[asterisk-users] SILK codec
hi folks. i've been experimenting with SILK codec and meet with some success on incorporating it in pjsip (an open source sip client). now i'm trying to do the same thing on Asterisk. any documentations, pointers, etc i should look into? any help is appreciated. -- Edwin Lam edwin@officegeneral.com Systems Engineer, OfficeWyze, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 -- _ -- 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] Forward voicemail not working
I still can't figure out why this isn't working. I updated to the latest version of Asterisk 1.8.1 with no luck. I am using Realtime for sipusers and vmusers if that makes any difference. I tested this on a new install and saw the following under the folder where I installed Asterisk I had /home/asterisk/asterisk-bin/spool/asterisk/voicemail so no directories had been created yet. I left a voicemail for 9xx2xx2...@irock.com. That created the directory irock.com under the voicemail folder and also created the directories for user 9XX2XX2009 (INBOX and all the other stuff). Then I called into 9XX2XX2009's mailbox and forwarded the voicemail to 9XX2XX2008. So under the voicemail directory there was no folder for 9XX2XX2008, but since I forwarded the voicemail to 9XX2XX2008 asterisk created the folder for that user and also created the subfolder INBOX, but it didn't copy the voicemail to that directory. If I can't get the forward voicemail option to work with the VoicemailMain() function is there any way to disable this option so that users don't try to use it? On Sun, Jan 2, 2011 at 11:44 AM, duane.lar...@gmail.com wrote: I have asterisk 1.8.0 installed and I am not able to forward a voicemail from one users mailbox to another user. I have the user log into their mailbox press 8 to forward a message enter the extension of the user I wish to forward too I don't prepend a audio message and press # to send the message to the other user from a debug perspective I don't see any errors. The only message I see is == Saving '/home/asterisk/asterisk-bin/spool/asterisk/voicemail/ irock.com/9XX2XX2009/Old/msg.txt': [Jan 2 11:24:18] NOTICE[17036]: app_voicemail.c:5154 copy_message: Copying message from 9xx2xx2...@irock.com to 2...@irock.com Yet when I look in 2011 spool directory I don't see any message at all. It is just not being copied. What could be the issue? -- -- *--*--*--*--*--* Duane *--*--*--*--*--* -- -- _ -- 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
[asterisk-users] system lockup when going into conference
Hi. I have an asterisk system under Debian Leni using asterisk 1.8 with no Digium hardware -- and when I go into a meetme conference the system either locks up or is 100% cpu utilized or something -- I can't type anything and I have to physically reboot the system. The dahdi module is loaded and the last log entry is the playing of you are the only person in this conference,. How would I even start to debug this one? Any ideas would be appreciated. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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] Benefit of PRI vs SIP trunk calls
On Thu, Jan 6, 2011 at 7:06 AM, Jim Dickenson dicken...@cfmc.com wrote: Are there reasons to prefer the use of PRI over SIP or SIP over PRI? Assuming you are talking to connect a PBX to the PSTN... PRI advantages: 1. Relatively little equipment between the PTSN and the PBX. Less to break or go wrong. 2. Simple to set up. No need for QoS, routing, authentication, etc. Of course if you only know IP, SIP is easier, but if you learn both, ISDN is easier. 3. If compared to SIP over internet, PRI has guaranteed quality. Granted, SIP *can* have just as good (and better) quality, just not guaranteed if done over the internet (it can be guaranteed over a private circuit). 4. Less latency/delay so there is less talk-over. 5. FAX, high speed modem, TTY, etc, pass-through actually works. (it *can* work over SIP, but Asterisk just isn't quite there yet) I run the PBX for my organization which has about 160 extensions. I wouldn't even think of doing anything but PRI for the main lines because (A) for our size organization where we are located, we're talking a couple hundred dollars a month difference between PRI and SIP in cost so it's nearly break-even in cost which means cost difference isn't a huge motivator, (B) it supports FAX, modems, and TTYs - perfectly, (C) Quality is 100% consistent. In addition, the reliability is good enough that I'm willing to use it for 911. Of course if this installation wasn't in downtown Denver, where ISDN PRI is very cheap (a full CLEC 23-channel ISDN PRI costs roughly what 6 or 7 ILEC POTS lines cost), then SIP would be interested. SIP advantages: 1. Cheap (at least SIP-over-internet) 2. Easy and quick to scale if you have bandwidth. 3. Great for disaster recovery if using SIP over internet 4. Very cheap to get local numbers from all around the world. 5. If using SIP over internet, easy to compare providers -- _ -- 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
[asterisk-users] Anyone have Festival application working?
