RE: [asterisk-users] Provisioning Linksys PAP2T ATA's
On Thu, 7 Jun 2007, Stewart Nelson wrote: You can reload via http using a command like: wget\ --output-document=/dev/null\ --quiet\ http://ip-address-of-pap/upgrade?http://ip-address-of-web- server:80/asterisk/spa000F66A83C90.cfg I tried it with my xml file and it complains about the file being corrupt. I'm guessing you need Sipura's configuration compiler. I managed to talk their support people out of the compiler for the spa3k several years ago. Maybe you can pratice your SE skills. I believe that you should use the 'resync' keyword instead of 'upgrade'; the latter is intended to specify the URL of a new firmware image. I'm guessing that in some cases it looks at the file contents to decide whether it's configuration data or firmware, so it works anyway. See http://www.sipura.com/Documents/faq/Section_2.html#11 Well, resync was a step closer. Checking my notes from when I was playing with it a year ago... You need /admin/ in the path as well. This works: wget\ --output-document=/dev/null\ --quiet\ http://pappy/admin/resync?http://tftp/asterisk/spa000F66A83C90.xml; Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --Bandwidth 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 MS RTC Library Ethernet Capacity
Hi guys, I was wondering whether there's anyone who could share his/her experiences with using Microsoft RTC Library. In particular I am wondering what Ethernet capacity should I have in scenario of 30 people using Microsoft RTC Library for SIP communication (PBX is obviously Asterisk :-) ) concurrently (alaw codec being used)? What problems can be expected in such scenario? Would a good 1 gBit switch be enough to handle that (Asterisk box would be connected to that switch with 1 gBit connection, and computers with Microsoft RTC Library would be connected with a 100 mBit connection)? Thanx! Alex ___ --Bandwidth 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] Bridged PRI calls - processor involvement?
The setup. Asterisk is on a 3G Zeon Dell 2850 running Fedora Core 5/6 (all yum updates applied), the TE410 lives on it's own interrupt. Asterisk sits between our telco and a PRI enabled PBX. These are the relevant versions installed: Linux: 2.6.20-1.2316.fc5smp Zaptel: 1:1.4.2.1-34.fc5 Asterisk: 1:1.4.0-34.fc5.at Libpri: 1:1.4.0-16.fc5.at Wildcard details: Found TE4XXP at base address fe3ffc00, remapped to f88bec00 TE4XXP version c01a016a, burst OFF, slip debug: OFF Octasic optimized! FALC version: 0005, Board ID: 00 Reg 0: 0x377bb400 Reg 1: 0x377bb000 Reg 2: 0x Reg 3: 0x Reg 4: 0x0001 Reg 5: 0x Reg 6: 0xc01a016a Reg 7: 0x1f00 Reg 8: 0x Reg 9: 0x00ff Reg 10: 0x004a TTE4XXP: Launching card: 0 TE4XXP: Setting up global serial parameters Found a Wildcard: Wildcard TE410P (3rd Gen) TE4XXP: Span 1 configured for CCS/HDB3/CRC4 TE4XXP: Span 2 configured for CCS/HDB3/CRC4 The problem: At random points during calls we lose 1-3 seconds of speech (both ways both callee and caller), this can be replicated (or at least a very good approximation!) by generating a high level of interrupt/cpu activity (for instance copying data from a USB caddy as we tried the other day in an attempt to reproduce this more reliably). The calls are bridged PRI:PRI calls, no VOIP involvement. This was not a problem until approx 3-4 weeks ago, but I can't tie it down to an exact date. Steve Interrupt sharing is not a problem anymore with those cards. What version of zaptel did you try installing? Can you explain more about your problems? Also, your configuration and setup would help out as well. --- Matthew Fredrickson Digium, Inc. The information contained in this email is intended for the personal and confidential use of the addressee only. It may also be privileged information. If you are not the intended recipient then you are hereby notified that you have received this document in error and that any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) Ltd Nevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UK Registered Office as above. Registered in England No. 2764339 See our current vacancies at www.brendata.co.uk ___ --Bandwidth 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] Provisioning Linksys PAP2T ATA's
Nick, Pretty much - it builds the XML output on the fly, and delivers it over HTTP (PHP/Apache). It works the same for all of the mainstream Linksys kit - including SPA phones. Generally, where we've installed IP phones, we've also installed an Asterisk appliance in the form of a Linux box, so those ones we provision via TFTP on-site. We've not yet had any problems with security. Please let's talk of that no further now :p We don't disable / lock anything. We don't want to become one of those providers. Our customers buy their CPE, and thus probably expect to completely own it. We have information on our web which shows the customer how to (reasonably) easily get the unit to start provisioning itself against our prov server again. Alternatively, we get them to establish it's LAN IP, then send them an email with a link to click which does this for them. Where available, the