Re: [Asterisk-Users] Asterisk on amd SERVER
Hello, No problems:) Cheers, Madhawa Kanishka Somaratne wrote: Hi I am going to install asterisk on an AMD server, did any one had problems installing it on an AMD server ? Regards Kani ___ --Bandwidth 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
SV: [Asterisk-Users] Polycom 501 - Disable DND feature?
Solved! In sip.cfg: Thanks to Derek for this solution! Regards, Jan Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För [EMAIL PROTECTED] Skickat: den 4 maj 2006 15:46 Till: asterisk-users@lists.digium.com Ämne: SV: [Asterisk-Users] Polycom 501 - Disable DND feature? Well, yes and no. I tested that before and it causes a silent ring instead of a call rejection. I actually want to disable the entire feature. So the phone always rings unless you're actually on the phone. Thanks for the reply though! Regards, Jan Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För Jerry Jones Skickat: den 4 maj 2006 15:00 Till: Asterisk Users Mailing List - Non-Commercial Discussion Ämne: Re: [Asterisk-Users] Polycom 501 - Disable DND feature? Attribute Values Default Interpretation call.rejectBusyOnDnd 0, 1 1 If set to 1, reject all incoming calls with the reason "busy" if do-not-disturb is enabled. Have not used, but looks like it may ignore the key if this is 0 Let us know... On May 4, 2006, at 2:22 AM, <[EMAIL PROTECTED]> <[EMAIL PROTECTED]> wrote: Hi, Is it possible to disable the DND feature on a Polycom 501? Regards, Jan ___ --Bandwidth 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] PCI voltage
Giordano Grandis wrote: Hi all, I have to bought a PCI with 4 PRI but on digium site I saw that there a re two different kind (3,3V and 5v). What’s the difference? 33MHz 32 bit PCI slots are 5V. PCI-X slots MAY support 5V and 3.3V depending on the age of the board. My understanding is that current PCI-X boards are 3.3V only now. The board you mention would appear to have PCI 5V slots. Regards, Richard ___ --Bandwidth 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: AW: [Asterisk-Users] DTMF detection when outgoing call to mobilephones
Marc Scheuffler wrote: > Yes I can hear the DTMF keys. I ve tried 2 different phones and 3 different > mobile network providers. Nothing. There was a bug in various versions of Asterisk when outbound calls were placed using spool files and then could not detect DTMF from the called party. Without more details, including the version of Asterisk you are running, it will be difficult to suggest anything to you other than upgrading to the latest release. ___ --Bandwidth 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 Phones behind dynamic IPs
Chris Bagnall wrote: > Okay, so assuming I've got to drop the re-registration to a much shorter > time than the default of every hour, what are the implications of doing so > (in terms of network traffic, load on the asterisk box, etc.)? What's the > lowest one can reasonably take it? 10 minutes? 1 minute? Network traffic is negligible; a few packets each way (REGISTER, 407, ACK, REGISTER, 200, ACK). There will be some additional load on the Asterisk server as it has to update the database each time a phone registers. The systems I have built all use a 5-minute registration timer and they seem to work just fine. ___ --Bandwidth 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] ISAC support?
Trond G. Andersen wrote: > Has there been done any work to support ISAC ? ISAC is a proprietary codec from Global IP Sound. There will not be any support for it in Asterisk unless GIPS wants to either open-source the codec (not likely) or allow Digium to license it in the same method as the G.729 codec is licensed. In addition, Skype uses ISAC in 16KHz mode, which Asterisk cannot support yet. ___ --Bandwidth 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: AW: [Asterisk-Users] DTMF detection when outgoing call tomobilephones
> There was a bug in various versions of Asterisk when outbound calls were > placed using spool files and then could not detect DTMF from the called > party. Without more details, including the version of Asterisk you are > running, it will be difficult to suggest anything to you other than > upgrading to the latest release. Works perfectly on mine. Using Digium Digital boards (ZAP -> UK Mobile). Asterisk version 1.2.1 with a few patches. Mark ___ --Bandwidth 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] DTMF Tones within my Asterisk on all type of Channels
Hi, i am using Asterisk 1.2.6 with Debian Kernel 2.6.8-2-386 and up 2 date Zaptel, Libpri and so on. My Hardware : 4 x SNOM 320 One Wildcard TDM40B 3 x Allnet 7950 1 x Fritz!Box 7050 - working as a ATA I use Asterisk in my Business stable since last year, the problem occurs since 2 or 3 months. I have detected some DTMF Tones within my conversations and turned on a DTMF Logfile to get more infos about this. The LOG reports Tones - they looks like appearing randomly on any type of channel (mISDN, ZAP, SIP) and i cant find any reason why. Here is a part of the LOG : Apr 13 10:18:59 DTMF[5007] channel.c: SIP/30-1352 : * Apr 13 10:39:52 DTMF[5165] channel.c: SIP/20-5073 : # [...] Apr 15 07:50:53 DTMF[14182] channel.c: Zap/1-1 : A Apr 15 07:51:44 DTMF[14182] channel.c: Zap/1-1 : A Apr 15 13:04:36 DTMF[14909] channel.c: SIP/50-7aa2 : 4 Apr 16 20:20:11 DTMF[19677] channel.c: mISDN/1-u61 : 9 [...] Apr 18 07:33:32 DTMF[26583] channel.c: Zap/1-1 : 5 Apr 18 07:38:50 DTMF[26663] channel.c: SIP/50-21c4 : 9 [...] As you can see, this happens on any type of available channel, the user can hear the tone and in the baddest case the "*" appears and the function to transfer in the DIAL is active (tT).. Can you give me a tip to get this problem solved or locate it ? Best Greets from Germany, Stefan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help with IRQ conflict between wct2xxp and eth0
Hi Phil, may sound a stupid advice buthave you tried to change PCI slots on your server? Giorgio Phil Menico wrote: I have a conflict problem with the eth0 card and wct2xxp digium board. The PRI can receive calls but my network connection is gone. When I "cat /proc/interrupts" I get the following: 1 .. 1 .. .. .. .. 169 0 IO-APIC-level wct2xxp, eth0 .. etc. even before I "modprobe wct2xxp" After I "modprobe wct2xxp" and "modprobe wctdm" and again run "cat /proc/interrupts" I then get: .. .. .. .. .. 169 118489 IO-APIC-level wct2xxp, eth0 201 118497 IO-APIC-level wctdm .. etc How can I force the wct2xxp to load on a separate IRQ? I tried moving the eth0 to IRQ 10 but could not. Any ideas? Thank you. _*/Phil Menico/*_ XTEND Communications 171 Madison Avenue, New York, NY 10016 212-951-7632 (Office) 212-951-7683 (Fax) www.xtend.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] Can I recreate a Fax from a recorded file?
Ok, I have to agree here. IF my simple fax server log/tiff archive is not enough to satisfy a client that the fax is genuine I would not want them as my customer. I don't care how much money they spend. Business is business and what I do is what I do. There has to be at least a little bit of trust between me and my clients otherwise we spend too much time bickering about stuff. Have you ever tried to bill "b*tch-time"? I have worked in "High-CYA" positions before and refuse to have that in my current company. Aloha, Mark C Scott Gifford wrote: I don't see the advantage to this; the client still has to trust that all of this is done correctly, and if they don't trust the fax recipient to put the correct fax in the paper file or keep the correct TIFF, why would they trust them to do this? Using a third party to receive and relay the fax, one which is trusted by both the client and the fax recipient, would solve the problem; the third party could create a document with the caller information (ideally from ANI, which is harder to forge), the time, and the message itself, then digitally sign it. This might even be an interesting business plan, for some applications where confirmed document transmittal is important. But it's hard for me to imagine this isn't overkill; if a client and a service provider distrust each other so thoroughly that they have to communicate through a third party to verify integrity, probably they just shouldn't do business with each other. Scott. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] why a perfectly fine iax2 host becomes UNREACHABLE?
2006/5/5, Eric ManxPower Wieling <[EMAIL PROTECTED]>: > whether that makes a difference. Should I switch from > hostname to IP address in the register string too? If Asterisk has a DNS lookup failure it will never retry that lookup. Well... I think it should be very easy to solve for the developpers what knows exactly where in the code to make the changes... -- Alejandro Vargas ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can I recreate a Fax from a recorded file?
2006/5/5, Steve Underwood <[EMAIL PROTECTED]>: In most cases, forensic analysis of the audio from another machine would easily show it was a fake. It would lack tell-tale fingerprints of the true path, unless it was done with extreme care. Certainly using exactly the same model of FAX machine that sent the genuine FAX would be a must. Not just for the vendor information it sends, but for the fine details in how its modems behave. To pass of the altered fax as being from the original sender would require careful control of the DSP. People must realize that fax is not a secure method to send information. There are protocols created to solve the legal problems, that uses digital signs, standard formats, and relies in a 3d party entity that certificates that the info was sent, recived and red, logging the exact times. These methods are actually used by many busines to send price listings, budgets, buy orders, invoices, bank money transfers, etc. all automatcaly and electronicaly. A stock system can detect a low count of some products, it sends price-listing request to the providers and waits for the answers, someone with authorities chooses the prefered one or request a bettre price. Then orders the system to buy the products. The provider system receives the request electronically, prepares the shipping, sends the invoice electronically, etc. etc. All automatic and all legal. Faxes are not the right way to do legal and reliable document transfers. It is only a quick and unreliable way to show a document to someone instead of dictating it by phone. -- Alejandro Vargas ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can I recreate a Fax from a recorded file?
