Re: [asterisk-users] VoIP Termination in Japan
On 5/5/10, Adrian Marsh adrian.ma...@ubiquisys.com wrote: Anyone have any experience with a Japanese local VoIP termination supplier? I've emailed a few companies looking to setup some PSTN to SIP and SIP to PSTN termination, but no luck so far. Thanks, Adrian -- Sent from my mobile device -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoIP Termination in Japan
On 5/5/10, Adrian Marsh adrian.ma...@ubiquisys.com wrote: Anyone have any experience with a Japanese local VoIP termination supplier? I've emailed a few companies looking to setup some PSTN to SIP and SIP to PSTN termination, but no luck so far. Thanks, Adrian -- Sent from my mobile device -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how does wrapuptime work in queue.conf
Hi Barry,
I used a while loop and Playback() like you suggested. It does the
job. Thank you for the suggestion. I just thought there might be
some built-in function or parameters in queue.conf that can do the
trick.
Thanks.
Andy
On Thu, Aug 27, 2009 at 12:32 PM, Barry L. Klineblkl...@attglobal.net wrote:
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Hash: SHA1
Andy Kuo wrote:
Hi Barry,
Thank you for the hint, but I forgot to mention that we have a few
advertisements, and we want the callers to listen to only one at a
time, and in a round robin or random order. Using Playback() doesn't
seem to serve that purpose. Is there any better way to achieve that?
Use the RAND function to generate or pick a filename.
exten = Set(advert=advert${RAND(1,10)})
exten = Playback(${advert})
That of course assumes that your advertisements are in files named
advert1.xxx through advert10.xxx (where xxx is wav,sln,etc)
Barry
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RhKepfm4CplaaeCHwtbpzWI=
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Re: [asterisk-users] how does wrapuptime work in queue.conf
Hi Lenz, That's what I was doing, putting the ad in MOH, but the callers only hear it when the agents are busy. When there are available agents, the callers just got connected to the agents without delay and hear no ads. The combination of a while loop and Playback() seem to be the only way to do it so far. Thanks. Andy On Wed, Sep 2, 2009 at 12:09 AM, Lenz Emilitrilenz.lo...@gmail.com wrote: Aht i would do is prepare a music on hold that has embedded the advertisements ( like one every 20 or 30 seconds) so that the caller hears more advertisements as the call progresses; and they are queued immediately, so no time is wasted. l. 2009/8/27 Andy Kuo aku...@gmail.com Hi Barry, Thank you for the hint, but I forgot to mention that we have a few advertisements, and we want the callers to listen to only one at a time, and in a round robin or random order. Using Playback() doesn't seem to serve that purpose. Is there any better way to achieve that? Thanks. Andy On Thu, Aug 27, 2009 at 11:56 AM, Barry L. Klineblkl...@attglobal.net wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Andy Kuo wrote: Hi list, I'd like to have the callers to listen to the advertisement (music on hold) before the agents answer them. So, I have wrapuptime=10 in queue.conf, but the call still goes straight to the agents without delay. Andy -- wrapuptime is the number of seconds that Asterisk waits between the time a agent hangs up with a caller and the next time that Asterisk sends a call to the newly-available agent. Wrap up time gives the agent a few moments to complete his last call and prepare for the next. What you need to do is use Playback() for your advertisement, then Queue() the call. Otherwise it acts just as you said, provided an agent is available. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFKltbjCFu3bIiwtTARAjE0AKCGFEchqYoGWyaeHqlIH+iNyzBKygCgqibn X/gSnE7W7EHnwiUpRC1FLRs= =pdMh -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Loway - home of QueueMetrics - http://queuemetrics.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] how does wrapuptime work in queue.conf
Hi list, I'd like to have the callers to listen to the advertisement (music on hold) before the agents answer them. So, I have wrapuptime=10 in queue.conf, but the call still goes straight to the agents without delay. Here's my queue.conf: [general] persistentmembers = yes [738] musiconhold = empty ;musiconhold = default ;announce = q-738 ;strategy = ringall strategy = rrmemory servicelevel = 60 ;announce-frequency = 60 ;periodic-announce-frequency=60 ;announce-holdtime = yes|no|once ;announce-holdtime = yes ;announce-round-seconds = 10 ; ; A context may be specified, in which if the user types a SINGLE ; digit extension while they are in the queue, they will be taken out ; of the queue and sent to that extension in this context. ;context = qoutcon timeout = 10 retry = 2 ; ;weight=0 ; ; After a successful call, how long to wait before sending a potentially ; free member another call (default is 0, or no delay) wrapuptime=10 ; maxlen = 0 ; ; How often to announce queue position and/or estimated holdtime to caller (0=off) ;announce-frequency = 60 ; How often to make any periodic announcement (see periodic-announce) ;periodic-announce-frequency=60 ; ; Should we include estimated hold time in position announcements? ; Either yes, no, or only once. ; Hold time will be announced as the estimated time, ; or less than 2 minutes when appropriate. ;announce-holdtime = yes|no|once ; ; What's the rounding time for the seconds? ; If this is non-zero, then we announce the seconds as well as the minutes ; rounded to this value. ; announce-round-seconds = 10 ; ; Use these sound files in making position/holdtime announcements. The ; defaults are as listed below -- change only if you need to. ; queue-youarenext = queue-youarenext ; (You are now first in line.) queue-thereare = queue-thereare; (There are) queue-callswaiting = queue-callswaiting ; (calls waiting.) queue-holdtime = queue-holdtime ; (The current est. holdtime is) queue-minutes = queue-minutes ; (minutes.) queue-seconds = queue-seconds ; (seconds.) queue-thankyou = queue-thankyou ; (Thank you for your patience.) queue-lessthan = queue-less-than; (less than) queue-reporthold = queue-reporthold ; (Hold time) periodic-announce = queue-periodic-announce; (All reps busy / wait for next) reportholdtime = no ; before the member hears any announcement messages), set this to the number of ; seconds to delay. ; memberdelay=1 ; ; If timeoutrestart is set to yes, then the timeout for an agent to answer is ; reset if a BUSY or CONGESTION is received. This can be useful if agents ; are able to cancel a call with reject or similar. ; ; timeoutrestart = no ; ; Each member of this call queue is listed on a separate line in ; the form technology/dialstring. member means a normal member of a ; queue. An optional penalty may be specified after a comma, such that ; entries with higher penalties are considered last. ; ;member = Zap/1 ;member = Zap/2 member = Agent/151 member = Agent/152 member = Agent/153 member = Agent/154 member = Agent/155 member = Agent/156 member = Agent/157 member = Agent/158 ;member = Agent/101 ;member = Agent/102 ;member = Agent/103 ; ; Note that using agent groups is probably not what you want. Strategies do ; not propagate down to the Agent system so if you want round robin, least ; recent, etc, you should list all the agents in this file individually and not ; use agent groups. ; ;member = Agent/@1 ; Any agent in group 1 ;member = Agent/:1,1 ; Any agent in group 1, wait for first ; available, but consider with penalty [vip] musiconhold = default announce = Q-vip strategy = rrmemory servicelevel = 60 timeout = 10 retry = 2 weight=5 wrapuptime=0 maxlen = 0 ; announce-frequency = 60 ;periodic-announce-frequency=60 announce-holdtime = yes announce-round-seconds = 10 queue-youarenext = queue-youarenext ; (You are now first in line.) queue-thereare = queue-thereare; (There are) queue-callswaiting = queue-callswaiting ; (calls waiting.) queue-holdtime = queue-holdtime ; (The current est. holdtime is) queue-minutes = queue-minutes ; (minutes.) queue-seconds = queue-seconds ; (seconds.) queue-thankyou = queue-thankyou ; (Thank you for your patience.) queue-lessthan = queue-less-than; (less than) queue-reporthold = queue-reporthold ; (Hold time) ;periodic-announce = queue-periodic-announce; (All reps busy / wait for next) ; reportholdtime = no ; ;;memberdelay=1 ;; timeoutrestart = no ; member = Agent/151 member =
Re: [asterisk-users] how does wrapuptime work in queue.conf
Hi Barry, Thank you for the hint, but I forgot to mention that we have a few advertisements, and we want the callers to listen to only one at a time, and in a round robin or random order. Using Playback() doesn't seem to serve that purpose. Is there any better way to achieve that? Thanks. Andy On Thu, Aug 27, 2009 at 11:56 AM, Barry L. Klineblkl...@attglobal.net wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Andy Kuo wrote: Hi list, I'd like to have the callers to listen to the advertisement (music on hold) before the agents answer them. So, I have wrapuptime=10 in queue.conf, but the call still goes straight to the agents without delay. Andy -- wrapuptime is the number of seconds that Asterisk waits between the time a agent hangs up with a caller and the next time that Asterisk sends a call to the newly-available agent. Wrap up time gives the agent a few moments to complete his last call and prepare for the next. What you need to do is use Playback() for your advertisement, then Queue() the call. Otherwise it acts just as you said, provided an agent is available. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFKltbjCFu3bIiwtTARAjE0AKCGFEchqYoGWyaeHqlIH+iNyzBKygCgqibn X/gSnE7W7EHnwiUpRC1FLRs= =pdMh -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] help - How to send hangup command to call in progress.
