, August 16, 2006 11:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Page() Dennis P. Clark wrote: I receive the following error in the Asterisk console when I try to execute the Page() application: WARNING[24360]: pbx.c:1700 pbx_extention_helper: No application 'Page' for extention (intercom, *, 1) What version of Asterisk are you running? Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] linuxdevices.com: Trolltech woos developers with open Linux phone Who'll be the first with * on a mobile?
Salve *! see: http://linuxdevices.com/news/NS8030785497.html It is based on a dual-core Marvell (formerly Intel) XScale processor clocked at 312MHz[and] The Greenphone's baseband processor/modem is a Broadcom BCM2121. I think the BCM chip is for the GSM stuff, for GUI and applications the XScale chip - so for running asterisk, the XScale will be the processor. Question, who will be the first to run asterik on a mobile phone, or does somebody already run it on a linux phone like A780? Greetings, rob ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Digium TDM400P Vs Sangoma A200
I was wondering which of these cards would be better for a 1-2 line SOHO. I would like room to grow as well as I am concerned with voice quality and life expectancy of the product. Any input into which one and why would be greatly appreciated. Thanks, Jon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] polycom config error 0x4020: possibly related to RE:Polycom upgrade issue?
I would get the same error when trying to use sftp. Switching to ftp eliminated the problem. Curt Shaffer wrote: I posted earlier about an application not found error. I have manually pointed the phone at the server but it just does not seem to ever even hit it. I am going to do some network captures here soon after I walk away from this computer for a while. But here is another question which I am not sure if it may be related. After loading the application successfully on other phones I get config error 0x4020 and it just keeps rebooting through this whole process. I have checked my configs and checked them twice against all documentation I could find, and from what I see they are OK. I have posted one here for you all to look at and maybe you can see something I am missing. MAC.cfg (located in /ftproot/ ?xml version=1.0 standalone=yes? !-- Default Master SIP Configuration File-- !-- Edit and rename this file to Ethernet-address.cfg for each phone.-- !-- $Revision: 1.13 $ $Date: 2004/11/26 23:30:44 $ -- APPLICATION APP_FILE_PATH=sip.ld CONFIG_FILES=x102/x102.cfg, sip.cfg MISC_FILES= LOG_FILE_DIRECTORY=x102/ ## X102.cfg (located in /ftproot/x102) ?xml version=1.0 standalone=yes?^M PHONE_CONFIG OVERRIDES reg.1.server.1.expires=60 reg.1.address=102 voIpProt.SIP.outboundProxy.port= log.level.change.cfg=0 _.0x20._log.level.change.sip=0 log.render.level=0 tcpIpApp.sntp.gmtOffset=-21600 tcpIpApp.sntp.address=xxx.xxx.xxx.xxx reg.1.server.1.address=xxx.xxx.xxx.xxx reg.1.auth.password=1234 reg.1.auth.userId=102 voIpProt.server.1.register= reg.1.displayName=Test voIpProt.server.1.address=xxx.xxx.xxx.xxx reg.1.ringType=8/ /PHONE_CONFIG I also have a .cfg file in this directory that has the following: ## .cfg ?xml version=1.0 standalone=yes? !-- Default Master SIP Configuration File-- !-- Edit and rename this file to Ethernet-address.cfg for each phone.-- !-- $Revision: 1.14 $ $Date: 2005/07/27 18:43:30 $ -- APPLICATION APP_FILE_PATH=sip.ld CONFIG_FILES=phone1.cfg, sip.cfg MISC_FILES= LOG_FILE_DIRECTORY= OVERRIDES_DIRECTORY= CONTACTS_DIRECTORY=/ Any help would be appreciated. And I realize this is more of a Polycom question rather than an Asterisk question so if anyone can point me to a good polycom list I would appreciate it as well. Thanks Curt ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] polycom config error 0x4020: possibly related to RE:Polycom upgrade issue?
On Wed, 2006-08-16 at 11:14 -0400, Curt Shaffer wrote: After loading the application successfully on other phones I get config error 0x4020 and it just keeps rebooting through this whole process. Which version of the bootrom does the phone have? I have been told that once you upgrade to version 3.x, you cannot go back (to say, version 2.6.1). Some of the newer phones are being shipped with 3.x. If your bootrom.ld file on your ftp boot server is version 2.x, and your phone already has a 3.x image, then you will see the 0x4020 error. It cannot replace the image, and the only thing it can do is reboot. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MFCR2 and Unicall PDF
Once in a while, the same questions are asked in this mailing list about Unicall and MFCR2. I wrote a document in spanis about 2 months ago about MFCR2 signaling and how to debug it with testcall. I have translated the document into english per users request, and made some other improvements. The document can be found at: http://moy.ivsol.net/unicall/mfcr2-asterisk-unicall-0.2-english.pdf Any corrections or suggestions to the document will be appreciated. Regards -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] modprobe wctdm fails in /etc/rc.local on FC5
Hi Tzafrir I'm still testing so I start asterisk manually after boot. You are right that 99 would normally be too late but as long as zaptel + modules are loaded first with a lesser boot sequence number and is running when asterisk starts on boot up it shouldn't matter where it is done in the boot process. - Martti Tienhaara wrote: Hi Robert Why don't you select the modules in /etc/sysconfig/zaptel and allow the system to boot via /etc/init.d/zaptel by placing a link S99zaptel in your appropriate run level directory. I found that S09zaptel, the default, didn't work for me in my setup because it was too early in the boot process. You have to install zaptel for these files to exist. Hope this helps. - Tzafrir Cohen wrote Why is 09 too early? And isn't 99 too late? How do you start asterisk? -- Martti Tienhaara ([EMAIL PROTECTED]) DASH Software Ltd. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.2.10 - g726 Issues
I'm running SVN-trunk-r16869M (compiled 2006-04-01) with iax trunks to five remote systems most of which are v1.2.10. No problems with any of those trunks using g726. Teliax is the only system that I've had any issues with using iax and g726. I've not tried sip to them and don't have any intentions of doing that right now. R. Cullin J. Wible wrote: Yeah, that's exactly the problem that I am having here (also switched to g729 and gsm). However, Teliax has told me that the g726 issue is with the * 1.2.10 release and as a result not an issue with their service. I'm not entirely convinced, but since we also use g726 for some of our internal phones we must support it and if it's broken in 1.2.10 then I won't upgrade. What version of * are you runing? Thanks, Cullin -Original Message- Cullin J. Wible wrote: I have hard that 1.2.10 has issues with voice quality through g726. Can anyone provide any feedback or point me in the right direction about the current status of this problem? Been using g726 between multiple * systems for some time and the quality has been very good. Recently, however, all calls via teliax.com using g726 have had very poor quality. Switching back to gsm for them cleared up the iax audio nicely. Not sure if teliax changed something or what, but had been working fine for several months. R. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Page()
Dennis P. Clark wrote: 1.2.10 Dennis Clark DENPRO WRK 207.618.1998 CEL 443.415.0527 FAX 1.888.811.8809 [EMAIL PROTECTED] -Original Message- Dennis P. Clark wrote: I receive the following error in the Asterisk console when I try to execute the Page() application: WARNING[24360]: pbx.c:1700 pbx_extention_helper: No application 'Page' for extention (intercom, *, 1) What version of Asterisk are you running? Doug The Page application is app_page.so (located in /usr/lib/asterisk/modules on RH systems). It is present in v1.2.10 and at least at SVN-trunk-r16869M (June 4, 2006). From the CLI, do a 'show modules like page' to see if it is loaded. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Page()