latest firmware is sent to the device as part of the pre-provisioning process. Differences between PAP2 and PAP2T are: - Plainer packaging - more OEM style - LED colour - No stand-up attachment (which could lead to overheats, given the vent placement) - PAP2T supports T.38 - PAP2T has more RAM - PAP2T is *supposed* to be able to handle 2 concurrent G729 calls - but it can't - Different chipset On Fri, 2007-06-08 at 00:34 -0400, Nick Seraphin wrote: On Fri, 8 Jun 2007, Mattt wrote: Doug, We just pre-provision the Linksys CPE (including PAP2(T)-NA's) in the lab over TFTP after barcode-scanning the relevant information for that unit into a web management interface and, once the unit is deployed onsite, it continues to pull it's config from our prov server over HTTP. The provisioning service itself is just a PHP engine which pulls the relevant settings for that CPE from a database - this way, we can have individual parameters for certain customers (who might be, for instance, having issues with echo or latency, etc, etc, or some need for a different config to our norm). Does your PHP engine just generate a plain text XML file to HTTP via stdout? Does it work the same for both PAP2's and the IP Phones (SPA942/SPA962, etc)? I've successfully provisioned an SPA942 via plain text XML from a tftp server, but I've never tried a PAP2 remotely, nor have I tried with HTTP. Do you find any problems with security with it being in plain text? Do you disable the restore to factory defaults thing? Do you upgrade to the latest firmware before shipping out your PAP2's? (more below) Works a treat, is easy as (once the coding is done), and takes about 10 seconds to pre-prov (also provides the opportunity to ensure the unit isn't a DOA). Customer simply receives the device and plugs it in, waits a few seconds (the pre-prov doesn't configure the unit, just prepares it for remote provisioning from the target site), then starts making calls. Oh - and, unless you can locate some new, old stock, you won't find the PAP2-NA (with the blue LEDs) anywhere. They were discontinued many months ago... What's the difference(s) between a PAP2 and a PAP2T? I've only got PAP2's, and I've got several spares in inventory that I'm hoping I won't regret having. :-) Thanks, -- Nick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Cheers, Mattt. - ROMATel - VoIP made easy - http://romatel.net - SpotSafe - WiFi Hotspot solution - http://spotsafe.net There are only 10 kinds of people. Those who understand binary, and those that don't... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Unexpected behaviour shown by meetme kick confno usernumber
Hi, I have Asterisk 1.4.4 on my linux box. Whenever i try to kick a participant in conference say 59681446 using following command meetme kick 59681446 1 where 1 is the participant number, following are the actions that asterisk takes * IVR You have been kicked from this conference is played. * Participant is taken out from that conference 59681446 * Plus the same participant is taken to a different conference with conf # h (always), as the partcipant's phone does not hang up. So, when the following command is issued meetme kick h 1 the participant's phone finally hangs up. I don't know why is this behaviour shown by the meetme module ? Could anybody help me in this regard ? Thanx, Raza ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk MS RTC Library Ethernet Capacity
On Fri, 8 Jun 2007, Asterisk wrote: Hi guys, I was wondering whether there's anyone who could share his/her experiences with using Microsoft RTC Library. In particular I am wondering what Ethernet capacity should I have in scenario of 30 people using Microsoft RTC Library for SIP communication (PBX is obviously Asterisk :-) ) concurrently (alaw codec being used)? What problems can be expected in such scenario? Would a good 1 gBit switch be enough to handle that (Asterisk box would be connected to that switch with 1 gBit connection, and computers with Microsoft RTC Library would be connected with a 100 mBit connection)? I don't know anything about MS RTC.. But if it's just SIP streams of alaw, then a rough guide is that you'll need 80Kbits/sec per stream. So 30 streams on a Gb network will barely be noticable over the usual background noise ... And each PC running a single stream will be just fine. The asterisk server will be seeing an aggregate bandwidth of (30*80*2)Kbits/sec, or 4.8Mbits/sec. Hardly anything to wory about. I have Linux routers based on 1Ghz processors that can sustain 20x that traffic, and a 1GHz processor running asterisk would easilly support double that number of connections, so any modern server type PC should be fine for your use. Gordon ___ --Bandwidth 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] Noise on FXS ports (Sangoma)
On Thu, Jun 07, 2007 at 04:50:55PM -0600, Stephen Bosch wrote: Tzafrir Cohen wrote: To generate a FXS dialtone without Asterisk, use fxstest (make fxstest) from the zaptel source directory. Can I break this dial tone with DTMF? No. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DUNDi and reinvites...