Hi Singh, Have you tried soft hangup? Andy On Wed, Mar 25, 2009 at 4:38 PM, Singh Saimbhi singh.saim...@palm.com wrote: Hi, I want to send hangup command to the call which was logged in earlier via cli. Lets say to '5aec0e7207b24c8e1bdb511a460f7...@callcentric.com Basically I want to hang up the call when ever I want but from the script. Thanks, Singh ___ -- 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] remove queue call
Hi, Try CLI soft hangup Local. Andy On 8/27/08, Rilawich Ango [EMAIL PROTECTED] wrote: Hi all, I have the following queue and members. I found that there is a call stuck in the queue so other call can't enter the queue. I want to know whether we can remove the call (by CLI) to free the queue. ango 2700 has 1 calls (max unlimited) in 'rrmemory' strategy (35s holdtime), W:0, C:134, A:48, SL:88.8% within 120s Members: Local/[EMAIL PROTECTED] (dynamic) (Not in use) has taken no calls yet Local/[EMAIL PROTECTED] (dynamic) (Not in use) has taken no calls yet Callers: 1. Local/[EMAIL PROTECTED],2 (wait: 113:19, prio: 0) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] karaoke functionality
Hi, Why not use MixMonitor(), so you have a single file with the singer and the music? Thanks. Andy On 5/20/08, Sherwood McGowan [EMAIL PROTECTED] wrote: Arjan Kroon | Mobillion wrote: Hi, Is it possible top use a form of Karaoke Functionality? When a caller calls a number, he hears a voicefile. During this voicefile he sings along with this voicefile. Is it possible to record what the caller is singing? Grt, ___ -- 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 Yes, this is entirely possible using Monitor(). http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Monitor When you record a conversation using the monitor command, the end result is two files, a [name]-in.[ext] file and a [name]-out.[ext] fileI believe you're looking for the input side, I always get them confused Just be sure not to use the m option, that would mix the two channels together into a single sound file. Hope this helps, Sherwood McGowan ___ -- 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] zapata.conf: cannot set txgain lower than -6.3 ?
Hi Steve, I tried txgain as low as -18 without any problem, but I never tried anything with decimal points. Andy On 12/12/06, Steve Hsieh [EMAIL PROTECTED] wrote: Greetings everyone, I have a Digium TDM400P card with both an FXO and FXS module to connect to the phone company and to a standard phone. The problem is that the volume of my voice is going out too loud. I tried lowering the txgain value in zapata.conf to compensate, but all audio drops out completely if I set txgain to -6.4 or lower. If I set it to -6.3, then everything works (but still too loud). Is there a limitation as to how low txgain can be set? From voip-info.org, I was under the impression that the range was -100 to 100. Even the page at http://www.voip-info.org/wiki/index.php?page=Asterisk+config+zapata.conf shows an example txgain of -15.9 Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Huh? An ilbc frame that isn't a multiple of 50 bytes long from RTP (4)?
Hi, Has anyone seen this before? Sep 12 22:31:38 WARNING[17472]: codec_ilbc.c:175 ilbctolin_framein: Huh? An ilbc frame that isn't a multiple of 50 bytes long from RTP (4)? I found a bug tracker http://bugs.digium.com/view.php?id=6333 talking about this, but didn't really understand why it happened and how to correct it. Any suggestions? Thanks. Andy ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] strange problem with calls between MGCP and SIP clients(ATA's)
Hi, We have experience problems with calls between MGCP ATA's and SIP ATA's (Linksys PAP2-NA). A call from MGCP or SIP to the other connects normally and the conversation can usually last around 30 seconds and it becomes one-way audio. What I don't understand is how the calls can be set up and talk for a few seconds without problems and suddenlly go wrong. If there are problems, such as misconfiguration, the call should not even be connected, or at least the on-way audio problem should start right from the beginning, shouldn't it? I know MGCP is not very popular here, but we have quite a few of them on hand that we would really like to use. Any comments/suggestions are greatly appreciated. Thanks. Andy ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk t38passthrough
Hi Ricardo, On a 1.2.4 with the T.38 patch, I tried both t38pt_udptl = yes t38pt_rtp = yes t38pt_tcp = yes and t38pt_udptl = yes t38pt_rtp = no t38pt_tcp = no but still got ...chan_sip.c:3716 process_sdp: Unknown SDP media type in offer: image 5144 UDPTL t38 Warnings I tried it on Kapanga Softphone as suggested, and I'll tried it on Grandstream ATA's later. Are there anything I'm missing? Thank you. Andy On 8/24/06, Ricardo Carvalho [EMAIL PROTECTED] wrote: Hi, I've installed Asterisk t38passthrough branch and I'm using one Grandstream ATA to connect Asterisk to a Fax machine. Every time I send a fax, it gets sent using codec G711, and never T.38. I added the following parameters in the [general] section as well as in device configurations: t38pt_udptl = yes t38pt_rtp = yes t38pt_tcp = yes I think that's the only thing that is needed to do to enable T.38 pass through... Why does Asterisk keeps sending in G711? Any help? Regards, Ricardo. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth 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] FAX questions
Hi Marco, I'm using T406P(with hardware EC) with a T1-PRI, and I'm having trouble sending fax out though SIP ATA in the same LAN subnet with the Asterisk box. I can send fax out using txfax in call file, but I did have to lower the rxgain and txgain. This is what I'm trying to do: Fax machine --- SIP ATA --LAN-- Asterisk --PRI-- PSTN Have you tried this? Do you have to disable Echo canneler? Thanks. Andy On 8/15/06, Marco Mouta [EMAIL PROTECTED] wrote: Hi, Another question. With latest version of asterisk softwares am I able using rxfax? I had read some remarks about incompatibility between TDM card and rxfax. Is it still exist? I've been using rx for fax reception with TE110P as well as X100P (this only for tests and learning) with very success. As far as i know what could be a problem is that SpanDSP doesn't implements ECM (error correction mode) For Fax reception, only with X100P i've had to play with rxgains, nothing else. I've had some problems only for tx fax lots of errors transmiting faxs, but i think that could be because my * is behind a legacy pbx and i could be facing time sinchronization problems. bye, Zsolt ___ --Bandwidth 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] Re: need a pointer regarding scripting asterisk
Hi, Can you give a quick example on how to query an EXTERNAL database? Thank you. Andy On 7/29/06, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Fri, Jul 28, 2006 at 04:08:19PM -0500, shawn bright wrote: i would use a dial plan, but we are monitoring about 1200 units in the field, i thought a dial plan would be a little long or complex for that. I suppose that i could use a dial plan and set guys up by editing the extensions.conf file for each one ? I just thought it might be easier to script it somehow. You can always generate part of extensions.conf automatically and #include it. It will be updated by, e.g., 'extensions reload'. Maybe you'll also find a smart way to do that using wildcards or whatever. You can also query the internal asteriskdb or an external dataase from the dialplan. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.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 ___ --Bandwidth 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] SOLVED: IAX jitter / clocking problem
Hi, I too would like to set a minimum jitterbuffer value, and that seems to mean that I need to use the old jitterbuffer implementation. Have you compared the 2 implementations? What are the advantages of using the new one and what are the disadvantages of using the old one? Thanks. Andy On 6/30/06, Pavel Jezek [EMAIL PROTECTED] wrote: I found my mistake jiterbuffer=yes vs. jitterbuffer=yes ;-) currently I have this settings, and seems this working quite well, only sometimes gaps appears, when jitter changes too much eg. 500ms - jitterbuffer probably can't adapt so quick, maybe good idea to set some minimum jitterbuffer value, but this is not possible in current new jitterbuffer implemenation PJ jitterbuffer=yes forcejitterbuffer=yes maxjitterbuffer=1500 maxjitterinterps=10 resyncthreshold=2000 Pavel Jezek wrote: hello, I'm still trying to tune iax jitterbuffer in asterisk 1.2.9.1, but without success, I'm using idefisk-asterisk over cdma network, where rtt is about 100-500ms, so jitter about 400ms but sound is very jerky, in diection idefisk-asterisk, in reverse direction is sound relatively smoth, so, my question: has iax same problem as in sip/rtp, where packets are generated along incomming packets (what is probably solved in trunk with: http://bugs.digium.com/view.php?id=5374 0005374: [patch] Asynchronous generation of outgoing frames when timing device available my iax jitterbuffer settings (iax.conf): [general] jiterbuffer=yes forcejitterbuffer=yes maxjitterbuffer=1500 resyncthreshold=-1 thanks for suggestions ;-) PJ ___ --Bandwidth 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] call quality statistics?