in the CLI do: show applications like page if you something there then you have it loaded, otherwise do: load app_page.so if that fails my guess is you need zaptel loaded first. On 8/16/06, Dennis P. Clark [EMAIL PROTECTED] wrote: 1.2.10 Dennis Clark DENPRO WRK 207.618.1998 CEL 443.415.0527 FAX 1.888.811.8809 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle Sent: Wednesday, August 16, 2006 11:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Page() Dennis P. Clark wrote: I receive the following error in the Asterisk console when I try to execute the Page() application: WARNING[24360]: pbx.c:1700 pbx_extention_helper: No application 'Page' for extention (intercom, *, 1) What version of Asterisk are you running? Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Extension for Incoming Call through Zap Channel
the Goto command is your friend On 8/16/06, Chan Kwang Mien [EMAIL PROTECTED] wrote: Hi, In my zapata.conf, I have the following lines signalling = fxs_ks context = fromfxs channel = 1 When there is an incoming Zap call at Zap channel 1, the context fromfxs is entered and the entry s extension in the context is executed. Would it be possible to jump to a particular extension in context fromfxs instead of the s extension ? for e.g. when there is an incoming Zap call at Zap channel 1, the 123 extension in the context fromfxs is executed ? Thank you. regards, Kwang Mien ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to reject a call without picking it up, (E1-T1-ISDN)
thanks CF, I did change the PRI CAUSE to unavailable, or reject. only that it still shows Accepting overlap call from. just before this -Executing SetVar(Zap/12-1, PRI_CAUSE=27) does anyone knows if this call being picked up at anytime? Problem is, this is a reverse charge line with more than 3000 calls per hour, and if it telco thinks it is picked up for a milisecond will charge for the whole minute. But I can't disconnect the service since it is needed during 2 hours a day on a TV show.(that's the only time when people should be calling, but they keep calling the whole day instead) C F escribió: Set the PRI cause: http://www.voip-info.org/wiki/index.php?page=Asterisk+variable+PRI_CAUSE On 8/15/06, Manrique Feoli [EMAIL PROTECTED] wrote: Hi, I´m in a bit of a hurry here, I need to reject calls before picking them up. If I do hangup on the first line, does anyone knows if the line counts as picked up for the Telco? how about if I register the incoming callerid, and then do hangup on the second line? thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium TDM400P Vs Sangoma A200
Jonathan Borden wrote: I was wondering which of these cards would be better for a 1-2 line SOHO. I would like room to grow as well as I am concerned with voice quality and life expectancy of the product. Any input into which one and why would be greatly appreciated. The sangoma a200d does a much better job in terms of audio quality (eg, no echo) compared to the tdm400p. The difference is primarily in the hardware echo canceler implementation on the a200d. The tdm400p does a fine job on pstn loops that are relatively short where the software echo canceler seems to be able to handle echo. Neither card seems to have any issues with early failures, etc, and there are no guarantees from either vendor as to future support. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium TDM400P Vs Sangoma A200
The A200 is a far better card. More forgiving of Motherboards, MUCH more expandable, slightly lower cost. Only real drawback is modules are in pairs, so if you want 4 FXO, you need to buy 4. It expands to 24 ports using one PCI slot Also, if you ever are rich, hardware echo cancel is an option John Novack Jonathan Borden wrote: I was wondering which of these cards would be better for a 1-2 line SOHO. I would like room to grow as well as I am concerned with voice quality and life expectancy of the product. Any input into which one and why would be greatly appreciated. Thanks, Jon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Server Hardware
I have setup a Quad T-1 on a Dell 850 but it trunks all calls to a main voip server that is a Dell 2850. We chose to use a Sangoma T-1 card to side step some of the possible problems with motherboards/Dell servers that Patrick mentioned. On 8/16/06, Patrick [EMAIL PROTECTED] wrote: On Wed, 2006-08-16 at 10:57 -0400, David Sampson wrote: Hello – I am curious as to what hardware folks are using successfully from HP or DELL.I will likely be running just a quad span T1 card with the system.HP DL380 G4, 4GB mem, 2x 146GB U320 in RAID1, dual hotswap PSHP DL360 G4, 2GB mem, 2x 146GB U320 in RAID1, dual hotswap PSSome Dell models may have issues. Check the Digium website for compatibility (and perhaps the list archives).Both HP boxes work fine with 2 or 4 port E1 cards (hyperthreading isturned off).Regards,Patrick___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- BruceNortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CallerID is not displaying for my incoming calls
On Wed, 2006-08-16 at 00:22 -0700, Crazy Boy wrote: extensions.conf file contents: [incoming] exten = s,1,Answer exten = s,2,SetMusicOnHold(default) exten = s,3,DigitTimeout,5 exten = s,4,ResponseTimeout,10 exten = s,5,Background(/tmp/virg2) exten = s,6,Goto(s,1) include = leader You need to wait for the second ring before answering the call. CallerID is sent between the first and second ring. Make make the s,1 entry Wait(2) and then answer. -- Carlos Chavez Prats Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue Management
Hi Eric, I'm working in a small call center, but with special requirements. We currently have a couple of clients, all of them have specific phone numbers configured in our system, so when we get a call for a specific client we take down the information via a webpage then it sent via email to them. One of the major problem that I'm seeing is the queue management. Right now with our current system, the agents are able to see what call are coming in, which one haven't been answered, which one are on hold. (That part is not so bad with Asterisk since it's already taking care of this) But the part I'm worring about is that the agent can see the Greeting message for the customer line. So the agent knows what to say before answering the line then IE popups with the URL for that client. Not sure if that can be replicated with Asterisk. We could probably adapt our selfs by doing a query about the DNIS and then store the DNIS associated with his greeting. Almost like what we do now actually... The only thing is to put all that together... hehehe There are serveral packages that can be used in your situation. You can certainly use FOP ( http://www.asternic.org ) to handle popups and displays status of your asterisk queues. There are also soft phones that can handle url passing, some feature complete callcenter packages (astguiclient). You can also do it yourself using the manager API or custom AGI scripts. Regards, -- Nicolás Gudiño Buenos Aires - Argentina ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Attended Transfer call return with asterisk + sipura spa2002
Hi! I have connected my analog phones to an asterisk box with sipura spa2002 devices. I can do an attended transfer by taking the call which should be transferred, pressing the flash button, dialing the number to which the call should be transferred and now i can hang up or talk to the person who receives the transferred call. Thats working perfectly. But if the other person (the person who gets the transferred call) isnt on his place and doesnt take the call, the call gets disconnected after about a minute. Does anyone have an idea how i could make that the call gets transferred back (to the person who initially did the transfer) automatically? thx, Tom ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to reject a call without picking it up, (E1-T1-ISDN)
