Re: [asterisk-users] Problems sending more than 2 SMS with asterisk / smsq
Am Montag, den 09.07.2007, 17:21 +0200 schrieb Matthias Huber: When i send more than one messages shortly after the other, my log (/var/spool/asterisk/sms ) looks like this and only two of four messages arrive. What am i doing wrong ? I am using an AVM B1 PCI with chan-capi and 1.4.4. and also, when sending with smsq -x only two of the messages are handled. (i thought, asterisk itself handles the queues ? ) Here the log: 2007-07-09T15:04:14 YOM04 0 - 0172xxx test11 2007-07-09T15:04:15 ?OM05 0 - 0172xxx test12 2007-07-09T15:07:51 YOM06 0 - 0172xxx test13 2007-07-09T15:07:53 ?OM07 0 - 0172xxx test14 sorry - i am a total newbie at asterisk. My experience with sending several subsequent short messages is that this might run you into a timing issue. Whyever, some calls will not successfully transmit the first two packets of the SMS handshake, resulting in a non-delivery. This can be seen on the CLI, so perhaps your problem shows up there as well: Try asterisk -r CLI set verbose 10 (keep CLI open) and send those messages. I would expect those failing messages to show a different pattern. I got this failing probability _way_ down by using an additional Wait(1) or Wait(2) in the dialplan where the SMS sending happens, after bringing up the line and before sending the SMS proper. HTH Anselm ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] gui for conferencing
Is there something simple like gastman that provides functionality to establishing conferencing? -- Eric Smith ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] open source screen pop software for asterisk
On Fri, Jul 13, 2007 at 10:54:29PM -0500, RENZZO SOTOMAYOR wrote: Hi! I am new here. Well I'm doing a call center using asterisk and I'm looking for an open source screen pop software to pop the caller's information, its call history and others things. i was looking around and find the U-rang2 the problem is that it isn't open source. if someone knows about an open source screen pop please tell me. There are a number of such programs. One major issue is how you link a phone to a computer system. Not that it is such a technically difficult problem, but you would usually use some sort of additional information. You have not mentioned the target platform. So start at http://voip-info.org/wiki/view/Asterisk+GUI#UserInterfaces -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Info about Providers
Al, If you don't mind I am actually writing to you about a little different matter. I am new to the asterisk biz and am not too far from where you are located in PA. I am interested in starting a VOIP business like your own and was wondering how you are finding the market for new customers? I noticed that you provide many services on your site. I am interested in providing call center services and am starting to see if there is a market for this still. I would like to know your thoughts since you are already out there in the trenches. Thanks, Al On 7/13/07, Al Bochter [EMAIL PROTECTED] wrote: To everyone on the list I put a site on line the URL is *http://bochterservices.com/phpbb/ *This is for any information on Good or Bad ITSP You can post any problems you had with the provider You can Vote on the provider This is for allowing multiple viewpoints to be heard. If a provider receives a bad review, they are more than welcome to post So long as the exchange is fairly open and truthful And this list will be carefully moderated Please do some posting! By the way I am looking for moderators for the list if you want to help let me know. -- Best regards, Al Bochterhttp://www.BochterServices.com http://www.bochterservices.com/ --- See what we are selling at auction http://www.epier.com/auctions.asp?bochterservices --- Take a look at our online storehttp://www.bochterservices.com/onlinestore/ --- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Zaptel/mISDN and call transfer
Hi list, I am searching for a possibility to do a certain call transfer method which is called path replacement in QSIG. But I want to do that in DSS1 (EuroISDN). If my asterisk does a call transfer I want the machine to signalize on dchan that the call path has to be replaced to a direct connect between the caller and the called, i.e. my machine is to hang up after the transfer and the channels are free again. Is it possible and with what card vendor (mISDN vs.zaptel) and how do I do that? Thanks in advance, Christophorus ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel/mISDN and call transfer