try iax2 show netstats On 6/23/06, Dr. Michael J. Chudobiak [EMAIL PROTECTED] wrote: Is it possible to set up some sort of call-quality statistics reporting/logging for IAX2 calls? Something that can keep track of dropped packet / jitter trends? (I know iax2 show channels shows this info for active calls.) Suggestions appreciated! - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth 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 t38 gateway tests
Hi Carlos, I missed the first part(s) of the conversation, but I think this is for Asterisk T.38 support. I tried the t.38 patch on Asterisk 1.2.4 about 2 months ago, but it still showed error messages when I tried t.38 fax on 2 t.38 enabled ATAs. (Grandstream HT286) Can I test T.38 with you? We have a 1.2.0 with PRI's on it, and 3 others running 1.2.4 and 1.2.7.1. They are all connected to each other through IAX2. Please let me know what you need from us to test with you. Thanks. Andy On 6/15/06, Carlos Alperin [EMAIL PROTECTED] wrote: Are you still interested on tests? Regards, Carlos Alperin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, April 25, 2006 2:46 PM To: asterisk-users@lists.digium.com; users@openser.org; [EMAIL PROTECTED] Subject: [Asterisk-Users] Sip t38 gateway tests Hello, I patched asterisk patched with the latest t38 support . I would need some people for tests. Regards harry ___ Faites de Yahoo! votre page d'accueil sur le web pour retrouver directement vos services préférés : vérifiez vos nouveaux mails, lancez vos recherches et suivez l'actualité en temps réel. Rendez-vous sur http://fr.yahoo.com/set ___ --Bandwidth 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] ast_sched_runq ran 281 scheduled tasks all at once
We get these messages too, but they don't seem to cause any problems. Are you connecting 2 * (with different versions) via IAX2? Are these messages only appear on the lower version one? I asked a similar question on the list, and the suggestion was to upgrade them to the same version. Hope this helps. Let us know how it goes. Andy On 4/13/06, Gareth Blades [EMAIL PROTECTED] wrote: Just noticed that I occasionally get these messages:- Apr 12 09:27:03 WARNING[11390] chan_iax2.c: chan_iax2: ast_sched_runq ran 281 scheduled tasks all at once Apr 13 09:13:18 WARNING[11390] chan_iax2.c: chan_iax2: ast_sched_runq ran 1987 scheduled tasks all at once Apr 13 12:47:56 WARNING[11390] chan_iax2.c: chan_iax2: ast_sched_runq ran 1804 scheduled tasks all at once Are they anything to be concerned about? ___ --Bandwidth 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] Steps to make trunked iax2
Hi, I'm connecting 2 Asterisk with IAX2. One is running 1.2.0, and the other is running 1.2.4 It has been working OK so far, except I get messages like these occassionally Apr 10 11:14:54 WARNING[13081]: chan_iax2.c:7971 network_thread: chan_iax2: ast_sched_runq ran 42 scheduled tasks all at once Apr 10 11:14:55 WARNING[13081]: chan_iax2.c:7971 network_thread: chan_iax2: ast_sched_runq ran 2206 scheduled tasks all at once Not sure what these means and how to avoid them, and not sure if these are caused by connecting 2 different versions. Any ideas? Thanks. Andy On 4/7/06, Rich Adamson [EMAIL PROTECTED] wrote: jonny hashem wrote: Hi: Is the difference of Asterisk verisons on two servers effect on the iax2 trunking between them ? Yes, without a doubt. However, I've not tried to keep track of the changes and would only be speculating on compatibility issues, etc. That happens to be one of the reasons why some itsp's provide less then quality audio on iax links (eg, older versions). ___ --Bandwidth 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] What causes deadlock?
I sometimes get these WARNINGs too. I would like to know what causes it and how to avoid them too. Thanks. Andy On 4/5/06, Chuck Bunn [EMAIL PROTECTED] wrote: Hi What causes deadlock? Apr 5 14:02:43 WARNING[2413] channel.c: Avoided initial deadlock for '0x82acb10', 10 retries! Apr 5 14:02:43 WARNING[2413] channel.c: Avoided initial deadlock for '0x8298160', 10 retries! Here is the portion of the log: Apr 5 14:02:42 NOTICE[23363] chan_zap.c: Got event 18 (Ring Begin)... Apr 5 14:02:42 VERBOSE[23363] logger.c: -- Executing Answer(Zap/5-1, ) in new stack Apr 5 14:02:42 VERBOSE[23363] logger.c: -- Executing SetMusicOnHold(Zap/5-1, default) in new stack Apr 5 14:02:42 VERBOSE[23363] logger.c: -- Executing DigitTimeout(Zap/5-1, 5) in new stack Apr 5 14:02:42 VERBOSE[23363] logger.c: -- Set Digit Timeout to 5 Apr 5 14:02:42 VERBOSE[23363] logger.c: -- Executing ResponseTimeout(Zap/5-1, 30) in new stack Apr 5 14:02:42 VERBOSE[23363] logger.c: -- Set Response Timeout to 30 Apr 5 14:02:42 VERBOSE[23363] logger.c: -- Executing GotoIfTime(Zap/5-1, 8:00-21:00|*|*|*?default|s|7) in new stack Apr 5 14:02:42 VERBOSE[23363] logger.c: -- Goto (default,s,7) Apr 5 14:02:42 VERBOSE[23363] logger.c: -- Executing Queue(Zap/5-1, extensions-home|tr|||25) in new stack Apr 5 14:02:42 VERBOSE[23363] logger.c: -- outgoing agentcall, to agent '3005', on 'Local/[EMAIL PROTECTED],1' Apr 5 14:02:42 VERBOSE[23363] logger.c: -- Called Agent/3005 Apr 5 14:02:42 VERBOSE[23363] logger.c: -- outgoing agentcall, to agent '3002', on 'Local/[EMAIL PROTECTED],1' Apr 5 14:02:42 VERBOSE[23363] logger.c: -- Called Agent/3002 Apr 5 14:02:42 VERBOSE[23363] logger.c: -- outgoing agentcall, to agent '3001', on 'Local/[EMAIL PROTECTED],1' Apr 5 14:02:42 VERBOSE[23363] logger.c: -- Called Agent/3001 Apr 5 14:02:42 VERBOSE[23365] logger.c: -- Executing Macro(Local/[EMAIL PROTECTED],2, stdexten|413|SIP/413) in new stack Apr 5 14:02:42 VERBOSE[23365] logger.c: -- Executing Set(Local/[EMAIL PROTECTED],2, DYNAMIC_FEATURES=automon) in new stack Apr 5 14:02:42 VERBOSE[23365] logger.c: -- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/413|20|Ttw) in new stack Apr 5 14:02:42 VERBOSE[23365] logger.c: -- Called 413 Apr 5 14:02:42 VERBOSE[23366] logger.c: -- Executing Macro(Local/[EMAIL PROTECTED],2, stdexten|510|SIP/510) in new stack Apr 5 14:02:42 VERBOSE[23366] logger.c: -- Executing Set(Local/[EMAIL PROTECTED],2, DYNAMIC_FEATURES=automon) in new stack Apr 5 14:02:42 VERBOSE[23366] logger.c: -- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/510|20|Ttw) in new stack Apr 5 14:02:42 VERBOSE[23366] logger.c: -- Called 510 Apr 5 14:02:42 VERBOSE[23367] logger.c: -- Executing Macro(Local/[EMAIL PROTECTED],2, stdexten|411|SIP/411) in new stack Apr 5 14:02:42 VERBOSE[23367] logger.c: -- Executing Set(Local/[EMAIL PROTECTED],2, DYNAMIC_FEATURES=automon) in new stack Apr 5 14:02:42 VERBOSE[23367] logger.c: -- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/411|20|Ttw) in new stack Apr 5 14:02:42 VERBOSE[23367] logger.c: -- Called 411 Apr 5 14:02:42 VERBOSE[23366] logger.c: -- SIP/510-1cb8 is ringing Apr 5 14:02:42 VERBOSE[23363] logger.c: -- Agent/3002 is ringing Apr 5 14:02:43 VERBOSE[23365] logger.c: -- SIP/413-d49e is ringing Apr 5 14:02:43 VERBOSE[23363] logger.c: -- Agent/3005 is ringing Apr 5 14:02:43 VERBOSE[23365] logger.c: -- SIP/413-d49e is ringing Apr 5 14:02:43 VERBOSE[23367] logger.c: -- SIP/411-1a1a is ringing Apr 5 14:02:43 VERBOSE[23363] logger.c: -- Agent/3001 is ringing Apr 5 14:02:43 VERBOSE[23366] logger.c: -- SIP/510-1cb8 answered Local/[EMAIL PROTECTED],2 Apr 5 14:02:43 VERBOSE[23363] logger.c: -- Agent/3002 answered Zap/5-1 Apr 5 14:02:43 VERBOSE[23365] logger.c: == Spawn extension (macro-stdexten, s, 2) exited non-zero on 'Local/[EMAIL PROTECTED],2' in macro 'stdexten' Apr 5 14:02:43 VERBOSE[23365] logger.c: == Spawn extension (macro-stdexten, s, 2) exited non-zero on 'Local/[EMAIL PROTECTED],2' Apr 5 14:02:43 VERBOSE[23367] logger.c: == Spawn extension (macro-stdexten, s, 2) exited non-zero on 'Local/[EMAIL PROTECTED],2' in macro 'stdexten' Apr 5 14:02:43 VERBOSE[23367] logger.c: == Spawn extension (macro-stdexten, s, 2) exited non-zero on 'Local/[EMAIL PROTECTED],2' Apr 5 14:02:43 VERBOSE[23366] logger.c: == Spawn extension (macro-stdexten, s, 2) exited non-zero on 'Local/[EMAIL PROTECTED],2' in macro 'stdexten' Apr 5 14:02:43 VERBOSE[23366] logger.c: == Spawn extension (macro-stdexten, s, 2) exited non-zero on 'Local/[EMAIL PROTECTED],2' Apr 5 14:02:43 WARNING[2413] channel.c: Avoided initial deadlock for '0x82acb10', 10 retries! Apr 5 14:02:43 WARNING[2413] channel.c: Avoided initial deadlock for '0x8298160', 10 retries! Apr 5 14:03:22 VERBOSE[2424] logger.c:
Re: [Asterisk-Users] ISDN and Analog DIAL UP Connection Through Asterisk and Digium TE405P
Hi, We have Rhino channel bank and TE406, and are thinking of doing the same thing. What modem or modem pool are you using? Can Asterisk serve as an access server/gateway to the internet? Please share your experience. Thank you. Andy On 3/28/06, Don Pobanz [EMAIL PROTECTED] wrote: Nico Giefing wrote: Is it possible to establish a ISDN DIAL up Connection and Analog Dial up Connection (V90) trough asterisk with Digium TE405? I do the v90 dial up. The modem is connected to an Adtran 750 channel bank. Our DID trunks are on a T1 line to the phone company. If you have analog lines to the phone company it will not work since only 1 A/D conversion is allowed! We aren't doing any IDSN. It ?may? be possible. Don Pobanz Nico Giefing ___ --Bandwidth 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] Dial plan question - exclamtion mark
Try using . instead of ! _001800NXX _X. _X. is more like a match the rest instead of match all Hope this helps. Andy On 3/22/06, Mike Hammett [EMAIL PROTECTED] wrote: http://www.voip-info.org/wiki/index.php?page=Asterisk+Dialplan+Patterns says: ! wildcard, matches zero or more characters immediately (only Asterisk 1.2 and later, see note) Note: The exclamation mark wildcard, which is available only in Asterisk 1.2 and later, behaves specially — it will match as soon as can without waiting for the dialing to complete, but it will not match until it is unambiguous, and the number being dialed cannot match any other extension in the context. It was designed for use as follows, so that as soon as the digits dialed don't match '001800...' the outgoing telephone line will be picked up and overlap dialing will be used (with full audio feedback from 'earlyb3' etc.) Context outgoing: Extension Description _001800NXXFree US calls made by VoIP _X! Outgoing calls via normal telco, with overlap dial. = So then can I have _!800NXX to match someone dialing 18005551212 and 8005551212? If not, what could I do in this situation? Mike Hammett Intelligent Computing Solutions http://www.ics-il.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 ___ --Bandwidth 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] Unable to forward frame
Hi,
I think there should be only one timing source, but you have 3 here...