Alejandro Vargas wrote: 2006/5/5, Steve Underwood <[EMAIL PROTECTED]>: In most cases, forensic analysis of the audio from another machine would easily show it was a fake. It would lack tell-tale fingerprints of the true path, unless it was done with extreme care. Certainly using exactly the same model of FAX machine that sent the genuine FAX would be a must. Not just for the vendor information it sends, but for the fine details in how its modems behave. To pass of the altered fax as being from the original sender would require careful control of the DSP. People must realize that fax is not a secure method to send information. There are protocols created to solve the legal problems, that uses digital signs, standard formats, and relies in a 3d party entity that certificates that the info was sent, recived and red, logging the exact times. These methods are actually used by many busines to send price listings, budgets, buy orders, invoices, bank money transfers, etc. all automatcaly and electronicaly. A stock system can detect a low count of some products, it sends price-listing request to the providers and waits for the answers, someone with authorities chooses the prefered one or request a bettre price. Then orders the system to buy the products. The provider system receives the request electronically, prepares the shipping, sends the invoice electronically, etc. etc. All automatic and all legal. Faxes are not the right way to do legal and reliable document transfers. It is only a quick and unreliable way to show a document to someone instead of dictating it by phone. FAXes *are* accepted for many legal things. It is one of the things keeping FAX from dying completely. They are completely forgable, yet accepted. Before them telex ket going long after its sell by date because a telex was acceptable in court. They were far more forgable than FAX. An old boss if mine has to show telexes were forged as an expert witness. The only reason he could do this is because people are sloppy, and didn't get the spacing exactly like a real telex would have. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
AW: AW: [Asterisk-Users] DTMF detection when outgoing call tomobilephones
Actually I am using Asterisk 1.2.7.1, zaptel 1.2.5, libpri 1.2.2 I ve tried many values for rx/txgain togeher with echocancel and relaxdtmf. The detection is not working with call file, manager originate and not with the dial command to the mobile. I have no ideas left. I got it sometimes to work if I use a specific channel (i.e. Dial(ZAP/14/...) But with the same vaules on a second call there is the DTMF problem again. Sometimes the DTMF tones are choppy at the other end of the line. There is also a Warning with outbound calls only: -- Called G1/016080... -- Moving call from channel 2 to channel 31 May 5 12:45:33 WARNING[1403]: chan_zap.c:7744 pri_fixup_principle: Can't fix up channel from 2 to 31 because 31 is already in use May 5 12:45:33 WARNING[1403]: chan_zap.c:9045 pri_dchannel: Unable to move channel 31! -- Zap/31-1 is ringing -- Zap/31-1 answered Zap/2-1 But everything is ok and working on not moblie connections My actual settings: Zaptel.conf # # Zaptel Configuration File # span=1,1,0,ccs,hdb3,crc4 span=2,0,0,ccs,hdb3,crc4 # bchan=1-15 dchan=16 bchan=17-31 bchan=32-46 dchan=47 bchan=48-62 # loadzone = de #loadzone = us-old #loadzone=gr #loadzone=it #loadzone=fr #loadzone=de #loadzone=uk #loadzone=fi #loadzone=jp #loadzone=sp #loadzone=no #loadzone=hu #loadzone=lt #loadzone=pl defaultzone=nl Zapata.conf ; ; Zapata telephony interface ; ; Configuration file ; [trunkgroups] [channels] ; context=default switchtype=euroisdn pridialplan=unknown prilocaldialplan=unknown ; ; sample 1 for Germany ;internationalprefix = 00 ;nationalprefix = 0 ;localprefix = ;privateprefix = ;unknownprefix = ;relaxdtmf=yes overlapdial=yes ;priindication = outofband ;rxwink=300 ; Atlas seems to use long (250ms) winks ;toneduration=100 ;usedistinctiveringdetection=yes ;busydetect=no ;immediate=no usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes ;echocancel=yes echocancelwhenbridged=yes ;rxgain=2.0 ;txgain=0.0 ; group=1 callgroup=1 pickupgroup=1 immediate=no ; ;callprogress=yes ;progzone=us ; ;jitterbuffers=4 ; group=1 ;switching=euroisdn signalling=pri_cpe context=from-pmx rxgain=2.0 txgain=0.0 relaxdtmf=yes echocancel=yes callerid=asreceived channel=>1-15, 17-31 group=2 ;switching=euroisdn signalling=pri_net context=to-pmx callerid=asreceived channel=>32-46, 48-62 -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Kevin P. Fleming Gesendet: Freitag, 5. Mai 2006 09:52 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: Re: AW: [Asterisk-Users] DTMF detection when outgoing call tomobilephones Marc Scheuffler wrote: > Yes I can hear the DTMF keys. I ve tried 2 different phones and 3 different > mobile network providers. Nothing. There was a bug in various versions of Asterisk when outbound calls were placed using spool files and then could not detect DTMF from the called party. Without more details, including the version of Asterisk you are running, it will be difficult to suggest anything to you other than upgrading to the latest release. ___ --Bandwidth 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] Re: Asterisk-Users Digest, Vol 22, Issue 26
Hi to all. My asterisk pbx has a tdm400p card with 2 FXO cards on it. I configured the extensions.conf to send all the call incoming from that zap channels to an IVR system. I see in the asterisk CLI the call incoming and the playback of the message custom/myfile but no sound is played on the channel, i cannot hear nothing. If i change the configuration and i send the call to an internal sip extensions alla works great. Also works well when i configure a sip trunks to answer and playback the ivr message, i can hear it very well. If I use the zap channel to do an outside call, all works well. Asterisk varsion is 1.2.7.1 I'm going crazy, please send me some suggestions. Thanks in advance. Bye ___ --Bandwidth 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] Festival , Cannot hear the words after ","
Hi John, are using enclosing in quotation mark your statement? (festival('text to speech') ) Giorgio John Joseph wrote: Hi I am trying to use festivall with asterisk , I am using RHEL4 , asterisk1.2.7.1 and festival-1.95-beta , I am able to hear the voice form the text file , when I dial to the extension, but when I have "," in my text file , it plays only the text upto "," and in the CLI , the "," is shown as "|" I had cut and pasted CLI messages for reference -- Executing Answer("SIP/326-78c7", "") in new stack -- Executing Festival("SIP/326-78c7", "Hello | This is Joseph | How are U ") in new stack == Parsing '/etc/asterisk/festival.conf': Found -- Executing Hangup("SIP/326-78c7", "") in new stack == Spawn extension (from-internal, 555, 3) exited non-zero on 'SIP/326-78c7' -- Executing Macro("SIP/326-78c7", "hangupcall") in new stack I had followed the link http://www.voip-info.org/wiki/view/Asterisk+festival+installation for the installation Thanks Joseph John Send instant messages to your online friends http://uk.messenger.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Help with IRQ conflict between wct2xxp and eth0
Go into the BIOS, disable every possible device such as floppy controller, usb, serial, parallel, etc. If that doesn't work, move card to another slot. > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Giorgio Incantalupo > Sent: Friday, May 05, 2006 1:39 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Help with IRQ conflict between > wct2xxp and eth0 > > Hi Phil, > may sound a stupid advice buthave you tried to change PCI > slots on your server? > > Giorgio > > Phil Menico wrote: > > > > I have a conflict problem with the eth0 card and wct2xxp > digium board. > > The PRI can receive calls but my network connection is gone. > > > > When I "cat /proc/interrupts" I get the following: > > > > 1 .. > > > > 1 .. > > > > .. > > > > .. > > > > .. > > > > 169 0 IO-APIC-level wct2xxp, eth0 > > > > .. > > > > etc. > > > > even before I "modprobe wct2xxp" > > > > After I "modprobe wct2xxp" and "modprobe wctdm" and again run "cat > > /proc/interrupts" > > > > I then get: > > > > > > > > .. > > > > .. > > > > .. > > > > .. > > > > .. > > > > 169 118489 IO-APIC-level wct2xxp, eth0 > > > > 201 118497 IO-APIC-level wctdm > > > > .. > > > > etc > > > > > > > > How can I force the wct2xxp to load on a separate IRQ? I > tried moving > > the eth0 to IRQ 10 but could not. > > > > Any ideas? > > > > > > > > > > Thank you. > > > > _*/Phil Menico/*_ > > > > XTEND Communications > > 171 Madison Avenue, New York, NY 10016 > > 212-951-7632 (Office) > > 212-951-7683 (Fax) > > www.xtend.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 > ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] why a perfectly fine iax2 host becomes UNREACHABLE?
Tom Engleward wrote: --- "Eric \"ManxPower\" Wieling" <[EMAIL PROTECTED]> wrote: If Asterisk has a DNS lookup failure it will never retry that lookup. "Never" meaning until the next "reload" command is issued, or until the next "restart" command is issued, or until the next time the OS reboots, or until the next time asterisk and its config files are deleted and reinstalled? Correct. Perhaps "never automatically" would be a better choice of words. Wait, that was a multiple choice question: which option is the least-drastic sufficiently drastic option to force asterisk to retry? If a mere "reload" is sufficient, then the periodic test-and-reload which I suggested (which Andrew Kohlsmith correctly said is just a band-aid) would at least be an effective band-aid for the short term, until the actual cause of the DNS lookup failure is found and fixed. A proper fix is preferable, but a band-aid is better than nothing. Any one of those will work. Of course if using IP address instead of hostname successfully avoids the problem, then that's a better band-aid, but only is practical if the host in question is not subject to having its IP address changed. Only if BOTH hosts are on dynamic IPs. If only one of them is on a dynamic IP, then the dynamic one should be registering with the static server. This is why registration was invented. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] why a perfectly fine iax2 host becomes UNREA CHABLE?
On Thursday 04 May 2006 18:44, Colin Anderson wrote: > Ah, true dat. However, if quality was crappy believe me my users would let > me know. They are salespeople and wholly intolerant of anything that keeps > them from yipping on the phone. I also run exactly the same rig at my house > backhauled to our main Asterisk box to force me to eat my own dog food, and :-) I do the *exact* same thing, and for the same reasons. :-) > time. The only time I get chop is when I use my cordless to the TDM400: > Cordless > TDM400 > * > IAX2 > * > PRI. Salespeople keep asking for > cordless phones, but I keep telling them forget it. Cordless to an FXS to > IAX over the Internet is just begging for trouble. I don't have any trouble with my cordless phone to a TDM400, but there could be a wealth of factors playing into the chop you're seeing, as I'm sure you're aware. I had a *really* bad digital cordless phone that I actually sent to Digium to help them figure out WHY this particular phone hated their FXS ports so badly, but I don't think I'd ever heard back from them. I've never had another phone act that way though. (Panasonic digital cordless, I have another one from Panasonic and it works perfectly fine.) -A. > ___ > --Bandwidth 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: AW: [Asterisk-Users] DTMF detection when outgoing call tomobilephones
Mark Ackroyd wrote: There was a bug in various versions of Asterisk when outbound calls were placed using spool files and then could not detect DTMF from the called party. Without more details, including the version of Asterisk you are running, it will be difficult to suggest anything to you other than upgrading to the latest release. Works perfectly on mine. Using Digium Digital boards (ZAP -> UK Mobile). Asterisk version 1.2.1 with a few patches. Also works for me with USA cell phones on at least two different Asterisk servers with version 1.2.x -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] why a perfectly fine iax2 host becomes UNREACHABLE?