Hi! I am searching for a possibility to do a certain call transfer method which is called path replacement in QSIG. But I want to do that in DSS1 (EuroISDN). They keyword to search for is explicit call transfer (ECT). At least chan_capi-com (http://www.melware.org/ChanCapi) comes with support for that. Don't know about mISDN. Cheers, Philipp ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel/mISDN and call transfer
Philipp von Klitzing schrieb: Hi! I am searching for a possibility to do a certain call transfer method which is called path replacement in QSIG. But I want to do that in DSS1 (EuroISDN). They keyword to search for is explicit call transfer (ECT). At least chan_capi-com (http://www.melware.org/ChanCapi) comes with support for that. Don't know about mISDN. Cheers, Philipp Thanks, but can I use chan_capi as frontend to mISDN or zaptel hardware? As I know I do have to choose between digium or beronet/junghanns hardware (E1) to use PRI with asterisk, right? Oh, I just caugh that I did not mention that before...sorry. Do I have to use chan_capi to access the zaptel hardware? Regards, Christophorus ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel/mISDN and call transfer
Hi! I am searching for a possibility to do a certain call transfer method which is called path replacement in QSIG. But I want to do that in DSS1 (EuroISDN). They keyword to search for is explicit call transfer (ECT). At least chan_capi-com (http://www.melware.org/ChanCapi) comes with support for that. Don't know about mISDN. Thanks, but can I use chan_capi as frontend to mISDN or zaptel hardware? You can run chan_capi on top of mISDN, but I have never done this myself (be careful to not confuse mISDN with chan_misdn). Anyway, I am not sure what mISDN hardware would be, whereas CAPI hardware would refer to cards that come along with a CAPI interface/driver like Eicon Diva or various AVM products, possibly also HST products. As I know I do have to choose between digium or beronet/junghanns hardware (E1) to use PRI with asterisk, right? Or Sangoma (which uses zaptel), or Sirrix (comes with its own channel driver). My personal suggestion would be that you take a closer look at Eicon and Sangoma. Do I have to use chan_capi to access the zaptel hardware? That won't work. By the way, you might want to also search for call deflection (CD) and partial reroute (for PtP = point-to-point connections = Anlagenanschluss) next to ECT. See: http://www.voip-info.org/wiki/view/ISDN+Features http://www.voip-info.org/wiki/view/Asterisk+CAPI+readme http://www.melware.org/ChanCapiCallReroute Cheers, Philipp ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 's' extension Asterisk 1.2.18
how can I fix this just started .. Jul 14 14:32:35 NOTICE[4983]: chan_zap.c:6223 ss_thread: Got event 18 (Ring Begin)... == Starting Zap/1-1 at bell,s,1 failed so falling back to exten 's' == Starting Zap/1-1 at bell,s,1 still failed so falling back to context 'default' Jul 14 14:32:35 WARNING[4983]: pbx.c:2377 __ast_pbx_run: Channel 'Zap/1-1' sent into invalid extension 's' in context 'default', but no invalid handler -- Hungup 'Zap/1-1' -- Starting simple switch on 'Zap/1-1' Otis ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] gui for conferencing
What do you mean establishing conferencing? There is a page for meetme conferences in the gui... -bk - Original Message - From: Eric Smith [EMAIL PROTECTED] To: Asterisk Users Mailing List asterisk-users@lists.digium.com Sent: Saturday, July 14, 2007 2:58:08 AM (GMT-0800) America/Tijuana Subject: [asterisk-users] gui for conferencing Is there something simple like gastman that provides functionality to establishing conferencing? -- Eric Smith ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Macro: s-NOANSWER, _s-.
Dear Mojo;
Looking to the below example again, there are two
lines for s-NOANSWER and s-BUSY, one line with priorty
1 and other with priority 2, and both lines are
calling the Voicemail application, so the question is:
when it will jump to priority 2 for s-NOASNWER and
s-NOBUSY?
Last thing, like what example we will use the line:
exten = _s-.,1,Goto(s-NOANSWER,1)? I mean: what the
dialed number that will suite this line (_s-.)?
[macro-voicemail]
exten = s,1,Dial(${ARG1},20)
exten = s,2,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail(u${MACRO_EXTEN})
exten = s-NOANSWER,2,Goto(incoming,s,1)
exten = s-BUSY,1,Voicemail(b${MACRO_EXTEN})
exten = s-BUSY,2,Goto(incoming,s,1)
exten = _s-.,1,Goto(s-NOANSWER,1)
Regards,
---
IP Telephony and Contact Center Engineer
Eng. Bilal Ghayad
Mobile: 009659849460
Don't pick lemons.