Zaptel.conf
span=1,1,0,ccs,hdb3,crc4
span=2,0,0,ccs,hdb3,crc4
span=3,0,0,ccs,hdb3,crc4
span=4,0,0,ccs,hdb3,crc4
Not sure if this is causing the problem though.
Andy
On 3/15/06, James Sturges [EMAIL PROTECTED] wrote:
Hi,
I get this error in the log file when I call from my mobile to the Asterisk
server, but hang up the mobile before anyone picks up.
Normally I would not worry about it, but I have been having some bad
experiences (only recently, after about 9 months of good operation) with
asterisk, although there have been related issues with Telco lines /
equipment and also some Asterisk initiated CRC errors after upgrading to
1.2. So I have downgraded to 1.0.9.
So I can isolate everything I have just installed a VERY VERY simple dial
plan.
The setup is
Telco --- TDM 4 Port BRI --- Ericsspn BP250
Extensions.conf (all of it)
[default]
exten = s,1,Dial(ZAP/g4/211,45,t)
[dialstring]
exten = i,1,Playback(invalid)
exten = i,2,Hangup
exten = t,1,Hangup
[te405p-frombp250]
exten = _3XX,1,Answer
exten = _3XX,2,Dial(Sip/${EXTEN},6000,t)
exten = _3XX,3,Hangup
exten = _X.,1,Answer
exten = _X.,2,Dial(Zap/g1/${EXTEN},6000,t)
exten = _X.,3,Hangup
[te405p-intelstra]
exten = _X.,1,Answer
exten = _X.,2,Dial(Zap/g4/${EXTEN},6000,t)
exten = _X.,3,Hangup
[from-sip]
exten = s,1,Dial(SIP/3332,45,t)
exten = _0X.,1,Answer
exten = _0X.,2,Dial(Zap/g1/${EXTEN:1},6000,t)
exten = _0X.,3,Hangup
exten = _X.,1,Answer
exten = _X.,2,Dial(Zap/g4/${EXTEN},6000,t)
exten = _X.,3,Hangup
Zapata.conf
[channels]
context=default
musiconhold=default
switchtype=euroisdn
usecallerid=yes
cidsignalling=v23
cidstart=polarity
hidecallerid=no
callwaiting=no
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=800
rxgain=0.0
txgain=0.0
group=1
context=te405p-intelstra
;context=te405p-ext
pridialplan=local
signalling=pri_cpe
;overlapdial=yes
callerid=asreceived
channel=1-15, 17-31
;channel=32-46, 48-62
group=4
context=te405p-frombp250
;context=te405p-in
pridialplan=local
signalling=pri_net
overlapdial=yes
callerid=asreceived
channel=94-108, 110-124
;channel=32-46, 48-62
Zaptel.conf
span=1,1,0,ccs,hdb3,crc4
bchan=1-15
dchan=16
bchan=17-31
span=2,0,0,ccs,hdb3,crc4
bchan=32-46
dchan=47
bchan=48-62
span=3,0,0,ccs,hdb3,crc4
bchan=63-77
dchan=78
bchan=79-93
span=4,0,0,ccs,hdb3,crc4
bchan=94-108
dchan=109
bchan=110-124
loadzone=au
defaultzone=au
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Re: [Asterisk-Users] Max retries exceeded to host...
What ATA's are you using? I've notice occassional occurance of the same messages, and they seem to be comming from only certain type of ATA's. I'm suspecting it's ATA related, but I don't have enough evidence to prove so yet. Andy On 3/14/06, Dan Morin [EMAIL PROTECTED] wrote: The past two days, I've been having issues with my two VoIP service providers where calls just suddenly hang up. The following is from the log: Mar 14 13:50:55 WARNING[5887] chan_iax2.c: Max retries exceeded to host 64.34.45.100 on IAX2/voipjet-3 (type = 6, subclass = 11, ts=25, seqno=80) Mar 14 13:50:55 DEBUG[10428] channel.c: Didn't get a frame from channel: IAX2/voipjet-3 Mar 14 13:50:55 DEBUG[10428] channel.c: Bridge stops bridging channels SIP/759052-e6e9 and IAX2/voipjet-3 Mar 14 13:50:55 DEBUG[10428] chan_iax2.c: We're hanging up IAX2/voipjet-3 now... Mar 14 13:50:55 VERBOSE[10428] logger.c: -- Hungup 'IAX2/voipjet-3' My Asterisk box is the only thing (behind a firewall) on a dedicated T1 (Internet - 1280K) with not more than 2 VoIP calls at a time. These calls are coming from a SIP ATA into asterisk and out an IAX2 trunk. I'm running asterisk 1.2.5. If you have any ideas I would really appreciate some assistance. Thanks in advance. Dan Disclaimer: This message is intended only for the use of the noted recipient(s) and may contain information that is privileged, proprietary and confidential. If you received this message in error, you must not, directly or indirectly, distribute, use, disclose, print, or copy any part of this message. Please notify the sender if you have received this message by mistake and erase this email and its contents from your system. Note that any views or opinions expressed in this message are solely those of the author and do not necessarily represent those of the company. Finally, note that email transmission cannot be guaranteed to be secure or error-free. Information could be intercepted, corrupted, lost, destroyed, received late or incomplete, or infected with a virus. Therefore, the sender does not accept liability for any errors in the content of this message which arise as a result of email transmission. ___ --Bandwidth 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 and T38 Fax
Hi, I tried to connect two T.38 capable SIP ATA's through Asterisk. I had canreinvite=yes and the 2 ATA's did talk directly to each other, but fax still failed. From Ethereal captures, I think the problem was when the originating ATA is ready to set up a T.38 session, it sends a message to the destination ATA to get ready for receiving T.38 packets. Unfortunately, Asterisk did not pass that message to the destination, so even though the actual T.38 packets did go directly to the destination, the destination ATA doesn't know it needs to switch to T.38 mode. What I am wondering is, if I can somehow configure the ATA's to be locked to T.38 only, no voice calls allowed, will it work? (since it is ONLY for receiving T.38 packets, we don't care if Asterisk pass on the message to the destination to switch to T.38) Any thoughts on whether this will work or this is just a crazy idea? Thanks. Andy On 2/21/06, Lee Howard [EMAIL PROTECTED] wrote: Carey Mould wrote: How can I get asterisk to work with faxes in my configuration? I have a WAN with Asterisk at the centre and Mediatrix 1104 gateways at the end nodes providing tone to legacy PBX's and fax machines. The Asterisk is connected to the PSTN via a Digium single port t1. The end nodes are connected via frame-relay 128kbps links. I want to use g.729 between the end nodes and the Asterisk box at the centre. TheMediatrix box supports T38. In my reading I am not seeing where asterisk supports T38. I believe that currently Asterisk SVN supports T.38 pass-through to some extent. Others here would be able to comment on that more fully. However, what you're seeking is T.38 gateway support, and Asterisk does not support that at the present. I know that there is some intention with spandsp to get T.38 gateway support there (so Asterisk would eventually support T.38 gateway via Unicall), but I suspect that's a long way off for your present needs. Again, I'm not really the one to be able to comment on this in detail. I'm not sure if the OpenH323 project (which I know supports at least T.38 pass-through) supports T.38 gateway, but even if it did, getting OpenH323 and Asterisk to work together may be difficult at best. I'm not really the one to be able to comment on that in detail, either. So for the time being I would consider that Asterisk does not support the T.38 features that you need to make this work and that you look at a different approach to getting your faxing working. 1) If a fax machine connected to the Meditrix box sends a fax to a location on the PSTN (legacy fax machine) where is the T38 converted back? That's what a T.38 gateway does. Asterisk doesn't do it. 2)Does Asterisk handle this? No. Not yet anyway. 3) How do I configure Asterisk to do this You can't presently. 4) What is the significance on the Digium web site where they have a CYA on the T1 board not supporting faxes ? You probably are referring to these support statements on virtually all of their products: The current state of faxing is incomplete and will not be supported. Well, it pretty much means what it says. I don't think you'll get any significant degree of support from Digium if you run into trouble with faxing using their hardware. All of that said, can you push a fax call through an Asterisk PBX? Yes. Search the archives and you'll see it can be done. Lee. ___ --Bandwidth 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] RE: ZAP extension, DTMF?
