[Asterisk-Users] SPA-3000 call pickup behind a PABX
Hi Folks, I am running a SPA-3000 behind a legacy PABX on an analog line. I have been able to set up a dial plan that sends outgoing calls out to the appropriate VSP depending on prefix, and that part and the incoming call handling works fine. I am now trying to implement call pickup (dial 6*) or manual call forwarding (flash, dial extension). On the first of these I have worked out how to get the 6* sent to the PSTN line - I had to allow the * to be dialed by changing the Dial Plan specified in Line 1 VoIP Caller DP on the PSTN Line tab from (xx.) to ([x#*].) but still no dice. The Line 1 dial plan includes 6*S0:@gw0 and from traces and various utilities I have concluded the digits are dialed to the PSTN line... Is there something else that need to be done to tell the SPA-3000 to connect after dialing? Any links or hints on working behind a PABX like this would be most welcome. I had to alter the Line-In-Use Voltage to 16 V to let the SPA work with our PABX but I'm hoping that is not related... Cheers, Dieter. ___ --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] Snap for Asterisk
[EMAIL PROTECTED] wrote: I've been working on a project for Asterisk for some time and it is finally ready for a beta release. Any feedback is well appreciated. At the basic core it's a Dialer for Windows. I'll be adding more features quickly, but I wanted to keep everything simple and stable in this first release. What is this for? I have set it up, trying to dial some number, a balloon tip says it is dialing but nothing happens. What am I doing wrong? -- Best regards, Bartosz Piec ___ --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 stops responding when internet is down
11 apr 2006 kl. 14.28 skrev Michael Strelnikov: I do have that line. I also have all my phones defined by IP address. But all providers are defined by names. On 4/10/06, Michiel van Baak [EMAIL PROTECTED] wrote:On 22:14, Mon 10 Apr 06, Michael Strelnikov wrote: Hi, My * refuses SIP registrations when internet is down. All is returing at the moment when outside connection is up. What is wrong? Try to set srvlookup=no in your sip.conf Or put all the phone ip's in the servers /etc/hosts This is clearly a resolving issue This has to do with the current DNS implementation in asterisk, which is not very asynchronus. we are working on fixing this. While waiting for that solution (hopefully in the release after 1.4) I would guess that running a local caching DNS server on your LAN would help. Asterisk will then get a DNS reply, even if it says sorry, have no answer. Sending DNS queries, not getting any response, kills Asterisk. /O --- * Olle E. Johansson - [EMAIL PROTECTED] * MeetAsterisk European Tour http://www.meetasterisk.com * Asterisk Training 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
Re: [Asterisk-Users] Why is the internet connection important to LAN and PSTN calls?
11 apr 2006 kl. 16.05 skrev Brent Torrenga: Out internet connection was out this morning. It seems that the SIP extensions on our LAN were affected. Behavior like: Call comes in over POTS to a TDM400P, there is a delay then before the Cisco 79[46]0's start to ring. If we were lucky enough to get a call through, then we could not transfer the call, or place the call on hold, or park the call. Outbound calls seemed to have a delay between the time they were dialed at the SIP phone and when they were connected. I know this has been brought up before, in fact there is a bit of a discussion going on now about DNS SRV (in sip.conf, set srvlookup=no, or put all the phone ip's on /etc/hosts). But what is really causing the issue here? Yes, it is DNS, or something related to DNS, but why does that have anything to do with * trying to make a phone ring on the LAN? The SRVLOOKUP setting has nothing to do with this, Asterisk will send DNS queries anyway. I just answered a similar question in another mail, so check that. If DNS does not work on your local network, Asterisk will lock up. /O ___ --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 stops responding when internet is down
What caching DNS do you recommend?On 4/12/06, Olle E Johansson [EMAIL PROTECTED] wrote: 11 apr 2006 kl. 14.28 skrev Michael Strelnikov: I do have that line. I also have all my phones defined by IP address. But all providers are defined by names. On 4/10/06, Michiel van Baak [EMAIL PROTECTED] wrote:On 22:14, Mon 10 Apr 06, Michael Strelnikov wrote: Hi, My * refuses SIP registrations when internet is down. All is returing at the moment when outside connection is up. What is wrong? Try to set srvlookup=no in your sip.conf Or put all the phone ip's in the servers /etc/hosts This is clearly a resolving issue This has to do with the current DNS implementation in asterisk, whichis not very asynchronus. we are workingon fixing this. While waiting for that solution (hopefully in therelease after 1.4) I would guess that running a local caching DNS server on your LAN would help. Asterisk will then get aDNS reply, even if it says sorry, have no answer.Sending DNS queries, not getting any response, kills Asterisk./O ---* Olle E. Johansson - [EMAIL PROTECTED] * MeetAsterisk European Tourhttp://www.meetasterisk.com* Asterisk Training 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 -- Best regards,Michael Strelnikov ___ --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] Trial Version of Asterisk Interface Available
Please use the asterisk-biz mailing list for all commercial offerings. Thank you. /Olle --- * Olle E. Johansson - [EMAIL PROTECTED] * Asterisk Training 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
RE: RE: [Asterisk-Users] MultiBRI in Australia - found one - maybe
I have one of the Draytek USB adaptors. Can someone point me in the right direction on how to get mISDN running with it? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Friday, 17 March 2006 12:17 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: RE: [Asterisk-Users] MultiBRI in Australia - found one - maybe Any idea where I can get some of these units in Melbourne? Paul Hales AsteriskIT Faxing received by SpanDSP seems to work fine with these units. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Phelan Sent: Tuesday, 14 March 2006 9:00 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] MultiBRI in Australia - found one - maybe HI Craig and all that is following this. I am running a Vanilla 2.6.11 From cli, misdn show config Misdn General-Config: - VERSION: 0.2.1 - DEBUG_LEVEL: 1 - TRACEFILE: not set - TRACE_CALLS: false - TRACE_DIR: /var/log/ - BRIDGING: no- STOP_TONE_AFTER_FIRST_DIGIT: yes - APPEND_DIGITS2EXTEN: yes- L1_INFO_OK: yes - CLEAR_L3: no- DYNAMIC_CRYPT: no - CRYPT_PREFIX: **- CRYPT_KEYS: test,muh So Far, no dropped calls etc Todays testing will be faxing. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Craig Guy Sent: Monday, 13 March 2006 6:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] MultiBRI in Australia - found one - maybe We went to kernel 2.6.15, and at the same time went to mISDN 0.3.0 rc25, we haven't had any lockups but users are reporting dropped calls. Unfortunately for us this means dropping chan_mISDN in favour of the Cisco router containing BRI cards and then SIP from the Cisco to Asterisk. It may still be possible to use chan_capi with the mISDN drivers for the Drayteks but for us we've run out of time which is a bit of a bummer. I believe the problem is in chan_mISDN which is admittedly still an experimental driver at this stage with release candidates every few days for the past couple weeks. I'm still interested to know how you guys get along with these adapters. As I said, I think the problem is within chan_mISDN at this stage rather than in the USB adapters, so maybe using chan_CAPI on top of mISDN hardware drivers or using chan_vISDN would be the way to go until chan_mISDN matures. Craig - Original Message - From: James Harper [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, March 13, 2006 3:16 PM Subject: RE: [Asterisk-Users] MultiBRI in Australia - found one - maybe Got my 2 dreytek adapters today... Dropped them on to my test system. After wadding thru my Memory of how to setup mISDN, I had it up and running within about 2 hours. You might be receiving an email from me shortly then if I get stuck. If it wasn't for these annoying public holidays (Labour day in Victoria) mine would probably have arrived today too :) Both of them operating in ptmp with no echo cancel turned on at this ___ --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 stops responding when internet is down
12 apr 2006 kl. 08.46 skrev Michael Strelnikov: What caching DNS do you recommend? Anyone you feel comfortable running. /O ___ --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: Received VNAK: resending outstanding frames?
On 11 Apr 2006, at 23:41, Carey O'Shea wrote: PA168S There is a manual at: http://www.centralitycomm.com/solutions/Download/documents/product/ PA168SUserguideEng.pdf If I understand it, you can use the 'set' key (followed by 'speaker') to navigate the settings menu. I guess the trick is to get the IP config sane then use the web browser to finish the job. Good luck. Tim. Tim Panton [EMAIL PROTECTED] ___ --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] still no solution for me, if one provider fails.
OK, your solution is fine but I'd like a more generic solution to adapt it
to my current [EMAIL PROTECTED] setup.
Thanks anyway
Mimmus
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Peter J Dean
We do it slightly different, rather than multiple macros, we
do it within a single macro.
On 11/04/2006, at 6:55 PM, Mimmus wrote:
I configured two trunks for my outgoing calls:
[outrt-001-out]
exten = _0.,1,Macro(dialout-trunk,2,${EXTEN:1},)
exten = _0.,2,Macro(dialout-trunk,5,${EXTEN:1},)
exten = _0.,3,Macro(outisbusy) ; No available circuits
If first fails, second is automatically used but I get a CDR with
disposition = 'FAILED'. How can I avoid this?
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Re: [Asterisk-Users] Why is the internet connection important to LAN and PSTN calls?
If DNS does not work on your local network, Asterisk will lock up. Out of curiosity - the async implementation you mentioned in the other thread - will it replace gethostbyname with something smarter or just run things in a different thread asynchronously? Thanks, Cristi ___ --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 stops responding when internet is down
I just never used one. Is BIND good enough?On 4/12/06, Olle E Johansson [EMAIL PROTECTED] wrote: 12 apr 2006 kl. 08.46 skrev Michael Strelnikov: What caching DNS do you recommend?Anyone you feel comfortable running./O___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best regards,Michael Strelnikov ___ --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] Cisco 7960 6.3 unlock/reset?
You can usually unlock the phone and then erase the config using the setting sbutton. Push the setting button, nafigate to the bottom of the list, select unlock. Use the keypad to enter the password which is cisco. Undwer network configuraiton there is an erase configuraiton option. Hope this helps. Joe ___ --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: Why is the internet connection important to LAN and PSTN calls?
JT == Joseph Tanner [EMAIL PROTECTED] writes: JT A slightly better (in my opinion) solution would be to code a pure JT caching dns server, whose sole purpose is to look up specific JT domains and resolve them to their ip address. It'll record the JT result, and will check every so often (once a minute, hour, day, JT whatever) and update its results. If it cannot get an answer, it JT keeps using the last known ip address. If anyone knows of a really JT bare-bones, standards-breaking dns server that would say, check a JT flat file database each time a request is made, we could run a JT daemon that would check the domains we need to resolve; if no JT answer is received, we just skip that line. That way the daemon JT will be sitting there waiting for a dns answer, and not asterisk. PowerDNS can do this (serve from a flat file). If I had to do this I'd probably go with SQLite, not a flat file backend, but either way it would work. PowerDNS can't do the first half, pre-query and put into the database, but a simple script could do that with SQLite. Just make a loop that hits one of your own recursive servers, fetching all the interesting records, and then maybe have a delay of a second or two between iterations. The delay isn't really necessary, serving from cache is fast. /Benny ___ --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] SipXPhone
Has anybody managed to get SipXPhone working with asterisk? I just cannot get it to work. It just keeps reporting an authentication failure even though all the details seem correct. The same settings work fine in X-Lite. Failing that, are there any opensource or reasonably priced SIP SDKs that people can recommend? I have the log from the SipXPhone. It says authentication required, yet it is definitely using the correct user/pass. Thanks, SIP Message Log SipUserAgent::sendUdp UDP SIP User Agent sent message: Remote Host:192.168.3.2 Port: 0 REGISTER sip:192.168.3.2 SIP/2.0 From: sip:[EMAIL PROTECTED];tag=a3f7929 To: sip:[EMAIL PROTECTED] Call-Id: 2e7cacadbcf327613f3430d1acf097c5 Cseq: 2146483648 REGISTER Contact: sip:[EMAIL PROTECTED];LINEID=adba43c7ad800946e9ca7305341f2df9;EXPIRES=0 Date: Wed, 12 Apr 2006 08:13:28 GMT Max-Forwards: 20 User-Agent: sipX/2.5.2 (WinNT) Accept-Language: en Supported: sip-cc, sip-cc-01, timer, replaces Via: SIP/2.0/UDP 192.168.2.190;branch=z9hG4bK-17752052e688c4514c7b61930a3f7132;rport Content-Length: 0 END Read SIP message: Remote Host:192.168.3.2 Port: 5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.2.190;branch=z9hG4bK-17752052e688c4514c7b61930a3f7132 From: sip:[EMAIL PROTECTED];tag=a3f7929 To: sip:[EMAIL PROTECTED];tag=as45b7a53c Call-ID: 2e7cacadbcf327613f3430d1acf097c5 CSeq: 2146483648 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 END SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.2.190;branch=z9hG4bK-17752052e688c4514c7b61930a3f7132 From: sip:[EMAIL PROTECTED];tag=a3f7929 To: sip:[EMAIL PROTECTED];tag=as45b7a53c Call-Id: 2e7cacadbcf327613f3430d1acf097c5 Cseq: 2146483648 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 Date: Wed, 12 Apr 2006 08:13:28 GMT END Read SIP message: Remote Host:192.168.3.2 Port: 5060 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.2.190;branch=z9hG4bK-17752052e688c4514c7b61930a3f7132 From: sip:[EMAIL PROTECTED];tag=a3f7929 To: sip:[EMAIL PROTECTED];tag=as45b7a53c Call-ID: 2e7cacadbcf327613f3430d1acf097c5 CSeq: 2146483648 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] WWW-Authenticate: Digest realm="asterisk", nonce="2409d1bd" Content-Length: 0 END SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.2.190;branch=z9hG4bK-17752052e688c4514c7b61930a3f7132 From: sip:[EMAIL PROTECTED];tag=a3f7929 To: sip:[EMAIL PROTECTED];tag=as45b7a53c Call-Id: 2e7cacadbcf327613f3430d1acf097c5 Cseq: 2146483648 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Www-Authenticate: Digest realm="asterisk", nonce="2409d1bd" Content-Length: 0 Date: Wed, 12 Apr 2006 08:13:28 GMT END SIP User agent delayed dispatch message: SIP/2.0 401 Unauthorized From: sip:[EMAIL PROTECTED];tag=a3f7929 To: sip:[EMAIL PROTECTED];tag=as45b7a53c Call-Id: 2e7cacadbcf327613f3430d1acf097c5 Cseq: 2146483648 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Www-Authenticate: Digest realm="asterisk", nonce="2409d1bd" Content-Length: 0 Date: Wed, 12 Apr 2006 08:13:28 GMT END SipUserAgent::sendUdp UDP SIP User Agent sent message: Remote Host:192.168.3.2 Port: 0 REGISTER sip:192.168.3.2 SIP/2.0 From: sip:[EMAIL PROTECTED];tag=49195c41 To: sip:[EMAIL PROTECTED] Call-Id: 2c0309cd5d962e6a6d773eb85aba9617 Cseq: 1 REGISTER Contact: sip:[EMAIL PROTECTED];LINEID=adba43c7ad800946e9ca7305341f2df9 Expires: 600 Date: Wed, 12 Apr 2006 08:13:29 GMT Max-Forwards: 20 User-Agent: sipX/2.5.2 (WinNT) Accept-Language: en Supported: sip-cc, sip-cc-01, timer, replaces Via: SIP/2.0/UDP 192.168.2.190;branch=z9hG4bK-0172b9b42586f951f1904d0072871fcd;rport Content-Length: 0 END Read SIP message: Remote Host:192.168.3.2 Port: 5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.2.190;branch=z9hG4bK-0172b9b42586f951f1904d0072871fcd From: sip:[EMAIL PROTECTED];tag=49195c41 To: sip:[EMAIL PROTECTED];tag=as574d6363 Call-ID: 2c0309cd5d962e6a6d773eb85aba9617 CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 END SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.2.190;branch=z9hG4bK-0172b9b42586f951f1904d0072871fcd From: sip:[EMAIL PROTECTED];tag=49195c41 To: sip:[EMAIL PROTECTED];tag=as574d6363 Call-Id: 2c0309cd5d962e6a6d773eb85aba9617 Cseq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS,
[Asterisk-Users] Where is the difference sip.conf - Real-time ?