Particularly interested if anyone has it working on Fedora 13+, with the Fedora RPMS. I've tried both F13 (asterisk-1.6.2.12-0.1.rc1.fc13.i686) and Rawhide (1.8 something), and both of them appear to be broken in the same way. Festival reports a connection, and a file is placed in the cache directory, but no audio is actually played. Additionally, the the call seems to get borked somehow; any applications that should be called after Festival never get called. I've burned too much time on this already, so I'm basically giving up. Would certainly be interested to know if anyone does have this working on a recent Fedora release (using the Festival application, not AGI). Thanks! -- Ian Pilcher arequip...@gmail.com -- _ -- 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] Too Few Fax Detections
On Jan 6, 2011, at 8:56 AM, Kevin P. Fleming wrote: On 01/05/2011 08:12 PM, Thomas Rymes wrote: OK, after my last message about fax detection, I feel a bit better informed and able to press forward. I started looking into this because I was getting lots of false positive fax detection errors in the logs with faxdetect=both set in chan_dahdi.conf. Anyhow, I do not currently use fax detection, and we have a dedicated Fax DID on our PRI, so setting faxdetect=no works fine. Having said that, I would like to sort it out as I may want to use fax detection in the future. Unfortunately, I seem to be having odd results. I set faxdetect=incoming last night and restarted dahdi and asterisk. Since that time, we have received 17 faxes, but I only have three fax detections in my asterisk log, so far as I can tell: # grep -i fax /var/log/asterisk/full [Jan 5 05:53:39] NOTICE[6686] chan_dahdi.c: Fax detected, but no fax extension [Jan 5 10:24:27] NOTICE[11834] chan_dahdi.c: Fax detected, but no fax extension [Jan 5 11:48:52] NOTICE[13804] chan_dahdi.c: Fax detected, but no fax extension All three calls listed are indeed fax calls, and since there is no fax extension in that context, the call just proceeds along as if nothing happened (which is appropriate). My question is this: If I have received 17 faxes since enabling fax detection, shouldn't I see ~17 entries in the log? How are you delivering the inbound FAX calls to your FAX machine? If you are sending them back out a DAHDI channel (to an FXS port on an analog card, for example), then as soon as the two channels are bridged the audio never comes up to Asterisk (under normal circumstances), it stays in DAHDI, so the Asterisk DSP can't detect the CNG tone. If the FAX machine answers the incoming call fairly quickly, there may not be any opportunity for the CNG to be detected. In addition, you may not be even receiving any audio from the calling FAX machine until you answer the incoming channel (depending on your PRI provider). If you want to have the best chance to detect each incoming FAX using the Asterisk DSP, you'll have to answer the incoming channel as soon as it hits the dialplan, then wait 3 or 4 seconds, then send the call onwards to your actual FAX machine. FAX detection is really expected to be used on calls that would otherwise be answered by a non-FAX endpoint (IVR, voicemail, user with a phone, etc.) That does make sense: the incoming calls are directed to a FreePBX ring group consisting of three IAXModems handled by HylaFAX, and I am fairly certain that they answer the call nearly instantaneously. I do realize that the detection is really intended for non-dedicated lines; I'm just trying to ensure it works before I start using it. Thanks for the response. Tom -- _ -- 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 PoE phones for restaurant (kitchen)
On Jan 6, 2011, at 10:05 AM, Andy Graybeal wrote: On 01/05/2011 01:51 PM, Tom Rymes wrote: On 01/05/2011 7:50 AM, Andy Graybeal wrote: We've got two noisy kitchens that need to talk back and forth. Andy, Why, exactly, are you trying to combine an inter-kitchen intercom and your phone system? Might it make more sense to have a non-phone-based intercom system, plus a phone for making phone calls? Tom Tom, Good question. I'm not sure, but maybe I was hoping to kill two birds with one stone. I will take your suggestion into account as I'm not sure what to do. Do you have any intercom system recommendations? Would it be POE also, and something I could manage with Asterisk? -Andy Unfortunately, I have no recommendations, but I was just thinking of a simple, dumb intercom for between the kitchens, plus a phone for when you need to make a call. Any old phone will do. Perhaps something simple and durable like this: http://cgi.ebay.com/2554-Single-Line-Wall-Phone-w-Amp-Handset-Cortelco-AT-T-/200552549353?pt=LH_DefaultDomain_0hash=item2eb1dd0be9#ht_3437wt_1141 Tom -- _ -- 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] Asterisk Outlook integration