Hi Dan, How is your echo can the issue? Did you disable the echo can and solve the DTMF issue? If you did, did it trade the DTMF issue with echo problem? It would nice if you can share your experience. Thanks. Andy On 2/14/06, Dan Elder [EMAIL PROTECTED] wrote: Please ignore my last query about DTMF on ZAP, turned out to be an echo can issue. ___ --Bandwidth 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] lists problem, Gmail????????
Hi, I'm using gmail, and I've been getting messages from the list and other people. Not sure what is the actual cause, but it looks to me like a subscription problem. Andy On 2/8/06, C F [EMAIL PROTECTED] wrote: Am I the only one having trouble with this list? Since the begining of the week I have not been receiving mail from the list like I used to, is this a gmail problem? or is it subscription problem? or is something wrong with the list? anybody else using gmail having any problems? ___ --Bandwidth 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] Anyway to do this?
Hi, Sorry to ask a slightly off topic question here, but I've been stuck on this for a while. My SIP ATA's are displaying callerID without problems. The problem is when a 2nd call comes in during a conversation, callwaiting callerID dosen't show up. I can only hear the callwaiting alert tones, but no callwaiting callerID. I have both callwating=yes and callwaitingcallerid=yes in my zapata.conf Can anyone please help me out here? Thanks. Andy On 2/1/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: If callerid is received, it will be displayed on the sip phones. My guess would be that it's not coming in on the analog line in the first place. PaulH Scott Geist [EMAIL PROTECTED] wrote: How do you retreive the caller id on incoming analog lines and display the id on the sip phones on the network? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to view Q.931 Disconnect code
I replied the following and got a returned to sender message from
the MAILER-DAEMON.
Not sure if you got it. Here it is again...
On 1/23/06, Andy Kuo [EMAIL PROTECTED] wrote:
Hi,
Try
exten = h,1,NoOp(${HANGUPCAUSE})
in your extensions.conf
Cheers.
Andy
On 1/23/06, Angelito Manansala [EMAIL PROTECTED] wrote:
Hi there,
Can anyone know how to view asterisk disconnect code.?
--
Best Regards,
Angelito Manansala
www.voicefidelity.net
Mobile: +63 917 542 5807
DID: (+63) 44 7906770
US DID: +1 619 399 0128
msn: [EMAIL PROTECTED]
skype: bulcrack
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Re: [Asterisk-Users] How to view Q.931 Disconnect code
Hi,
Try
exten = h,1,NoOp(${HANGUPCAUSE})
in your extensions.conf
Cheers.
Andy
On 1/23/06, Angelito Manansala [EMAIL PROTECTED] wrote:
Hi there,
Can anyone know how to view asterisk disconnect code.?
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[Asterisk-Users] Call Waiting CallerID
Hi, According to the wiki, we need to have both callwaiting=yes and callwaitingcallerid=yes , and that's what I have in zapata.conf. I can hear the call waiting alert tone when a 2nd call comes in during an established call, and I can switch between the calls without problems. However, CallerID on the 2nd call does not show up with the call waithing alert tones. Am I missing something? Can anyone help? Thank you in advance. Andy ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Can TE406P provide PRI to other VoIP gateways?
Hi, Does anyone know if it's possible to have 1 port of TE406P to provide T1 PRI to other VoIP gateways? I am trying to provide T1 PRI to one of our old Clarent gateways to integrate it to Asterisk. Can anyone help? Thank you. Andy ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Waiting CallerID not showing up
Hi All, According to the wiki, we need to have both callwaiting=yes and callwaitingcallerid=yes , and that's what I have in zapata.conf. I can hear the call waiting alert tone when a 2nd call comes in during an established call, and I can switch between the calls without problems. However, CallerID on the 2nd call does not show up with the call waithing alert tones. Can anyone help? Thank you in advance. Andy ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SpanDSP not sending to fax extension.
Hi, Are you using exten = fax,1,rxfax(. in extensions.conf and faxdetect=both in zapata.conf? If yes, have you tried assigning an extension number for receiving fax? (instead of the fax extension) Andy On 1/18/06, Ken D'Ambrosio [EMAIL PROTECTED] wrote: Hi, all. I've got a fax extension in my extensions.conf, but spandsp never sends my faxes there. Both applications -- txfax and rxfax -- are registered by Asterisk, so they compiled and installed correctly. I've got a Sangoma A104 card, and (as some people had suggested) have loaded ztdummy. Everything seems fine, except that it never gets recognized as a fax. I've even turned off echo cancellation -- same deal. Any ideas? I'm pretty stumped, here. Thanks! -Ken D'Ambrosio ___ --Bandwidth 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] Call Forwarding with Account Code.. can it be done?
I use SetAccount(${EXTEN}) when the extension gets the call. The original dialed extension will be recorded as AccountCode in CDR, before the call is forwarded. The 1st field in CDR will be the extension your customer, the 2nd will be the caller (source), the 3rd will be the forwared number.
It works for me pretty well.
Andy
On 12/6/05, Matt [EMAIL PROTECTED] wrote:
I want to allow my users to be able toCall Forward UnconditionalCall Forward Busy
Call Forward No AnswerAnd curently I am doing this via my ATA and phone settings, howeverthis has the problem that when a call is forwarded it goes out withoutan accountcode (Even though the ATA is forwarding the call), and hence
I can't track the call!Can someone suggest a way to either fix this so that accountcodes gointo the CDRs when the ATA/phone forwards the call, or to do the threeforwarding types directly on asterisk?
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Re: [Asterisk-Users] Context confict question??
Hi, The one in [big-business] has higher priority than the one in [small-business]Included context has lower priority. Hope this helps. Andy On 12/2/05, Chuck Bunn [EMAIL PROTECTED] wrote: Hi,If I have an extension in a context and I have another context with thesame extension and I include the second context in the first does this cause a conflict or does Asterisk know that there is a 600 extension ineach context[big-business]exten = 600,1,Dial(ZAP/1,20)include = small-business[small-business]exten = 600,1,Dial(ZAP/2,15) Thanks___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users]can I have T1 and E1 on the same TE406 card?
Hi, We have aDigiumTE406P connected to 1 T1/PRI now. Can I put in an E1 to one of the unused ports on the same card? Thanks. Andy ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users]can I have T1 and E1 on the same TE406 card?
Thank you Steve / Kevin. I'll look for the jumper on the card when I go to our co-lo. Andy On 11/25/05, Kevin P. Fleming [EMAIL PROTECTED] wrote: Andy Kuo wrote: We have a Digium TE406P connected to 1 T1/PRI now. Can I put in an E1 to one of the unused ports on the same card? Yes, just change the selection jumpers on the card appropriately.___--Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best Communications Line for VoIP
If you need to make calls in and out to the PSTN, you need a T1/PRI. Unless you send the calls to other VoIP provider. Andy On 11/22/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: We are putting in an Asterisk VoIP solution and was wondering what the bestcommunications medium would be for this implementation.We are going to need 20 telephone lines in/out of our business.We currently have a dataT1.Could we put another data T1 to use for Asterisk, or would it bebetter to put in a Voice T1 or a PRI line?Also, when we do put this T1 or PRI line in, what would be the best equipment to use with the Asterisk box?Any other recommendations would be appreciated?Thank you,Jyran GluckyAdvisory ProgrammerBlueWare, Inc.Strategic HealthWare Solutions3060 W. 13th Street Cadillac, MI 49601Phone:(231) 779-0224 ext. 111Fax: 231-779-1002mailto:[EMAIL PROTECTED]http://www.blueware.net DID YOU KNOW?BlueWare is the Grand Prize Winner of the 2005 IBM Beacon Award BEST DB2(Document Management) Application Worldwide.BlueWare Market Share for Hospital Document Management Systems is in 25states in the US. ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk to Fax Server
What Fax server are you using? On 11/21/05, Arcady Litmanovich [EMAIL PROTECTED] wrote: Hi I'm looking for following solution: Asterisk is connected to PSTN by Digium or some another card which has Fax Detection If incoming call is a fax I woud like to transfer it to External Fax server by SIP or H323 for getting a Fax. If incoming call is a voice to direct it to another trunk. Is it possible to make it on Asterisk? If yes which E1 card is preferable? Thanks in advance Arcady ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] GotoIf always goes to true?
Hi all,
I recently found GotoIf not working right in my extensions.conf, so I write a simple test and test it on my newly installed v1.2 box.