The call is not being picked up. Manrique Feoli wrote: thanks CF, I did change the PRI CAUSE to unavailable, or reject. only that it still shows Accepting overlap call from. just before this -Executing SetVar(Zap/12-1, PRI_CAUSE=27) does anyone knows if this call being picked up at anytime? Problem is, this is a reverse charge line with more than 3000 calls per hour, and if it telco thinks it is picked up for a milisecond will charge for the whole minute. But I can't disconnect the service since it is needed during 2 hours a day on a TV show.(that's the only time when people should be calling, but they keep calling the whole day instead) C F escribió: Set the PRI cause: http://www.voip-info.org/wiki/index.php?page=Asterisk+variable+PRI_CAUSE On 8/15/06, Manrique Feoli [EMAIL PROTECTED] wrote: Hi, I´m in a bit of a hurry here, I need to reject calls before picking them up. If I do hangup on the first line, does anyone knows if the line counts as picked up for the Telco? how about if I register the incoming callerid, and then do hangup on the second line? thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 'Hosting'
Has anyone ever tried to run multiple instances of Asterisk on a single system, running each with a different username, and each in a separate base directory? Something like /home/pbx/business-1, home/pbx/business-2 etc? Did it work? I assume for every service that Asterisk runs, on each instance, you'd have to use a different port numbers, which may get confusing. Each businesses phones would have to be configred with different SIP ports then too. What about processes? I notice that Asterisk runs about 26 processes (or are they threads?) for a single instance. Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Digium TDM400P Vs Sangoma A200
Sangoma has a better PCI interface so no interrupt or compatibility issues like you get with the Digium card. Sangoma will also upgrade the card to a version with hardware echo cancellation if you cannot solve your echo problems with the software echo cancellers. I believe you send the card in and pay thecostof the hardware echo can module. From: Jonathan Borden [mailto:[EMAIL PROTECTED] Sent: Wednesday, August 16, 2006 8:41 AMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: [asterisk-users] Digium TDM400P Vs Sangoma A200 I was wondering which of these cards would be better for a 1-2 line SOHO. I would like room to grow as well as I am concerned with voice quality and life expectancy of the product. Any input into which one and why would be greatly appreciated. Thanks, Jon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom upgrade issue
Doug, Note: Don't take this email serious, I'm just messing with you, but it sure as poop is ;). In version 1.6.x released 18th of July 2005 in section 2.2.1.4, Reset the Factory Defaults "To perform this function on all phones except the IP4000, simultaneously press and hold 4,6,8 and * dial pad keys until the password prompt appears." However, depending on which version you are looking at it may be in a different section. Cheers, Kevin Douglas Garstang wrote: How did you find out about 468*??? It's sure as poop not documented in the Polycom Admin Guide anywhere. -Original Message- From: Dovid Bender [mailto:[EMAIL PROTECTED]] Sent: Tuesday, August 15, 2006 11:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom upgrade issue I believe 468* resets the phone but dosent return it to the orig. firmware. Also try to name the files with the phones mac id and see what happens. I am doing this with 1.6.6 and its working fine. - Original Message - From: Curt Shaffer To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Tuesday, August 15, 2006 10:07 PM Subject: [asterisk-users] Polycom upgrade issue OK, I may have done something stupid. I was trying to upgrade my Polycom to the newest firmware I could find (1.6.7). I am also trying to get provisioning working from a central server. I tired to reset with holding 468* down and it kept the settings the phone had on the phone. From what I understand the settings on the phone override all. So I went into reset it from the phone and choose to format the firmware. Now when I try to boot it I am getting the following in the *-boot.log 0527180621|cfg |4|00|Could not get all 512 bytes of the header. 0527181013|cfg |4|00|Could not get all 512 bytes of the header. 0527181014|app1 |6|00|Error application is not present. 0527181014|app1 |6|00|Uploading boot log, time is SAT MAY 27 18:10:14 2006 I tried to put the old firmware and configs back in the directory but I get the same thing. Any help out there? Thanks! Curt ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 7970 SIP image
Anyone help :-( I did find one but s I said it only had png's and xml's in it Thanks - Original Message - From: Paul A Brown To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Tuesday, August 15, 2006 9:41 PM Subject: [asterisk-users] 7970 SIP image Hi Guys, I found a file on the Chisco site for 7970 Sip image (a cop file) but all it had in was xml and png files. No .loads or .sbn Anyone know the exact link? Thanks ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX2 Peer
I have two asterisk (trixbox) connected by IAX2 Trunk. Both of them have interfaces TE205P configured and working fine. I can places calls to PSTN on both of them. I can place calls from SIP phones connected on asterisk one, using the IAX2 Trunk, to SIP phones connected on the asterisk two. I can not place calls from SIP phones connect on asterisk one to ZAP Trunk connected on the other. Is it possible?? Please help me, I am getting crazy. Hernany Oliveira ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 'Hosting'
You might be able to use virtual NICs to eliminate the problem with non-standard ports for a company's SIP phones. Or real NICs using a couple of multi-homed cards.I haven't tried it, though. On 8/16/06, Douglas Garstang [EMAIL PROTECTED] wrote: Has anyone ever tried to run multiple instances of Asterisk on a single system, running each with a different username, and each in a separate base directory? Something like /home/pbx/business-1, home/pbx/business-2 etc? Did it work? I assume for every service that Asterisk runs, on each instance, you'd have to use a different port numbers, which may get confusing. Each businesses phones would have to be configred with different SIP ports then too. What about processes? I notice that Asterisk runs about 26 processes (or are they threads?) for a single instance.Doug.___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 'Hosting'
On 8/16/06, Douglas Garstang [EMAIL PROTECTED] wrote: Has anyone ever tried to run multiple instances of Asterisk on a single system, running each with a different username, and each in a separate base directory? Something like /home/pbx/business-1, home/pbx/business-2 etc? Did it work? I assume for every service that Asterisk runs, on each instance, you'd have to use a different port numbers, which may get confusing. Each businesses phones would have to be configred with different SIP ports then too. What about processes? I notice that Asterisk runs about 26 processes (or are they threads?) for a single instance.Doug.You can put Asterisk to hear in the same default port, but you must use another IP address, theoretically. -- Ralph LiebessohnICQ: 74835911Skype: liebessohn ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Run As User Asterisk
There is a good page on the wiki about this: http://www.voip-info.org/wiki-Asterisk+non-root CP On Aug 14, 2006, at 6:44 PM, Forrest Beck wrote: Does anyone have a listing on file/directories that asterisk needs ownership of to run as a user other than root? I know about the major items --- /etc/asterisk, /var/spool/asterisk/, /var/lib/asterisk, etc... Anyone have a script to fix all the directories? Thanks in advance. FB ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] 1.2.10 - g726 Issues
Thanks. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Wednesday, August 16, 2006 11:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 1.2.10 - g726 Issues I'm running SVN-trunk-r16869M (compiled 2006-04-01) with iax trunks to five remote systems most of which are v1.2.10. No problems with any of those trunks using g726. Teliax is the only system that I've had any issues with using iax and g726. I've not tried sip to them and don't have any intentions of doing that right now. R. Cullin J. Wible wrote: Yeah, that's exactly the problem that I am having here (also switched to g729 and gsm). However, Teliax has told me that the g726 issue is with the * 1.2.10 release and as a result not an issue with their service. I'm not entirely convinced, but since we also use g726 for some of our internal phones we must support it and if it's broken in 1.2.10 then I won't upgrade. What version of * are you runing? Thanks, Cullin -Original Message- Cullin J. Wible wrote: I have hard that 1.2.10 has issues with voice quality through g726. Can anyone provide any feedback or point me in the right direction about the current status of this problem? Been using g726 between multiple * systems for some time and the quality has been very good. Recently, however, all calls via teliax.com using g726 have had very poor quality. Switching back to gsm for them cleared up the iax audio nicely. Not sure if teliax changed something or what, but had been working fine for several months. R. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Real Time and sip.conf file used at the same time