I'm talking out my rear so someone please apply an attitude adjustment if I'm way off base. But, if you are using Dundi as a lookup engine it should know the contact information both endpoints and how to reach them perhaps not ONLY knowing how to comunicate via another asterisk box. Much like simply initializing a base dns infrastructure for the CPE devices. If the CPE devices are configured to accept SIP transactions from $domain or both asterisk servers server A should be able to send a invite directly to client B and bring up the inbound call. As far as the client knows it's still talking and placing outbound calls with server B. IE: Client A calls Client B Client A hits Serv A. Serv A does lookup finds it knows about Client B Serv A sends the call direct to Client B's IP. I'm assuming that both servers are acting as mirrors of eachother, in that voicemail and all that is a //shared// resource.. so if Client B rings unavail/busy that your serv A knows what to do with the call. In general as long as a client device knows to understand and accept sip messages from $host an inbound call does not have to come from the server they registered to. If you look at a linksys adapter this is one of the reasons they have that domain parameter which controls the list of hosts that are allowed to send SIP transactions to the unit. Am I wrong on this? The only other artifact I can think of is the fact of NAT traversal, where if client B that's to recieve the call is behind a NAT firewall and you are not doing port forwarding of the SIP signaling then ofcourse it won't get the call because server A has not established the NAT association. But assuming you are using a common 'sbc' or gatekeeper (ser) that box would know the association and things would be happy. On Jun 7, 2007, at 7:11 PM, Jared Smith wrote: On 6/7/07, Douglas Garstang [EMAIL PROTECTED] wrote: That's all fine and good until it becomes the receiving phone, and the other phone (as an originator) also has canreinvite set to yes. Then, your back to both Asterisk servers being completely taken out of the loop again! While I haven't taken the time to actually try this, I might suggest that you could set up separate user and peer sections in sip.conf, so that you can handle inbound calls differently that outbound calls. -Jared ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- Bryan Laird, Sr. Manager CM Operations -+- Cablemodems are the gateway to the Internet. The Internet is a gateway to some things that are better left un-seen. ___ --Bandwidth 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] Noise on FXS ports (Sangoma)
In this troubleshooting case, it probably is better that there is NO dialtone, which would make the hiss easier to hear. I am curious what the OP found When Asterisk is stopped, does the hiss continue? That would help to narrow down the location of the problem It sure sounds to me as if it is a hardware problem within the card or even the module, rather than on the PCI bus or within Asterisk John Novack Tzafrir Cohen wrote: On Thu, Jun 07, 2007 at 04:50:55PM -0600, Stephen Bosch wrote: Tzafrir Cohen wrote: To generate a FXS dialtone without Asterisk, use fxstest (make fxstest) from the zaptel source directory. Can I break this dial tone with DTMF? No. ___ --Bandwidth 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] agi with java?
Hello Matthew, Java is not a great solution for AGIs because they are script you should fire up and terminate very fast, while the overhead of launching a JVM, loading all classes, etc, is pretty large. Also, you don't want multiple JVMs in parallel loading everything multiple times. This is not to say AGI is not feasible: you should look for FastAGI, where * connects to an external server with resident proceses, and that suits the Java model much better. Of course, if all you want to do is lookup the callerid on mysql or something just as trivial, go for a 20-line Perl script. Just my two eurocents, l. On Thu, 07 Jun 2007 23:32:12 +0200, Matthew Pease [EMAIL PROTECTED] wrote: Hi all - Searching for java agi in the mailing list archives turns up ancient posts. Anyone else using java for their AGI? How well is it working what are you using? My script is pretty simple, and I could write it with perl easy enough, but I just would feel better if I can keep most programming code for our system in java. Thank you- Matt -- Loway Research - Home of QueueMetrics http://queuemetrics.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] Noise on FXS ports (Sangoma)
John Novack wrote: In this troubleshooting case, it probably is better that there is NO dialtone, which would make the hiss easier to hear. I am curious what the OP found When Asterisk is stopped, does the hiss continue? That's tough to assess because the other problem I have had with it is that, often, after restarting wanrouter, I lose talk battery on one of the two FXS ports. If I restart it enough times, I can get it back; alternatively, if I place a call *to* the port, it will sometimes ring and then I have talk battery and dial tone. But I've never heard the noise *without* first hearing a dial tone. To solve that problem I've been using the beta driver from Sangoma (because that is what is recommended on the wiki) but I've been asked to revert to a stable driver. I haven't had a chance to try that yet. That would help to narrow down the location of the problem Does it matter, really? I mean -- whether or not Asterisk activates it, the dial tone is still generated by the card/module; I think that would be a red herring. It sure sounds to me as if it is a hardware problem within the card or even the module, rather than on the PCI bus or within Asterisk Well, I doubt it's the module, as we've already replaced it once. It might be the card. Sangoma is suggesting the noise is coming into the card from the mainboard. -Stephen- ___ --Bandwidth 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 problem...