See all the new 2007 cars at Yahoo! Autos.
http://autos.yahoo.com/new_cars.html
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[asterisk-users] tT in callparking
Hi List; [incoming] include = parkedcalls exten=103,1,Dial(SIP/Bob,,tT) exten=104,1,Dial(SIP/Charlie,,tT) When we use tT and when we use t alone or T alone, I know this for call parking, but I do not know what the tT does? Regards Bilal Sucker-punch spam with award-winning protection. Try the free Yahoo! Mail Beta. http://advision.webevents.yahoo.com/mailbeta/features_spam.html ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Slow list
On 16:28, Thu 05 Jul 07, Philipp Kempgen wrote: Since the list was switched over to API-Digital almost every message I get is older than a week. Coincidence? Is anyone else having trouble? Regards, Philipp I got this message today July 14 Yes, I have the same. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] gui for conferencing
Check out this gui for meetme. http://sourceforge.net/projects/web-meetme/ On 7/14/07, Eric Smith [EMAIL PROTECTED] wrote: Is there something simple like gastman that provides functionality to establishing conferencing? -- Eric Smith ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 1.4 Crashes
So last night thanks to the help of the fine folks on this list I got the newest Zap 1.4 and Asterisk 1.4 compiled and installed and seemingly working except that after a few calls and a few keystrokes I get a kernel panic. The first time I had just typed reload after cleaning up some deprecated commands, the second time I had just pressed Alt+2 to get to another session. So 1.2 is back and working as it always has. This is the same problem I had when I installed the first version of 1.4 so there is clearly something wrong with 1.4 or my computer or install. Any and all suggestions welcome. I'd rather not reinstall Linux, but if it's the only choice, I will. It's CentOS with yum update run recently and all updates applied. It was originally installed from a TrixBox disk about 2 years ago so I guess it's CentOS 4. Ira ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Slow list
Well, this is now the third active thread on this subject, but I guess you won't see this message for a while. Has anyone dissected the headers of a delayed message yet? We should be able to tell for sure where the holdup is. All of the messages are coming through on time for me, so it won't do much good for me to look. Looks like mail is getting held up between INXS.digium.internal and lists.digium.com INXS.digium.internal received it the first of July, lists.digium.com received it on the 4th. drdos.info (ME) received it from lists.digium.com on that same day (Today). Attached: From - Wed Jul 04 14:19:03 2007 X-Account-Key: account2 X-UIDL: 86007 X-Mozilla-Status: 0001 X-Mozilla-Status2: X-Mozilla-Keys: Return-Path: asterisk-users-bounces at lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users Received: from lists.digium.com ([192.168.145.1]) by drdos.info with hMailServer ; Wed, 4 Jul 2007 14:11:14 -0400 Received: from lists.digium.com ([216.207.245.17] helo=lists.digium.com) by ASSP-nospam; 4 Jul 2007 14:12:33 -0301 Received: from localhost ([127.0.0.1] helo=INXS.digium.internal) by lists.digium.com with esmtp (Exim 4.63) (envelope-from asterisk-users-bounces at lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users) id 1I53rV-0005GQ-A7; Sun, 01 Jul 2007 13:09:29 -0500 Received: from exprod8mx6.postini.com ([64.18.3.106] helo=psmtp.com) by lists.digium.com with smtp (Exim 4.63) (envelope-from some.email at yahoo.com http://lists.digium.com/mailman/listinfo/asterisk-users) id 1I53rM-0005Fx-Sc for asterisk-users at lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users; Sun, 01 Jul 2007 13:09:21 -0500 -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] surge protector?