I have two phones (111 and 112) on a LAN, and I have on a users site a phone 333. phone 111 uses sip.conf, while 112 uses real-time set-up. 111 can call 333 AND the audio is working 112 can call 333 but audio is just white noise. 333 can call 111 or 112 and audio is working. The phones are identically set-up (just user name = phone number and password are different) sip.conf (for 111 - all remarked lines removed) [general] context=default; Default context for incoming calls port=5060; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes; Enable DNS SRV lookups on outbound calls tos=lowdelay ; lowdelay,throughput,reliability,mincost,none maxexpirey=7200; Max length of incoming registration we allow defaultexpirey=3600; Default length of incoming/outoing registration videosupport=yes; Turn on support for SIP video disallow=all; First disallow all codecs allow=ulaw; Allow codecs in order of preference allow=alaw allow=g729 allow=gsm rtcachefriends=yes rtnoupdate=yes rtautoclear=yes externip = 59.14.2.1 localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks [111] type=friend username=hotline secret=I-know-it canreinvite=no host=dynamic dtmfmode=rfc2833 [EMAIL PROTECTED] nat=yes callgroup=1 pickupgroup=1 callerid=Ronald Hotline,111 qualify=1000 Real-time for 112: name=112 callerid=Ronald Hotline,112 canreinvite=yes context=default dtmfmode=rfc2833 host=dynamic language=en [EMAIL PROTECTED] nat=yes pickupgroup=1 port=5060 qualify=1000 secret=I-know-it type=friend username=112 disallow=all allow=ulaw;alaw;g729;gsm cancallforward=yes Which of the settings cause the different behaviour? Which settings should I change (maybe not related to the problem)? bye Ronald ___ --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] call center running Asterisk - sound quality- critical!
Hi, how do you record calls? Monitor app. or MixMonitor or something else? How does your storage backend looks like? What kind of channels do you use? Do you record IAX2 channels? Regards, Tamas Wai Wu wrote: You got to be kidding about 53 calls being recorded at sametime is an issue. I have done at least twice as many on my dual xeon 3.4Ghz system and had no problem as clients like to record every call that goes through the system. Then again, in my system, the in and out channels are mixed first before they are written to the disk. From: [EMAIL PROTECTED] on behalf of Matt Roth Sent: Tue 4/11/2006 5:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] call center running Asterisk - sound quality- critical! On 4/10/06, Dov Bigio [EMAIL PROTECTED] wrote: Hi, I am using Asterisk for a call center on a Dual Xeon machine.. I currently have 109 active channels 53 active calls Every body is complaining about quality and cpu is around 80% idle. Is there any tuning I can do??? Besides that, Asterisk normally goes down once or twice per day... Thank you Dov C F wrote: From what you say it sounds that the problem is not with asteisk, but the way it's configured. Asterisk should *never* go down that often. Asterisk as a normal PBX should run without a restart for as long as there is power to the box, in the case of a call center if I would hear of a restart once a week I would accept it, but still would look for ways of improving it beyond that. You complain about call quality, what type of phones are thes? What codec? are they all local? Dov, I agree with the first response. Your system is failing at an abnormal rate. Please share more information about your setup so that we can help you. Hardware, software, OS, configuration...there's no such thing as too many details when trying to work out these problems via a mailing list. Information about what tasks you are asking Asterisk to perform and how you have it configured to do so is vital. In particular, I'm curious to know if you're recording the calls using the Monitor() application? 53 concurrent calls being recorded directly to disk is about where things start to go south (it's an I/O bottleneck, not CPU). If you have a Digium card in the box, make sure that it's not sharing an interrupt with any other hardware. The list and the wiki both have plenty of information to help you with that. Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer ___ --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 channel unavailable/busy/really not there
Steve Kennedy wrote:
Is there a way to differentiate between a SIP address which hasn't
registered (but is within sip.conf) and one that's not there at all
(i.e. not in sip.conf) using a straight dialplan.
I'd like to differentiate actions depending the state of a SIP device
and whether it's in my config or not (if that makes sense, basic automap
of dial-in lines to sip phones, but if they've turned it off, different
error from a line not mapped to a phone).
Asterisk-1.2.6 ...
Steve
Might do the trick for you:
-= Info about application 'ChanIsAvail' =-
[Synopsis]
Check channel availability
[Description]
ChanIsAvail(Technology/resource[Technology2/resource2...][|options]):
This application will check to see if any of the specified channels are
available. The following variables will be set by this application:
${AVAILCHAN} - the name of the available channel, if one exists
${AVAILORIGCHAN} - the canonical channel name that was used to create
the channel
${AVAILSTATUS} - the status code for the available channel
Options:
s - Consider the channel unavailable if the channel is in use at all
j - Support jumping to priority n+101 if no channel is available
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[Asterisk-Users] iax2 show netstats
Hi guys, i've been using iax2 show netstats and i wonder if someone could explain what all these means, just in case i have them wrong. Because i am looking for something that tells me that there is delay , and/or packet loss. LOCAL - REMOTE Channel RTT Jit Del Lost % Drop OOO Kpkts Jit Del Lost % Drop OOO Kpkts IAX2/iaxBBG-16384 1000 -10-1 -1 0 -1 000 0 0 00 0 IAX2/iaxBBG-16386 16 -10-1 -1 0 -1 10 40 0 0 00 0 -- thanks, yusuf ___ --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] Where is the difference sip.conf - Real-time ?
Hello, Have you tried to delete rtnoupdate and rtautoclear lines? I'm using realtime, with caching for sip but without those 2 lines, and works perfectly. Another point : verify that you have the field fullcontact in your realtime sip table. Bye, Alban Elziere I have two phones (111 and 112) on a LAN, and I have on a users site a phone 333. phone 111 uses sip.conf, while 112 uses real-time set-up. 111 can call 333 AND the audio is working 112 can call 333 but audio is just white noise. 333 can call 111 or 112 and audio is working. The phones are identically set-up (just user name = phone number and password are different) sip.conf (for 111 - all remarked lines removed) [general] context=default; Default context for incoming calls port=5060; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes; Enable DNS SRV lookups on outbound calls tos=lowdelay ; lowdelay,throughput,reliability,mincost,none maxexpirey=7200; Max length of incoming registration we allow defaultexpirey=3600; Default length of incoming/outoing registration videosupport=yes; Turn on support for SIP video disallow=all; First disallow all codecs allow=ulaw; Allow codecs in order of preference allow=alaw allow=g729 allow=gsm rtcachefriends=yes rtnoupdate=yes rtautoclear=yes externip = 59.14.2.1 localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks [111] type=friend username=hotline secret=I-know-it canreinvite=no host=dynamic dtmfmode=rfc2833 [EMAIL PROTECTED] nat=yes callgroup=1 pickupgroup=1 callerid=Ronald Hotline,111 qualify=1000 Real-time for 112: name=112 callerid=Ronald Hotline,112 canreinvite=yes context=default dtmfmode=rfc2833 host=dynamic language=en [EMAIL PROTECTED] nat=yes pickupgroup=1 port=5060 qualify=1000 secret=I-know-it type=friend username=112 disallow=all allow=ulaw;alaw;g729;gsm cancallforward=yes Which of the settings cause the different behaviour? Which settings should I change (maybe not related to the problem)? bye Ronald ___ --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] billing with PostgreSQL
Hello to all Im looking for a billing tool for Asterisk, that works with PostgreSQL. All the tools I found in www.asteriskbilling.com just work with MySQL :( Do you know a nice billing tool for Asterisk with PostgreSQL? Thanks Joao Pereira ___ --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: Received VNAK: resending outstanding frames?
On Wed, 2006-04-12 at 07:58 +0100, Tim Panton wrote: There is a manual at: http://www.centralitycomm.com/solutions/Download/documents/product/ PA168SUserguideEng.pdf Tim Panton [EMAIL PROTECTED] I'm now outside the network again and have run iax2 debug. Below are the results. Notice how after the Raw Hangup there is a 30 second pause, then it retries, and when it gets to the VNAK then it repeats the same message constantly for another 30 seconds (snipped the 5000+ lines of course), and then gets the Raw Hangup again. Ad infinitum. I have uploaded the log here: http://www.users.on.net/~lncoshea/carey/asterisk-log.txt Does the log help? Anyone have any ideas going from the log? Regards, Carey O'Shea. PS: Thanks Tim, I worked out how to reset the phone a few hours ago, the manual was wrong for my particular model, I had to press hash (#) _before_ power on, see here: http://forums.whirlpool.net.au/forum-replies.cfm?t=504889 ___ --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] SIP call hangup from asterisk CLI
Hi,We are using Vicidial and sometime even when agent disconnects, outgoing call originated by dialer is still active. Since call was initiated by dialer and then bought into meetme conference of agent and we can't corelate this call to any agent channel. When agents are dialing, channels doesn't show calls vicidial2*CLI show channels Channel Location State Application(Data) Local/[EMAIL PROTECTED] [EMAIL PROTECTED]:1 Up MeetMe(8600051|q) Local/[EMAIL PROTECTED] [EMAIL PROTECTED]:3 Up Wait(3600) SIP/primus-8f43 (None) Ringing AppDial((Outgoing Line)) Local/[EMAIL PROTECTED] [EMAIL PROTECTED] Ring Dial(SIP/[EMAIL PROTECTED]||t Local/[EMAIL PROTECTED] [EMAIL PROTECTED]:1 Down (None) Local/[EMAIL PROTECTED] [EMAIL PROTECTED]:1 Up MeetMe(8600053|q) Local/[EMAIL PROTECTED] [EMAIL PROTECTED]:3 Up Wait(3600) SIP/primus-00fe (None) Ringing AppDial((Outgoing Line)) Local/[EMAIL PROTECTED] [EMAIL PROTECTED] Ring Dial(SIP/[EMAIL PROTECTED]||t Local/[EMAIL PROTECTED] [EMAIL PROTECTED]:1 Down (None) Local/[EMAIL PROTECTED] [EMAIL PROTECTED]:1 Up MeetMe(8600054|q) Local/[EMAIL PROTECTED] [EMAIL PROTECTED]:3 Up Wait(3600) SIP/primus-95db [EMAIL PROTECTED]:1 Up MeetMe(8600051) Zap/pseudo-122590356 [EMAIL PROTECTED]:1 Rsrvd (None) SIP/agent7-44fa [EMAIL PROTECTED]:1 Up MeetMe(8600055) SIP/primus-0a7c [EMAIL PROTECTED]:1 Up MeetMe(8600053) SIP/primus-7c73 [EMAIL PROTECTED]:1 Up MeetMe(8600054) Local/[EMAIL PROTECTED] [EMAIL PROTECTED]:1 Up MeetMe(8600052|q) Local/[EMAIL PROTECTED] [EMAIL PROTECTED]:3 Up Wait(3600) SIP/primus-2ed8 [EMAIL PROTECTED]:1 Up MeetMe(8600052) Zap/pseudo-104079549 [EMAIL PROTECTED]:1 Rsrvd (None) SIP/agent1-32b5 [EMAIL PROTECTED]:1 Up MeetMe(8600054) Zap/pseudo-204709889 [EMAIL PROTECTED]:1 Rsrvd (None) SIP/agent8-d3ab [EMAIL PROTECTED]:1 Up MeetMe(8600056) SIP/agent5-ec77 [EMAIL PROTECTED]:1 Up MeetMe(8600051) Zap/pseudo-92046 [EMAIL PROTECTED]:1 Rsrvd (None) SIP/agent3-2df5 [EMAIL PROTECTED]:1 Up MeetMe(8600053) Zap/pseudo-204290210 [EMAIL PROTECTED]:1 Rsrvd (None) SIP/agent2-4ff6 [EMAIL PROTECTED]:1 Up MeetMe(8600052) SIP/primus-fc90 [EMAIL PROTECTED]:1 Up MeetMe(8600051) Zap/pseudo-170346238 [EMAIL PROTECTED]:1 Rsrvd (None) 31 active channels After agents have logged outvicidial2*CLI show channelsChannel Location State Application(Data)SIP/primus-fc90 [EMAIL PROTECTED]:1 Up MeetMe(8600051)Zap/pseudo-170346238 [EMAIL PROTECTED]:1 Rsrvd (None)Calls doesn't show channelsvicidial2*CLI sip show channelsPeer User/ANR Call ID Seq (Tx/Rx) Form Hold Last Message 203.63.248.197 122001 20a58a2e251 00651/0 unkn No203.196.128.56 6135625116 5420f80176e 00102/0 g729 No Tx: ACKcalls doesn't show channel CLIsip show channel 5420f80176e * SIP Call Direction: Outgoing Call-ID: [EMAIL PROTECTED] Our Codec Capability: 256 Non-Codec Capability: 1 Their Codec Capability: 256 Joint Codec Capability: 256 Format g729 Theoretical Address: 203.196.128.56:5060 Received Address: 203.196.128.56:5060 NAT Support: RFC3581 Audio IP: 220.227.174.4 (local) Our Tag: as7a55ac7a Their Tag: 29258 SIP User agent: Username: 61356251162 Peername: 90340 Original uri: sip:[EMAIL PROTECTED]:5060 Need Destroy: 0 Last Message: Tx: ACK Promiscuous Redir: No Route: sip:[EMAIL PROTECTED];ftag=as7a55ac7a;lr=on DTMF Mode: rfc2833 SIP Options: (none)BUT ONE THING IS COMMON IS THAT OLDEST SIP CALL WILL COME IN THE BOTTOM OF THE LIST of COMMAND sip show channels (agents will be above it) so it is hung and needs to be destroyed manually. Also channel corresponding to this call will also come in the bottom of SHOW Channels command for same technology i.e. it will be last SIP/XYZ entry so to destroy this call lets try destroy last SIP channel entry.vicidial2*CLI soft hangup SIP/primus-fc90Requested Hangup on channel 'SIP/primus-fc90' -- Hungup 'Zap/pseudo-1703462386' == Spawn extension (default, 8600051, 1) exited non-zero on 'SIP/primus-fc90' -- Executing DeadAGI(SIP/primus-fc90, call_log.agi|h) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/call_log.agi -- AGI Script call_log.agi completed, returning 0 -- Executing DeadAGI(SIP/primus-fc90, VD_hangup.agi|h) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/VD_hangup.agi -- AGI Script VD_hangup.agi completed, returning 0vicidial2*CLI sip show channelsPeer User/ANR Call ID Seq (Tx/Rx) Form Hold Last Message 0 active SIP channelsIT WORKS!! A crude way but very important to save 100 of dollars of hung call while agent are dialing. You can always do stop now but then whole operations will stop.Dont know why this happens in first place but atleast I have seen it coming twice and now keep a vigil that no call is below the agents in sip show channels, it there is any it means its a hung call costing you money Abhimanyu ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options
Re: [Asterisk-Users] Where is the difference sip.conf - Real-time ?