On Fri, 2006-05-05 at 07:47 -0500, Eric "ManxPower" Wieling wrote: > Only if BOTH hosts are on dynamic IPs. If only one of them is on a > dynamic IP, then the dynamic one should be registering with the static > server. This is why registration was invented. Until I got static IPs for all of them I had 3 *s intercommunicating. Solved the problem by doing a reload from ipup-local when the ip changes it's handled automatically. -- Dave Cotton <[EMAIL PROTECTED]> ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Hold and Retrieve
Our current PBX allows us to put a call on hold and then anyone in the building can dial #9XXX and pick up that call. I know that I can replicate a similar function by parking. But I would really like to replicate the existing setup. Something about having to train people to hit more than one button. Any suggestions? Sean ___ --Bandwidth 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] number that starts with star on PAP2
I wrote: > In the PAP2's setup there are all of these "Vertical Service Activation > Codes" that start with star and "Outbound Call Codec Selection Codes", > also the setup menu is accessed by pressing star four times, could they > be intefering with dialing numbers that start with a star? And is there > any way to get *8 and *XXX to dial? Time Bandit wrote: > Why I did to mine is modify all the internal "Vertical Service > Activation Codes" to be "**x" instead of "*x". There is probably a > better way, but this worked for me. We tried that, but users report they are still having the same problem (the site is located in a different country so I can't check myself). Philippe Lindheimer wrote: Yes - that's your problem. You need to porgram the dialpan in the PAP2 appropriately, disable functions you don't want, etc. We were trying to dial *100, and there wasn't anything in either of the Codes section that started with *1. Do we have to disable every function that starts with a star to get anything to work? Also, is a function disabled by clearing it? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Repost: External voicemail and MWI on internal phone
Hi, I didn't get any response to this posting so thought I'd post again in the hope that anyone who missed the posting the first time may be able to offer some ideas. This problem isn't specific to the particular model of phone - I can see from sip debug that the local extension itself is not being sent any Messages-Waiting headers. Any alternative strategy would also be welcomed if I'm off on the wrong track ;-) Thanks, Reuben Original Message Subject: External voicemail and MWI on internal phone Date: Mon, 17 Apr 2006 10:35:07 +1200 From: Reuben Farrelly <[EMAIL PROTECTED]> To: asterisk-users@lists.digium.com Hi, I have a Cisco 7941 phone running SIP, and for a variety of reasons [1] have configured this in the short term to use my local Asterisk server to register to, and then have my local Asterisk server in turn register to the upstream SIP proxy at my voice provider. Voicemail is supplied by my VoIP provider, so I've no real need to set up voicemail locally. My provider also runs Asterisk, and correctly sends me down these headers: Messages-Waiting: yes Voice-Message: 1/0 (0/0) However the local phones never see the headers and thus never know that there is voicemail waiting upstream. Is there a way to forward on or regenerate the Messages-Waiting and Voice-Message headers from upstream, to each and every local phone in the same context so that any and all phones on the local network see the voicemail present and have their MWI light up? Thanks, Reuben [1] So far I've been unable to get the 7941 to do authenticate to my provider, unsure quite why, the phone seems to just ignore WWW-Authenticate requests from upstream. A Cisco 7940 and ATA-188 work fine so it may be a bug or a feature that needs to be configured - dunno. ___ --Bandwidth 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] Registering Remote Sipura to Asterisk (both behind firewall)
Can anybody point or provide working configuration how to register Sipura to Asterisk over the Internet. Both Sipura and Asterisk are behind firewalls. I'll be force to use SIP as that is the only protocol that Sipura is using. Do I need to enter any "STUN Server:" setting in "SIP" tab. On Asterisk I think I only need to make changes in sip.conf isn't it? What ports do I need to open on my firewall? -- #Joseph ___ --Bandwidth 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] number that starts with star on PAP2
> Why I did to mine is modify all the internal "Vertical Service > Activation Codes" to be "**x" instead of "*x". There is probably a > better way, but this worked for me. We tried that, but users report they are still having the same problem (the site is located in a different country so I can't check myself). Sorry, I don't have my PAP2 under hand, but this is all I did, changed every *xx to **xx and it worked. Something that may help you is http://www.netphonedirectory.com/pap2_dialplan.htm Philippe Lindheimer wrote: > Yes - that's your problem. You need to porgram the dialpan in the PAP2 > appropriately, disable functions you don't want, etc. > We were trying to dial *100, and there wasn't anything in either of the Codes section that started with *1. Do we have to disable every function that starts with a star to get anything to work? Also, is a function disabled by clearing it? I didn't try that so I don't know. Just make sure that you changed every single "Vertical Service Activation Codes" to a double *. If you still can't fix it, let me know and I will get back my PAP2 and try to help you hth ___ --Bandwidth 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] Dumping queue_log to MySQL
Anyone have a working solution for this? I played with the demo that came with QueueMetrics to see how they were doing it and it was working for a bit but now somehow every night it stopped. Perl and Tail are still running on the server but the information is not dumping to the MySQL database. I don’t get any error messages anywhere telling me why it stops. As far as tail and perl are concerned everything is fine. We will be using this for a call center and need more reliability. Anyone got one working? Thanks _ Kevin Savoy Business Unit Telecom Analyst 2218 4th Ave W Williston, ND 58801 Ph: 701-774-4023 Fax: 701-774-2901 http://www.novo1.com Novo 1 is a service mark of Novo 1, Inc ___ --Bandwidth 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] problem g729
Hello , I’m have this problem before copy codec in the /usr/lib/asterisk/modules before registration ….. My asterisk is Asterisk SVN-trunk-r20297 built by xxx@ xxx on a i686 running Linux on 2006-04-20 01:02:07 UTC This erro : codec_g729a.so]May 5 21:39:16 WARNING [6950]: loader.c:731 __load_resource: misstng mod_data for codec_g729a.so Segmentation fault Thanks Cristian Latapiat ___ --Bandwidth 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] problem g729
cristian wrote: > My asterisk is Asterisk SVN-trunk-r20297 built by xxx@ xxx on a i686 running > Linux on 2006-04-20 01:02:07 UTC This has been covered many times on this list already. The G.729 codec binary is not compatible with current SVN trunk. If you are running SVN trunk in production, stop doing so now :-) ___ --Bandwidth 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] Dumping queue_log to MySQL
We use a Perl script called acd_logger.pl. I can't remember when I got it from now but if you want a copy then mail me directly.RegardsJon Jon FarmerTelford, Shropshire, UK- Original Message From: Kevin Savoy <[EMAIL PROTECTED]>To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Friday, 5 May, 2006 2:57:06 PMSubject: [Asterisk-Users] Dumping queue_log to MySQL Anyone have a working solution for this? I played with the demo that came with QueueMetrics to see how they were doing it and it was working for a bit but now somehow every night it stopped. Perl and Tail are still running on the server but the information is not dumping to the MySQL database. I don’t get any error messages anywhere telling me why it stops. As far as tail and perl are concerned everything is fine. We will be using this for a call center and need more reliability. Anyone got one working? Thanks _ Kevin Savoy Business Unit Telecom Analyst 2218 4th Ave W Williston, ND 58801 Ph: 701-774-4023 Fax: 701-774-2901 http://www.novo1.com Novo 1 is a service mark of Novo 1, Inc ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FW: NuFone Update: DIDs
i got the same email. They emailed everyone later that there was an "error". --- Steve Prior <[EMAIL PROTECTED]> wrote: > Rich Adamson wrote: > > > > > At least outbound calls still work, even though > they changed IP > > addresses (and probably colo locations). > > > > > > Maybe not so much now. I just got a disconnect > notice from nufone > which states that I have a positive balance in my > account, but still > need to add money to bring it up above zero for my > account to be > re-enabled. Somebody just broke the billing code I > think... > > Steve > ___ > --Bandwidth and Colocation provided by Easynews.com > -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Transfer Disconnect (CT-5)
Hi, Anyone has experience in using Call Transfer Disconnect (CT-5) over a PRI with Asterisk ? Call Transfer Disconnect allows you to transfer a call to a third party and disconnect yourself from the communication and also freeing your PRI channels. Here is a document that explains how it works: http://www.callamericacom.com/pdf/ctd_instructions.pdf My question is how can I do this with Asterisk, especially with a softphone (currently using SJPhone). Andre Courchesne ___ --Bandwidth 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] CARD.XML for MGCP cisco phone
Anyone has a working CARD.XML for cisco MGCP phones? The one on the cisco site is old and it's not working with the new firmwares. Thanks Nico___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] why a perfectly fine iax2 host becomes UNREACHABLE?
On Friday 05 May 2006 08:47, Eric "ManxPower" Wieling wrote: > Any one of those will work. I disagree; this is NOT an Asterisk DNS issue. I specify my hosts by IP and this still happens (in fact, it happened this morning). -A. ___ --Bandwidth 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] Realtime, 2 server setup problem?
All, we're running realtime and ast 1.2.7.1 stable. The problem I'm having is when you register a SIP device on ServerA, works fine, sip show peer works fine. When you dial the SIP device from ServerB however, it tries to dial, even does a MySQL query like it should but comes back saying no route to dest. When I do a 'sip show peer ..' on ServerB, the peer comes back fine, it even shows all the proper fields listed in the database (ie, codec, dtmf mode, etc). However, the IP address field is coming back as "unspecified". Now, naturally I ensured server time was acurate with NTP for starters. This was to avoid the registration appearing as EXPIRED on ServerB. Sip.conf snippet (same on ServerA and ServerB) ... rtcachefriends=yes rtupdate=yes rtautoclear=yes rtignoreexpire=yes So, as you can all see it's most importantly caching the peer and updating MySQL. And yes if you do a 'realtime load sipusers name PEERNAME', it shows the ipaddr field with the correct value. Is there something obvious I am missing? I googled this to death and cannot find anyone with a similar issue. I remember running into this issue in the past and quite frankly am unsure if it has *ever* worked. Thanks in advance, Matt Schulte ___ --Bandwidth 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] Dumping queue_log to MySQL
On 5/5/06, Jon Farmer <[EMAIL PROTECTED]> wrote: We use a Perl script called acd_logger.pl. I can't remember when I got it from now but if you want a copy then mail me directly.Regards Jon Jon FarmerTelford, Shropshire, UK - Original Message From: Kevin Savoy < [EMAIL PROTECTED]>To: Asterisk Users Mailing List - Non-Commercial DiscussionSent: Friday, 5 May, 2006 2:57:06 PMSubject: [Asterisk-Users] Dumping queue_log to MySQL Anyone have a working solution for this? I played with the demo that came with QueueMetrics to see how they were doing it and it was working for a bit but now somehow every night it stopped. Perl and Tail are still running on the server but the information is not dumping to the MySQL database. I don't get any error messages anywhere telling me why it stops. As far as tail and perl are concerned everything is fine. We will be using this for a call center and need more reliability. Anyone got one working? We've had a fair amount of success using the script attached with the following mysql schema. mysql> desc queue_actions;+---+-+--+-+-+---+| Field | Type | Null | Key | Default | Extra |+---+-+--+-+-+---+ | queueid | int(11) | | MUL | 0 | || uniqueid | varchar(25) | | MUL | | || queuename | varchar(45) | | MUL | | || agentname | varchar(45) | | | | | | action | varchar(25) | | MUL | | || alt1 | varchar(10) | | | | || alt2 | varchar(10) | | | | || userfield | varchar(75) | | | | | | calltype | int(4) | YES | | NULL | || nodename | varchar(25) | YES | | NULL | |+---+-+--+-+-+---+10 rows in set (0.00 sec) -- Bird's The Word Technologies, Inc.http://www.btwtech.com/ insert-queue-info.pl Description: Binary data ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk 1.2 & hisax teles 16.3 isa
Hi I was using asterisk 1.0.7 until now. For my connection with the PSTN I used a good old isa hisax teles 16.3 isdn card. Now while compiling 1.2 I nothiced that i4l and chan_modem is no longer supported? Is there any way of getting my good old ISA card to work under 1.2? The docs keep saying use misdn, capi, etc from now on. But I don't think my card is CAPI capable arne ___ --Bandwidth 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] Realtime, 2 server setup problem?