However, in all cases, GotoIf seems to always result in true. This happens to me in both ABE and V1.2
my extensions.conf :
[globals]Music=123
[default]
exten = ${Music},1,Answerexten = ${Music},2,SetVar(t=1)exten = ${Music},3,NoOp(${TIMESTAMP} - ${T})exten = ${Music},4,MP3Player(/var/lib/asterisk/mohmp3/deck.mp3)exten = ${Music},5,SetVar(t=$[${T} + 1])
exten = ${Music},6,GotoIf(${T}3?3:7)exten = ${Music},7,Hangup
CLI output :
*CLI -- Executing Answer(SIP/100-274e, ) in new stack -- Executing Set(SIP/100-274e, t=1) in new stack -- Executing NoOp(SIP/100-274e, 20051118-153136 - 1) in new stack
-- Executing MP3Player(SIP/100-274e, /var/lib/asterisk/mohmp3/deck.mp3)in new stackNov 18 15:31:39 NOTICE[5414]: app_mp3.c:108 timed_read: Poll timed out/errored out with 0 -- Executing Set(SIP/100-274e, t=2) in new stack
-- Executing GotoIf(SIP/100-274e, 23?3:7) in new stack -- Goto (default,123,3) -- Executing NoOp(SIP/100-274e, 20051118-153139 - 2) in new stack
-- Executing MP3Player(SIP/100-274e, /var/lib/asterisk/mohmp3/deck.mp3)in new stackNov 18 15:31:42 NOTICE[5414]: app_mp3.c:108 timed_read: Poll timed out/errored out with 0 -- Executing Set(SIP/100-274e, t=3) in new stack
-- Executing GotoIf(SIP/100-274e, 33?3:7) in new stack -- Goto (default,123,3) -- Executing NoOp(SIP/100-274e, 20051118-153142 - 3) in new stack
-- Executing MP3Player(SIP/100-274e, /var/lib/asterisk/mohmp3/deck.mp3)in new stackNov 18 15:31:45 NOTICE[5414]: app_mp3.c:108 timed_read: Poll timed out/errored out with 0 -- Executing Set(SIP/100-274e, t=4) in new stack
-- Executing GotoIf(SIP/100-274e, 43?3:7) in new stack -- Goto (default,123,3) -- Executing NoOp(SIP/100-274e, 20051118-153145 - 4) in new stack
-- Executing MP3Player(SIP/100-274e, /var/lib/asterisk/mohmp3/deck.mp3)in new stackNov 18 15:31:48 NOTICE[5414]: app_mp3.c:108 timed_read: Poll timed out/errored out with 0
I have been trying to figure this out for the past few days. I think it must be some stupid mistake of mine, but just can't figure out what/where.
Please help.
Thank you very much.
Andy
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Re: [Asterisk-Users] receive fax with asterisk
Can you tell me how you send/receive fax? ie. Fax conneted to ATA - Asterisk - Fax on PSTN? or Fax connected to Digium card - Asterisk -- Fax on PSTN? or ??? Thanks. AK On 11/16/05, Ben Higley [EMAIL PROTECTED] wrote: I have downloaded the iaxmodem package, and incorporated it with hylafaxwith great sucess... http://www.hylafax.org/ Harry --- Doug Lytle [EMAIL PROTECTED] a écrit : Jason Brashear wrote: Receiving faxes with Asterisk. Is there a good resource for learning how to set this up? www.soft-switch.org Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Outgoing sound very low
have you played around with rxgain and txgain in zapata.conf? AK On 11/16/05, Abdock [EMAIL PROTECTED] wrote: Hello,I have setup asterisk with Digium TDM04B, with FXO ports. I dial in using a local line and the asterisk connects me to the SIP phone network. I can hear them loudly, but when i talk they are not able to hear, the sound is very low. How can i adjust,i am using g729 codec and using IAX to connect to the asterisk exchange. ( which holds the SIP phone network ) Any help !Thanks,___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to check how many G729 codec licenseinstalled
I think you can specify the priority in sip.conf in the [general] section disallow=all allow=g729 allow=ulaw allow=alaw . . . During call setup, Asterisk will negotiate for g729 first, if it's not available on the other end, it'll try ulaw next, then alaw... I'm not sure what happens when your g729 license isused up. I'm assuming it should go on the next codec on the priority. Need to confirm on that. So far, we have morce g729 license than what we are using now. Cheers. AK On 11/15/05, Mark Quitoriano [EMAIL PROTECTED] wrote: hmmm... so there's no way i can prioritize g.729 before ulaw and alaw? i just want to do is use all g.729 first if all g.729 already used now thats the time ulaw will be used. On 11/16/05, trixter aka Bret McDanel [EMAIL PROTECTED] wrote: On Wed, 2005-11-16 at 01:20 +0800, Mark Quitoriano wrote: how can i force asterisk to use g.729 codec first. i try to call outside sometimes the g.729 is being used sometimes not. Asterisk will negotitate based on your list of available codecs for achannel and the remote end points available codecs.If you set it to only g.729 and the other end doesnt support g.729 the call will fail. Keep in mind that g.729 takes more cpu than some other codecs.Takeulaw, while it uses quantinization it doesnt really compress (itsarguable that quantinization is compression but that is a symanticargument more than anything).So its a trade off between cpu and network resources.If you truely want to force all calls to use g.729 or nothing only allow g.729 and disallow everything else.I do not think you will be happyunless you use only providers that support g.729 and have enoughlicenses available for every call that you make.If you directlyconnect to other servers that you dont know whether or not they do g.729or if your provider doesnt have enough g.729 licenses you may see both inbound and outbound calls fail from time to time.--Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402US +1 360 207 0479 or +1 516 687 5200FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group -BEGIN PGP SIGNATURE-Version: GnuPG v1.4.1 (GNU/Linux)iD8DBQBDeh2V+1olxlzQw5cRAtF3AKCykanu2BnxlIvD9YD6Se4s6yztfgCfQL7l vKqil6wmZbgG2d6v8bFZKrc==jqaB-END PGP SIGNATURE-___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Regards,Mark Quitoriano, CCNA http://www.atamanetworks.com Fan the flame...http://www.spreadfirefox.com/?q=user/registerr=19441 ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fail over?
in extensions.conf exten = _X.,1,Dial(SIP/[EMAIL PROTECTED]) exten = _X.,2,Dial(SIP/[EMAIL PROTECTED]) On 11/11/05, John E. Elkin [EMAIL PROTECTED] wrote: Maybe its already been posted, but i cant find it... I have an asterisk box running agilevoice (Customer signup and provisioning system) I have two sip termination providers. One provides did and termination. The other provides just my termination. My big question is. If the termination on provider A goes out.. i want my asterisk box to route calls to provider B how do i make this happen automaticly? John___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Fax support using T.38
Lilantha I've been looking for fax solutions with Asterisk too. Unfortunately, it seems like there's no T.38support for Asterisk so far. In fact, I think there's only fax-to-email solution for * now. I'm gettingsome SIP ATA's with T.38 support next week, but I am not sure if I can somehow get it towork with Asterisk. I wonder if there's anyone in the Asterisk community able to do fax between 2 SIP/IAX ATA's, or it's just simply not possible. AK On 11/9/05, Cory Andrews [EMAIL PROTECTED] wrote: Nice analogy, can I borrow that one?Cory J AndrewsPartner / Purchasing+++VOIPSupply.com - Everything you need for VOIP454 Sonwil DriveBuffalo, NY 14225+++tf voice - 800-398-VOIP X22l voice - 716.630.1555 X22f - 716.630.1548e - [EMAIL PROTECTED] AIM - b2CoryC F wrote: The same way the best roads can handle landing and takeoffs of 747s but weren't meant for it, a runway is what's needed, VoIP could have faxing with it, but TDM is really whats needed. Please search the archives for this question, it has been asked over and over and over, again and again. On 11/9/05, Lilantha Karunaratne [EMAIL PROTECTED] wrote: Hi, Just wondering whether anyone has done fax relaying or pass-through using Asterisk T.38 Please let me know your thoughts as I need to come up with a fax server using Asterisk with T.38 possible? Cheers! Lilantha ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users___ --Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sensing fax with txfax
Hi, I think you should change your codec on both legs to ulaw/alaw. I've been trying to get txfax going myself too. I can receive fax with Spandsp 0.0.2pre19 on Asterisk 1.0.9 with rxfax, but whenI tried sending the received fax fileto a fax machine, I either get line error or just a blank page. -- Attempting call on SIP/[EMAIL PROTECTED] for applicationtxfax(/root/testfax.tif) (Retry 1)Channel SIP/yyy-3c49 was answered.Lauching txfax(/root/testfax.tif) on SIP/yyy-3c49 I got the same CLI output and a blank page.(or line error on fax machine display) I'm trying to send fax to a regular fax machine on SIP ATA (Linksys PAP2-NA). Is anyone having any success on this? (or any other ATA's)? Can anyone help us out here? Thanks. AK On 11/8/05, Bartosz Piec [EMAIL PROTECTED] wrote: Matt Riddell napisał(a): You cant do fax in g723.So what to do? Change the fax machine? --Best regards,Bartosz Piec___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisks talking to asterisks
I do that through SIP. Assuming your TX extensions are 10XX, and NJ extensons are 20XX On your NJ box... sip.conf [gwtx] type=friend secret=x host=10.11.12.13(your TX IP) extensions.conf [toTX] exten = _10XX,1,Dial(SIP/[EMAIL PROTECTED]) On your TX box sip.conf [gwnj] type=friend secret=x host=20.21.22.23 (your IP for your NJ gateway) extensions.conf [toNJ] exten = _20XX,1,Dial(SIP/[EMAIL PROTECTED]) I think you should be able to do similar using IAX too. I don't know if this helps. I'm still quite new to Asterisk too. Good Luck. AK On 11/6/05, Jason Brashear [EMAIL PROTECTED] wrote: I have a request. I have a server in TexasAnd one in NJ.Is it possible for the system in Texas to log into the system in NJ so that Extensions can call each other?-J___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users]how to send fax using Spandsp