Alban wrote: Hello, Have you tried to delete rtnoupdate and rtautoclear lines? I'm using realtime, with caching for sip but without those 2 lines, and works perfectly. Another point : verify that you have the field fullcontact in your realtime sip table. Bye, Alban Elziere While I compiled the message, I discovered the difference already in canreinvite=yes/no I could test it, and it was the problem! I found than that if you have the phones behind asterisk you MUST have canreinvite=no to force, the rtp stream to go through asterisk and not to try to bypass it. I use bypass so that the users are directly connected to the gateways without bothering my servers bandwidth. rtnoupdate=yes ; do not send the update request over realtime. rtautoclear=yes ; Auto-Expire friends created on the fly on the same schedule ; as if it had just registered when the registration expires ; the friend will vanish from the configuration until requested ; again. If set to an integer, friends expire ; within this number of seconds instead of the ; same as the registration interval I have two phones (111 and 112) on a LAN, and I have on a users site a phone 333. phone 111 uses sip.conf, while 112 uses real-time set-up. 111 can call 333 AND the audio is working 112 can call 333 but audio is just white noise. 333 can call 111 or 112 and audio is working. The phones are identically set-up (just user name = phone number and password are different) sip.conf (for 111 - all remarked lines removed) [general] context=default; Default context for incoming calls port=5060; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes; Enable DNS SRV lookups on outbound calls tos=lowdelay ; lowdelay,throughput,reliability,mincost,none maxexpirey=7200; Max length of incoming registration we allow defaultexpirey=3600; Default length of incoming/outoing registration videosupport=yes; Turn on support for SIP video disallow=all; First disallow all codecs allow=ulaw; Allow codecs in order of preference allow=alaw allow=g729 allow=gsm rtcachefriends=yes rtnoupdate=yes rtautoclear=yes externip = 59.14.2.1 localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks [111] type=friend username=hotline secret=I-know-it canreinvite=no host=dynamic dtmfmode=rfc2833 [EMAIL PROTECTED] nat=yes callgroup=1 pickupgroup=1 callerid=Ronald Hotline,111 qualify=1000 Real-time for 112: name=112 callerid=Ronald Hotline,112 canreinvite=yes context=default dtmfmode=rfc2833 host=dynamic language=en [EMAIL PROTECTED] nat=yes pickupgroup=1 port=5060 qualify=1000 secret=I-know-it type=friend username=112 disallow=all allow=ulaw;alaw;g729;gsm cancallforward=yes Which of the settings cause the different behaviour? Which settings should I change (maybe not related to the problem)? bye Ronald ___ --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 -- Ronald Wiplinger (CEO of ELMIT) http://www.elmit.com http://voip.elmit.com http://e-paper.elmit.com Tel. (M) +886.939.775.516 (O) +886.2.2835.7765 (ENUM) or FWD 511208 - I'm a SpamCon Foundation Member, #694, Verify it at http://www.spamcon.org PS: Spam prevention! Our system is protected with a spam prevention program. If you send us an e-mail, our system will send you a confirmation message back. Just reply to this confirmation message please. After receiving this confirmation message, our system will send the hold message (one) and all future messages (after the received confirmation message) to me without asking you again. ___ --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] Failed to recieve Fax: Asterisk - IAXModem - Hylafax
Hi, I've tired to forward a Fax from Asterisk to Hylafax. It works so far until I tried with a Fax machine. I just got error shown in the log below. I'm not sure why. I've tested it with other 6 machines and they all work fine. Do you have any idea why? Pim Hylafax Session log: Apr 12 11:16:48.82: [ 5933]: SESSION BEGIN 00078 492212601860 Apr 12 11:16:48.82: [ 5933]: HylaFAX (tm) Version 4.2.5 Apr 12 11:16:48.82: [ 5933]: CallID: 2283683381 NONE NONE 444 Apr 12 11:16:48.82: [ 5933]: MODEM set XON/XOFF/FLUSH: input ignored, output disabled Apr 12 11:16:48.82: [ 5933]: -- [4:ATA\r] Apr 12 11:16:52.93: [ 5933]: -- [7:CONNECT] Apr 12 11:16:52.93: [ 5933]: ANSWER: FAX CONNECTION DEVICE '/dev/ttyIAX7' Apr 12 11:16:52.93: [ 5933]: STATE CHANGE: ANSWERING - RECEIVING Apr 12 11:16:52.93: [ 5933]: RECV FAX: begin Apr 12 11:16:52.93: [ 5933]: -- HDLC32:FF C0 04 AD 00 55 12 9E 36 86 62 82 1A 04 14 2E B6 94 04 6A A6 4E CE 96 F6 76 04 2C 74 4C 74 AC Apr 12 11:16:52.93: [ 5933]: -- data [32] Apr 12 11:16:52.93: [ 5933]: -- data [2] Apr 12 11:16:53.99: [ 5933]: -- [7:CONNECT] Apr 12 11:16:53.99: [ 5933]: -- HDLC23:FF C0 02 EC B6 A6 26 F6 B6 1A 82 92 04 04 04 04 04 04 04 04 04 04 04 Apr 12 11:16:53.99: [ 5933]: -- data [23] Apr 12 11:16:53.99: [ 5933]: -- data [2] Apr 12 11:16:54.81: [ 5933]: -- [7:CONNECT] Apr 12 11:16:54.81: [ 5933]: -- HDLC13:FF C8 01 00 73 5F 23 01 FB C1 01 01 18 Apr 12 11:16:54.81: [ 5933]: -- data [13] Apr 12 11:16:54.81: [ 5933]: -- data [2] Apr 12 11:16:55.41: [ 5933]: -- [2:OK] Apr 12 11:16:55.41: [ 5933]: -- [9:AT+FRH=3\r] Apr 12 11:17:02.41: [ 5933]: -- [0:] Apr 12 11:17:02.41: [ 5933]: MODEM Empty line Apr 12 11:17:02.41: [ 5933]: MODEM TIMEOUT: waiting for v.21 carrier Apr 12 11:17:02.41: [ 5933]: -- data [1] Apr 12 11:17:02.43: [ 5933]: -- [2:OK] Apr 12 11:17:02.43: [ 5933]: DELAY 1500 ms Apr 12 11:17:03.93: [ 5933]: -- [9:AT+FTH=3\r] Apr 12 11:17:03.93: [ 5933]: -- [7:CONNECT] Apr 12 11:17:03.93: [ 5933]: -- HDLC32:FF C0 04 AD 00 55 12 9E 36 86 62 82 1A 04 14 2E B6 94 04 6A A6 4E CE 96 F6 76 04 2C 74 4C 74 AC Apr 12 11:17:03.93: [ 5933]: -- data [32] Apr 12 11:17:03.93: [ 5933]: -- data [2] Apr 12 11:17:05.01: [ 5933]: -- [10:NO CARRIER] Apr 12 11:17:05.01: [ 5933]: MODEM No carrier Apr 12 11:17:05.01: [ 5933]: DELAY 1500 ms Apr 12 11:17:06.51: [ 5933]: -- [9:AT+FTH=3\r] Apr 12 11:17:06.51: [ 5933]: -- [10:NO CARRIER] Apr 12 11:17:06.51: [ 5933]: DELAY 1500 ms Apr 12 11:17:08.01: [ 5933]: -- [9:AT+FTH=3\r] Apr 12 11:17:08.01: [ 5933]: -- [5:ERROR] Apr 12 11:17:08.01: [ 5933]: RECV FAX: RSPREC error/got EOT Apr 12 11:17:08.01: [ 5933]: RECV FAX: end Apr 12 11:17:08.01: [ 5933]: SESSION END HylaFAX(tm) Users Mailing List ___ To subscribe/unsubscribe, click http://lists.hylafax.org/cgi-bin/lsg2.cgi On UNIX: mail -s unsubscribe [EMAIL PROTECTED] /dev/null *To learn about commercial HylaFAX(tm) support, mail [EMAIL PROTECTED] ___ --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 BRI in the USA
I looked into it last year, and in Texas BRIs are only about $55/mo and include the optional calling features for which you pay extra with POTS. (Caller ID, call forwarding, etc) The roblem I ran into was that Euro standard hardware does not work on US standard BRI lines. And I could find literally no workable hardware. Since FXOs historixally have been a great weakness for Asterisk I think that BRIs would be a great alternative...if the hardware existed. I ended up simply call forwarding my remaining POTS lines to DID privided by an ITSP. These come in over my DSL line via IAX2. Michael Graves On Tue, 11 Apr 2006 13:38:07 -0400, Rusty Dekema wrote: I dunno if it's THAT bad. I had a BRI line in the (relatively) podunk town of Kalamazoo, Michigan back in 1998. Sure, it took the phone company a couple of weeks to provision the service, but it takes the phone company a couple of weeks to do most anything in my experience. The price was something like $45/mo for two channels and the same per-call/per-minute pricing scheme as POTS (no per-minute fee for incoming and local calls, regular LD pricing for LD, and 800 local outgoing calls included after which it was something like 6 cents per call). The switch on ILEC's end was a DMS-100 implementing National ISDN-1. I really put the ISDN line through its paces too -- voice, data, bonded data, automatic bonding and de-bonding to allow for voice calls -- and everything always worked flawlessly. I don't know what today's pricing is like for ISDN BRI what with all of the various mergers (at the time, I had service from Ameritech), but unless it has gone up significantly, BRI seems like the perfect type of trunk for an Asterisk system too small for a T1/PRI to be an affordable option. -Rusty ___ --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] SIP call hangup from asterisk CLI
Hi all,My architecture is:PSTN-E1OldPBXE1-AsteriskI've a similar problem, SIP user agents using X-Lite:Sip User Agent A calls to PSTN user BB user hangs the call A user starts listening busy indications on the phone, and if he doesn't hangup correctly on Xlite The calls seems to be alive Only solved it with soft hangup, and that is not an acceptable solution. I have on user that seems to have turned off the pc ( at least he reports me that) and the call (at least on Asterisk CDR) remained alivedidn't disconnectIt is working fine only if SIP user agents dials to an extension in the Old PBX, that case if the called party Hangs, the Old Pbx immediately sends a DISCONNECT message to Asterisk and the call hangs... I hope someone could help US.Best regards,Marco MoutaOn 4/12/06, Abhimanyu Rapria [EMAIL PROTECTED] wrote:Hi,We are using Vicidial and sometime even when agent disconnects, outgoing call originated by dialer is still active. Since call was initiated by dialer and then bought into meetme conference of agent and we can't corelate this call to any agent channel. When agents are dialing, channels doesn't show calls vicidial2*CLI show channels Channel Location State Application(Data) Local/[EMAIL PROTECTED] [EMAIL PROTECTED]:1 Up MeetMe(8600051|q) Local/[EMAIL PROTECTED] [EMAIL PROTECTED]:3 Up Wait(3600) SIP/primus-8f43 (None) Ringing AppDial((Outgoing Line)) Local/[EMAIL PROTECTED] [EMAIL PROTECTED] Ring Dial(SIP/[EMAIL PROTECTED]||t Local/[EMAIL PROTECTED] [EMAIL PROTECTED]:1 Down (None) Local/[EMAIL PROTECTED] [EMAIL PROTECTED]:1 Up MeetMe(8600053|q) Local/[EMAIL PROTECTED] [EMAIL PROTECTED]:3 Up Wait(3600) SIP/primus-00fe (None) Ringing AppDial((Outgoing Line)) Local/[EMAIL PROTECTED] [EMAIL PROTECTED] Ring Dial(SIP/[EMAIL PROTECTED]||t Local/[EMAIL PROTECTED] [EMAIL PROTECTED]:1 Down (None) Local/[EMAIL PROTECTED] [EMAIL PROTECTED]:1 Up MeetMe(8600054|q) Local/[EMAIL PROTECTED] [EMAIL PROTECTED]:3 Up Wait(3600) SIP/primus-95db [EMAIL PROTECTED]:1 Up MeetMe(8600051) Zap/pseudo-122590356 [EMAIL PROTECTED]:1 Rsrvd (None) SIP/agent7-44fa [EMAIL PROTECTED]:1 Up MeetMe(8600055) SIP/primus-0a7c [EMAIL PROTECTED]:1 Up MeetMe(8600053) SIP/primus-7c73 [EMAIL PROTECTED]:1 Up MeetMe(8600054) Local/[EMAIL PROTECTED] [EMAIL PROTECTED]:1 Up MeetMe(8600052|q) Local/[EMAIL PROTECTED] [EMAIL PROTECTED]:3 Up Wait(3600) SIP/primus-2ed8 [EMAIL PROTECTED]:1 Up MeetMe(8600052) Zap/pseudo-104079549 [EMAIL PROTECTED]:1 Rsrvd (None) SIP/agent1-32b5 [EMAIL PROTECTED]:1 Up MeetMe(8600054) Zap/pseudo-204709889 [EMAIL PROTECTED]:1 Rsrvd (None) SIP/agent8-d3ab [EMAIL PROTECTED]:1 Up MeetMe(8600056) SIP/agent5-ec77 [EMAIL PROTECTED]:1 Up MeetMe(8600051) Zap/pseudo-92046 [EMAIL PROTECTED]:1 Rsrvd (None) SIP/agent3-2df5 [EMAIL PROTECTED]:1 Up MeetMe(8600053) Zap/pseudo-204290210 [EMAIL PROTECTED]:1 Rsrvd (None) SIP/agent2-4ff6 [EMAIL PROTECTED]:1 Up MeetMe(8600052) SIP/primus-fc90 [EMAIL PROTECTED]:1 Up MeetMe(8600051) Zap/pseudo-170346238 [EMAIL PROTECTED]:1 Rsrvd (None) 31 active channels After agents have logged outvicidial2*CLI show channelsChannel Location State Application(Data)SIP/primus-fc90 [EMAIL PROTECTED]:1 Up MeetMe(8600051)Zap/pseudo-170346238 [EMAIL PROTECTED]:1 Rsrvd (None)Calls doesn't show channelsvicidial2*CLI sip show channelsPeer User/ANR Call ID Seq (Tx/Rx) Form Hold Last Message 203.63.248.197 122001 20a58a2e251 00651/0 unkn No203.196.128.56 6135625116 5420f80176e 00102/0 g729 No Tx: ACK calls doesn't show channel CLIsip show channel 5420f80176e * SIP Call Direction: Outgoing Call-ID: [EMAIL PROTECTED] Our Codec Capability: 256 Non-Codec Capability: 1 Their Codec Capability: 256 Joint Codec Capability: 256 Format g729 Theoretical Address: 203.196.128.56:5060 Received Address: 203.196.128.56:5060 NAT Support: RFC3581 Audio IP: 220.227.174.4 (local) Our Tag: as7a55ac7a Their Tag: 29258 SIP User agent: Username: 61356251162 Peername: 90340 Original uri: sip:[EMAIL PROTECTED]:5060 Need Destroy: 0 Last Message: Tx: ACK Promiscuous Redir: No Route: sip:[EMAIL PROTECTED];ftag=as7a55ac7a;lr=on DTMF Mode: rfc2833 SIP Options: (none)BUT ONE THING IS COMMON IS THAT OLDEST SIP CALL WILL COME IN THE BOTTOM OF THE LIST of COMMAND sip show channels (agents will be above it) so it is hung and needs to be destroyed manually. Also channel corresponding to this call will also come in the bottom of SHOW Channels command for same technology i.e. it will be last SIP/XYZ entry so to destroy this call lets try destroy last SIP channel entry.vicidial2*CLI soft hangup SIP/primus-fc90Requested Hangup on channel 'SIP/primus-fc90' -- Hungup 'Zap/pseudo-1703462386' == Spawn extension (default, 8600051, 1) exited non-zero on 'SIP/primus-fc90' -- Executing DeadAGI(SIP/primus-fc90, call_log.agi|h) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/call_log.agi -- AGI Script call_log.agi completed,
[Asterisk-Users] free video (soft) phone available?