Matt Schulte wrote: > Is there something obvious I am missing? I googled this to death and > cannot find anyone with a similar issue. I remember running into this > issue in the past and quite frankly am unsure if it has *ever* worked. Seriously? This has been discussed many times on this list and the -dev list. It is not expected to work at this time. The current version of Asterisk does not have the ability to share realtime registrations between two servers (they can share the same table as long as they have separate lists of peers). In other words, your last comment above is correct :-) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dumping queue_log to MySQL
http://forums.digium.com/viewtopic.php?t=4073 I modified it some, but it integrates with my cms and provides realtime call stats. Contact me off list if you’ve like to see how it works. Wes Baehr Ability Business Computing, Ltd. Office: 330.882.0455 x25 Cell: 330.882.0455 x35 Fax: 330.882.0455 [EMAIL PROTECTED] From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin Savoy Sent: Friday, May 05, 2006 9:57 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Dumping queue_log to MySQL Anyone have a working solution for this? I played with the demo that came with QueueMetrics to see how they were doing it and it was working for a bit but now somehow every night it stopped. Perl and Tail are still running on the server but the information is not dumping to the MySQL database. I don’t get any error messages anywhere telling me why it stops. As far as tail and perl are concerned everything is fine. We will be using this for a call center and need more reliability. Anyone got one working? Thanks _ Kevin Savoy Business Unit Telecom Analyst 2218 4th Ave W Williston, ND 58801 Ph: 701-774-4023 Fax: 701-774-2901 http://www.novo1.com Novo 1 is a service mark of Novo 1, Inc -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.392 / Virus Database: 268.5.4/332 - Release Date: 5/4/2006 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.392 / Virus Database: 268.5.4/332 - Release Date: 5/4/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] why a perfectly fine iax2 host becomes UNREACHABLE?
Andrew Kohlsmith wrote: On Friday 05 May 2006 08:47, Eric "ManxPower" Wieling wrote: Any one of those will work. I disagree; this is NOT an Asterisk DNS issue. I specify my hosts by IP and this still happens (in fact, it happened this morning). Then you are experiencing a different problem. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Registering Remote Sipura to Asterisk (both behind firewall)
Can anybody point or provide working configuration how to register Sipura to Asterisk over the Internet. Both Sipura and Asterisk are behind firewalls. I'll be force to use SIP as that is the only protocol that Sipura is using. Do I need to enter any "STUN Server:" setting in "SIP" tab. On Asterisk I think I only need to make changes in sip.conf isn't it? What ports do I need to open on my firewall? -- #Joseph ___ --Bandwidth 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] Problem on Zap Channel with IVR
Hi to all. My asterisk pbx has a tdm400p card with 2 FXO cards on it. I configured the extensions.conf to send all the call incoming from that zap channels to an IVR system. I see in the asterisk CLI the call incoming and the playback of the message custom/myfile but no sound is played on the channel, i cannot hear nothing. If I change the configuration and i send the call to an internal sip extensions alla works great. Also works well when i configure a sip trunks to answer and playback the ivr message, i can hear it very well. If I use the zap channel to do an outside call, all works well also. Asterisk varsion is 1.2.7.1 This is my zapata.conf : *Code:* channels] language=en context=from-pstn signalling=fxs_ks rxwink=300 ; Atlas seems to use long (250ms) winks ; ; Whether or not to do distinctive ring detection on FXO lines ; ;usedistinctiveringdetection=yes busydetect=yes busycount=4 usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no echotraining=800 rxgain=0.0 txgain=0.0 group=0 callgroup=1 pickupgroup=1 immediate=no ;faxdetect=both faxdetect=incoming ;faxdetect=outgoing ;faxdetect=no and this is what I see in the asterisk cli during the call : *Code:* -- Starting simple switch on 'Zap/1-1' -- Executing Set("Zap/1-1", "LOOPCOUNT=0") in new stack -- Executing Answer("Zap/1-1", "") in new stack -- Executing Wait("Zap/1-1", "1") in new stack -- Executing Set("Zap/1-1", "TIMEOUT(digit)=10") in new stack -- Digit timeout set to 10 -- Executing Set("Zap/1-1", "TIMEOUT(response)=10") in new stack -- Response timeout set to 10 -- Executing BackGround("Zap/1-1", "custom/benvenut0") in new stack -- Playing 'custom/benvenut0' (language 'en') == Spawn extension (from-pstn, s, 6) exited non-zero on 'Zap/1-1' -- Executing Hangup("Zap/1-1", "") in new stack == Spawn extension (from-pstn, h, 1) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' I'm going crazy, please send me some suggestions. Thanks in advance. Bye ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7970 running SIP question
Group I have a Cisco 7970 Running the newest SIP image. I'm running Asterisk SVN-trunk-r7498 on 2006-04-30 15:11:39 UTC When I get a call the callerid number show something like [EMAIL PROTECTED] I thought I seen somewhere what that was but I'm unable to find the correct wording when searching Google to find that post again. Can anyone help me out here. How can I remove the asterisk servers IP from the phone number? Also I'm unable to get the time zone correct on the phone. It is in UTC and I'm in EST I see in the file where it looks like it goes but what I have tried has not worked as of yet. Here is what it looks like M/D/Y EST Thanks again for all your help!!! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7970 running SIP question
I don't remember exactly what the reasoning on Cisco's part is of having the IP address on there, but it happens on ours too. It shouldn't cause any problems with making outgoing calls from the directory, it's just annoying to see it pop up. As for the date time settings... this is what we have in ours: CMLocal M/D/YA Central Standard/Daylight Time I'm guessing you should be able to change it to say Eastern instead of Central On Fri, 5 May 2006, Hall, Eric M. wrote: Group I have a Cisco 7970 Running the newest SIP image. I'm running Asterisk SVN-trunk-r7498 on 2006-04-30 15:11:39 UTC When I get a call the callerid number show something like [EMAIL PROTECTED] I thought I seen somewhere what that was but I'm unable to find the correct wording when searching Google to find that post again. Can anyone help me out here. How can I remove the asterisk servers IP from the phone number? Also I'm unable to get the time zone correct on the phone. It is in UTC and I'm in EST I see in the file where it looks like it goes but what I have tried has not worked as of yet. Here is what it looks like M/D/Y EST Thanks again for all your help!!! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unwanted conference with snom320 and asterisk1.07bristuffed
Title: Messaggio We use a large number of Snom 320's and we have this same problem even with Call join on Xfer set to off. I had not previously linked it decisively to the Snoms, but it sounds like that is likely our issue. We've had to stay at the 4.5 firmware because otherwise we get additional incomming calls when our reps put someone on hold to make a transfer. If I find any solution I'll be sure to share it with the list. Thanks for sharing your experiences, Frank Webb Assistant Project Leader Inter Media Marketing Solutions - Original Message - From: Alexander Lopez To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Thursday, May 04, 2006 7:35 AM Subject: RE: [Asterisk-Users] Unwanted conference with snom320 and asterisk1.07bristuffed Under Advanced make sure this is set: Call join on Xfer (2 calls): to off From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tommaso CalosiSent: Thursday, May 04, 2006 4:02 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Unwanted conference with snom320 and asterisk 1.07bristuffed I have 13 Snom 320 with asterisk 1.07 bristuffed. The problem is that sometimes on random basis, when one customer is placed on hold and another call arrives, the customers are put in conference with each other. This look very strange to me, but I've disabled the confernce button on the snom phones to prevent the human errors, but it still occurs. Investigating I've discovered that a similar problem was fixed with the Snom320 Release 5.2 (http://www.snom.com/snom320_release_notes.html ) It says: fixed unwanted conference bug in offhook/enter during ringback with an incoming call BUT my phones are already running 5.2 firmware. Any idea? Am I the only one with this problem? Do you think is the usual buggy-snom firmware problem? Or it might be an Asterisk problem? ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dumping queue_log to MySQL
Kevin Savoy wrote: Anyone have a working solution for this? I played with the demo that came with QueueMetrics to see how they were doing it and it was working for a bit but now somehow every night it stopped. Perl and Tail are still running on the server but the information is not dumping to the MySQL database. I don’t get any error messages anywhere telling me why it stops. As far as tail and perl are concerned everything is fine. We will be using this for a call center and need more reliability. Anyone got one working? Kevin, We are using QueueMetrics to provide reporting for our inbound call center activities. On a heavy day we're handling 13,000+ calls spread across roughly 6 to 10 queues. Due to some of the specifics or our setup, we are using a custom program to load queue_log data into MySQL. There may be a little tweaking involved to get things up and running (it's rare to find a product that is a *perfect* fit for your business), but once you're there I think you'd be happy with QueueMetrics. The developers are very responsive to feedback and our floor managers are extremely pleased with the reporting. Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: Spam? Re: [Asterisk-Users] Cisco 7970 running SIP question
Aaron Yes it is very annoying! Thanks for the date time settings. That worked GREAT!!! Thanks - Eric -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Aaron Daniel Sent: Friday, May 05, 2006 11:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Spam? Re: [Asterisk-Users] Cisco 7970 running SIP question I don't remember exactly what the reasoning on Cisco's part is of having the IP address on there, but it happens on ours too. It shouldn't cause any problems with making outgoing calls from the directory, it's just annoying to see it pop up. As for the date time settings... this is what we have in ours: CMLocal M/D/YA Central Standard/Daylight Time I'm guessing you should be able to change it to say Eastern instead of Central On Fri, 5 May 2006, Hall, Eric M. wrote: > > Group > I have a Cisco 7970 Running the newest SIP image. > I'm running Asterisk SVN-trunk-r7498 on 2006-04-30 15:11:39 UTC > > When I get a call the callerid number show something like > [EMAIL PROTECTED] I thought I seen somewhere what that was but I'm > unable to find the correct wording when searching Google to find that > post again. Can anyone help me out here. How can I remove the asterisk > servers IP from the phone number? > > > Also I'm unable to get the time zone correct on the phone. It is in > UTC and I'm in EST I see in the file where it looks like it goes but > what I have tried has not worked as of yet. Here is what it looks like > > > M/D/Y > EST > > > > Thanks again for all your help!!! > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth 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] Music on Hold from Soundcard
Hi everyone, Sent this out previously, but it didn't seem to show up. My apologies if this is a duplicate! I've been trying to get MoH to work from the line-in on my soundcard, but as of yet have had no success. I found this script that should allow for it to happen: http://www.sineapps.com/news.php?rssid=722 The script, when run as the asterisk user, works properly and streams sound to stdin. I can use arecord to record wavs which playback fine. But when Asterisk starts MoH it stops it immediately afterwards with no explanation. Has anyone gotten this to work? Or does anyone have any ideas on how to debug why MoH stops immediately after starting? Thanks in advance!Alex Robar-- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] number that starts with star on PAP2