Is it possible to use Asterisk RealTime and also config files (like sip.conf) at the same time? As much as I know, only one thing can be used and I need them both working!... Thanks, Ricardo. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Restricting Incoming SIP Calls Without call-limit
set group is your friend: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+SetGroup On 8/16/06, Damien Gabrielson [EMAIL PROTECTED] wrote: I have been trying to restrict incoming calls for some time and I have not had any luck yet so I hope someone may have done this already. I receive calls from a SIP provider with whom I do not register. I merely accept the call to a certain DID and route the call elsewhere. I want to limit the amount of concurrent calls for a certain group of DIDs for example: 555-, 555-, and 555- are all in one group which are allowed to have 2 concurrent calls. I can always add logic to my dial plan to count the calls as they come in and hang them up if they are over the limit but I was hoping to find a more native way in Asterisk. Thanks, Damien ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 'Hosting'
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Douglas Garstang wrote: Has anyone ever tried to run multiple instances of Asterisk on a single system, running each with a different username, and each in a separate base directory? Something like /home/pbx/business-1, home/pbx/business-2 etc? Did it work? I assume for every service that Asterisk runs, on each instance, you'd have to use a different port numbers, which may get confusing. Each businesses phones would have to be configred with different SIP ports then too. What about processes? I notice that Asterisk runs about 26 processes (or are they threads?) for a single instance. Why not just use different contexts for each company? - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFE416ZS6d5vy0jeVcRAkkJAJ9ePGEV4H5GNOljhx+syWb42IdoRACfcSet 6dTJAdgseqkUk63mGTOONik= =2M0q -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 'Hosting'
Douglas Garstang wrote: Has anyone ever tried to run multiple instances of Asterisk on a single system, running each with a different username, and each in a separate base directory? Something like /home/pbx/business-1, home/pbx/business-2 etc? Did it work? I assume for every service that Asterisk runs, on each instance, you'd have to use a different port numbers, which may get confusing. Each businesses phones would have to be configred with different SIP ports then too. What about processes? I notice that Asterisk runs about 26 processes (or are they threads?) for a single instance. Why do you need multiple instances? Just setup your Asterisk configuration to separate the various 'customers' or 'tenants'. CAKE Jeremy McNamara ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Speed dials on Polycom IP601?
I just got my first IP601 and put together my first * system (yay!) I have the first 2 buttons set up to be for the extension for the phone. I was wondering how I could make the remaining 4 into speed dials? IE: label button 3 Sales mgr and have it dial extension 246. TIA, Warren ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk 'Hosting'
Well, we're talking about several dozen, maybe 100, companies, per Asterisk box here. -Original Message-From: David Freeman [mailto:[EMAIL PROTECTED]Sent: Wednesday, August 16, 2006 11:36 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [asterisk-users] Asterisk 'Hosting'You might be able to use virtual NICs to eliminate the problem with "non-standard" ports for a company's SIP phones. Or real NICs using a couple of multi-homed cards.I haven't tried it, though. On 8/16/06, Douglas Garstang [EMAIL PROTECTED] wrote: Has anyone ever tried to run multiple instances of Asterisk on a single system, running each with a different username, and each in a separate base directory? Something like /home/pbx/business-1, home/pbx/business-2 etc?Did it work? I assume for every service that Asterisk runs, on each instance, you'd have to use a different port numbers, which may get confusing. Each businesses phones would have to be configred with different SIP ports then too. What about processes? I notice that Asterisk runs about 26 processes (or are they threads?) for a single instance.Doug.___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Real Time and sip.conf file used at the same time
Yes. On Wed, 2006-08-16 at 19:03 +0100, Ricardo Carvalho wrote: Is it possible to use Asterisk RealTime and also config files (like sip.conf) at the same time? As much as I know, only one thing can be used and I need them both working!... Thanks, Ricardo. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Server Hardware
Dell PE 1800s are our standard build. They are tower or Rack capable, have 3 open slots for expansion (2 if you get the remote access card). They are big though (5U) which is both a good and a bad thing. Good in that they have GREAT air flow inside the system so there is rarely any concern of overheating Digium cards. Bad in that they are friggin huge. Realistically though, if comparing in size to your average Avaya PBX for up to 100 users, than really the size is about the same. Regards, Raymond McKayPresidentRAYNET Technologies LLChttp://www.raynettech.com(860) 693-2226 x 31Toll Free (877) 693-2226 - Original Message - From: David Sampson To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Wednesday, August 16, 2006 10:57 AM Subject: [asterisk-users] Server Hardware Hello I am curious as to what hardware folks are using successfully from HP or DELL. I will likely be running just a quad span T1 card with the system. I appreciate your input. Thanks, Dave ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 'Hosting'
You beat me to it Matt. =)-brandonOn 8/16/06, Matt Riddell (NZ) [EMAIL PROTECTED] wrote: -BEGIN PGP SIGNED MESSAGE-Hash: SHA1Douglas Garstang wrote: Has anyone ever tried to run multiple instances of Asterisk on a single system, running each with a different username, and each in a separate base directory? Something like /home/pbx/business-1, home/pbx/business-2 etc? Did it work? I assume for every service that Asterisk runs, on each instance, you'd have to use a different port numbers, which may get confusing. Each businesses phones would have to be configred with different SIP ports then too. What about processes? I notice that Asterisk runs about 26 processes (or are they threads?) for a single instance.Why not just use different contexts for each company?- --Cheers, Matt Riddell___http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community)http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss)-BEGIN PGP SIGNATURE-Version: GnuPG v1.4.2 (MingW32)Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.orgiD8DBQFE416ZS6d5vy0jeVcRAkkJAJ9ePGEV4H5GNOljhx+syWb42IdoRACfcSet6dTJAdgseqkUk63mGTOONik= =2M0q-END PGP SIGNATURE-___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Brandon GalbraithEmail: [EMAIL PROTECTED]AIM: brandong00Voice: 630.400.6992A true pirate starts drinking before the sun hits the yard-arm. Ya. --thelost ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Restricting Incoming SIP Calls Without call-limit