I am using eyebeam and I am happy with it. However, it is boring just to talk to my son in the other room. Whenever I try to convince somebody to buy eyebeam, they are scared of the price. Is there a free video soft phone available, that will work with eyebeam / asterisk? bye Ronald Wiplinger ___ --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] Texas User Group
I'm in Houston as well. Would be very interested. Michael On Tue, 11 Apr 2006 21:00:49 -0500 (CDT), Aaron Daniel wrote: I'm in Huntsville... close enough to Houston. Aaron On Tue, 11 Apr 2006, Lacy Moore - Aspendora wrote: I'm in Houston. On 4/11/06, Ryan Burke [EMAIL PROTECTED] wrote: I'm interested but I'm in the Dallas area. Are there any in the Dallas area anyone knows of? Ryan - Original Message - *From:* Bruce Reeves [EMAIL PROTECTED] *To:* Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com *Sent:* Monday, April 10, 2006 12:51 PM *Subject:* [Asterisk-Users] Texas User Group I am wondering if any of the Texas user groups have members in the North West part of the state. I am in the Amarillo area and would like to find some othere in this area, maybe even start a user group in this area. -- Bruce Reeves Nortex Networks -- ___ --Bandwidth and Colocation provided by Easynews.com http://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 http://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 ___ --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] g.726 codec not working in one direction
Hi, Iam using Asterisk Asterisk 1.2.5 Iam calling: NOT OK: phone A -ulaw - Asterik-A - gsm - Asterisk-B - g.726 - POTS phone B NO sound from from phone A to phone B, phone B to phone A works If iam using ulaw to connect from Asterisk-B to POTS phone B everythink is OK: OK: phone A -ulaw - Asterik-A - gsm - Asterisk-B - ulaw -POTS phone B Any idea how this can happened? If additional information required please ask.. best regards Thomas ___ --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] help -- voicemail
Hi, Did someone experience that Asterisk OS 1.2.5 voicemail issues? Problem description: Some one call to the extensions 200, After 10 sec ring then go to voicemail [EMAIL PROTECTED] Announcement Please leave me a messages.blar blar.. When I completed to leave a message... IF : I press the pound #key ... Then it says Transfer IF : I Press the zero 0key Then it say Please confirm your recording IF : I hangup after leaving a message...then things get normal. What is this Funny. Pls some one reply. ___ --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] Snap for Asterisk
Bartosz, When set up correctly the phone on your desk should ring and then when you pick it you will be connected to the number you dialed. This is all done via the origination command. Did you configure the Asterisk management interface both in Asterisk and Snap? The best approach to debugging is to log into Asterisk via asterisk -vr and watch what is happening when you try to dial. Best of luck to you, Mitchel On 4/11/06, Bartosz Piec [EMAIL PROTECTED] wrote: [EMAIL PROTECTED] wrote: I've been working on a project for Asterisk for some time and it is finally ready for a beta release. Any feedback is well appreciated. At the basic core it's a Dialer for Windows. I'll be adding more features quickly, but I wanted to keep everything simple and stable in this first release. What is this for? I have set it up, trying to dial some number, a balloon tip says it is dialing but nothing happens. What am I doing wrong? -- Best regards, Bartosz Piec ___ --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 BRI in the USA
Mark Coccimiglio wrote: Hey all, It such a shame that BRI technology is such a flop in the USA. For a small office such as mine it would be a great product. So her goes my question What is a known asterisk working BRI card that will operate in the USA. I need to weigh price/quality. I need to do DID/DDI (or what ever you want to call it). Asterisk will do everything else I need. The ILEC has at the other end a DMS-100. I have been having all kinds of problems using POTS lines that I will consider it an investment to move to a more digital connection. I am considering going the VoIP route (Vonage, Broadvoice, etc...) but before I commit either way I'm exploring all my options. Your opnion matter here to please let me know. Mark Coccimiglio [EMAIL PROTECTED] sip:[EMAIL PROTECTED] In theory, bri makes a lot of sense, flawless disconnect detection, 8 directory numbers, placing a caller on hold is done by the switch and doesn't tie up a line, and on. But my experience hasn't been all that encouraging. I've had a bri line in Seattle for about 4 years or so. The local Qwest co switch is a 5ESS. It took about 3 months to get it properly provisioned for a couple of Lucent 970 phones. And that's only because one of their techs felt sorry for me, came in on a Saturday and followed the provisioning instructions I found on a telecom site. I'm now convinced that if I had provided a copy of the 5ESS screens with my order and they actually got to the tech, I would not have had a problem. Next, I got a Eicon Diva board and tried to get the hisax kernel driver working. It's ni-1 implementation, the only one I could find, isn't very complete. It was written by a guy in Australia using only an isdn simulator, a significant accomplishment. It appears that it's intent was to just place outgoing data calls. At best, it would signal my POTS line, but give up during call setup. Unfortunately, our layer 3 protocol is secret and the specs have to be purchased from Telcordia. The last time I checked, assuming I chose the right publication, it was about $600. Adding ni-1 to either Junghanns' work or visdn probably wouldn't be that difficult given the specs. Both of these drivers happily talk to my $10 HFC-PCI card and negotiate, then assign a tei to the phone. So, the existing layer 1 and 2 stuff configured as point-to-multipoint seems to work fine. My understanding is that all bri's, both here and in Europe, use the same Q.920/921 standards. It's layer 3 that's different. Given an ni-1 protocol stack, hardware like Junghanns' 4 and 8 port cards should work with Asterisk here too. So, - Telcordia NI-1 specs - Some code - Detailed provisioning instructions for at least a 5ESS and a DMS-100 Anybody interested? Steve ___ --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] Oh323 inband DTMF
Hi group! Does DTMF inband work with oh323 channel driver ver. 0.6.7? If yes, how to enable it, make it work? I have tried with inBandDTMF=yes in general context of oh323.conf, but I get this message when I * is starting. [chan_oh323.so] = (InAccess Networks OpenH323 Channel Driver) == Parsing '/etc/asterisk/rtp.conf': Found == Parsing '/etc/asterisk/oh323.conf': Found Apr 12 13:38:17 NOTICE[3622]: chan_oh323.c:4813 reload_config: Ignoring unknown H.323 [general] keyword 'silenceSuppression', line 49. Apr 12 13:38:17 NOTICE[3622]: chan_oh323.c:4813 reload_config: Ignoring unknown H.323 [general] keyword 'inBandDTMF', line 53. == Registered channel type 'OH323' (InAccess Networks OpenH323 Channel Driver) == OpenH323 Channel Ready (v0.6.7) So, how to enable inband? -- Tomislav Parcina tparcina#lama.hr *CLI oh323 show conf Configuration of OpenH323 channel driver -- Version: 0.6.7 Listening on address: 85.114.48.254:1720 Gatekeeper used: No gatekeeper FastStart/H245Tunnelling/H245inSetup: ON/ON/ON Supported formats in pref. order: alaw0 Jitter buffer limits (min/max): 20-100 ms TCP port range: 1 - 2 UDP (RAS) port range: 1 - 2 UDP (RTP) port range: 1 - 2 IP Type-of-Service value: 0 User input mode: tone Max number of inbound H.323 calls: 10 Max number of outbound H.323 calls: 10 Max number of simultaneous H.323 calls: 10 Max call rate (ingress direction): 1.00/30 Default language: en Default music class: default Default context: incomingh323 ___ --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] Why is the internet connection important to LAN and PSTN calls?
12 apr 2006 kl. 09.08 skrev Cristian Draghici: If DNS does not work on your local network, Asterisk will lock up. Out of curiosity - the async implementation you mentioned in the other thread - will it replace gethostbyname with something smarter or just run things in a different thread asynchronously? I am not personally involved in the details, but as far as I know, it will replace gethostbyname with something smarter. /O ___ --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] TE410P upgrade to TE411P = (solution to) no more fax carrier detection !
[EMAIL PROTECTED] wrote: I changed from a TE410P to a TE411P and fax carriers weren't detected anymore ! I have tried everything (recompile zaptel+asterisk+spandsp ; echocancel=yes/no ; rxgain=0/txgain=0 ; increase Wait() ; ...), nothing worked. The only solution that worked for me was to install and use NVFaxDetect. For the moment, if you need FAX tone detection, you will need to use 'vpmdtmfsupport=0' in your module configuration for loading the wct4xxp module; this will not disable the echo canceler, just stop using it for tone detection. ___ --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_log timestamp?