You need to modify the dialplan within the PAP2 unit to allow that as a valid number or it won't pass it on. Take a look at the following, it is not specifically for the PAP2 but all the dialplan information should apply: http://www.sipura.com/Documents/SipuraSPAUserGuidev2.0.9.pdf From: "Time Bandit" <[EMAIL PROTECTED]>To: "Asterisk Users Mailing List - Non-Commercial Discussion"Date: Fri, 5 May 2006 09:48:41 -0400Subject: Re: [Asterisk-Users] number that starts with star on PAP2> > Why I did to mine is modify all the internal "Vertical Service> > Activation Codes" to be "**x" instead of "*x". There is probably a> > better way, but this worked for me.>> We tried that, but users report they are still having the same problem> (the site is located in a different country so I can't check myself).Sorry, I don't have my PAP2 under hand, but this is all I did, changedevery *xx to **xx and it worked.Something that may help you ishttp://www.netphonedirectory.com/pap2_dialplan.htm>> Philippe Lindheimer wrote:> > Yes - that's your problem. You need to porgram the dialpan in the PAP2> > appropriately, disable functions you don't want, etc.> >> We were trying to dial *100, and there wasn't anything in either of the> Codes section that started with *1. Do we have to disable every> function that starts with a star to get anything to work? Also, is a> function disabled by clearing it?I didn't try that so I don't know. Just make sure that you changedevery single "Vertical Service Activation Codes" to a double *. If youstill can't fix it, let me know and I will get back my PAP2 and try tohelp youhth Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. Great rates starting at 1¢/min.___ --Bandwidth 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] Access to sip.conf username field from dialplan
I have two SIP clients defined in sip.conf, as follows: [dave] username=dave type=friend secret=dave host=dynamic context=local accountcode=deskoptional [dave-laptop] username=dave type=friend secret=dave host=dynamic context=local accountcode=deskoptional I want to refer to the username field in extensions.conf, but cannot find a function to do it. I know Asterisk can see this value, because both "SIP SHOW PEER dave" and "SIP SHOW PEER dave-laptop" display the desired value under "Def. Username". PS: I am new to Asterisk and this newsgroup, so a cc to my email address (nntp at deskoptional.com) would be appreciated. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: Spam? Re: [Asterisk-Users] Cisco 7970 running SIP question
Aaron Any idea how to change it from 24hr to 12hr ? Thanks again! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hall, Eric M. Sent: Friday, May 05, 2006 11:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: Spam? Re: [Asterisk-Users] Cisco 7970 running SIP question Aaron Yes it is very annoying! Thanks for the date time settings. That worked GREAT!!! Thanks - Eric -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Aaron Daniel Sent: Friday, May 05, 2006 11:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Spam? Re: [Asterisk-Users] Cisco 7970 running SIP question I don't remember exactly what the reasoning on Cisco's part is of having the IP address on there, but it happens on ours too. It shouldn't cause any problems with making outgoing calls from the directory, it's just annoying to see it pop up. As for the date time settings... this is what we have in ours: CMLocal M/D/YA Central Standard/Daylight Time I'm guessing you should be able to change it to say Eastern instead of Central On Fri, 5 May 2006, Hall, Eric M. wrote: > > Group > I have a Cisco 7970 Running the newest SIP image. > I'm running Asterisk SVN-trunk-r7498 on 2006-04-30 15:11:39 UTC > > When I get a call the callerid number show something like > [EMAIL PROTECTED] I thought I seen somewhere what that was but I'm > unable to find the correct wording when searching Google to find that > post again. Can anyone help me out here. How can I remove the asterisk > servers IP from the phone number? > > > Also I'm unable to get the time zone correct on the phone. It is in > UTC and I'm in EST I see in the file where it looks like it goes but > what I have tried has not worked as of yet. Here is what it looks like > > > M/D/Y > EST > > > > Thanks again for all your help!!! > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth 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: Spam? Re: [Asterisk-Users] Cisco 7970 running SIP question
The A at the end of the dateTemplate sets that. Should read M/D/YA instead of M/D/Y. Aaron On Fri, 5 May 2006, Hall, Eric M. wrote: Aaron Any idea how to change it from 24hr to 12hr ? Thanks again! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hall, Eric M. Sent: Friday, May 05, 2006 11:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: Spam? Re: [Asterisk-Users] Cisco 7970 running SIP question Aaron Yes it is very annoying! Thanks for the date time settings. That worked GREAT!!! Thanks - Eric -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Aaron Daniel Sent: Friday, May 05, 2006 11:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Spam? Re: [Asterisk-Users] Cisco 7970 running SIP question I don't remember exactly what the reasoning on Cisco's part is of having the IP address on there, but it happens on ours too. It shouldn't cause any problems with making outgoing calls from the directory, it's just annoying to see it pop up. As for the date time settings... this is what we have in ours: CMLocal M/D/YA Central Standard/Daylight Time I'm guessing you should be able to change it to say Eastern instead of Central On Fri, 5 May 2006, Hall, Eric M. wrote: Group I have a Cisco 7970 Running the newest SIP image. I'm running Asterisk SVN-trunk-r7498 on 2006-04-30 15:11:39 UTC When I get a call the callerid number show something like [EMAIL PROTECTED] I thought I seen somewhere what that was but I'm unable to find the correct wording when searching Google to find that post again. Can anyone help me out here. How can I remove the asterisk servers IP from the phone number? Also I'm unable to get the time zone correct on the phone. It is in UTC and I'm in EST I see in the file where it looks like it goes but what I have tried has not worked as of yet. Here is what it looks like M/D/Y EST Thanks again for all your help!!! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth 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 -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FW: NuFone Update: DIDs
I didn't get that message, but I got a "we're sorry for sending out the bogus messages" message, saying that it was an error.. -Steve -Original Message- From: Steve Prior [mailto:[EMAIL PROTECTED] Sent: Thursday, May 04, 2006 6:40 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] FW: NuFone Update: DIDs Rich Adamson wrote: > > At least outbound calls still work, even though they changed IP > addresses (and probably colo locations). > > Maybe not so much now. I just got a disconnect notice from nufone which states that I have a positive balance in my account, but still need to add money to bring it up above zero for my account to be re-enabled. Somebody just broke the billing code I think... Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] why a perfectly fine iax2 host becomes UNREACHABLE?
Just as another datapoint, I have a cheap (~$17) walmart cordless phone at home hooked to my digium dual FXS card, and it works great, with the possible exception that there is a buzz (I perceive it as ground hum) for about the first 4 seconds of EVERY call, and slowly it diminishes. I don't know if there's a component of the echo cancellation that is fixing it, or if it's some capacitor draining that fixes it, but even my toughest asterisk critic (my wife) hasn't complained a bit about it. :-) -Original Message- From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED] Sent: Friday, May 05, 2006 8:47 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] why a perfectly fine iax2 host becomes UNREACHABLE? I don't have any trouble with my cordless phone to a TDM400, but there could be a wealth of factors playing into the chop you're seeing, as I'm sure you're aware. I had a *really* bad digital cordless phone that I actually sent to Digium to help them figure out WHY this particular phone hated their FXS ports so badly, but I don't think I'd ever heard back from them. I've never had another phone act that way though. (Panasonic digital cordless, I have another one from Panasonic and it works perfectly fine.) -A. > ___ > --Bandwidth 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] AASTRA 9133i and PIX Firewall
Hi, Does anyone have experience with the Aastra 9133i and a PIX firewall? My problem is that it passes data with no problem at all (that being audio goes fine)... however my voicemail notifications do not work. If I take the phone OUT from behind the PIX firewall, voicemail notifications work. What do I need to do (on phone, firewall, or asterisk) to get the msgwaiting indicator to work? ___ --Bandwidth 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] Registering Remote Sipura to Asterisk (both behind firewall)
Hi, you have to forward port 5060 udp and all the others udp ports that asterisk uses for RTP, 1 to 2 by default, see rtp.conf. At the sipura device in the server configuration you have to put the LAN ip of your Asterisk box as sip proxy server and firewall's ip as out-bound sip proxy. Joseph wrote: Can anybody point or provide working configuration how to register Sipura to Asterisk over the Internet. Both Sipura and Asterisk are behind firewalls. I'll be force to use SIP as that is the only protocol that Sipura is using. Do I need to enter any "STUN Server:" setting in "SIP" tab. On Asterisk I think I only need to make changes in sip.conf isn't it? What ports do I need to open on my firewall? ___ --Bandwidth 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] Registering Remote Sipura to Asterisk (both behind firewall)
On Fri, 2006-05-05 at 15:37 -0300, Lucas Alvarez wrote: > Hi, you have to forward port 5060 udp and all the others udp ports that > asterisk uses for RTP, 1 to 2 by default, see rtp.conf. Thank you for the hint. The above part is clear. > At the sipura device in the server configuration you have to put the LAN > ip of your Asterisk box as sip proxy server and firewall's ip as > out-bound sip proxy. Where is the "server configuration" on the Sipura box, is it under "SIP" tab? Note: Sipura is running as a stand alone device behind NAT so there is no Asterisk. If I had an asterisk on the other end I would use IAX2. Just to be clear I will be connecting: Asterisk <--> NAT <--> Internet <--> NAT <--> Sipura-3K (No Asterisk) I'm force to to this way as FWD IAX is not reliable. -- #Joseph > > Joseph wrote: > > >Can anybody point or provide working configuration how to register > >Sipura to Asterisk over the Internet. Both Sipura and Asterisk are > >behind firewalls. > > > >I'll be force to use SIP as that is the only protocol that Sipura is > >using. > >Do I need to enter any "STUN Server:" setting in "SIP" tab. > > > >On Asterisk I think I only need to make changes in sip.conf isn't it? > >What ports do I need to open on my firewall? ___ --Bandwidth 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 Center Phone with Auto Answer
Can anyone recommend a phone to use in an inbound call center environment that has an auto answer feature? We don’t want the agents having to acknowledge the call. The call should just activate on the headphones. We have tried Grandstream 2000, Polycom 301, 501 and 601. None of these support it. Thanks _ Kevin Savoy Business Unit Telecom Analyst 2218 4th Ave W Williston, ND 58801 Ph: 701-774-4023 Fax: 701-774-2901 http://www.novo1.com Novo 1 is a service mark of Novo 1, Inc ___ --Bandwidth 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] Passing SIP Subscriptions???