Excellent! I'm not sure how I missed this before. Thanks, Damien C F wrote: set group is your friend: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+SetGroup On 8/16/06, Damien Gabrielson [EMAIL PROTECTED] wrote: I have been trying to restrict incoming calls for some time and I have not had any luck yet so I hope someone may have done this already. I receive calls from a SIP provider with whom I do not register. I merely accept the call to a certain DID and route the call elsewhere. I want to limit the amount of concurrent calls for a certain group of DIDs for example: 555-, 555-, and 555- are all in one group which are allowed to have 2 concurrent calls. I can always add logic to my dial plan to count the calls as they come in and hang them up if they are over the limit but I was hoping to find a more native way in Asterisk. Thanks, Damien ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk 'Hosting'
-Original Message- From: Matt Riddell (NZ) [mailto:[EMAIL PROTECTED] Sent: Wednesday, August 16, 2006 12:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 'Hosting' -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Douglas Garstang wrote: Has anyone ever tried to run multiple instances of Asterisk on a single system, running each with a different username, and each in a separate base directory? Something like /home/pbx/business-1, home/pbx/business-2 etc? Did it work? I assume for every service that Asterisk runs, on each instance, you'd have to use a different port numbers, which may get confusing. Each businesses phones would have to be configred with different SIP ports then too. What about processes? I notice that Asterisk runs about 26 processes (or are they threads?) for a single instance. Why not just use different contexts for each company? Because Asterisk wasn't designed with carrier class features in mind. It was designed for a single enterprise. The dialplan, and config files, start to get very very complicated after you add more than a few companies. Combine that with having to have multiple extensions for a single function (our Queues are accessed by a regular extension but then have to dial another 'virtual' extension so that DUNDi can work out the 'primary' server for a queue) and so on. Anyway, it's becoming unmanagable. Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP asterisk over Linksys VPN
I usually run the RV series of router for this. Much better thoughourput on the VPN. Remember these low end devices can usually only handle about 1Mbps - 3Mbps of encryption max depending on the unit. Other than that, I have had up to 8 behind a VPN such as this. I do generally recommend though that a small appliance style asterisk box sit on any side of a remote connection with a 1 port FXO card installed for timing, emergency 911 capability, and trucking and jitterbuffer support over IAX2. This, IMHO, tends to provide for better reliability. I generally recommend some kind of HDD less Compact Flash based system. Less mechanicals to break and you can pick one up generally for $600-$800 with the digium card depending on speed and number of phones to support. Regards, Raymond McKay President RAYNET Technologies LLC http://www.raynettech.com (860) 693-2226 x 31 Toll Free (877) 693-2226 - Original Message - From: C F [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, August 15, 2006 1:23 PM Subject: [asterisk-users] SIP asterisk over Linksys VPN Has anybody tried using a VPN and around 10 phones behind the tunnel to connect to an asterisk server using Linksys VPN routers? Like this one: http://www.linksys.com/servlet/Satellite?c=L_Product_C2childpagename=US%2FLayoutcid=1115416832495pagename=Linksys%2FCommon%2FVisitorWrapper ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk 'Hosting'
-Original Message- From: Jeremy McNamara [mailto:[EMAIL PROTECTED] Sent: Wednesday, August 16, 2006 12:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 'Hosting' Douglas Garstang wrote: Has anyone ever tried to run multiple instances of Asterisk on a single system, running each with a different username, and each in a separate base directory? Something like /home/pbx/business-1, home/pbx/business-2 etc? Did it work? I assume for every service that Asterisk runs, on each instance, you'd have to use a different port numbers, which may get confusing. Each businesses phones would have to be configred with different SIP ports then too. What about processes? I notice that Asterisk runs about 26 processes (or are they threads?) for a single instance. Why do you need multiple instances? Just setup your Asterisk configuration to separate the various 'customers' or 'tenants'. It's obvious that Asterisk was designed more for the enterprise (ie a single company), rather than for the carrier (ie multiple companies). It's a bit hard to explain here, but even with more than a few companies, the config files and dial plan start to become horribly complex. Our first customer has 15 contexts (right now) in extensions.conf (we've broken each company into a separate files included from extensions.conf and sip.conf for some manageability). At several hundred companies, that's several thousand contexts. We have three Asterisk boxes, and can add more, but the config is (almost) idential between them for redundancy, and this means that each Asterisk box has to have a dialplan configured for all companies. Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 'Hosting'
Doug,I'd suggest using contexts, but then having two servers for redundancy also. That way, if one asterisk box goes down, you don't have 50-100 clients completely down.-brandon On 8/16/06, Douglas Garstang [EMAIL PROTECTED] wrote: Well, we're talking about several dozen, maybe 100, companies, per Asterisk box here. -Original Message-From: David Freeman [mailto:[EMAIL PROTECTED]]Sent: Wednesday, August 16, 2006 11:36 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [asterisk-users] Asterisk 'Hosting'You might be able to use virtual NICs to eliminate the problem with non-standard ports for a company's SIP phones. Or real NICs using a couple of multi-homed cards.I haven't tried it, though. On 8/16/06, Douglas Garstang [EMAIL PROTECTED] wrote: Has anyone ever tried to run multiple instances of Asterisk on a single system, running each with a different username, and each in a separate base directory? Something like /home/pbx/business-1, home/pbx/business-2 etc?Did it work? I assume for every service that Asterisk runs, on each instance, you'd have to use a different port numbers, which may get confusing. Each businesses phones would have to be configred with different SIP ports then too. What about processes? I notice that Asterisk runs about 26 processes (or are they threads?) for a single instance.Doug.___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Brandon GalbraithEmail: [EMAIL PROTECTED] AIM: brandong00Voice: 630.400.6992A true pirate starts drinking before the sun hits the yard-arm. Ya. --thelost ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 'Hosting'
Then virtual would be the way to go...I'm no expert, so you'd have to do some research on how many virtual interfaces you could use reliably.But some of the other suggestions I've seen might be a better option? Separate contexts for each entity, etc. On 8/16/06, Douglas Garstang [EMAIL PROTECTED] wrote: Well, we're talking about several dozen, maybe 100, companies, per Asterisk box here. -Original Message-From: David Freeman [mailto:[EMAIL PROTECTED]]Sent: Wednesday, August 16, 2006 11:36 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [asterisk-users] Asterisk 'Hosting'You might be able to use virtual NICs to eliminate the problem with non-standard ports for a company's SIP phones. Or real NICs using a couple of multi-homed cards.I haven't tried it, though. On 8/16/06, Douglas Garstang [EMAIL PROTECTED] wrote: Has anyone ever tried to run multiple instances of Asterisk on a single system, running each with a different username, and each in a separate base directory? Something like /home/pbx/business-1, home/pbx/business-2 etc?Did it work? I assume for every service that Asterisk runs, on each instance, you'd have to use a different port numbers, which may get confusing. Each businesses phones would have to be configred with different SIP ports then too. What about processes? I notice that Asterisk runs about 26 processes (or are they threads?) for a single instance.Doug.___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Comfort noise support incomplete in Asterisk (RFC 3389).