Thanks but I doubt it does pop-up of outlook contacts. It probably only does outbound calling. My main need is to have an outlook contact pop-up when a call comes in. I also favor open source if possible. Thanks On Wed, Jan 5, 2011 at 4:02 AM, Giorgio Incantalupo gincantal...@fgasoftware.com wrote: Hi BB, you could try this: http://asterisk-outlook-dialer.voip-singapore.qarchive.org/ Never tested it deeply but apparently seems to work fine. Giorgio Incantalupo Bruce B wrote: Hi Guys, What is out there other than OutCall that works with M$ Outlook and Asterisk 1.4/1.6 ? I prefer opensource and free (as in free in price) but can consider low price - working - programs as well. OutCall is giving issues with various versions of Outlook and it always brings up NEW CONTACT even if contact exists. Thanks, -- _ -- 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 -- _ -- 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 -- _ -- 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] Force different codecs on call base
On Thu, Dec 30, 2010 at 10:10:18AM +0100, Stefan Schmidt wrote: my idea was if i can find a way that the first call of a peer has g711a codec (like normally) and if a second call comes in, or has to be placed for this peer i only offer g726 (40kbit) so i dont have a bandwith issue. https://issues.asterisk.org/view.php?id=13243 explains settings codecs (and its difficulties, but since you have full control there shouldn't be a problem). Use GROUP and GROUP_COUNT to find out how many channels are are active, use this to decide whether to use alaw or g726. -- Daniel Tryba -- _ -- 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
[asterisk-users] Call queues on load-balanced asterisks
Hello, I have been asked to implement the following design: Load-balanced Kamailio servers handling registrations and routing. Load-balanced asterisk feature servers handling voicemail and other things Kamailio cannot do. Plus several load-balanced gateways, but they are not relevant to my question. All this is working fine. I've now been asked to start implementing calling queues, and my question is this: How can I implement the same queue on multiple Asterisk servers? Let's say that 10 people call the same queue. These calls would then currently be distributed 5 to Asterisk A and 5 to Asterisk B. How can I make Asterisk A respect the 5 people queued on the other server and vice versa? Will the customer need to change their design to make the feature servers master-slave with failover instead of load-balanced? Mvh Pan-- _ -- 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
[asterisk-users] Channel name changed in asterisk 1.8
Hi, The channel name for DAHDI channels has changed in 1.8 with no information that I can find in the ChangeLog. The old format was DAHDI/XX-Y where XX was the real channel number. It has changed to DAHDI/iZ/XX-YYY where XX is the callerid. And Z is the number of the span in /etc/dahdi/system.conf Our channel names look like this DAHDI/i8/0517383600-229 DAHDI/i1/0031650545840-329 DAHDI/i4/0512515245-20f DAHDI/i6/0517417488-1fb But we want to know which channel number of these four channels is used. P.S. This is our system.conf: span=1,1,0,ccs,hdb3,yellow bchan=1-15,17-31 dchan=16 span=2,1,0,ccs,hdb3,yellow bchan=32-46,48-62 dchan=47 span=3,1,0,ccs,hdb3,yellow bchan=63-77,79-93 dchan=78 span=4,1,0,ccs,hdb3,yellow bchan=94-108,110-124 dchan=109 span=5,1,0,ccs,hdb3,yellow bchan=125-139,141-155 dchan=140 span=6,1,0,ccs,hdb3,yellow bchan=156-170,172-186 dchan=171 span=7,1,0,ccs,hdb3,yellow bchan=187-201,203-217 dchan=202 span=8,1,0,ccs,hdb3,yellow bchan=218-232,234-248 dchan=233 We use two seperate cards. (TE4/1/3 T4XXP (PCI)) Arjan Kroon Mobillion BV -- _ -- 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