Can anyone tell me if they know if it's possible to pass/copy sip subscriptions from one Asterisk system to another? Can IAX do this? What about regexten? Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Call Center Phone with Auto Answer
The Snom 360 spec says it has an auto-answer feature. You can download the data sheet here http://www.snom.com/snom360_voip_phone.html Cory Andrews Executive Vice President ++ VoIPSupply.com PBXSelect.com ++ 454 Sonwil Drive Buffalo, NY 14225 voice - 800.398.VoIP X3402 fax - 716.630.1548 e - [EMAIL PROTECTED] m - 716.907.4059 aim - B2Cory From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin Savoy Sent: Friday, May 05, 2006 3:41 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Call Center Phone with Auto Answer Can anyone recommend a phone to use in an inbound call center environment that has an auto answer feature? We don’t want the agents having to acknowledge the call. The call should just activate on the headphones. We have tried Grandstream 2000, Polycom 301, 501 and 601. None of these support it. Thanks _ Kevin Savoy Business Unit Telecom Analyst 2218 4th Ave W Williston, ND 58801 Ph: 701-774-4023 Fax: 701-774-2901 http://www.novo1.com Novo 1 is a service mark of Novo 1, Inc ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Center Phone with Auto Answer
Kevin Savoy wrote: Can anyone recommend a phone to use in an inbound call center environment that has an auto answer feature? We don’t want the agents having to acknowledge the call. The call should just activate on the headphones. We have tried Grandstream 2000, Polycom 301, 501 and 601. None of these support it. My Polycom phones support auto-answer. This link should get you started. http://www.voip-info.org/wiki-Polycom+auto-answer+config 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
Re: [Asterisk-Users] Call Center Phone with Auto Answer
Kevin Savoy wrote: Can anyone recommend a phone to use in an inbound call center environment that has an auto answer feature? We don’t want the agents having to acknowledge the call. The call should just activate on the headphones. We have tried Grandstream 2000, Polycom 301, 501 and 601. None of these support it. Thanks May be I am not understanding... Why not use agentlogin and have the agents always logged in with MOH... they get a beep and they are connected.. Change ackcall=no in agents.conf Then you don't need auto-answer ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Call Center Phone with Auto Answer
The problem with what is in wiki is that these calls are being sent to a queue. There is no way to have the queue dial the preceding digit that I can think of that would trigger this. In the example shown he has an 8 dialed before the extension. How would I get Asterisk to dial an 8 before sending the call to the logged in agent in the queue? Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Clark Sent: Friday, May 05, 2006 2:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Call Center Phone with Auto Answer Kevin Savoy wrote: > Can anyone recommend a phone to use in an inbound call center > environment that has an auto answer feature? We don't want the agents > having to acknowledge the call. The call should just activate on the > headphones. We have tried Grandstream 2000, Polycom 301, 501 and 601. > None of these support it. > My Polycom phones support auto-answer. This link should get you started. http://www.voip-info.org/wiki-Polycom+auto-answer+config 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] Call Center Phone with Auto Answer
We are using agent login but we don't want MOH on the line at all times as some of these phones could and probably will be connected in remote locations. We don't want to stream MOH across frames chewing up bandwidth when there are no calls to present to that phone. We do have the ackcall=no in the agents.conf and it seems to have no affect. Am I missing something here? Appreciate any help -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sean Cook Sent: Friday, May 05, 2006 3:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Call Center Phone with Auto Answer Kevin Savoy wrote: > > Can anyone recommend a phone to use in an inbound call center > environment that has an auto answer feature? We don't want the agents > having to acknowledge the call. The call should just activate on the > headphones. We have tried Grandstream 2000, Polycom 301, 501 and 601. > None of these support it. > > Thanks > May be I am not understanding... Why not use agentlogin and have the agents always logged in with MOH... they get a beep and they are connected.. Change ackcall=no in agents.conf Then you don't need auto-answer ___ --Bandwidth 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] Cisco 7970 running SIP question
I don't remember exactly what the reasoning on Cisco's part is of having the IP address on there, but it happens on ours too. It shouldn't cause any problems with making outgoing calls from the directory, it's just annoying to see it pop up. It's so the phone routes the call to the correct server especially in a multiple server environment (ex: dialing a missed call) _ Mobilcom http://www.mobilcom.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Code parsing error?
This code executes just fine, and leaves the SIP peer's mailbox setting from sip.conf in variable target. exten => 1,1,Set(target=${CHANNEL:4}-) exten => 1,n,Set(target=${SIPPEER(${CUT(target,,1)}:mailbox}) exten => 1,n,VoiceMailMain(${target}) However, every time it runs I get an error in the CLI as follows WARNING[5629]: pbx.c:1366 ast_func_read: Can't find trailing parenthesis? This happens right after it executes the first line of code, then execution continues normally. I've looked at this until my eyes crossed at don't see any unbalanced parens or brackets. Perhaps I shouldn't worry since it seems to work, but what's going on here? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 300 DID's required in Alpine Texas Area code 432
Have customer who requires 300 DID'd in Alpine Tx 432 if the price is right. Email with details. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk <--> NAT <--> Internet <--> NAT <--> Sipura-3K (No Asterisk)
So far I have gathered that on my NAT (where asterisk server is) I have to port forward UDP ports: 5060 and range 1 - 2 to my asterisk server But I'm still stuck how to configure Sipura (behind NAT) what sip proxy and out-bound sip proxy, under which tab to change. -- #Joseph ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Code parsing error?
Did you mean to have the dash inside the braces -- this may be your problem. on Friday 05/05/2006 David L. West([EMAIL PROTECTED]) wrote > This code executes just fine, and leaves the SIP peer's mailbox setting from > sip.conf in variable target. > > exten => 1,1,Set(target=${CHANNEL:4}-) > exten => 1,n,Set(target=${SIPPEER(${CUT(target,,1)}:mailbox}) > exten => 1,n,VoiceMailMain(${target}) > > However, every time it runs I get an error in the CLI as follows > > WARNING[5629]: pbx.c:1366 ast_func_read: Can't find trailing parenthesis? > > This happens right after it executes the first line of code, then execution > continues normally. I've looked at this until my eyes crossed at don't see > any unbalanced parens or brackets. Perhaps I shouldn't worry since it seems > to work, but what's going on here? > > > > > > > > This code executes just fine, and > leaves > the SIP peer's mailbox setting from sip.conf in variable target. > > exten => > 1,1,Set(target=${CHANNEL:4}-) exten => > 1,n,Set(target=${SIPPEER(${CUT(target,,1)}:mailbox}) > exten > => 1,n,VoiceMailMain(${target}) > > However, every time it runs I get an > error > in the CLI as follows > > WARNING[5629]: > pbx.c:1366 ast_func_read: Can't find trailing parenthesis? > > This happens right after it > executes > the first line of code, then execution continues normally. > face=Arial size=2>I've looked at this until > my > eyes crossed at don't see any unbalanced parens or > brackets. Perhaps I shouldn't worry since it seems to work, but what's going > on > here? > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Bandwidth via my Asterisk PBX
Am new to Asterisk - have it up and running & connected to a couple service providers (telasip & teliax). Nice! Am running our Asterisk PBX on a static DSL connection (~600 kbps up/~4mbps down), and would like to extend VoIP service to 10 non-profits we're working with. Am I correct in assuming that all calls from each organization would route through our Asterisk server & be passed off to the service provider ( i.e. TelaSip, Teliax) thus keeping our bandwidth requirements to a minimum? Or - are ALL calls that route through our Asterisk PBX consuming our bandwidth through the duration of the call? Thanks, Dakota ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk behind load-balancing switch
Hi, I have 2 SER and 2 Asterisk boxes behind a load-balancing switch. I need Asterisk to initiate the RTP streams to both endpoints. Can that be done? Right now, Asterisk does it part of the time, so I have to create NATs for the Asterisk boxes for the RTP streams to get in. Thanks. Jack ___ --Bandwidth 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] ODBC Voicemail storage and app_directory
I've checked the wiki, searched the mailing list, Mantis (bug tracker), and looked through the source code, but found not mention of this issue: If using ODBC Storage for voicemail messages, name-playback by app_directory breaks. app_directory has no ODBC Storage code and therefore only looks for the greet.wav file (recording of user's name) in the filesystem, whereas app_voicemail saves the recording in the database. (1.2.6 stable) I'm hopnig I missed something somewhere and someone will point me to a patch or the bug number in Mantis! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] why a perfectly fine iax2 host becomes UNREA CHABLE?