I lost one channel on an FXO module on a Sangoma A200 card due to a lightening zap in the area (well - it died the same night as a major thunder storm came through)Is there a recommended/standard surge protector for phone lines I should be using? My server has 2 POTS lines. thanks Todd ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] open source screen pop software for asterisk
I like ADM as it has a URL popup feature (open a URL with a DID or CallerID in URL). The problem is that for each call, I tend to get 4 or 5 popups... But as the other author said, there are many programs to choose from... Todd On Jul 13, 2007, at 11:54 PM, RENZZO SOTOMAYOR wrote: Hi! I am new here. Well I'm doing a call center using asterisk and I'm looking for an open source screen pop software to pop the caller's information, its call history and others things. i was looking around and find the U-rang2 the problem is that it isn't open source. if someone knows about an open source screen pop please tell me. thanks in advance renzzo ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [EUQR] Re: North American voice BRI - Informal survey
Dave Donovan wrote: On 7/10/07, *Stephen Bosch* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Jason Aarons (US) wrote: Since many CLECs (Competitve Local Exchange Carriers in NA) offer fractional PRI, combined with Internet/Data, I haven't seen any demand for ISDN BRIs for voice or data since early 90s. I don't know what it's like in your area, but here, fractional PRI is just not cost competitive if you need fewer than 10 channels. -Stephen- That's because, as Jason says, it's often offered with internet in the US. In Canada we see those offered as two discrete services. It adds a bit more money to the pot in terms of the economics of displacing analog. In which case, my question would be -- is the Internet service provided over fractional PRI cost competitive with DSL? Say I take a full T1 for data here. The economics are definitely arguable. I'm looking at five or six hundred clams for 1.5 Mbps. Even though it's symmetrical, I'm not sure the extra 300 over DSL is defensible. Whether you run voice or data over them, the cost per channel -- at least here -- is the same. Let's take 10 channels at an average of 40 dollars per channel (about typical on a fractional PRI without a contract). We'll be generous and say we are using 5 channels for voice. Our voice lines are competitive, but the data? 5 x 64 = 320 kbps 5 x 40 = $200 Hmn. Would I take 2.5 Mbps with static IP at $120/month, or 5 B for 320 kbps at $200 a month? If it's for servers, I can just as easily put them in a half cabinet colocation for the same price. I get Internet with power, SLA, security and fire protection thrown in. -Stephen- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] open source screen pop software for asterisk
Hi, I work with gnudialer vicidal Best Regards On 7/14/07, Todd H [EMAIL PROTECTED] wrote: I like ADM as it has a URL popup feature (open a URL with a DID or CallerID in URL). The problem is that for each call, I tend to get 4 or 5 popups... But as the other author said, there are many programs to choose from... Todd On Jul 13, 2007, at 11:54 PM, RENZZO SOTOMAYOR wrote: Hi! I am new here. Well I'm doing a call center using asterisk and I'm looking for an open source screen pop software to pop the caller's information, its call history and others things. i was looking around and find the U-rang2 the problem is that it isn't open source. if someone knows about an open source screen pop please tell me. thanks in advance renzzo ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 's' extension Asterisk 1.2.18
Never mind the 1.2.18 messed and did not recognize the s extension any more so I just upgrade to 1.2.21.1 and fixed the problem,.weird. otis OCOSA ListAcct wrote: how can I fix this just started .. Jul 14 14:32:35 NOTICE[4983]: chan_zap.c:6223 ss_thread: Got event 18 (Ring Begin)... == Starting Zap/1-1 at bell,s,1 failed so falling back to exten 's' == Starting Zap/1-1 at bell,s,1 still failed so falling back to context 'default' Jul 14 14:32:35 WARNING[4983]: pbx.c:2377 __ast_pbx_run: Channel 'Zap/1-1' sent into invalid extension 's' in context 'default', but no invalid handler -- Hungup 'Zap/1-1' -- Starting simple switch on 'Zap/1-1' Otis ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TimeStamp a Recording
Has anyone come up with to timestamp a Recording? I am using a pretty simple dialplan to record a audio file for a hotline. I'd like to store the date and time it was recorded somewhere, Ast DB or MySQL DB. Then when the audio file is played back to a caller, the system will say something like. This message was recorded January 14th at 10 42 pm Thanks for any ideas you may have. -- *** Forrest Beck IAXTEL: 17002871718 [EMAIL PROTECTED] http://www.shift8.biz ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] calling from ACT
Yes, just get a MSTAPI compliant program that integrates with asterisk. On 7/13/07, Al lists [EMAIL PROTECTED] wrote: I was wondering if any of you guys are aware of ability to call customers by click on customer's phone number in ACT? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] surge protector?
How about connecting it to UPS (uninterpretable power supply) Standard of the shelf $40.00-$80.00 will do the trick. -- #Joseph On Sat, 2007-07-14 at 22:17 -0400, Todd H wrote: I lost one channel on an FXO module on a Sangoma A200 card due to a lightening zap in the area (well - it died the same night as a major thunder storm came through)Is there a recommended/standard surge protector for phone lines I should be using? My server has 2 POTS lines. thanks Todd ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