Or you can make it a bit simple in this way (number at the end of line is your timestamp) : [EMAIL PROTECTED] perl -le 'print scalar localtime 1112336460' |It's a unixtime stamp. It's the number of seconds since the |epoch(Jan 1, 1970). | |[EMAIL PROTECTED] wrote: | How do I read (make sense of) the timestamp in the queue_log? I'm | probably just slow but I don't understand it. -- s pozdravem / regards +-+ Stribrny Tomas [ technician of Network Operation Center ] SkyNet, a.s. Na Rybnicku 5 CZ-12000 Praha 2 Czech Republic tel:+4202 9636 8633 fax:+4202 3901 7633 http://www.SkyNet.cz ___ --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] SIP conections, with RTP not going trough Asterisk
Hi, Ie been looking for some time how to use asterisk to initiate SIP connections between 2 IP phones, but afetr initiated the communication making the RTP go directly from one telephone to the other, without passing by asterisk. Unfortunately I found no explanations of how to do it. Does anyone care to give a pointer to any explanation about how to do it? Thanks in advance ___ --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 conections, with RTP not going trough Asterisk
Tiago Stein D`Agostini wrote: Hi, Ie been looking for some time how to use asterisk to initiate SIP connections between 2 IP phones, but afetr initiated the communication making the RTP go directly from one telephone to the other, without passing by asterisk. Unfortunately I found no explanations of how to do it. Does anyone care to give a pointer to any explanation about how to do it? canreinvite=yes and look at the options for dial() Thanks in advance bye Ronald Wiplinger ___ --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 BRI in the USA
I dunno if it's THAT bad. I had a BRI line in the (relatively) podunk town of Kalamazoo, Michigan back in 1998. Sure, it took the phone company a couple of weeks to provision the service, but it takes the phone company a couple of weeks to do most anything in my experience. The price was something like $45/mo for two channels and the same per-call/per-minute pricing scheme as POTS (no per-minute fee for incoming and local calls, regular LD pricing for LD, and 800 local outgoing calls included after which it was something like 6 cents per call). The switch on ILEC's end was a DMS-100 implementing National ISDN-1. I really put the ISDN line through its paces too -- voice, data, bonded data, automatic bonding and de-bonding to allow for voice calls -- and everything always worked flawlessly. I don't know what today's pricing is like for ISDN BRI what with all of the various mergers (at the time, I had service from Ameritech), but unless it has gone up significantly, BRI seems like the perfect type of trunk for an Asterisk system too small for a T1/PRI to be an affordable option. It's still similar. Out here, we get a lot of RF interference, and it turns out that BRI is actually cheaper than equivalent POTS lines with Caller-ID (a feature I require), and you can do neat stuff like having 56K dial-in with a USR I-Modem. However, CPE has always been very limited here in the States, and there was no good way to hook up direct to Asterisk. I've heard a few stories that reported partial success with an Eicon Diva Server card, but always with the caveat that it doesn't work quite right or something along those lines. CPE like the USR I-Modem won't deliver Caller-ID to the POTS port. Other CPE like the Motorola BitSurfr Pro is sensitive to RF noise. We were using Netgear RT338's for a number of years, but they are all burnt out now and impossible to replace (actually most CPE is virtually irreplaceable, as so few mfr's make ISDN gear anymore). And while most CPE was OK with our old POTS based phone system, almost none of it worked reliably with POTS-VOIP gateways, such as the Sipura SPA-3000. Further, BRI has two channels, and the U interface pretty much dictates that you feed both of them to the same place. Putting them into an Asterisk box, I would lose the ability to use the USR I-Modem, for example... Despairing, I thought I might have to abandon the beautiful digital delivery of ISDN, which is stupid when you have a digital (VoIP) phone system. But: After talking with a friend up in Minneapolis, I bought an Adtran Atlas 550 off of eBay, which is a versatile Swiss Army Knife for telecom needs. With a quad port ISDN BRI and an octal FXS, it's the killer CPE device, but the best part is that it also does T1/PRI, so you can /convert/ BRI to PRI, etc. I've not actually done that just yet, though I do have a Digium T1 card around here somewhere and want to try it out one of these days. So, I can't actually say it /works/, but it's supposed to. ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DID'S Romania - Bucharest
Dear List, We have Romania Bucharest DIDs available with area code 4021 and 4031 For more information go to www.didx.org Best Regards, Oliver Vermeulen World Venture Group Telecom Tech / Admin Corporate Address: Str Avionului Nr 35/bl16J/3 Bucharest, 014333 Romania Office : +(40) 21-569-4700 Office2 : +(40) 31-860-0030 Fax: +(40) 31-860-0031 USA DID: + 1 (305) 722-1457 BELGIUM DID: +(32) 9 395-5620 UK DID: +(44) 870-478-8896 SIP : [EMAIL PROTECTED] website : http://www.wvg-tele.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
RE: [Asterisk-Users] Texas User Group
I'm in Lubbock. A little closer to Amarillo than Dallas. Thanks, Greg From: Ryan Burke [mailto:[EMAIL PROTECTED] Sent: Tuesday, April 11, 2006 7:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Texas User Group I'm interested but I'm in the Dallas area. Are there any in the Dallas area anyone knows of? Ryan - Original Message - From: Bruce Reeves To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Monday, April 10, 2006 12:51 PM Subject: [Asterisk-Users] Texas User Group I am wondering if any of the Texas user groups have members in the North West part of the state. I am in the Amarillo area and would like to find some othere in this area, maybe even start a user group in this area. -- Bruce Reeves Nortex 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 ___ --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] Automatic 3 Way Call
Dear Group, I'm working on a call recording solution and would like to have the ability to initiate a 3 way call based on an incoming call. One party will be an AGI that I have other will be an outbound call via a second T1 interface. Does anyone have a working configuration for an Asterisk initiated 3 way call? Thanks and Regards Shad Mortazavi -- Nexus Group Technical Manager n|m Nexus Management Inc SIP: [EMAIL PROTECTED] ___ --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] URL in Queue App / Determining the DID/Queue at Agent's Phone
I'd like for our custom soft phone to be able to know what queue, and/or what DID is calling an Agent's phone before the agent picks up. The agent is using the AGENTCALLBACKLOGIN. One agent can be in multiple queues so it would be nice if they could get a pop up window telling them who's on the line before they pick up and hear the announcement telling them that. I'd like to lose the announcment all together. It seems like that the phone can easily know what extension was dialed to make it ring, but at best that's the phone client's extension (The server dialed it via the Local/ interface), and at worst it's 's'. Is there anyway I can know the DID of the person who called into the Queue? I've done ethereal traces and it seems like the DID, that actually called the agent/phone is no where to be found. I've tried also to use the URL string in the Queue() application, but the server doesn't seem to send it. (I've also tried having the client send a URL, and it seems to get sent, yet the server doesn't seem to forward it. It seems to just get lost). Has anyone gotten the URL in the Queue application to work? And if it does, it it delivered to the phone before, or after the phone answers? Any hacks,tips,tricks,pointers, would be most appreciated. Thanks, Steve Feinstein GatherWorks Inc. begin:vcard fn:Steve Feinstein n:Feinstein;Steve org:GatherWorks Inc. adr:;;27 Dupaw Gould Road;Brookline;NH;03033;USA email;internet:[EMAIL PROTECTED] tel;work:+1 (603) 672-1472 x-mozilla-html:TRUE url:http://www.gatherworks.com/ version:2.1 end:vcard ___ --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] DID'S Romania - Bucharest
Oliver Vermeulen schrieb: ... We have ... Hi, I'm sure, there are a lot of providers of very interesting and useful and helpful products and offers reading and writing to this group - including our company. Nevertheless, noone is offering his products here, because it is not fair, if someone is offering and others are not, respecting this mailing list's rules. I don't know the right word in english, in german Oliver Vermeulen's behaviour is called unlauter, which means, that he is granting himself better chances by using forbidden means. Roger. ___ --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] billing with PostgreSQL
Hi Joao, some billing solutions are listed here - http://www.voip-info.org/wiki/view/Open+Source+Billing+Systems IIRC, none works with PGSQL. My opinion is that considering the importance of billing, it's better to develop a customised solution. That way, you would have full understanding and confidence in it. References to other systems can be useful also. Hope it helps. Regards Andy Tan On Wed, 12 Apr 2006 11:15:24 +0100, Joao Pereira [EMAIL PROTECTED] said: Hello to all Im looking for a billing tool for Asterisk, that works with PostgreSQL. All the tools I found in www.asteriskbilling.com just work with MySQL :( Do you know a nice billing tool for Asterisk with PostgreSQL? Thanks Joao Pereira ___ --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 -- Andy Tan [EMAIL PROTECTED] -- http://www.fastmail.fm - Faster than the air-speed velocity of an unladen european swallow ___ --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] Performance: Xeon or Opteron?
Yep, there is a lot of chatter about how hardware x performs with Asterisk and while I/O is the primary mover, most designs today will handle the modest Asterisk install easily. I've got a site where they use 6 lines and 15 users on a 500Mhz CPU w/512MB RAM and boot off a 2GB flash disk. VERY modest and absolutely dominates that particular install. Only in the larger installs will hardware be an issue, but even then it doesn't take much hardware (from a server perspective) to handle a LOT of Asterisk traffic. RandyW Waldo Rubinstein wrote: AFAIK, it doesn't make much of a difference if all you are going to be mainly using is the TE card. From what I've heard and seen, a single P4 3GHz machine will handle a fully loaded TE4XX board with no problem. - Waldo On Apr 11, 2006, at 10:30 PM, Tim Connolly wrote: I was offered an upgrade path for my two Dell 1750's (2.8 Dual Xeon) to get into a pair of new Dual Core Dual Opteron servers. Assuming I can get the IRQ BS worked out so my TE411XP doesn't flip out, this should be a pretty significant upgrade. Has anyone been able to quantify any benefits to using one processor over the other? Should I wait for the newer Intel processors this summer or go for the AMD DC DO? Thanks Tim ___ --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] URL in Queue App / Determining the DID/Queue at Agent's Phone
On 4/12/06, Steve Feinstein [EMAIL PROTECTED] wrote: I'd like for our custom soft phone to be able to know what queue, and/or what DID is calling an Agent's phone before the agent picks up. The agent is using the AGENTCALLBACKLOGIN. One agent can be in multiple queues so it would be nice if they could get a pop up window telling them who's on the line before they pick up and hear the announcement telling them that. I'd like to lose the announcment all together. It seems like that the phone can easily know what extension was dialed to make it ring, but at best that's the phone client's extension (The server dialed it via the Local/ interface), and at worst it's 's'. Is there anyway I can know the DID of the person who called into the Queue? I've done ethereal traces and it seems like the DID, that actually called the agent/phone is no where to be found. I've tried also to use the URL string in the Queue() application, but the server doesn't seem to send it. (I've also tried having the client send a URL, and it seems to get sent, yet the server doesn't seem to forward it. It seems to just get lost). Has anyone gotten the URL in the Queue application to work? And if it does, it it delivered to the phone before, or after the phone answers? Any hacks,tips,tricks,pointers, would be most appreciated. http://bugs.digium.com/view.php?id=6843 Here's code to fire off an AGI to do pretty much anything you need to do on the calling channel after a Queue Member has been assigned to it. -- Bird's The Word Technologies, Inc. http://www.btwtech.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
Re: [Asterisk-Users] help -- voicemail
Check your features.conf file for conflicting key set. # is the default key for blind transfer feature. []'s MM chan (Alpha Trilogies Networks) wrote: Hi, Did someone experience that Asterisk OS 1.2.5 voicemail issues? Problem description: Some one call to the extensions 200, After 10 sec ring then go to voicemail [EMAIL PROTECTED] Announcement Please leave me a messages.blar blar.. When I completed to leave a message... IF : I press the pound #key ... Then it says Transfer IF : I Press the zero 0key Then it say Please confirm your recording IF : I hangup after leaving a message...then things get normal. What is this Funny. Pls some one reply. ___ --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] Performance: Xeon or Opteron?
RandyW wrote: Yep, there is a lot of chatter about how hardware x performs with Asterisk and while I/O is the primary mover, most designs today will handle the modest Asterisk install easily. I've got a site where they use 6 lines and 15 users on a 500Mhz CPU w/512MB RAM and boot off a 2GB flash disk. VERY modest and absolutely dominates that particular install. Only in the larger installs will hardware be an issue, but even then it doesn't take much hardware (from a server perspective) to handle a LOT of Asterisk traffic. RandyW The worst problem will be older hardware that doesn't play well with Digium cards. The TDM400 is the one I have some experience with, and even motherboards that are PCI 2.2 don't always see the TDM400 The Sangoma A200 seems more forgiving. I have to wonder if the T1/E1 cards suffer in a similar manner? John Novack Waldo Rubinstein wrote: AFAIK, it doesn't make much of a difference if all you are going to be mainly using is the TE card. From what I've heard and seen, a single P4 3GHz machine will handle a fully loaded TE4XX board with no problem. - Waldo On Apr 11, 2006, at 10:30 PM, Tim Connolly wrote: I was offered an upgrade path for my two Dell 1750's (2.8 Dual Xeon) to get into a pair of new Dual Core Dual Opteron servers. Assuming I can get the IRQ BS worked out so my TE411XP doesn't flip out, this should be a pretty significant upgrade. Has anyone been able to quantify any benefits to using one processor over the other? Should I wait for the newer Intel processors this summer or go for the AMD DC DO? Thanks Tim ___ --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] Trunking Protocols
Hi, understand that Asterisk supports a variety of signaling protocols like SIP, IAX2 etc. As a ITSP, which would be the best or most appropiate protocol to use as trunk to wholesale providers? Know that IAX2 can conserve bandwidth, but I believe media and signaling are carried with the same channel/path. That would make off-loading bandwidth utilization for media impossible. Appreciate any input. Thanks. Regards Any Tan -- Andy Tan [EMAIL PROTECTED] -- http://www.fastmail.fm - Access all of your messages and folders wherever you are ___ --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] Macro-hangupcall - has a Wait(5) - [EMAIL PROTECTED] --- why?
[macro-hangupcall]exten = s,1,ResetCDR(w)exten = s,2,NoCDR()exten = s,3,Wait(5)exten = s,4,HangupHi all, currently i've been getting troubles with SIpphone Sjphone and Xlite seems also to get delay but no crash on hanging. I found that [EMAIL PROTECTED] is executing this Wait(5) and it seems to me that Sjphone is giving timeout error because of it...Why is this 5 seconnds? any one knows?best regards,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
RE: [Asterisk-Users] GXP-2000 phones stop registering
Mark, Do you have the Flash Operator Panel or anything else installed? I only had 1 phone stop registering in the first 2 weeks that I used them and then after I installed FOP I had 3 phones stop registering in the next couple of days. I have now disabled FOP and have gone just over 2 days without any problems. Its probably just a coincidence but I am going to run without FOP for another week and then try enabling it again. On Mon, 2006-04-10 at 12:14, Mark Edwards wrote: Yes. Me. I don't have a fix unfortunately - like you I seek one, however I have had a better experience by far though with the new 102x firmware branch. I would definitely recommend it to you. Mark -Original Message- From: Gareth Blades [mailto:[EMAIL PROTECTED] Sent: Monday, 10 April 2006 8:49 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] GXP-2000 phones stop registering I have about 30 GXP-2000 phones running 1.0.1.9 which have all been configured using the provisioning feature so the configuration is all identical. The problem I am having is that they randomly seem to stop registering with asterisk. When they stop registering they can still make calls but oviously asterisk cannot ring the phone so all incoming calls go to voicemail. Has anyone else had similar problems? example sip.conf entry:- 6015] type=friend secret=x username=6015 callerid=users name 6015 host=dynamic nat=no canreinvite=yes disallow=all allow=ulaw allow=alaw deny=0.0.0.0/0.0.0.0 permit=10.0.0.0/255.0.0.0 context=voipuk mailbox=6015 The phone config is fairly standard. the registration expiry is set to 60 minutes ___ --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] Texas User Group
It sounds like what might be best is a Texas User group, since most of us are spread out across our great state. With Astircon 2006 coming to Dallas this year, we could all probably get together at that time. Mainly I would like to see a user group in Texas because I am deploying a wide spread asterisk setup in several cities across the state and Oklahoma and Louisiana. It would be nice to know some possible local asterisk contacts. I am willing to setup some space on my website or a new domain for a user group if a state wide group sounds good. Any suggestions as to features for the site? Wiki? Forums? ??BruceNortex Networks On 4/12/06, Greg Camp [EMAIL PROTECTED] wrote: I'm in Lubbock. A little closer to Amarillo than Dallas. Thanks, Greg ___ --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] Performance: Xeon or Opteron?