>toughest asterisk critic (my wife) yes, my wife is worth more in real-world "this sucks" feedback than an entire office. keeps me on my toes. ___ --Bandwidth 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] Passing Callerid
How do I setup a dial plan that will allow a customer to set their own callerid? No matter what I try, the username that I have for them in sip.conf is passed because I have no callerid setup for them. If I setup a callerid in sip.conf that is passed & I can change it and it works. That is well and fine but I want to send what the customer is passing as their callerid... what am I doing wrong? To eliminate the customer being wrong, I tried with a pair of my asterisk boxes and I have the same problem. I even try setcallerid(1234567890) and it still doesn't pass. Could it be because I am having them register? Thanks, bp ___ --Bandwidth 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] Running Asterisk as non-root
I saw where one should not run Asterisk as root: http://www.voip-info.org/wiki/index.php?page=Asterisk+non-root How important is this? Thanks, just curious whether it's worth the trouble. H ___ --Bandwidth 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] Running Asterisk as non-root
I saw where one should not run Asterisk as root: How important is this? It's probably not very important at the moment, however, it's not that hard to do either. I run Asterisk non-root and in a chrooted environment -- it keeps all necessary files nicely separated (easily portable, easy to switch versions), doesn't clog up common directories. Just make a new directory like /usr/local/asterisk and use that as the root for the chrooted environment. Chown all /var and /etc/asterisk files in there to the asterisk user and you're good to go. The tough part is to get all the shared libraries copies over -- ldd is your friend. --Luki ___ --Bandwidth 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] Running Asterisk as non-root
If you run Asterisk as non root, you may have problems installing G729 licenses. The digium registration utility has certain hard coded stuff, and doesn't behave well when things aren't installed in the standard location. Doug. > -Original Message- > From: Luki [mailto:[EMAIL PROTECTED] > Sent: Friday, May 05, 2006 4:53 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Running Asterisk as non-root > > > > I saw where one should not run Asterisk as root: > > How important is this? > > It's probably not very important at the moment, however, it's not that > hard to do either. I run Asterisk non-root and in a chrooted > environment -- it keeps all necessary files nicely separated (easily > portable, easy to switch versions), doesn't clog up common > directories. Just make a new directory like /usr/local/asterisk and > use that as the root for the chrooted environment. Chown all /var and > /etc/asterisk files in there to the asterisk user and you're good to > go. The tough part is to get all the shared libraries copies over -- > ldd is your friend. > > --Luki > ___ > --Bandwidth 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] Re: Running Asterisk as non-root
Thanks for the swift response Luki. Have you checked out: http://www.voip-info.org/wiki/index.php?page=Asterisk+non-root recently? Does it seem up to date to you? It indicates it was updated in March of this tear, but I did this once and don't want to go thru all the hassle if the directions are outdated!! Cheers, Howard On 5/5/06, Luki <[EMAIL PROTECTED]> wrote: > I saw where one should not run Asterisk as root: > How important is this? It's probably not very important at the moment, however, it's not that hard to do either. I run Asterisk non-root and in a chrooted environment -- it keeps all necessary files nicely separated (easily portable, easy to switch versions), doesn't clog up common directories. Just make a new directory like /usr/local/asterisk and use that as the root for the chrooted environment. Chown all /var and /etc/asterisk files in there to the asterisk user and you're good to go. The tough part is to get all the shared libraries copies over -- ldd is your friend. --Luki ___ --Bandwidth 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] Re: Running Asterisk as non-root
Oh boy - thanks. I use g729... H On 5/5/06, Douglas Garstang <[EMAIL PROTECTED]> wrote: If you run Asterisk as non root, you may have problems installing G729 licenses. The digium registration utility has certain hard coded stuff, and doesn't behave well when things aren't installed in the standard location. Doug. > -Original Message- > From: Luki [mailto:[EMAIL PROTECTED] > Sent: Friday, May 05, 2006 4:53 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Running Asterisk as non-root > > > > I saw where one should not run Asterisk as root: > > How important is this? > > It's probably not very important at the moment, however, it's not that > hard to do either. I run Asterisk non-root and in a chrooted > environment -- it keeps all necessary files nicely separated (easily > portable, easy to switch versions), doesn't clog up common > directories. Just make a new directory like /usr/local/asterisk and > use that as the root for the chrooted environment. Chown all /var and > /etc/asterisk files in there to the asterisk user and you're good to > go. The tough part is to get all the shared libraries copies over -- > ldd is your friend. > > --Luki > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] REGISTER that isn't a register
Anyone come across this message: WARNING[2203]: chan_sip.c:9633 handle_response_register: Got 200 OK on REGISTER that isn't a register I get it from time to time but don't know what to do about it! Thanks, h ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ASTERISK DISA FOR INCOMING DID CALL
Hi, I am trying to create a situation where I call the DID number which is 1140636249 and I receive a dial tone to dial. I d like also to autenticate the number 1130851536. I can see that asterisk decode this number but I dont know how to authenticate this number only. This is what I am doing Sip.conf [globo] type=friend username=itn111 fromuser=itn111 secret=123456 insecure=very host= globo.net.br context=fromttt fromdomain= globo.net.br dtmfmode=rfc2833 disallow=all allow=g729 register => itn111:[EMAIL PROTECTED]:5060/itn111 where itn111 is the LOGIN for DID and the virtual extension for Extensions.conf file Extensions.conf [fromttt] exten => itn111,1,Dial(SIP/29650,60,Ttr) exten => itn111,2,Hangup() This settings above can can garantee that every call to 1140636249 goes to extension 29650. Do DID part is working ok. Now I would like to get a second dial tone when I call 1140636249 for asterisk DISA. I d like also to autenticate the number 1130851536 (caller number) and only this number can receive the call This is what I am trying to do exten => itn111,1,Dial(SIP/29650,60,Ttr) exten => 29650,2,DISA(no-password|brasil) ; I use no-password for this for now and Brasil context exten => 29650,3,Hangup() or exten => itn111,1,DISA(no-password|brasil) In sip show channels I see SIP/itn123456-cdfe (fromgvt itn123456 1 ) Up DISA no-password|brasil But there s no dial tone. And I don t know how to authenticate this number 1130851536. I see that asterisk collect this number Can you pls help me to do this settings ? Atenciosamente Diretoria Comercial - Newton Medina PABX 11.3085.1536 MSN [EMAIL PROTECTED] Rua Augusta 2.212 SL 26 Jardins 01412001 São Paulo - Brasil ___ --Bandwidth 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] Code parsing error?
At 02:06 PM 5/5/2006, you wrote: exten => 1,1,Set(target=${CHANNEL:4}-) However, every time it runs I get an error in the CLI as follows WARNING[5629]: pbx.c:1366 ast_func_read: Can't find trailing parenthesis? This happens right after it executes the first line of code, then execution continues normally. I've looked at this until my eyes crossed at don't see any unbalanced parens or brackets. Perhaps I shouldn't worry since it seems to work, but what's going on here? I guess I'd have to ask, what's in CHANNEL? Ira ___ --Bandwidth 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: Running Asterisk as non-root
> If you run Asterisk as non root, you may have problems installing G729 > licenses. The digium registration utility has certain hard coded stuff, and > doesn't behave well when things aren't installed in the standard location. Good point. However, in the chrooted environment there is no need to make any changes to any paths or recompile asterisk. Asterisk thinks it's dealing with /var/lib/whatever while in reality it's accessing /usr/local/asterisk/var/lib/whatever. While I do not use g929, I don't think you would have a problem with the license install as long as you run the installation in the chrooted environment as well: chroot /usr/local/asterisk license-installation-script I don't have time to write up the steps to chroot asterisk, but if anyone is interested then I will tonight. --Luki ___ --Bandwidth 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] Code parsing error?
On Fri, May 05, 2006 at 03:06:43PM -0600, David L. West spake thusly: > This code executes just fine, and leaves the SIP peer's mailbox setting from > sip.conf in variable target. > > exten => 1,1,Set(target=${CHANNEL:4}-) > exten => 1,n,Set(target=${SIPPEER(${CUT(target,,1)}:mailbox}) ^^ ^ " " There's definitely a ) missing in this line! A good text editor with bracket matching (I'm using vim now) makes it a lot easier to find this sort of thing. -- Jon-o Addleman - http://redowl.dyndns.org ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Call Center Phone with Auto Answer
I don't see any reason you can't use a polycom. You should be able to solve your problem multiple ways. You can simply put the default ring on the Polycom to autoanswer if that is the sole purpose, give it a second extension to be used in the queue that is programmed to autoanswer, as a couple of examples, or design your dialplan such that the appropriate _ALERT_INFO variable is set where the queue is concerned. p From: "Kevin Savoy" <[EMAIL PROTECTED]>To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"Date: Fri, 5 May 2006 15:31:41 -0500Subject: RE: [Asterisk-Users] Call Center Phone with Auto AnswerThe problem with what is in wiki is that these calls are being sent to aqueue. There is no way to have the queue dial the preceding digit that I canthink of that would trigger this. In the example shown he has an 8 dialedbefore the extension. How would I get Asterisk to dial an 8 before sendingthe call to the logged in agent in the queue?Thanks-Original Message-From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED] On Behalf Of Mike ClarkSent: Friday, May 05, 2006 2:55 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Call Center Phone with Auto AnswerKevin Savoy wrote:> Can anyone recommend a phone to use in an inbound call center > environment that has an auto answer feature? We don't want the agents > having to acknowledge the call. The call should just activate on the > headphones. We have tried Grandstream 2000, Polycom 301, 501 and 601. > None of these support it.>My Polycom phones support auto-answer. This link should get you started.http://www.voip-info.org/wiki-Polycom+auto-answer+configMike___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users Get amazing travel prices for air and hotel in one click on Yahoo! FareChase ___ --Bandwidth 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] Silent Attendant
I'd like to set up a "silent attendant". By this I mean when someone calls me I'd like for them to hear the comfort ringing tones, but for the first 5 seconds I'd like to give them the option of pressing 9 to send the call to an alternate extension; if they don't press 9, the call goes to a default extension. For most callers I just want standard PSTN behaviour, only a select few will be told about the silent 9 option I'm stuck tho because it seems I need the Background command for this to work, but this stops the comfort tones. Any ideas? Here's my attendant script, as you can see I currently have Background commented out so the silent option is non functional. Thanks, H exten => s,1,NoOp(${CONTEXT}) exten => s,n,Set(TIMEOUT(digit)=5) exten => s,n,Set(TIMEOUT(response)=5) ;Menu prompt disabled for now. Just let ringing continue for caller and have it pass thru ;to General Delivery after timeout. ;exten => s,n,Background(AttendantMainMenu) ;exten => s,n,Background(silence/5) ; exten => 9,1,Dial(Local/[EMAIL PROTECTED]) ; exten => ${EXT_CONFERENCEROOM},1,GoTo(conference-call-menu,s,1) ; ; If they take too long, send to general delivery exten => t,1,Dial(Local/[EMAIL PROTECTED]) ; ; "That's not valid, try again" exten => i,1,Playback(invalid) exten => i,2,Goto(Attendant-mainmenu,s,1) ; exten => h,1,Hangup ___ --Bandwidth 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] Info
Hi all, anyone could pls explain me what does it means ? It a part of zaptel.conf file. LBO= Line Build Out 0: 0 dB (CSU) / 0 - 133 feet (DSX-1) 1: 133 - 266 feet (DSX-1) 2: 266 - 299 feet (DSX-1) 3: 399 - 533 feet (DSX-1) 4: 533 - 655 feet (DSX-1) 5: -7.5 dB (CSU) 6: -15 dB (CSU) 7: -22.5 dB (CSU) Thanks Giordano -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.392 / Virus Database: 268.5.5/333 - Release Date: 05/05/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to determine if a device is in a call
I have gotten intercom working on my office phones (Linksys SPA-942s), but I have noticed that if someone is in a call, it places the call on hold and sends the intercom audio to the person holding the phone that is being paged. I'd like to add logic to my dialplan that doesn't send a page to a phone that is currently in a call. But to do this I need a function that will tell me if a device is in a call. Any suggestions? Thanks, Carl ___ --Bandwidth 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] CW options not changing
I have a weird issue with a system, running freepbx in devices and users mode (same as dozens of systems) but this one when you hit *70 and get "call waiting activated" it is not storing the setting in the database. I can manually set CW 900 ENABLED but then *71 does not disable it. Any suggestions? Kerry GarrisonDirector of Technical ServicesTech Data Pros - Orange County's Mobile IT Service Provider(949) 502-7819 x200 - [EMAIL PROTECTED]http://www.techdatapros.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Running Asterisk as non-root
At 04:22 PM 5/5/2006, you wrote: I don't have time to write up the steps to chroot asterisk, but if anyone is interested then I will tonight. Personally, I'd be very interested. Thanks so much, Ira ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Code parsing error?