I trying to setup a outbound trunk with IPSmarx. It's working, but when I make a call, the ring dialtone stills ringing on my side, even after the other side picksup the phone. I got a NOTICE message from Asterisk that I hope you can help me: -- Called [EMAIL PROTECTED] -- SIP/ipsmarx-out-0995f270 is making progress passing it to IAX2/1010-14 -- SIP/ipsmarx-out-0995f270 is ringing -- SIP/ipsmarx-out-0995f270 is making progress passing it to IAX2/1010-14 Aug 16 15:39:21 NOTICE[16215]: rtp.c:331 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 64.34.224.230 ipsmarx-out is my outbound route. I got two SIP passing process. So I listen 2 ringtone and when the second ringtone start with a delay I got this NOTICE from asterisk:Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 64.34.224.230. I googled this error but could find a fix to this bug. Thank you in advance. Luc Moreira __ Logic Telecom Fortaleza, Brasil +55 (85) 3263-0372 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Real Time and sip.conf file used at the same time
On Wed, 2006-08-16 at 19:03 +0100, Ricardo Carvalho wrote: Is it possible to use Asterisk RealTime and also config files (like sip.conf) at the same time? As much as I know, only one thing can be used and I need them both working!... Yes, you can use both at the same time. The only restriction is that you cannot use the realtime static configuration and realtime configuration. -- Carlos Chavez Prats Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk and Speakeasy VOIP
Greetings, has anyone ever set up Asterisk and Speakeasy VOIP? It uses a Motorola VT1005 - any luck with this? TIA -Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Asterisk load testing
Nitin Gupta wrote: Hi, did anyone try do load-testing on asterisk, for sip channel calls? I want to have a rough estimate about - how many calls, an asterisk server, running on say dual 240 opteron with 1 GB memory, can handle? Also how much internet bandwidth does a typical call requires? I heard around 20Kbps with typical codecs, is that right? Thanks in advance, Nitin http://www.erlang.com/calculator/lipb/ Go read up on some codecs http://www.vocal.com/data_sheets/codecs_voip.html Too many variables to answer your question. -- =+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ J. Oquendo http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743 sil infiltrated . net http://www.infiltrated.net How a man plays the game shows something of his character - how he loses shows all - Mr. Luckey ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 'Hosting'
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Douglas Garstang wrote: Well, we're talking about several dozen, maybe 100, companies, per Asterisk box here. Surely all the more reason to do it with contexts than instances. - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFE422LS6d5vy0jeVcRAhQQAJ9XLDlNHe2Xv7oBA568nvaPbnKI1wCeM+t4 5geXNT+XaPj1gSxdSROQKYE= =AZoL -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 'Hosting'
You could use Xen on Fedora Core 6 and virtualize each instance if you feel the need is there.On 8/16/06, Douglas Garstang [EMAIL PROTECTED] wrote: -Original Message- From: Matt Riddell (NZ) [mailto: [EMAIL PROTECTED]] Sent: Wednesday, August 16, 2006 12:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 'Hosting' -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Douglas Garstang wrote: Has anyone ever tried to run multiple instances of Asterisk on a single system, running each with a different username, and each in a separate base directory? Something like /home/pbx/business-1, home/pbx/business-2 etc? Did it work? I assume for every service that Asterisk runs, on each instance, you'd have to use a different port numbers, which may get confusing. Each businesses phones would have to be configred with different SIP ports then too. What about processes? I notice that Asterisk runs about 26 processes (or are they threads?) for a single instance. Why not just use different contexts for each company?Because Asterisk wasn't designed with carrier class features in mind. It was designed for a single enterprise. The dialplan, and config files, start to get very very complicated after you add more than a few companies. Combine that with having to have multiple extensions for a single function (our Queues are accessed by a regular extension but then have to dial another 'virtual' extension so that DUNDi can work out the 'primary' server for a queue) and so on. Anyway, it's becoming unmanagable. Doug.___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Brandon GalbraithEmail: [EMAIL PROTECTED]AIM: brandong00Voice: 630.400.6992A true pirate starts drinking before the sun hits the yard-arm. Ya. --thelost ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 'Hosting'
and each in a separate base directory? Something like /home/pbx/business-1, home/pbx/business-2 etc? Use VPSs, like www.openvz.org Pablo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 'Hosting'
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Douglas Garstang wrote: Because Asterisk wasn't designed with carrier class features in mind. It was designed for a single enterprise. The dialplan, and config files, start to get very very complicated after you add more than a few companies. Combine that with having to have multiple extensions for a single function (our Queues are accessed by a regular extension but then have to dial another 'virtual' extension so that DUNDi can work out the 'primary' server for a queue) and so on. Anyway, it's becoming unmanagable. So write better management software, that's what we've and many others have done. - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFE43DHS6d5vy0jeVcRAnEyAJ9yNLv+vDF2esy4S6Hik8C46POiDQCeKd5X 6BND4aXxRw5nxifVC1oQM6U= =pfPk -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP asterisk over Linksys VPN
I tried to do sip over vpn with with a linksys router handle just one phone. When I tried it, it worked fine. Once i shipped it out we had all types of problems. at first it was fine, then 1 out of 5 calls would sound like cell phones. Now I can call him be he can't hear anything. Everything else works fine through the vpn. Most importantly, I can't trouble shoot correctly. I finally gave up and got him callvantage. Now all I have to worry about is forwarding a DID number. Raymond McKay wrote: I usually run the RV series of router for this. Much better thoughourput on the VPN. Remember these low end devices can usually only handle about 1Mbps - 3Mbps of encryption max depending on the unit. Other than that, I have had up to 8 behind a VPN such as this. I do generally recommend though that a small appliance style asterisk box sit on any side of a remote connection with a 1 port FXO card installed for timing, emergency 911 capability, and trucking and jitterbuffer support over IAX2. This, IMHO, tends to provide for better reliability. I generally recommend some kind of HDD less Compact Flash based system. Less mechanicals to break and you can pick one up generally for $600-$800 with the digium card depending on speed and number of phones to support. Regards, Raymond McKay President RAYNET Technologies LLC http://www.raynettech.com (860) 693-2226 x 31 Toll Free (877) 693-2226 - Original Message - From: C F [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, August 15, 2006 1:23 PM Subject: [asterisk-users] SIP asterisk over Linksys VPN Has anybody tried using a VPN and around 10 phones behind the tunnel to connect to an asterisk server using Linksys VPN routers? Like this one: http://www.linksys.com/servlet/Satellite?c=L_Product_C2childpagename=US%2FLayoutcid=1115416832495pagename=Linksys%2FCommon%2FVisitorWrapper ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk 'Hosting'