=) in new stack -- Executing Set(SIP/4000-163c, RT=) in new stack -- Executing Macro(SIP/4000-163c, record-enable|2000|IN) in new stack -- Executing GotoIf(SIP/4000-163c, 0?2:4) in new stack -- Goto (macro-record-enable,s,4) -- Executing DeadAGI(SIP/4000-163c, recordingcheck|20070608-131412|1181308451.0) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck -- AGI Script recordingcheck completed, returning 0 -- Executing NoOp(SIP/4000-163c, No recording needed) in new stack -- Executing Macro(SIP/4000-163c, dial||tr|2000) in new stack -- Executing DeadAGI(SIP/4000-163c, dialparties.agi) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi -- AGI Script dialparties.agi completed, returning 0 -- Executing NoOp(SIP/4000-163c, Returned from dialparties with no extensions to call) in new stack -- Executing NoOp(SIP/4000-163c, DIALSTATUS is ) in new stack -- Executing Set(SIP/4000-163c, SV_DIALSTATUS=) in new stack -- Executing GosubIf(SIP/4000-163c, 0?docfu|1) in new stack -- Executing GosubIf(SIP/4000-163c, 0?docfb|1) in new stack -- Executing Set(SIP/4000-163c, DIALSTATUS=) in new stack -- Executing NoOp(SIP/4000-163c, Voicemail is novm) in new stack -- Executing GotoIf(SIP/4000-163c, 1?s-|1) in new stack -- Goto (macro-exten-vm,s-,1) -- Executing PlayTones(SIP/4000-163c, congestion) in new stack -- Executing Congestion(SIP/4000-163c, 10) in new stack == Spawn extension (macro-exten-vm, s-, 2) exited non-zero on 'SIP/4000-163c' in macro 'exten-vm' == Spawn extension (macro-exten-vm, s-, 2) exited non-zero on 'SIP/4000-163c' -- Executing Macro(SIP/4000-68ca, exten-vm|novm|2000) in new stack -- Executing Macro(SIP/4000-68ca, user-callerid) in new stack -- Executing NoOp(SIP/4000-68ca, user-callerid: device 4000) in new stack -- Executing GotoIf(SIP/4000-68ca, 0?report) in new stack -- Executing GotoIf(SIP/4000-68ca, 0?start) in new stack -- Executing Set(SIP/4000-68ca, REALCALLERIDNUM=4000) in new stack -- Executing NoOp(SIP/4000-68ca, REALCALLERIDNUM is 4000) in new stack -- Executing Set(SIP/4000-68ca, AMPUSER=4000) in new stack -- Executing Set(SIP/4000-68ca, AMPUSERCIDNAME=outro2) in new stack -- Executing GotoIf(SIP/4000-68ca, 0?report) in new stack -- Executing Set(SIP/4000-68ca, CALLERID(all)=outro2 4000) in new stack -- Executing Set(SIP/4000-68ca, REALCALLERIDNUM=4000) in new stack -- Executing NoOp(SIP/4000-68ca, TTL: ARG1: novm) in new stack -- Executing GotoIf(SIP/4000-68ca, 0?continue) in new stack -- Executing Set(SIP/4000-68ca, __TTL=64) in new stack -- Executing GotoIf(SIP/4000-68ca, 1?continue) in new stack -- Goto (macro-user-callerid,s,21) -- Executing NoOp(SIP/4000-68ca, Using CallerID outro2 4000) in new stack -- Executing Set(SIP/4000-68ca, FROMCONTEXT=exten-vm) in new stack -- Executing Set(SIP/4000-68ca, VMBOX=novm) in new stack -- Executing Set(SIP/4000-68ca, EXTTOCALL=2000) in new stack -- Executing Set(SIP/4000-68ca, CFUEXT=) in new stack -- Executing Set(SIP/4000-68ca, CFBEXT=) in new stack -- Executing Set(SIP/4000-68ca, RT=) in new stack -- Executing Macro(SIP/4000-68ca, record-enable|2000|IN) in new stack -- Executing GotoIf(SIP/4000-68ca, 0?2:4) in new stack -- Goto (macro-record-enable,s,4) -- Executing DeadAGI(SIP/4000-68ca, recordingcheck|20070608-131415|1181308455.1) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck -- AGI Script recordingcheck completed, returning 0 -- Executing NoOp(SIP/4000-68ca, No recording needed) in new stack -- Executing Macro(SIP/4000-68ca, dial||tr|2000) in new