> There's definitely a ) missing in this line! A good text editor with Thanks( I finally spotted it after fortifying myself with a good Thai dinner). I've been using gEdit, so should probably start shopping for something vim-ish. ___ --Bandwidth 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 determine if a device is in a call
Check in voip-info.org about sip.conf parameter "call-limit". It may help you. Regards On 5/5/06, Carl Youngblood <[EMAIL PROTECTED]> wrote: I have gotten intercom working on my office phones (Linksys SPA-942s), but I have noticed that if someone is in a call, it places the call on hold and sends the intercom audio to the person holding the phone that is being paged. I'd like to add logic to my dialplan that doesn't send a page to a phone that is currently in a call. But to do this I need a function that will tell me if a device is in a call. Any suggestions? Thanks, Carl ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- "Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org"; ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How to determine if a device is in a call
Show application chanisavail Should be what you need Wes Baehr Ability Business Computing, Ltd. Office: 330.882.0455 x25 Cell: 330.882.0455 x35 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carl Youngblood Sent: Friday, May 05, 2006 8:35 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] How to determine if a device is in a call I have gotten intercom working on my office phones (Linksys SPA-942s), but I have noticed that if someone is in a call, it places the call on hold and sends the intercom audio to the person holding the phone that is being paged. I'd like to add logic to my dialplan that doesn't send a page to a phone that is currently in a call. But to do this I need a function that will tell me if a device is in a call. Any suggestions? Thanks, Carl ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.392 / Virus Database: 268.5.4/332 - Release Date: 5/4/2006 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.392 / Virus Database: 268.5.4/332 - Release Date: 5/4/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Multiple periodic announcements in queues? Possible?
Hi all! Curious if there was a way to introduce multiple announcments in the queues. Rather than just: "Thank you for holding. Your call is important to us" ...wait 60 sec... [repeat] Is it possible to: "Thank you for holding. Your call is important." ...wait 60 sec... "Did you know you can.." ...wait 60 sec... "We are currently experiencing high call volumes." ...wait 60 sec... [repeat] Maybe a: Periodic-announce = AGI(rotate-msg) Periodic-announce-frequency = 60 Thanks for any advice!! ~~Aaron ___ --Bandwidth 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] Silent Attendant
Simply generate or record a 5-second sample of ringing. Then use Background() to play that ringing file - if someone presses 9, they will be routed accordingly, or otherwise sent to your default extension. Wes Baehr Ability Business Computing, Ltd. Office: 330.882.0455 x25 Cell: 330.882.0455 x35 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of hugolivude Sent: Friday, May 05, 2006 7:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Silent Attendant I'd like to set up a "silent attendant". By this I mean when someone calls me I'd like for them to hear the comfort ringing tones, but for the first 5 seconds I'd like to give them the option of pressing 9 to send the call to an alternate extension; if they don't press 9, the call goes to a default extension. For most callers I just want standard PSTN behaviour, only a select few will be told about the silent 9 option I'm stuck tho because it seems I need the Background command for this to work, but this stops the comfort tones. Any ideas? Here's my attendant script, as you can see I currently have Background commented out so the silent option is non functional. Thanks, H exten => s,1,NoOp(${CONTEXT}) exten => s,n,Set(TIMEOUT(digit)=5) exten => s,n,Set(TIMEOUT(response)=5) ;Menu prompt disabled for now. Just let ringing continue for caller and have it pass thru ;to General Delivery after timeout. ;exten => s,n,Background(AttendantMainMenu) ;exten => s,n,Background(silence/5) ; exten => 9,1,Dial(Local/[EMAIL PROTECTED]) ; exten => ${EXT_CONFERENCEROOM},1,GoTo(conference-call-menu,s,1) ; ; If they take too long, send to general delivery exten => t,1,Dial(Local/[EMAIL PROTECTED]) ; ; "That's not valid, try again" exten => i,1,Playback(invalid) exten => i,2,Goto(Attendant-mainmenu,s,1) ; exten => h,1,Hangup ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.392 / Virus Database: 268.5.4/332 - Release Date: 5/4/2006 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.392 / Virus Database: 268.5.4/332 - Release Date: 5/4/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Code parsing error?
Oh, on the off chance that it's interesting to anybody what I'm actually doing here. I have two SIP devices, "dave" and "dave-laptop". I want to have one voicemail box but be able to check it from either one. In sip.conf, both devices have "mailbox=dave". It's this setting that I'm retrieving with the code below and passing into VoiceMailMain. "David L. West" <[EMAIL PROTECTED]> wrote in message news:[EMAIL PROTECTED]... This code executes just fine, and leaves the SIP peer's mailbox setting from sip.conf in variable target. exten => 1,1,Set(target=${CHANNEL:4}-) exten => 1,n,Set(target=${SIPPEER(${CUT(target,,1)}:mailbox}) exten => 1,n,VoiceMailMain(${target}) However, every time it runs I get an error in the CLI as follows WARNING[5629]: pbx.c:1366 ast_func_read: Can't find trailing parenthesis? This happens right after it executes the first line of code, then execution continues normally. I've looked at this until my eyes crossed at don't see any unbalanced parens or brackets. Perhaps I shouldn't worry since it seems to work, but what's going on here? ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Silent Attendant
I changed my mind. This worked for me, using AEL: Context test { 825 => { Set(TIMEOUT(response)=5); Answer(); Ringing(); WaitExten(5); Goto(menu-main|s|1); }; }; Or in regular format: exten => 825,1,Noop exten => 825,n,Set(TIMEOUT(response)=5) exten => 825,n,Answer(); exten => 825,n,Ringing(); exten => 825,n,WaitExten(5); exten => 825,n,Dowhatever Wes Baehr Ability Business Computing, Ltd. Office: 330.882.0455 x25 Cell: 330.882.0455 x35 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of hugolivude Sent: Friday, May 05, 2006 7:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Silent Attendant I'd like to set up a "silent attendant". By this I mean when someone calls me I'd like for them to hear the comfort ringing tones, but for the first 5 seconds I'd like to give them the option of pressing 9 to send the call to an alternate extension; if they don't press 9, the call goes to a default extension. For most callers I just want standard PSTN behaviour, only a select few will be told about the silent 9 option I'm stuck tho because it seems I need the Background command for this to work, but this stops the comfort tones. Any ideas? Here's my attendant script, as you can see I currently have Background commented out so the silent option is non functional. Thanks, H exten => s,1,NoOp(${CONTEXT}) exten => s,n,Set(TIMEOUT(digit)=5) exten => s,n,Set(TIMEOUT(response)=5) ;Menu prompt disabled for now. Just let ringing continue for caller and have it pass thru ;to General Delivery after timeout. ;exten => s,n,Background(AttendantMainMenu) ;exten => s,n,Background(silence/5) ; exten => 9,1,Dial(Local/[EMAIL PROTECTED]) ; exten => ${EXT_CONFERENCEROOM},1,GoTo(conference-call-menu,s,1) ; ; If they take too long, send to general delivery exten => t,1,Dial(Local/[EMAIL PROTECTED]) ; ; "That's not valid, try again" exten => i,1,Playback(invalid) exten => i,2,Goto(Attendant-mainmenu,s,1) ; exten => h,1,Hangup ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.392 / Virus Database: 268.5.4/332 - Release Date: 5/4/2006 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.392 / Virus Database: 268.5.4/332 - Release Date: 5/4/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Code parsing error?
Title: Re: [Asterisk-Users] Re: Code parsing error? On 5/5/06 10:45 PM, "David L. West" wisely said: exten => 1,n,Set(target=${SIPPEER(${CUT(target,,1)}:mailbox}) Well: Exten => 1,n,Set(target=${SIPPEER(${CUT(target,,1)}):mailbox}) It looks like you didn’t end the parenthesis for SIPPEER(). Maybe? ~~Aaron ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Re: Code parsing error?
Title: Re: [Asterisk-Users] Re: Code parsing error? >It looks like you didn¹t end the parenthesis for SIPPEER(). Maybe? Yup, edited to look like this and it works w/o generating an error: ; Send to the user's vmail box by looking up the "mailbox" setting in sip.conf exten => 1,1,Set(target=${CHANNEL:4}-) exten => 1,n,Set(target=${CUT(target,,1)}:mailbox}) exten => 1,n,VoiceMailMain(${target}) I'm doing some tricky things in the dialplan and sip.conf so that I can have sip devices "dave" and "dave-laptop" point to the same voicemail box. Also rigging the "local" dialplan context to force calls to "dave" or "dave-laptop" to go to "dave", so that if dave-laptop calls somebody who then returns the call after dave-laptop has signed off it actually gets routed back to "dave". The idea is to abstract devices and users as separate layers rather than rely on the assumption that one user = one device. "A.J. Paxson" <[EMAIL PROTECTED]> wrote in message news:[EMAIL PROTECTED]...On 5/5/06 10:45 PM, "David L. West" wisely said: exten => 1,n,Set(target=${SIPPEER(${CUT(target,,1)}:mailbox})Well:Exten => 1,n,Set(target=${SIPPEER(${CUT(target,,1)}):mailbox})It looks like you didnt end the parenthesis for SIPPEER(). Maybe?~~Aaron ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Silent Attendant
Wes Baehr wrote: Simply generate or record a 5-second sample of ringing. Then use Background() to play that ringing file - if someone presses 9, they will be routed accordingly, or otherwise sent to your default extension. Or use the Ringing app, along with the WaitExten app. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users