Hi, I've been trying to get fax going for the last few days. I can receive fax with Spandsp 0.0.2pre19 on Asterisk 1.0.9 with rxfax, but whenI tried sending the received fax fileto a fax machine, I either get line error or just a blank page. Is anyone using Spandsp to send fax to fax machines on PSTN? I've run out of things to try now, and I'd really appreciate if anyone can share some ideas/experiences here. Thank you. AK ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fax between Asterisk SIP clients
Hi all, I'm looking for a fax solution with Asterisk. I would like the users to be able to hook up regular fax machines to their SIP ATA's and send/receive fax from PSTN and/or other SIP clients. My goal is: fax machines - SIP ATA - Asterisk - T1(TE406E) - fax on PSTN It looks likeHylafax will allow me to receive fax from PSTN, but not send to PSTN. I also tried Spandsp, and it seems to receive fax ok from ATA's, but I can't figure out how to have it automatically forward the fax file to fax machines on PSTN or other SIP extensions. Can I have Spandsp dial and send the fax to the destination automatically? Are there other software / hardware solutions that can help me achieve my goal? Please advise. Thanks to any help/ideas. AK ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OS for ABE
Weuse Fedora 3 and ABE-A.1 The pair has been workinggreat for usso far. AK On 11/2/05, Eric Alexander [EMAIL PROTECTED] wrote: We are setting up ABE for a client of ours. This is not our first Asterisk install, far from it, but it is our first time using ABE. Here is the problem, ABE only supports Fedora 3 and Red Hat EL3, we typically use CentOS. Our problem with this scenario is that RHEL3 is an old release, we would rather use 4 if we have to, and we have not had good experiences with Fedora. We tried to use Fedora but we are running into some problems with our SCSI card. What distro do most people use with ABE? What happens if we use CentOS? Will it render our ABE purchase useless? -Eric AlexanderSenior IT Systems ConsultantThe Uptime Group, Inc.(303) 757-4611, Ext. 402- ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] spandsp patch
Hi all, I'm trying to install spandsp. I followed the instructions on http://www.soft-switch.org/installing-spandsp.html, and when I applied the patch, I got the following errors: [EMAIL PROTECTED] apps]# patch apps_makefile.patchpatching file MakefileHunk #1 FAILED at 55.Hunk #2 FAILED at 93.2 out of 2 hunks FAILED -- saving rejects to file Makefile.rej I tried spandsp-0.0.2pre19 on Asterisk 1.1x and also on Asterisk 1.0.9. I also tried spandsp-0.0.2pre21a on Asterisk 1.1x, but nothing worked for me. I'm not a C programmer, and I really don't know what to modify in the patch. I searched around, but didn't really find anything on it. Can anyone please help? Thanks. AK ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] problem with receiving faxes over cisco as5300
Hi, Sorry this does not answer your question. As I am trying to implement fax on Asterisk, can you please tell me if you are using spandsp? Are you sending fax from SIP ATA's? Thank you. AK On 10/27/05, Florian Meister [EMAIL PROTECTED] wrote: Hi,does anybody have a working sample configuration of a cisco as53xx forreceiving faxes ? Sending faxes over the as5300 works fine, but if I send a fax from pstn toasterisk (over the as5300 as pstn/voip gateway) it does not work.Thx, florian-- (__) Florian Meister (oo)/---\/ [EMAIL PROTECTED]/ | ||+43 5572 501 134 *||w---|| The only problem with troubleshooting is that sometimes trouble shoots back.-- ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cdr_odbc with tds
What database server are you using? If you are using MSSQL, just use freetds without unixODBC. AK On 10/20/05, Ben merrills [EMAIL PROTECTED] wrote: Does anyone know why, using latest cvs head, freetds 0.62.1-0 andunixODBC, when running cdr_odbc, it says it's logged the call successfully, however, when checking the table, nothing is there!I checked through the bug tracker; and a problem very much like mine wasin there, with status resolved as of last year (1339).Can anyone shed some light on this please? Cheers,Ben___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip accounts
Hi, Try add [1234] ... host=dynamic or host=xxx.xxx.xxx.xxx (the client's IP) ... ... AK On 10/14/05, Kong [EMAIL PROTECTED] wrote: hi, i facing a problem here. in my sip.conf, i specify a account like this,[1234]type=friendcontext=from-sip username=1234secret=1234nat=nocanreinvite=yesdtmfmode=info[EMAIL PROTECTED]disallow=allallow=ulawso i am able to login with username 1234 and password 1234but ther weird part is, i can also register as any number (account) without having to specify in sip.conf. thus anybody can just use mysystem to call others.. lets say i do have set that some certainaccount can make some certain calls only.how can i solve this problem? thank you.___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk log
Hi Andrea, How do you start a weekly-rotation log? Do we need to do it manually through CLI? or can we set it somewhere? Thanks. AK On 10/12/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Thank you very muchI decided not lo lower the log information (leaving all: full =notice,warning,error,debug,verbose) I started a weekly-rotation of the full log.Andreagincantalupo[EMAIL PROTECTED]software.comTo Sent by:Asterisk Users Mailing List -asterisk-users-bo Non-Commercial Discussion[EMAIL PROTECTED] asterisk-users@lists.digium.comm.comccSubject 12/10/2005 12.01Re: [Asterisk-Users] asterisk logPlease respond to Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED]ists.digium.comHi,[EMAIL PROTECTED] wrote: Is there a way to1) disable asterisk from writing in the fulllog? (/var/log/asterisk/full )Take a look at /etc/asterisk/logger.confor2) implement a log rotation or similar of the full log ? I see the full log grows a lot (about 100 MB per Month)Use logrotate.thanks in advance,AndreaChi ricevesse questa mail per errore e' gentilmente pregato di cancellarla.Visitate il sito http://www.frameweb.it___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] call waiting not working on PAP2
Hi, I have callwaiting=yes in my zapata.conf, and Call Waiting Serv: Yes in the PAP2s. However,there's sitllno callwaitingon the PAP2s. Everything else work fine. Any ideas? Am I missing something somewhere? Thank you. AK ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call waiting not working on PAP2
Hi, I can't seem to find CW Act Code: and CW Per Call Act Code: in PAP2. Does anyone know what they are in PAP2? thanks. AK On 10/13/05, Tom Vile [EMAIL PROTECTED] wrote: Have a look at the CW Act Code: CW Per Call Act Code: and remove the entries in there. I have a sipura so I dont know if they are using the same terminology buts it the same hardware. On 10/13/05, Andy Kuo [EMAIL PROTECTED] wrote: Hi, I have callwaiting=yes in my zapata.conf, and Call Waiting Serv: Yes in the PAP2s. However,there's sitllno callwaitingon the PAP2s. Everything else work fine. Any ideas? Am I missing something somewhere? Thank you. AK___--Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Tom VileBaldwin Technology Solutions, IncConsulting - Web Design - VoIP Telephony www.baldwintechsolutions.comPhone: 518-631-2855 x205Fax: 518-631-2856 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Modifying cmd VoicemailMain
Hi, Maybe you can record the sound file vm-five.gsm as five hour in Japanese, instead of just five. AK On 10/12/05, Kuniyoshi Murata [EMAIL PROTECTED] wrote: Dear Asterisk Users,I'm a Japanese and now configuring Voicemail.Now I need to modify the way cmd VoicemailMain works to fix language difference and other my conveniences.What I want to do are...1) Add words used in message retrieving guidance.I need to add different suffixes to numeric words due to Japanese way ofmentioning time. ( e.g. in English, you can say Five forty-five for 5:45,but in Japanese, we have to put hour and minute for respective time unit(meaning, VoicemailMain should pronounce as Five hours and forty-five minutes in Japanese). So, is there any way to add words modifying theregular word order?2) Disable most of the key function guidance for retrieving the message.I don't want too much function guidance of VoicemailMain saying such as 3 for advanced options and the like. I just want to hear just a few importantkeys to press. So, is there any way I can separately disable guidance foreach key functionsAny input is welcome. TIAKuni--Kuniyoshi Murata.iChat/AIM:macwebcasterEnglish-Japanese Interpreter mailto:[EMAIL PROTECTED]Macintosh Webcast Specialist http://www.macwebcaster.com___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Noise using TE410P Rhino Channel Bank
I'm not sure where the noise is coming from, but you can change the timing source in zaptel.conf in zaptel.conf: span=1,0,0,esf,b8zs --- Asterisk is using external timing source span=1,1,0,esf,b8zs -Asterisk is providing timing to the channel bank AK On 10/7/05, Eddie [EMAIL PROTECTED] wrote: Anyone ever experience noise instead of a dial tone. We are using a Digium TE410P and 2 units of Rhino FXO FXS Channel Bank. Everything works well when they are freshly turned on or rebooted, but after a while there's only noise can be heard instead of dial tone.Does this have anything to do with the timing configuration?Where is this time is generated from, is it from the TE410P card? How can get rid of this noise? Please advice.___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Codec issue? Dropping incompatible voice frame ...