Yep, there is a lot of chatter about how hardware x performs with Asterisk and while I/O is the primary mover, most designs today will handle the modest Asterisk install easily. I've got a site where they use 6 lines and 15 users on a 500Mhz CPU w/512MB RAM and boot off a 2GB flash disk. VERY modest and absolutely dominates that particular install. Only in the larger installs will hardware be an issue, but even then it doesn't take much hardware (from a server perspective) to handle a LOT of Asterisk traffic. RandyW The worst problem will be older hardware that doesn't play well with Digium cards. The TDM400 is the one I have some experience with, and even motherboards that are PCI 2.2 don't always see the TDM400 The Sangoma A200 seems more forgiving. I have to wonder if the T1/E1 cards suffer in a similar manner? While talking with one of the sangoma folks very recently, he was rather emphatic the pci bus was designed to share interrupts. I was a little concerned as a test server had the wanpipe driver sharing an interrupt with libata and uhc1_hcd. His comment was that's the way its suppose to work, sharing interrupts as needed. I've not had any recognizable issues with the A200D card at all, and faxing via a A200D fxs port to a A200D fxo (pstn) port functions 100% reliably. What that would suggest is the TDM400 pci firmware (whether on card logic or whatever) is the source of at least part of the TDM400 shared interrupt issue. I don't have any digium T1/E1 cards laying around, but if memory serves correctly, the T1/E1 cards do not use the same pci controller chip. That would suggest the T1/E1 cards are less of an issue then with the TDM400 card. ___ --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] Performance: Xeon or Opteron?
Rich Adamson wrote: Yep, there is a lot of chatter about how hardware x performs with Asterisk and while I/O is the primary mover, most designs today will handle the modest Asterisk install easily. I've got a site where they use 6 lines and 15 users on a 500Mhz CPU w/512MB RAM and boot off a 2GB flash disk. VERY modest and absolutely dominates that particular install. Only in the larger installs will hardware be an issue, but even then it doesn't take much hardware (from a server perspective) to handle a LOT of Asterisk traffic. RandyW The worst problem will be older hardware that doesn't play well with Digium cards. The TDM400 is the one I have some experience with, and even motherboards that are PCI 2.2 don't always see the TDM400 The Sangoma A200 seems more forgiving. I have to wonder if the T1/E1 cards suffer in a similar manner? While talking with one of the sangoma folks very recently, he was rather emphatic the pci bus was designed to share interrupts. I was a little concerned as a test server had the wanpipe driver sharing an interrupt with libata and uhc1_hcd. His comment was that's the way its suppose to work, sharing interrupts as needed. I've not had any recognizable issues with the A200D card at all, and faxing via a A200D fxs port to a A200D fxo (pstn) port functions 100% reliably. What that would suggest is the TDM400 pci firmware (whether on card logic or whatever) is the source of at least part of the TDM400 shared interrupt issue. I don't have any digium T1/E1 cards laying around, but if memory serves correctly, the T1/E1 cards do not use the same pci controller chip. That would suggest the T1/E1 cards are less of an issue then with the TDM400 card. The single port T1/E1 card (te110p) and the TDM400 both use the TigerJet 320. -- Kristian Kielhofner ___ --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] Bandwidth Management
I think this belongs to the development mail-list. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jean-Michel Hiver Sent: Wednesday, April 12, 2006 12:05 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Bandwidth Management Andy Tan a écrit : Hi Alex, thanks for the suggestion. Did some checks, and thought that I could set a global variable to track the utilized bandwidth. Wish that there are plans for support to include variables like SIP_CODEC in other protocols. Actually this sounds like a really nice idea. It would be cool to have a way to start using less intensive bandwith codecs for new calls when bandwith reaches a certain threshold. For example: - 0-40% bandwith: g711 - 40-60% bandwith: g729 - 60%-80% bandwith: g723 - 80%-100% bandwith: drop new calls, or maybe use lpc10 It wouldn't help in SOHO usage but when using Asterisk as a call termination gateway, it would help making the most out of available bandwith. g711 is certainly better than g729 when you have the bandwith, and i'm pretty sure that even lpc10 sounds better when on non-saturated bandwith compared with g729 with some packet loss... How would you go about implementing this? Cheers, Jean-Michel. -- Jean-Michel Hiver - http://ykoz.net/ Découvrez la Réunion des Technologies IP Telecom TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE ___ --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] SipXPhone
Mark, I could not get SipXPhone working either. We've been using this SDK and really like it: http://www.worksoutsoftware.com/ The pricing is seems decent as well. Thanks, Greg From: Mark Hayward [mailto:[EMAIL PROTECTED] Sent: Wednesday, April 12, 2006 3:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] SipXPhone Has anybody managed to get SipXPhone working with asterisk? I just cannot get it to work. It just keeps reporting an authentication failure even though all the details seem correct. The same settings work fine in X-Lite. Failing that, are there any opensource or reasonably priced SIP SDKs that people can recommend? I have the log from the SipXPhone. It says authentication required, yet it is definitely using the correct user/pass. 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
RE: [Asterisk-Users] call center running Asterisk - sound quality-critical!
Just good old monitor with no mixing onto the scsi drive. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tamas Sent: Wednesday, April 12, 2006 4:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] call center running Asterisk - sound quality-critical! Hi, how do you record calls? Monitor app. or MixMonitor or something else? How does your storage backend looks like? What kind of channels do you use? Do you record IAX2 channels? Regards, Tamas Wai Wu wrote: You got to be kidding about 53 calls being recorded at sametime is an issue. I have done at least twice as many on my dual xeon 3.4Ghz system and had no problem as clients like to record every call that goes through the system. Then again, in my system, the in and out channels are mixed first before they are written to the disk. From: [EMAIL PROTECTED] on behalf of Matt Roth Sent: Tue 4/11/2006 5:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] call center running Asterisk - sound quality- critical! On 4/10/06, Dov Bigio [EMAIL PROTECTED] wrote: Hi, I am using Asterisk for a call center on a Dual Xeon machine.. I currently have 109 active channels 53 active calls Every body is complaining about quality and cpu is around 80% idle. Is there any tuning I can do??? Besides that, Asterisk normally goes down once or twice per day... Thank you Dov C F wrote: From what you say it sounds that the problem is not with asteisk, but the way it's configured. Asterisk should *never* go down that often. Asterisk as a normal PBX should run without a restart for as long as there is power to the box, in the case of a call center if I would hear of a restart once a week I would accept it, but still would look for ways of improving it beyond that. You complain about call quality, what type of phones are thes? What codec? are they all local? Dov, I agree with the first response. Your system is failing at an abnormal rate. Please share more information about your setup so that we can help you. Hardware, software, OS, configuration...there's no such thing as too many details when trying to work out these problems via a mailing list. Information about what tasks you are asking Asterisk to perform and how you have it configured to do so is vital. In particular, I'm curious to know if you're recording the calls using the Monitor() application? 53 concurrent calls being recorded directly to disk is about where things start to go south (it's an I/O bottleneck, not CPU). If you have a Digium card in the box, make sure that it's not sharing an interrupt with any other hardware. The list and the wiki both have plenty of information to help you with that. Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer ___ --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] Newbie MOH and call transfer question
Hi, I use the AAH2.7 (asterisk version 1.2.5). When someone call me and I pickup the phone. If I want to transfer to another extension. Then I dial the # key the system will play the onhold music. After I dial the extension number. The system stop play onhold music and play ringtone. Is it possiable keep play onhold music until someone pickup the phone? Appreciate any input. Thanks. Kevin ___ --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] Performance: Xeon or Opteron?
Kristian Kielhofner wrote: Rich Adamson wrote: Yep, there is a lot of chatter about how hardware x performs with Asterisk and while I/O is the primary mover, most designs today will handle the modest Asterisk install easily. I've got a site where they use 6 lines and 15 users on a 500Mhz CPU w/512MB RAM and boot off a 2GB flash disk. VERY modest and absolutely dominates that particular install. Only in the larger installs will hardware be an issue, but even then it doesn't take much hardware (from a server perspective) to handle a LOT of Asterisk traffic. RandyW The worst problem will be older hardware that doesn't play well with Digium cards. The TDM400 is the one I have some experience with, and even motherboards that are PCI 2.2 don't always see the TDM400 The Sangoma A200 seems more forgiving. I have to wonder if the T1/E1 cards suffer in a similar manner? While talking with one of the sangoma folks very recently, he was rather emphatic the pci bus was designed to share interrupts. I was a little concerned as a test server had the wanpipe driver sharing an interrupt with libata and uhc1_hcd. His comment was that's the way its suppose to work, sharing interrupts as needed. I've not had any recognizable issues with the A200D card at all, and faxing via a A200D fxs port to a A200D fxo (pstn) port functions 100% reliably. What that would suggest is the TDM400 pci firmware (whether on card logic or whatever) is the source of at least part of the TDM400 shared interrupt issue. I don't have any digium T1/E1 cards laying around, but if memory serves correctly, the T1/E1 cards do not use the same pci controller chip. That would suggest the T1/E1 cards are less of an issue then with the TDM400 card. The single port T1/E1 card (te110p) and the TDM400 both use the TigerJet 320. I guess they both would have the same issues then. ;) ___ --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] update - 512 Simultaneous Calls with Digital Recording
Another solution would be to use a dedicated recording server sniffing
RTP and signalling packets in the media path using software such as
http://www.oreka.org. Oreka automatically mixes both legs of an RTP
conversation to disk and GSM encodes the result in a separate thread
so that capture always has priority.
Cheers
Henri
On 05/04/06, Isaac Xiao [EMAIL PROTECTED] wrote:
Matthew, thanks for your feedback and advice.
what I actually experienced was the complete breakdown of Asterisk at
around 60 concurrent recordings without it (the reality).
The drive for saving your voice recordings is the same as your OS
(Asterisk)? What do you think that save the voice recordings to a
dedicated drive rather than the one which Asterisk program (OS) locates?
I also think about using GSM format (Monitor(gsm,${CALLFILENAME}, mb))
rather than WAV, PCM. In this case, it will use more CPU, but I/O of
hard disk is reduced dramatically as you mentioned that it is I/O
bottleneck issue, not CPU (In my case, I want to use P4 Dual core CPU or
extreme edition). In order to reduce the CPU usage, we can have two leg
files mixed after peak time.
Matt mentioned about fragmented free space. I googled about Linux
defragment topic. People always talk about that Linux doesn't need to
defragment, it can handle it by itself very well. Not sure how true it
is.
I am looking a solution to record expanding simultaneous calls in the
future in a call centre which accepts calls from our global branches. If
I find the good solution, I definitely post it to the community.
Cheers,
Isaac Xiao
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Re: [Asterisk-Users] URL in Queue App / Determining the DID/Queue at Agent's Phone
Thanks!, I will definitely take a look at that. We were hoping not to have to do AGI in the client, but if we have to, we have to. It'll probably be useful for other things down the road. -Steve Feinstein GatherWorks Inc. BJ Weschke wrote: On 4/12/06, Steve Feinstein [EMAIL PROTECTED] wrote: I'd like for our custom soft phone to be able to know what queue, and/or what DID is calling an Agent's phone before the agent picks up. The agent is using the AGENTCALLBACKLOGIN. One agent can be in multiple queues so it would be nice if they could get a pop up window telling them who's on the line before they pick up and hear the announcement telling them that. I'd like to lose the announcment all together. It seems like that the phone can easily know what extension was dialed to make it ring, but at best that's the phone client's extension (The server dialed it via the Local/ interface), and at worst it's 's'. Is there anyway I can know the DID of the person who called into the Queue? I've done ethereal traces and it seems like the DID, that actually called the agent/phone is no where to be found. I've tried also to use the URL string in the Queue() application, but the server doesn't seem to send it. (I've also tried having the client send a URL, and it seems to get sent, yet the server doesn't seem to forward it. It seems to just get lost). Has anyone gotten the URL in the Queue application to work? And if it does, it it delivered to the phone before, or after the phone answers? Any hacks,tips,tricks,pointers, would be most appreciated. http://bugs.digium.com/view.php?id=6843 Here's code to fire off an AGI to do pretty much anything you need to do on the calling channel after a Queue Member has been assigned to it. -- Bird's The Word Technologies, Inc. http://www.btwtech.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 begin:vcard fn:Steve Feinstein n:Feinstein;Steve org:GatherWorks Inc. adr:;;27 Dupaw Gould Road;Brookline;NH;03033;USA email;internet:[EMAIL PROTECTED] tel;work:+1 (603) 672-1472 x-mozilla-html:TRUE url:http://www.gatherworks.com/ version:2.1 end:vcard ___ --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] call center running Asterisk - sound quality - critical!
If you don't want to worry about * handling the full recording of all traffic, you can potentially do this on a separate server on the RTP path using http://www.oreka.org. Cheers Henri On 10/04/06, Dov Bigio [EMAIL PROTECTED] wrote: Hi, I am using Asterisk for a call center on a Dual Xeon machine.. I currently have 109 active channels 53 active calls Every body is complaining about quality and cpu is around 80% idle. Is there any tuning I can do??? Besides that, Asterisk normally goes down once or twice per day... Thank you Dov ___ --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] Performance: Xeon or Opteron?
Rich Adamson wrote: While talking with one of the sangoma folks very recently, he was rather emphatic the pci bus was designed to share interrupts. I was a little concerned as a test server had the wanpipe driver sharing an interrupt with libata and uhc1_hcd. His comment was that's the way its suppose to work, sharing interrupts as needed. I've not had any recognizable issues with the A200D card at all, and faxing via a A200D fxs port to a A200D fxo (pstn) port functions 100% reliably. What that would suggest is the TDM400 pci firmware (whether on card logic or whatever) is the source of at least part of the TDM400 shared interrupt issue. I don't have any digium T1/E1 cards laying around, but if memory serves correctly, the T1/E1 cards do not use the same pci controller chip. That would suggest the T1/E1 cards are less of an issue then with the TDM400 card. That's good to know, but considering the response from Digium on the TDM400 ( try another motherboard) when there didn't seem to even be an int. sharing issue, the card just couldn't be seen at all , and the support I received from Sangoma on a recent FXS issue that was resolved within a few days, I would tend to go with Sangoma for the T1 card, if and when I have the need. John Novack ___ --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] call center running Asterisk - sound quality-critical!