Brandon, Thanks. We're a litle past that stage of complexity. I'm just throwing the question out there because it's becoming obvious that trying to provision hundreds of customers on a cluster of Asterisk systems is going to be very hard to manage. -Original Message-From: Brandon Galbraith [mailto:[EMAIL PROTECTED]Sent: Wednesday, August 16, 2006 12:53 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [asterisk-users] Asterisk 'Hosting'Doug,I'd suggest using contexts, but then having two servers for redundancy also. That way, if one asterisk box goes down, you don't have 50-100 clients completely down.-brandon On 8/16/06, Douglas Garstang [EMAIL PROTECTED] wrote: Well, we're talking about several dozen, maybe 100, companies, per Asterisk box here. -Original Message-From: David Freeman [mailto:[EMAIL PROTECTED]]Sent: Wednesday, August 16, 2006 11:36 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [asterisk-users] Asterisk 'Hosting'You might be able to use virtual NICs to eliminate the problem with "non-standard" ports for a company's SIP phones. Or real NICs using a couple of multi-homed cards.I haven't tried it, though. On 8/16/06, Douglas Garstang [EMAIL PROTECTED] wrote: Has anyone ever tried to run multiple instances of Asterisk on a single system, running each with a different username, and each in a separate base directory? Something like /home/pbx/business-1, home/pbx/business-2 etc?Did it work? I assume for every service that Asterisk runs, on each instance, you'd have to use a different port numbers, which may get confusing. Each businesses phones would have to be configred with different SIP ports then too. What about processes? I notice that Asterisk runs about 26 processes (or are they threads?) for a single instance. Doug. ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Brandon GalbraithEmail: [EMAIL PROTECTED] AIM: brandong00Voice: 630.400.6992"A true pirate starts drinking before the sun hits the yard-arm. Ya. --thelost" ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Manager Interface API's
Actually, because there's no documentation, I don't have anything that I can use. -Original Message- From: Dovid Bender [mailto:[EMAIL PROTECTED] Sent: Tuesday, August 15, 2006 12:54 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Manager Interface API's Some of them write it for them selves and out of the goodness of thier heart will put out there for free. They dont need doc's since they wrote it them selves. Be happy that you got it for free. Do you want people to stop releasing code because others complain ? - Original Message - From: John Novack [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, August 15, 2006 12:39 PM Subject: Re: [asterisk-users] Manager Interface API's I, for one, didn't take his comment as anything other than constructive Lack of documentation is an issue, open source or not. It is an unfortunate situation that many very smart coders understand what they have created, but are unwilling or unable to supply enough information for many others to make effective use of their creation How many have struggled through the years with uncommented or poorly commented code when the original creator is off to greener pastures? JMO John Novack Moises Silva wrote: Douglas. Please take this as a constructive comment. I have followed your questions in asterisk-dev and users lists, and you always seem to make non constructive comments about the people giving code/work for Free. And you focus in the negative part, never giving importance to the positive things about it. If you dont like something, then change it yourself, they are not providing a payed service. The source is available AS-IS if you want it, and if you like it, take it; If you dont, just ignore it, try to not make peyorative comments. Regards On 8/15/06, Douglas Garstang [EMAIL PROTECTED] wrote: Well, I don't know about you, but if I have to read the source code to work out how it works, I'm going to go and look at someone elses, that may have some BASIC documentation and examples. -Original Message- From: Don [mailto:[EMAIL PROTECTED] Sent: Tuesday, August 15, 2006 9:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Manager Interface API's Probably cause it is someone like most of us sitting at home doing it...releasing it for free...so why would we write pages of documentation for it? If it's open source and it's free...Then offer them some money to make documentation for it hehe... - Original Message - From: Douglas Garstang To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Tuesday, August 15, 2006 11:05 AM Subject: [asterisk-users] Manager Interface API's Can anyone recommend the best Manager Interface API, putting language preferences aside? The python and perl ones have bupkiss documentation. I can't understand why anyone would even write an api and make it publically available without documenting it. Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.405 / Virus Database: 268.10.10/419 - Release Date: 8/15/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] calling in-out
Hello people, I am having some issues with my new SIP provider. The sip provider gives me only an IP address to configure my sip account, since they do allow by IP address and not by username password. This all configuration appears to work well, since I can originate a call and it will ring the destination, and I can originate a call from PSTN and * will see it. But none of both call difections will be stabilished. If I originate a call from * to a PSTN number, with a sip debug I get: Destroying call '[EMAIL PROTECTED]' pbx*CLI -- SIP read from 200.123.190.50:5060: SIP/2.0 500 Server Internal Error To: sip:[EMAIL PROTECTED];tag=3364745030-621025 From: CrossFone sip:[EMAIL PROTECTED];tag=as4d4398b9 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Contact: sip:[EMAIL PROTECTED]:5060 Via: SIP/2.0/UDP 200.59.45.210:5060;branch=z9hG4bK6c94441d;rport Content-Length: 0 --- (8 headers 0 lines)--- -- Got SIP response 500 Server Internal Error back from 200.123.190.50 Transmitting (no NAT) to 200.123.190.50:5060: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 200.59.45.210:5060;branch=z9hG4bK6c94441d;rport From: CrossFone sip:[EMAIL PROTECTED];tag=as4d4398b9 To: sip:[EMAIL PROTECTED];tag=3364745030-621025 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/CrossFone-087b3e40 is circuit-busy Destroying call '[EMAIL PROTECTED]' == Everyone is busy/congested at this time (1:0/1/0) -- Executing Goto(SIP/1501-087acbf8, s-CONGESTION|1) in new stack -- Goto (macro-dialout-trunk,s-CONGESTION,1) -- Executing NoOp(SIP/1501-087acbf8, Dial failed due to CONGESTION) in new stack -- Executing Macro(SIP/1501-087acbf8, outisbusy|) in new stack -- Executing Playback(SIP/1501-087acbf8, all-circuits-busy-now) in new stack If I make a call to my SIP number, it will ring till I pickup the phone, when I pickup the phone, I get: -- SIP read from 201.216.206.221:62477: --- (0 headers 1 lines)--- pbx*CLI -- SIP read from 200.123.190.50:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Max-Forwards: 70 Session-Expires: 3600;Refresher=uac Supported: timer To: 1159174200 sip:[EMAIL PROTECTED] From: sip:[EMAIL PROTECTED]:5060;tag=3364745421-27664 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE Via: SIP/2.0/UDP 200.123.190.50:5060;branch=b86b531fb60caa03195a218a6e8947fe Contact: sip:[EMAIL PROTECTED]:5060 Content-Type: application/sdp Content-Length: 170 v=0 o=NexTone-MSW 1234 0 IN IP4 200.123.190.53 s=sip call c=IN IP4 200.123.190.53 t=0 0 m=audio 21660 RTP/AVP 18 4 4 0 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes --- (12 headers 8 lines)--- Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 200.123.190.50 : 5060 (non-NAT) Found peer 'CrossFone' Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 4 Found RTP audio format 0 Peer audio RTP is at port 200.123.190.53:21660 Found description format G729 Capabilities: us - 0x10f (g723|gsm|ulaw|alaw|g729), peer - audio=0x105 (g723|ulaw|g729)/video=0x0 (nothing), combined - 0x105 (g723|ulaw|g729) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Looking for 1159174200 in from-sip-external (domain 200.59.45.210) list_route: hop: sip:[EMAIL PROTECTED]:5060 Transmitting (no NAT) to 200.123.190.50:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 200.123.190.50:5060;branch=b86b531fb60caa03195a218a6e8947fe;received=200.123 .190.50 From: sip:[EMAIL PROTECTED]:5060;tag=3364745421-27664 To: 1159174200 sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- -- Executing NoOp(SIP/5060-087ace18, Received incoming SIP connection from unknown peer to 1159174200) in new stack -- Executing Set(SIP/5060-087ace18, DID=1159174200) in new stack -- Executing Goto(SIP/5060-087ace18, s|1) in new stack -- Goto (from-sip-external,s,1) -- Executing GotoIf(SIP/5060-087ace18, 0?from-trunk|1159174200|1) in new stack -- Executing Set(SIP/5060-087ace18, TIMEOUT(absolute)=15) in new stack -- Channel will hangup at 2006-08-16 19:27:24 UTC. -- Executing Answer(SIP/5060-087ace18, ) in new stack We're at 200.59.45.210 port 19920 Adding codec 0x100 (g729) to SDP Adding codec 0x1 (g723) to SDP Adding codec 0x4 (ulaw) to SDP Reliably Transmitting (no NAT) to 200.123.190.50:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 200.123.190.50:5060;branch=b86b531fb60caa03195a218a6e8947fe;received=200.123 .190.50 From: sip:[EMAIL PROTECTED]:5060;tag=3364745421-27664 To: 1159174200 sip:[EMAIL PROTECTED];tag=as7635cbf2 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Type: application/sdp Content-Length: 231 v=0 o=root 2815 2815 IN IP4 200.59.45.210
RE: [asterisk-users] Manager Interface API's
-Original Message- From: Moises Silva [mailto:[EMAIL PROTECTED] Sent: Tuesday, August 15, 2006 10:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Manager Interface API's Douglas. Please take this as a constructive comment. I have followed your questions in asterisk-dev and users lists, and you always seem to make non constructive comments about the people giving code/work for Free. And you focus in the negative part, never giving importance to the positive things about it. In my opinion, and it seems perfectly logical to me, if someone writes some code, but provides no documentation, such that no one can use it, then what is the point? They have not provided a solution to anyones problem except their own, and have no added value to the open source community in any way, except to create 'vapourware' whereby software appears to be available, but is unusable, because no one can work out how to make it work. If you dont like something, then change it yourself, they are not providing a payed service. The source is available AS-IS if you want it, and if you like it, take it; If you dont, just ignore it, try to not make peyorative comments. I'll refer to my opinion above. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk 'Hosting'
Use a virtual private asterisk system. You'll be happier if you did. http://www.telephreak.org/papers/vpa/ Has anyone ever tried to run multiple instances of Asterisk on a single system, running each with a different username, and each in a separate base directory? Something like /home/pbx/business-1, home/pbx/business-2 etc? Did it work? I assume for every service that Asterisk runs, on each instance, you'd have to use a different port numbers, which may get confusing. Each businesses phones would have to be configred with different SIP ports then too. What about processes? I notice that Asterisk runs about 26 processes (or are they threads?) for a single instance. It's obvious that Asterisk was designed more for the enterprise (ie a single company), rather than for the carrier (ie multiple companies). It's a bit hard to explain here, but even with more than a few companies, the config files and dial plan start to become horribly complex. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 'Hosting'
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Douglas Garstang wrote: It's obvious that Asterisk was designed more for the enterprise (ie a single company), rather than for the carrier (ie multiple companies). It's a bit hard to explain here, but even with more than a few companies, the config files and dial plan start to become horribly complex. Our first customer has 15 contexts (right now) in extensions.conf (we've broken each company into a separate files included from extensions.conf and sip.conf for some manageability). At several hundred companies, that's several thousand contexts. We have three Asterisk boxes, and can add more, but the config is (almost) idential between them for redundancy, and this means that each Asterisk box has to have a dialplan configured for all companies. And so you're thinking it would be better to run several hundred Asterisk instances?! Good luck. I think your project would work a lot better if you worked like this: 1) Get requirements 2) Map features and limitations of products 3) Write PseudoCode 4) Work out ways to load test your ideas 5) Write real code 6) Load test again with real code Hint: Layer your system so that each component is not doing too much Hint #2: Read: http://www.astricon.net/files/David_Zimmer_EUR06.pdf - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFE43N2S6d5vy0jeVcRAj0OAJ4vgp3aMeBiEsVsU+zqhyouu8CPlgCffPAv 0SccdLfefS8GUtkxZpIMpU4= =ciMO -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 'Hosting'
Douglas Garstang wrote: Well, we're talking about several dozen, maybe 100, companies, per Asterisk box here. Ok - And the problem is? Jeremy McNamara ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Manager Interface API's
-Original Message- From: John Novack [mailto:[EMAIL PROTECTED] Sent: Tuesday, August 15, 2006 10:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Manager Interface API's I, for one, didn't take his comment as anything other than constructive Lack of documentation is an issue, open source or not. It is an unfortunate situation that many very smart coders understand what they have created, but are unwilling or unable to supply enough information for many others to make effective use of their creation How many have struggled through the years with uncommented or poorly commented code when the original creator is off to greener pastures? Green pastures for sure. I think people develop the code, thinking they will write docs later on. By the time they get close to releasing their code, they've lost interest, or the priority of this project has decreased. It's human nature. The open source community then ends up with software thats unusable. Is it so ludicrous that if you develop an API that you document it? We're not talking about developing a fahrenheight-celcius converter in basic here. We're talking about an Application Programming Iinterface! It's a programming interface. It's not the same as some GUI where you can get an idea of how it works by using it. If an API doesn't have any docs, it's completely useless. Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