Hi, When I call forward on PAP2, the incoming call will right the forwarded number. However, there is one-way voice problem. The caller can hear the destination(the forwarded number), but after the called party answers, the caller can't hear anything. Then the CLI produce continuous errors as following: Oct 6 10:57:45 NOTICE[11026]: channel.c:1409 ast_read: Dropping incompatible voice frame on Local/[EMAIL PROTECTED],2 of format gsm since our native format h as changed to g729Oct 6 10:57:45 NOTICE[11032]: channel.c:1409 ast_read: Dropping incompatible voice frame on Local/[EMAIL PROTECTED],2 of format ulaw since our native form at has changed to slin I searched the list and found similar topic on http://lists.digium.com/pipermail/asterisk-users/2005-May/104942.html, and used their advice by adding Answer before Dial in extensions.conf, and canreinvite=no in sip.conf. It worked in the way that I was able to get 2-way communication, but pages and pages of the above messages are still there. Anyone has similar experience? Please advice. Thanks. AK ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] how do I know what codec is being used
Hi, This may be a stupid/easy question for many of you. Q. how do I know what codec is being used for a particular call or call leg? Thanks. AK ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] success story: TE406P (quadspan with hardware echocan)
Hi Andrew, I'm using a TE406P too, and I have echocancel=yes in zapata.conf. Is this redundant? Should I take the line out? Please advice. Thanks. AK On 10/3/05, Rod Bacon [EMAIL PROTECTED] wrote: Which version of asterisk and zaptel are you using?Will they work with 1.0.9 ?== Rod BaconEmpowered CommunicationsGround Floor, 102 York St. South MelbourneVictoria, Australia. 3205Phone: +613 99401600Fax: +613 99401650FWD: 512237 ICQ: 5662270== Andrew Kohlsmith wrote: I just wanted to post here and let everyone know that the TE406P (quadspan T1/E1 with hardware echo can) kicks some serious ass. We've been running a PRI now for over a year with Asterisk (every single call in and out is through two Asterisk boxes, including faxes) and while the software based echo cancellation is more than adequate, we'd get the occassional edgy echo and very infrequently get full-out holy shit echo. So far the TE406 has eliminated that entirely. Anyway as I said I just wanted to post here and tell the world that at least as far as I have been able to determine, the extra cost of the hardware echo can is *well* worth the money. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FreeTDS 0.63
Hi, People on the list just told me that we can only use 0.62.x AK On 10/3/05, Richard Cook [EMAIL PROTECTED] wrote: Hello, Is anyone using FreeTDS version 0.63 with *? --Richard Cook[EMAIL PROTECTED]T: 705-223-2000 x2010 ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CDR and TDS
Hi, I'm trying to install FreeTDS. I followed the instructions on http://www.voip-info.org/tiki-index.php?page=FreeTDS , but still can't get it to work. I serched around trying to find instructions on it, and it seems the same info (even wording) appear on all sites I found. I downloaded freetds-0.63, and followed the instructions step by step, and when I try to re-compile Asterisk (#make in /usr/src/asterisk), I got these errors: cdr_tds.c: In function `mssql_connect':cdr_tds.c:429: error: `TDSCONNECTINFO' undeclared (first use in this function)cdr_tds.c:429: error: (Each undeclared identifier is reported only oncecdr_tds.c:429: error: for each function it appears in.) cdr_tds.c:429: error: `connection' undeclared (first use in this function)cdr_tds.c:474: warning: implicit declaration of function `tds_free_connect'make[1]: *** [cdr_tds.o] Error 1make[1]: Leaving directory `/usr/src/asterisk/cdr' make: *** [subdirs] Error 1 I think I might need to put in info such as the IP, DB name, user, password, etc. of my SQL server, but I'm not sure what the correct format is. Can anyone please give me an example? Are there anything else I do wrong here? or Are there different versions of instructions on the topic? Thank you all in advance. AK ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] please help on FreeTDS (writing CDR to MS-SQL or MySQL)
Hi, I'm trying to install FreeTDS. I followed the instructions on http://www.voip-info.org/tiki-index.php?page=FreeTDS, but still can't get it to work. I serched around trying to find instructions on it, and it seems the same info (even wording) appear on all sites I found. I downloaded freetds-0.63, and followed the instructions step by step, and when I try to re-compile Asterisk (#make in /usr/src/asterisk), I got these errors: cdr_tds.c: In function `mssql_connect':cdr_tds.c:429: error: `TDSCONNECTINFO' undeclared (first use in this function)cdr_tds.c:429: error: (Each undeclared identifier is reported only oncecdr_tds.c:429: error: for each function it appears in.) cdr_tds.c:429: error: `connection' undeclared (first use in this function)cdr_tds.c:474: warning: implicit declaration of function `tds_free_connect'make[1]: *** [cdr_tds.o] Error 1make[1]: Leaving directory `/usr/src/asterisk/cdr' make: *** [subdirs] Error 1 I think I might need to put in info such as the IP, DB name, user, password, etc. of my SQL server, but I'm not sure what the correct format is. Can anyone please give me an example? Are there anything else I do wrong here? or Are there different versions of instructions on the topic? Thank you all in advance. AK ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Delay in dial
Hi, Try taking out Answer in your extensions.conf. You don't need to answer before dialing a SIP channel. cheers. AK On 9/28/05, yusuf [EMAIL PROTECTED] wrote: Hi all,I am using Asterisk CVS, and I am getting a huge delay in dialing SIP.This Asterisk box is taking calls from a PABX over ZAP, then dialing SIP users.So, a user '0251' dials from his phone, the PABX sends it the myAsterisk box, no delay, then I get a 15 sec delay, before it actuallydials the end SIP user.1 -- Accepting call from '0251' to '0834541083' on channel 0/1, span 2-- Executing NoOp(Zap/1-1, 0834541083) in new stack3-- Executing Answer(Zap/1-1, ) in new stack4-- Executing Dial(Zap/1-1,SIP/[EMAIL PROTECTED] :5060)in 5 new stack6-- Called [EMAIL PROTECTED]:50607-- SIP/192.168.11.111:5060-1699 is ringing8-- SIP/192.168.11.111:5060-1699 is ringing9-- SIP/192.168.11.111:5060-1699 is ringing When the call is placed, Asterisk executes until Line 5.There is a 10-15 sec delay between Line 5 and Line 6. Once Line 6 has executed, onlythen does it dialCan anyone helpyusuf___ --Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T.38 Faxing
Hi, It sounds like Asterisk should handle T.38 fax (at least at very light load). However, I am having troubles faxing through a Linksys PAP2 NA. It doesn't seem to get through at all. Everything else (voice calls) works just fine. I am using Asterisk Business Edition ( ABE-A.1). I have faxdetect=both in zapata.conf, and I wonder if there's anything else I'm missing. I know I won't get 100% with fax over IP, but the success rate for fax on my Asterisk box is just way too low (about 5%, I would say). Can you please tell me what ATA you are using? Could it be that the PAP2 doesn't support fax? Please advice. Thank you. AK On 9/28/05, Roger Schreiter [EMAIL PROTECTED] wrote: Carlos Alperin schrieb: ... Bank, and send all the faxes through that system, and forget to try anything through Asterisk. We're already tired of the complainings from customers that never can send faxes, or sometimes some pages.Wow, strong words!Since our experiences with asterisk are mainly positive, we won't forget faxing through asterisk.As long as asterisk is just used as pbx or switch, it isobviously capable of transporting 100 and more faxessimultaniously without problems.The only thing missing seems to be a reliable modulator/ demodulator and T.30 machine.rxfax/txfax+spandsp is still far away from being reliable underhigh load, but developing. Our experience is, that it doesa very good job when sending or receiving single faxes. As soon as three or more faxes are processed at once,problems do occour.We are currently testing a commercial fax library with the goal,to implement an IAX client as fax server, in order the send or receive some dozen faxes simultaniously. It is too early tosay, if we'll succeed, but we hope so, and thus won't forgetfaxing through asterisk.This is _no_ announcement of a product, since we currently won't have the right to distribute the library, but we think,it would be very interesting to know, whether it works or not.Those, looking for a GNU licenced tool will have to wait forprogresses of spandsp anyway. By the way, the producer of the library announces T.38 support insome weeks. asterisk just started to support T.38 pass through,and probably will do it reliably in some weeks. Maybe this willenable T.38 support for asterisk - though unfortunately not for free noropen source.Roger.___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura 2000 Dial Plan
5551212 1212121212 will be matched by [2-9]xxS0, and that allows only 7 digits. That's why It does not dial for a while and then it dials 555 1212 try appending a . at the end of the dial pattern, it means repeat. ie, x.allowsrepeats of x AK On 9/27/05, Michael Blood [EMAIL PROTECTED] wrote: Anybody ever run into a case where the Sipura Dial Plan will not work with the S0 option to immediately connect? My Dial plan reads (*xx|[3469]11S0|0|00|[2-9]xxS0|1xxx[2-9]xxS0) and I can dial ONLY then numbers in the dial plan so I know that it works. For some reason when I dial 5551212 1212121212 It does not dial for a while and then it dials 555 1212 Anyone have any ideas? Thanks Michael Blood ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ZAP ISDN losing digits
I had similar problem using Digium TE406 card. Try update the driver. I worked for me. Good luck. AK On 9/23/05, maka [EMAIL PROTECTED] wrote: Hi all, I got into a strange problem here. I've got an asterisk box with bristuff-0.2.0-RC7k, and a HFC PCI ISDN card, running in NT mode. The ISDN card is connected to a S0 bus and to a Siemens ISDN PBX. Two phones are connected to the ISDN PBX and are successfully getting calls from the asterisk box. When dialling from one of the phones, the ZAP channel seems to be missing out on some of the dialled digits everytime, i.e. if I dial 099557896, the asterisk box receives 09955896 sometimes, or 0995789, or something like that. This only happens on one of the phones, the other one is dialling fine and digits are being recognized well. I already tried setting relaxdtmf=yes in zapata.conf, but to no effect. If anyone has any idea what might be wrong, appreciate the feedback..Cheers ___ --Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] problems with sending fax from SIP channels
Hi All, I'm having problem sending fax fromSIP extensions (Linksys PAP2) through Asterisk Zap channels (ISDN PRI). The SIP extensions can receive fax without problems, but sending fax fails most of the time. Does anyone have this problem? Please advice. Thank you. AK ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