Wai Wu wrote: You got to be kidding about 53 calls being recorded at sametime is an issue. I have done at least twice as many on my dual xeon 3.4Ghz system and had no problem as clients like to record every call that goes through the system. Nope. We took our system to MCI's development lab and ran it against an Abacus 5000. Things fell apart on the 64 call test. We looked at the logs and saw a massive amount of disk I/O, so we set up a RAM disk to write the recordings to. We were then able to successfully test up to 512 simultaneous calls. Looking at this list and the wiki, you'll see that many other users ran into the same issue at around 60 simultaneous recordings via Monitor(). Tamas wrote: how do you record calls? Monitor app. or MixMonitor or something else? Wai Wu wrote: Then again, in my system, the in and out channels are mixed first before they are written to the disk. Just good old monitor with no mixing onto the scsi drive. These statements seem contradictory. I know of no way (short of a custom patch) to tell Monitor() to mix the in and out legs prior to writing them to disk. On the other hand, MixMonitor() does just that and I believe it also buffers the writes in a way that circumvents the I/O bottleneck associated with Monitor(). Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer ___ --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] Texas User Group
That may be the best idea. Unfortunately we're such a huge state that it's going to be pretty hard to get everyone in the same room unless there's some big event going on. Astricon may be a good time to get together in person though. As for the site, a simple wiki may be best, and if everyone wants a forum (personally prefer mailing lists, easier to filter through, but that's just me) that'd be nifty as well. Perhaps when this gets started, we may find more users in the state and do mini-sessions in different parts of the state. Aaron On Wed, 12 Apr 2006, Bruce Reeves wrote: It sounds like what might be best is a Texas User group, since most of us are spread out across our great state. With Astircon 2006 coming to Dallas this year, we could all probably get together at that time. Mainly I would like to see a user group in Texas because I am deploying a wide spread asterisk setup in several cities across the state and Oklahoma and Louisiana. It would be nice to know some possible local asterisk contacts. I am willing to setup some space on my website or a new domain for a user group if a state wide group sounds good. Any suggestions as to features for the site? Wiki? Forums? ?? Bruce Nortex Networks On 4/12/06, Greg Camp [EMAIL PROTECTED] wrote: I'm in Lubbock. A little closer to Amarillo than Dallas. Thanks, Greg -- -- 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
[Asterisk-Users] CAPI Installation Eicon Diva Server
Hi I've got a dell 2550 with an Eicon Diva server PRI card plugged into it. I can call out using the acopy2 test utility. I'm having trouble with asterisk making calls however... my capi.conf and modules.conf looks correct by the wiki instructions - does anyone have any advice on where to look ? I can attach conf files etc. if needed. Asterisk says it has 30 capi channels available, but my mistake may be in configuring the trunks... Nick ** Any information in this communication which is confidential must not be disclosed to others without our consent. Such consent is not required where the information is publicly available and intended for onward distribution. If the information is confidential and if you are not the intended recipient, you are not authorised to and must not disclose, copy, distribute, or retain this message or any part of it. You are requested to return this message to the sender immediately. Due to the electronic nature of e-mail, there is a risk that the information contained in this message has been modified. Consequently Man Investments can accept no responsibility or liability as to the completeness or accuracy of the information. Visit us at: www.maninvestments.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] Setting Codecs on the Fly
Does anyone know if it's possible to set the codecs for a number via an Asterisk command? Ie, yes you can set the codecs in sip.conf for a user, but I'd like to have a command that can set the same thing so that it can be done without having to change sip.conf. Essentially I want the user to be able to prefix a code to their dialled number to set their preferred codec for that call. Possible? 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] call center running Asterisk - sound quality-critical!
Matt Roth wrote: These statements seem contradictory. I know of no way (short of a custom patch) to tell Monitor() to mix the in and out legs prior to writing them to disk. On the other hand, MixMonitor() does just that and I believe it also buffers the writes in a way that circumvents the I/O bottleneck associated with Monitor(). Both of these statements are correct. ___ --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] iax2 show netstats
i've been using iax2 show netstats and i wonder if someone could explain what all these means, just in case i have them wrong. Because i am looking for something that tells me that there is delay , and/or packet loss. LOCAL - REMOTE Channel RTT Jit Del Lost % Drop OOO Kpkts Jit Del Lost % Drop OOO Kpkts IAX2/iaxBBG-16384 1000 -10-1 -1 0 -1 000 0 0 00 0 IAX2/iaxBBG-16386 16 -10-1 -1 0 -1 10 40 0 0 00 0 The new Jitterbuffer in Asterisk Steve Kann .. 5) Testing and monitoring: -- You can test the effectiveness of PLC and the new jitterbuffer's detection of loss by using the new CLI command iax2 test losspct n. This will simulate n percent packet loss coming _in_ to chan_iax2. You should find that with PLC and the new JB, 10 percent packet loss should lead to just a tiny amount of distortion, while without PLC, it would lead to silent gaps in your audio. iax2 show netstats shows you statistics for each iax2 call you have up. The columns are RTT which is the round-trip time for the last PING, and then a bunch of s tats for both the local side (what you're receiving), and the remote side (what the other end is telling us they are seeing). The remote stats may not be complete if the remote end isn't using the new jitterbuffer. The stats shown are: * Jit: The jitter we have measured (milliseconds) * Del: The maximum delay imposed by the jitterbuffer (milliseconds) * Lost: The number of packets we've detected as lost. * %: The percentage of packets we've detected as lost recently. * Drop: The number of packets we've purposely dropped (to lower latency). * OOO: The number of packets we've received out-of-order * Kpkts: The number of packets we've received / 1000. ... Benchev ___ --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] TE410P upgrade to TE411P = (solution to) no more fax carrier detection !
Kevin - if you stop it from tone detection with 'vpmdtmfsupport=0' will it detect the fax cgn?RegardsRobOn 12/04/06, Kevin P. Fleming [EMAIL PROTECTED] wrote:[EMAIL PROTECTED] wrote: I changed from a TE410P to a TE411P and fax carriers weren't detected anymore ! I have tried everything (recompile zaptel+asterisk+spandsp ; echocancel=yes/no ; rxgain=0/txgain=0 ; increase Wait() ; ...), nothing worked. The only solution that worked for me was to install and use NVFaxDetect.For the moment, if you need FAX tone detection, you will need to use'vpmdtmfsupport=0' in your module configuration for loading the wct4xxp module; this will not disable the echo canceler, just stop using it fortone detection.___--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] TE410P upgrade to TE411P = (solution to) no more fax carrier detection !
Rob Lith wrote: Kevin - if you stop it from tone detection with 'vpmdtmfsupport=0' will it detect the fax cgn? Yes, that was the point of my message; with that setting, the software tone detector will be used, just as it was before the OP's VPM got installed. ___ --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] call center running Asterisk - sound quality-critical!
Kevin P. Fleming wrote: Matt Roth wrote: These statements seem contradictory. I know of no way (short of a custom patch) to tell Monitor() to mix the in and out legs prior to writing them to disk. On the other hand, MixMonitor() does just that and I believe it also buffers the writes in a way that circumvents the I/O bottleneck associated with Monitor(). Both of these statements are correct It seems, MixMonitor is usable again (since yesterday's svn commit) so it can save at least saving of one channel. We would test now MixMonitor for this reason. Kevin, does MixMonitor have buffering? How big is the buffer? Is it possible to change the size? I guess, we are talking about buffering voice samples and writing only a bulk of them to disk (e.g. in every 50 packets - 1second). If there is no such buffer, do you think implementing it can be a real solution? Storing into RAM needs too big RAM which can be a problem (e.g. if we want to use monitor() for storing 60 concurrent calls for min. 15-20 minutes). Regards, T. ___ --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] update - 512 Simultaneous Calls with Digital Recording
Hey Henri, Long time no talk. How far have you been able to scale oreka up to? How many simultaneous calls have you been able to record and under what hardware config? Thanks, Waldo On Apr 12, 2006, at 11:12 AM, Henri Herscher wrote: Another solution would be to use a dedicated recording server sniffing RTP and signalling packets in the media path using software such as http://www.oreka.org. Oreka automatically mixes both legs of an RTP conversation to disk and GSM encodes the result in a separate thread so that capture always has priority. Cheers Henri ___ --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] call center running Asterisk - sound quality-critical!
Tamas wrote: Kevin, does MixMonitor have buffering? How big is the buffer? Is it possible to change the size? I guess, we are talking about buffering voice samples and writing only a bulk of them to disk (e.g. in every 50 packets - 1second). It buffers the data in memory, there is no fixed size. It _will_ attempt to write out a mixed audio frame each time a matching pair of frames has been received from both sides; changing that to only write after a certain amount of data has been received would not be a significant task. There is a risk of data loss, though, if you do that... but at least MixMonitor does not sit in the channel read/write path like Monitor does, so delays in writing the audio don't impact the audio being bridged across the channels. ___ --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] * 1.2.4 1.2.6: Ringing anamoly
I was alerted the other day by of all people, my mom, that she wasn't hearing a ring when she dialed my number. Puzzled, I tried calling myself. The call connects,but there's dead silence until voicemail picks up. Calling internally, extensions worked perfectly. So, I figured, another damned Broadvoice issue. For kicks, I upgraded to 1.2.6 today, and the end result is the same. So, I went to the dialplan playground, and removed a few lines for testing. It turns out that if I playback a file before ringing an extension, ringing works fine. Without, dead silence. Any ideas? ___ --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] Setting Codecs on the Fly
Douglas Garstang wrote:
Does anyone know if it's possible to set the codecs for a number via an
Asterisk command?
Ie, yes you can set the codecs in sip.conf for a user, but I'd like to have a
command that can set the same thing so that it can be done without having to
change sip.conf.
Essentially I want the user to be able to prefix a code to their dialled number
to set their preferred codec for that call.
Possible?
Humm..I wonder if what google returned for:
asterisk set codec on a call
http://www.voip-info.org/wiki-Asterisk+variables
Would help...Seeems that in fact, google is my friend:
${SIP_CODEC}: Used to set the SIP codec for a call
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RE: [Asterisk-Users] Setting Codecs on the Fly
Ahhh a variable. I was looking for a command. Thanks, I'll try it out.
-Original Message-
From: Julio Arruda [mailto:[EMAIL PROTECTED]
Sent: Wednesday, April 12, 2006 10:42 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Setting Codecs on the Fly
Douglas Garstang wrote:
Does anyone know if it's possible to set the codecs for a
number via an Asterisk command?
Ie, yes you can set the codecs in sip.conf for a user, but
I'd like to have a command that can set the same thing so
that it can be done without having to change sip.conf.
Essentially I want the user to be able to prefix a code to
their dialled number to set their preferred codec for that call.
Possible?
Humm..I wonder if what google returned for:
asterisk set codec on a call
http://www.voip-info.org/wiki-Asterisk+variables
Would help...Seeems that in fact, google is my friend:
${SIP_CODEC}: Used to set the SIP codec for a call
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RE: [Asterisk-Users] Setting Codecs on the Fly
Simply check out the READMEs in asterisk/doc/ in your source directory.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Julio Arruda
Sent: Wednesday, April 12, 2006 12:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Setting Codecs on the Fly
Douglas Garstang wrote:
Does anyone know if it's possible to set the codecs for a
number via an Asterisk command?
Ie, yes you can set the codecs in sip.conf for a user, but
I'd like to have a command that can set the same thing so
that it can be done without having to change sip.conf.
Essentially I want the user to be able to prefix a code to
their dialled number to set their preferred codec for that call.
Possible?
Humm..I wonder if what google returned for:
asterisk set codec on a call
http://www.voip-info.org/wiki-Asterisk+variables
Would help...Seeems that in fact, google is my friend:
${SIP_CODEC}: Used to set the SIP codec for a call
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RE: [Asterisk-Users] Bandwidth Management
Brought over from -users, Please reply to the -dev list.
I agree, lets move the discusstion over to that list as it has to be discussed
there. After we reach an accord on how it should be done we will open up a
issue on Mantis.
I see this as being two distinctive parts that would need to be tied together:
First: We need to make the selection of CODECS technology agnostic, There
currently exist a facility for CODEC selection (SIP_CODEC) in the sip channel
but not in others.
Second: Discuss is this sould be an outside application that is called from
within Asterisk or if it should become a function
Set(CODEC=${OPTIMALCODEC(quality)})
available options could be:
quality
bandwidth
license
Any comments.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wai Wu
Sent: Wednesday, April 12, 2006 10:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Bandwidth Management
I think this belongs to the development mail-list.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Jean-Michel Hiver
Sent: Wednesday, April 12, 2006 12:05 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Bandwidth Management
Andy Tan a écrit :
Hi Alex,
thanks for the suggestion.
Did some checks, and thought that I could set a global variable to
track the utilized bandwidth.
Wish that there are plans for support to include variables like
SIP_CODEC in other protocols.
Actually this sounds like a really nice idea. It would be
cool to have a way to start using less intensive bandwith
codecs for new calls when bandwith reaches a certain threshold.
For example:
- 0-40% bandwith: g711
- 40-60% bandwith: g729
- 60%-80% bandwith: g723
- 80%-100% bandwith: drop new calls, or maybe use lpc10
It wouldn't help in SOHO usage but when using Asterisk as a
call termination gateway, it would help making the most out
of available bandwith. g711 is certainly better than g729
when you have the bandwith, and i'm pretty sure that even
lpc10 sounds better when on non-saturated bandwith compared
with g729 with some packet loss...
How would you go about implementing this?
Cheers,
Jean-Michel.
--
Jean-Michel Hiver - http://ykoz.net/
Découvrez la Réunion des Technologies IP Telecom
TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE
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[Asterisk-Users] Polycom VLANs
So has anyone had any experience working with the polycom 501 or 301 and vlans? We run dell managed switches here, so we don't have the luxury of running CDP to force the VOIP vlan. I haven't been able to get the polycom phones to talk on a manually set vlan. I have some junky sipura phones that work fine-(get dhcp, register to asterisk etc) when i manually set them to vlan4. Any advice you guys have would be greatly appreciated! ___ --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] playback soundfile stored in mysql database
Hi Guys, I want to playback a sound file stored in mysql database in my perl scriptpls can anyone help with an idea? response would be greatly appreciated Rgds _ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ --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] call center running Asterisk -sound quality-critical!
Except that mixmonitor still has a bug in it. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin P. Fleming Sent: Wednesday, April 12, 2006 11:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] call center running Asterisk -sound quality-critical! Matt Roth wrote: These statements seem contradictory. I know of no way (short of a custom patch) to tell Monitor() to mix the in and out legs prior to writing them to disk. On the other hand, MixMonitor() does just that and I believe it also buffers the writes in a way that circumvents the I/O bottleneck associated with Monitor(). Both of these statements are correct. ___ --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] playback soundfile stored in mysql database
Look at using EAGI. Hi Guys, I want to playback a sound file stored in mysql database in my perl scriptpls can anyone help with an idea? response would be greatly appreciated Rgds ___ --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 VLANs
On 4/12/06, Rob Terhaar [EMAIL PROTECTED] wrote: So has anyone had any experience working with the polycom 501 or 301 and vlans? We run dell managed switches here, so we don't have the luxury of running CDP to force the VOIP vlan. I haven't been able to get the polycom phones to talk on a manually set vlan. I have some junky sipura phones that work fine-(get dhcp, register to asterisk etc) when i manually set them to vlan4. Any advice you guys have would be greatly appreciated! Yes. Works fine. You need to make sure there the VLAN ID in the phone matches the VLAN ID you've got set in your PowerConnect switches and you should be good to go. Well, that, and the fact that the VLAN for the phone should be added to that port as a tagged port. :) -- Bird's The Word Technologies, Inc. http://www.btwtech.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
Re: [Asterisk-Users] call center running Asterisk -sound quality-critical!
On 4/12/06, Wai Wu [EMAIL PROTECTED] wrote: Except that mixmonitor still has a bug in it. Had. Corrected yesterday. -- Bird's The Word Technologies, Inc. http://www.btwtech.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] Recording queue transfers
Regarding this article (1) I have one question to make. What can I do to record the call if the agent makes a transfer using the flash button instead of transfer button or using blindxfer or atxfer defined in features. conf If the agent makes the transfer with flash, the comunication between the person who is calling and is already in the queue and the target person who receive the call doesn't get recorded. e.g. Client/ Costumer (P1), contact the Call Center and he is assisted by an agent (P2), (P2) transfers the call to his supervisor (P3) by pressing flash plus extension # of his supervisor. The comunication between P1 and P3 doesn't get recorded. What can I do to get this recorded? [1] http://www.oinko.net/astrecipes/index.php?from=1q=astrecipes/recording+queue+transfers+to+disk Maxi ___ --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] call center running Asterisk -sound quality-critical!
Wai Wu wrote: Except that mixmonitor still has a bug in it. What kind of bug? Issue number? FYI: yesterday one issue has been fixed :D http://bugs.digium.com/view.php?id=6457 Did you mean that type of bug? If something else, please let us know... T. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin P. Fleming Sent: Wednesday, April 12, 2006 11:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] call center running Asterisk -sound quality-critical! Matt Roth wrote: These statements seem contradictory. I know of no way (short of a custom patch) to tell Monitor() to mix the in and out legs prior to writing them to disk. On the other hand, MixMonitor() does just that and I believe it also buffers the writes in a way that circumvents the I/O bottleneck associated with Monitor(). Both of these statements are correct. __ ___ --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] Macro-hangupcall - has a Wait(5) - [EMAIL PROTECTED] --- why?
On 4/12/06, Marco Mouta [EMAIL PROTECTED] wrote: [macro-hangupcall] exten = s,1,ResetCDR(w) exten = s,2,NoCDR() exten = s,3,Wait(5) exten = s,4,Hangup Hi all, currently i've been getting troubles with SIpphone Sjphone and Xlite seems also to get delay but no crash on hanging. I found that [EMAIL PROTECTED] is executing this Wait(5) and it seems to me that Sjphone is giving timeout error because of it... Why is this 5 seconnds? any one knows? You may want to pose that question to an Asterisk @ Home forum. -- Bird's The Word Technologies, Inc. http://www.btwtech.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] Config with TE210P, Asterisk and Legacy PBX and FreePBX?
Hi list! Has anyone ever tried the following installation : I want to replace our legacy PBX with Asterisk but... I still need the legacy PBX as a 'channel bank' for fax (I need E1 not T1) I will put a dual port PRI card in the Asterisk box, and for incoming and outgoing faxes I want to use native bridging on the TE210P and route fax calls (based on DID and prefix when dialling) to / from the legacy PBX. I guess I do not need to modify anything in the PBX (Alcatel Novo Supreme) because I can simply use dialling prefixes to catch outbound calls. Does anyone have example config files how to implement this config? This would be the setup : PRI - Asterisk -- Legaxy PBX on TE210P |- SIP phones Would it be possible to use FreePBX to setup such routing (inbound and outbound), if anyone could guide me in the basic direction for this I would be most grateful. 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
Re: [Asterisk-Users] call center running Asterisk -sound quality-critical!
Matt Roth wrote: These statements seem contradictory. I know of no way (short of a custom patch) to tell Monitor() to mix the in and out legs prior to writing them to disk. On the other hand, MixMonitor() does just that and I believe it also buffers the writes in a way that circumvents the I/O bottleneck associated with Monitor(). Kevin P. Fleming wrote: Both of these statements are correct. Wai Wu wrote: Except that mixmonitor still has a bug in it. Wai, Please explain how the in and out channels are mixed first before they are written to the disk using monitor with no mixing onto the scsi drive. I'd love to implement this on our system to cut in half the I/O associated with Monitor(). Also, what bug does MixMonitor() have? It is my understanding that MixMonitor() is based on ChanSpy() and we seem to be having an issue with ChanSpy() where the legs of a call fall out of synch. My hunch is that it has to do with a caller being muted or placed on hold. Do these issues seem related? Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer ___ --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] call center running Asterisk-sound quality-critical!
Yes. That's is the one. It is resolved now. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tamas Sent: Wednesday, April 12, 2006 1:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] call center running Asterisk-sound quality-critical! Wai Wu wrote: Except that mixmonitor still has a bug in it. What kind of bug? Issue number? FYI: yesterday one issue has been fixed :D http://bugs.digium.com/view.php?id=6457 Did you mean that type of bug? If something else, please let us know... T. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin P. Fleming Sent: Wednesday, April 12, 2006 11:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] call center running Asterisk -sound quality-critical! Matt Roth wrote: These statements seem contradictory. I know of no way (short of a custom patch) to tell Monitor() to mix the in and out legs prior to writing them to disk. On the other hand, MixMonitor() does just that and I believe it also buffers the writes in a way that circumvents the I/O bottleneck associated with Monitor(). Both of these statements are correct. __ ___ --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] Cisco 7960 won't dial (sccp)
I'm trying to setup a couple of Cisco 7960's in asterisk. I have asterisk
working fine for sip clients, and can call the 7960's just fine, but
I can't seem to dial out on them.
As soon as I enter the first digit, the phone attempts to dial it without
waiting for the rest. I've changed timeout settings, etc but can't seem to
get it to work. Any ideas?
Asterisk SVN-trunk-r7498
chan_sccp-20060207
[general]
servername = asterisk
keepalive = 60
debug = 10
context = from-sccp-internal
dateFormat = M/D/YA
port = 2000
disallow=all
allow=alaw
allow=ulaw
allow=g723
firstdigittimeout = 60
digittimeout = 8
autoanswer_ring_time = 0
autoanswer_tone = 0x32
remotehangup_tone = 0x32
transfer_tone = 0
callwaiting_tone = 0x2d
musicclass=default
language=en
rtptos = 184
echocancel = on
silencesuppression = off
trustphoneip = no
tos = 0x68
[devices]
type = 7960
autologin = 2002
description = phone2002
dtmfmode = inband
imageversiom = P00307020200
dnd = on
trustphoneip = no
speeddial = 2000
private = on
device = SEP00036BC3852B
[lines]
id = 2002 ; future use
pin = 1234 ; future use
label = 2002 ; button line label
description = Line 2002 ; top diplay description
context = from-sccp-intenal
incominglimit = 2
transfer = on
mailbox = 1001
vmnum = 2999
cid_name = Phone2002; caller id name
cid_num = 2002
trnsfvm = 1000
secondary_dialtone_digits = 9
secondary_dialtone_tone = 0x22 ; outside dialtone
musicclass=default
language=en
rtptos = 18
echocancel = on
silencesuppression = off
line = 2002
extensions.conf
[from-sccp-internal]
include = local-extensions
include = always-out-pots
include = local-calls-pots
include = ld-calls
exten = h,1,Hangup
exten = i,1,Congestion
exten = i,2,Hangup
[always-out-pots]
exten = _9XXX.,1,Dial(Zap/1/${EXTEN}:1)
exten = _9XXX.,2,Goto(102)
exten = _9XXX.,102,Congestion
exten = _9XXX.,103,Hangup
[local-extensions]
exten = 2002,1,Dial(SCCP/2002)
exten = 2002,2,Voicemail(u2002)
exten = 2002,102,Voicemail(b2002)
exten = 2002,103,Hangup
asterisk*CLI
-- SEP00036BC3852B: New call on line 2002
-- SEP00036BC3852B: New call on line 2002
-- SEP00036BC3852B: Cisco Digit: 0009 (9) on line 2002
-- SEP00036BC3852B: Cisco Digit: 0009 (9) on line 2002
-- SEP00036BC3852B: Ending call 1 on line 2002
-- SCCP: Asterisk request to hangup Outbound channel SCCP/2002-0001
-- SEP00036BC3852B: Ending call 1 on line 2002
-- SCCP: Asterisk request to hangup Outbound channel SCCP/2002-0001
1
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Re: [Asterisk-Users] * 1.2.4 1.2.6: Ringing anamoly
Ronald Lewis wrote: I was alerted the other day by of all people, my mom, that she wasn't hearing a ring when she dialed my number. Puzzled, I tried calling myself. The call connects, but there's dead silence until voicemail picks up. Calling internally, extensions worked perfectly. So, I figured, another damned Broadvoice issue. For kicks, I upgraded to 1.2.6 today, and the end result is the same. So, I went to the dialplan playground, and removed a few lines for testing. It turns out that if I playback a file before ringing an extension, ringing works fine. Without, dead silence. Any ideas? Just out of curiosity did you happen to put an Answer() before playing audio or ringing? I use BroadVoice also and I used to have the exact same problem but putting Answer() as the first step in the context before playing my menu solved the problem. -Chris -- Chris Shaw IT Manager Precision Pump, Inc 150 N Main St Banks, OR 97106 Phone: 503-324-2361 Fax: 503-324-2203 E-Mail: [EMAIL PROTECTED] ___ --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] Callback Agents and Dial 'g' option
I'm unable to get the Dial option 'g' to work with callback agents. The plan is to use it so that I can redirect a customer to a menu so they can rate the call they just had with the agent. However, when the agent hangs up the call does not continue in the dialplan. I login with the agent. Call joins the queue. The agent and call get connected. The agent hangs up and the call should continue to the Playback(beep) and the Noop(), however the call is hung up on both sides. Extensions.conf: [default] ; Handle login and logout exten = ,1,Agentcallbacklogin(1,,[EMAIL PROTECTED]) exten = ,1,AgentCallbackLogin(1,s) ; join the queue exten = ,1,Answer exten = ,2,Queue(testing) [queue] exten = 1,1,Dial(Sip/4000||got) exten = 1,2,Playback(beep) exten = 1,3,Noop(Jump to the QA menu now) Any ideas? --johann ___ --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] playback soundfile stored in mysql database
.want to playback a raw binary file without writing into an intermediate file which would increase latency From: Alexander Lopez [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] playback soundfile stored in mysql database Date: Wed, 12 Apr 2006 13:17:13 -0400 Look at using EAGI. Hi Guys, I want to playback a sound file stored in mysql database in my perl scriptpls can anyone help with an idea? response would be greatly appreciated Rgds ___ --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 _ Express yourself instantly with MSN Messenger! Download today - it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ --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] free video (soft) phone available?
We use Neos from neosmt.com to connect to our interoffice jabber server and I noticed recently that it can do video and audio via a h.323 gatekeeper. Haven't tried it out yet but you might. Ronald Wiplinger wrote: I am using eyebeam and I am happy with it. However, it is boring just to talk to my son in the other room. Whenever I try to convince somebody to buy eyebeam, they are scared of the price. Is there a free video soft phone available, that will work with eyebeam / asterisk? bye Ronald Wiplinger ___ --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 -- Mojo [EMAIL PROTECTED] Office Manger, Horan Company, LLC (907) 747- x112 ___ --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] DUNDi with SIP
Anyone out there have a functional DUNDi configuration using SIP for the inter-Asterisk transport? I've gotten it to work with IAX2, but if I change it to SIP it does not pass the call over even though it knows where to send it. Thanks. The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ --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] URL in Queue App / Determining the DID/Queue at Agent's Phone
Have you tried something like:exten = 2,1,SetCIDName(QUEUENAME: ${CALLERIDNAME})exten = 2,n,Queue(QUEUENAME)On 4/12/06,
Steve Feinstein [EMAIL PROTECTED] wrote:
Thanks!, I will definitely take a look at that.We were hoping not tohave to do AGI in the client, but if we have to, we have to.It'llprobably be useful for other things down the road.-Steve Feinstein
GatherWorks Inc.BJ Weschke wrote: On 4/12/06, Steve Feinstein [EMAIL PROTECTED] wrote: I'd like for our custom soft phone to be able to know what queue, and/or
what DID is calling an Agent's phone before the agent picks up.The agent is using the AGENTCALLBACKLOGIN.One agent can be in multiple queues so it would be nice if they could get a pop up window telling
them who's on the line before they pick up and hear the announcement telling them that.I'd like to lose the announcment all together. It seems like that the phone can easily know what extension was dialed
to make it ring, but at best that's the phone client's extension (The server dialed it via the Local/ interface), and at worst it's 's'.Is there anyway I can know the DID of the person who called into the Queue?
I've done ethereal traces and it seems like the DID, that actually called the agent/phone is no where to be found. I've tried also to use the URL string in the Queue() application, but
the server doesn't seem to send it.(I've also tried having the client send a URL, and it seems to get sent, yet the server doesn't seem to forward it.It seems to just get lost).
Has anyone gotten the URL in the Queue application to work?And if it does, it it delivered to the phone before, or after the phone answers? Any hacks,tips,tricks,pointers, would be most appreciated.
http://bugs.digium.com/view.php?id=6843Here's code to fire off an AGI to do pretty much anything you need to
do on the calling channel after a Queue Member has been assigned to it. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/
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