stack -- Executing DeadAGI(SIP/4000-68ca, dialparties.agi) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi -- AGI Script dialparties.agi completed, returning 0 -- Executing NoOp(SIP/4000-68ca, Returned from dialparties with no extensions to call) in new stack -- Executing NoOp(SIP/4000-68ca, DIALSTATUS is ) in new stack -- Executing Set(SIP/4000-68ca, SV_DIALSTATUS=) in new stack -- Executing GosubIf(SIP/4000-68ca, 0?docfu|1) in new stack -- Executing GosubIf(SIP/4000-68ca, 0?docfb|1) in new stack -- Executing Set(SIP/4000-68ca, DIALSTATUS=) in new stack -- Executing NoOp(SIP/4000-68ca, Voicemail is novm) in new stack -- Executing GotoIf(SIP/4000-68ca, 1?s-|1) in new stack -- Goto (macro-exten-vm,s-,1) -- Executing PlayTones(SIP/4000-68ca, congestion) in new stack -- Executing Congestion(SIP/4000-68ca, 10) in new stack == Spawn extension (macro-exten-vm, s-, 2) exited non-zero on 'SIP/4000-68ca' in macro 'exten-vm' == Spawn extension (macro-exten-vm, s-, 2) exited non-zero on 'SIP/4000-68ca' hernandezz-laptop*CLI *** *** *** can you help me
Re: [asterisk-users] Bridged PRI calls - processor involvement?
Did it accompany an update you made? If you can find out what version the problem started occurring, that would help in fixing the problem. Matthew Fredrickson Software/Hardware Engineer Digium, Inc. On Jun 8, 2007, at 2:59 AM, Steve Hanselman wrote: The setup. Asterisk is on a 3G Zeon Dell 2850 running Fedora Core 5/6 (all yum updates applied), the TE410 lives on it's own interrupt. Asterisk sits between our telco and a PRI enabled PBX. These are the relevant versions installed: Linux: 2.6.20-1.2316.fc5smp Zaptel: 1:1.4.2.1-34.fc5 Asterisk: 1:1.4.0-34.fc5.at Libpri: 1:1.4.0-16.fc5.at Wildcard details: Found TE4XXP at base address fe3ffc00, remapped to f88bec00 TE4XXP version c01a016a, burst OFF, slip debug: OFF Octasic optimized! FALC version: 0005, Board ID: 00 Reg 0: 0x377bb400 Reg 1: 0x377bb000 Reg 2: 0x Reg 3: 0x Reg 4: 0x0001 Reg 5: 0x Reg 6: 0xc01a016a Reg 7: 0x1f00 Reg 8: 0x Reg 9: 0x00ff Reg 10: 0x004a TTE4XXP: Launching card: 0 TE4XXP: Setting up global serial parameters Found a Wildcard: Wildcard TE410P (3rd Gen) TE4XXP: Span 1 configured for CCS/HDB3/CRC4 TE4XXP: Span 2 configured for CCS/HDB3/CRC4 The problem: At random points during calls we lose 1-3 seconds of speech (both ways both callee and caller), this can be replicated (or at least a very good approximation!) by generating a high level of interrupt/cpu activity (for instance copying data from a USB caddy as we tried the other day in an attempt to reproduce this more reliably). The calls are bridged PRI:PRI calls, no VOIP involvement. This was not a problem until approx 3-4 weeks ago, but I can't tie it down to an exact date. Steve Interrupt sharing is not a problem anymore with those cards. What version of zaptel did you try installing? Can you explain more about your problems? Also, your configuration and setup would help out as well. --- Matthew Fredrickson Digium, Inc. The information contained in this email is intended for the personal and confidential use of the addressee only. It may also be privileged information. If you are not the intended recipient then you are hereby notified that you have received this document in error and that any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) Ltd Nevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UK Registered Office as above. Registered in England No. 2764339 See our current vacancies at www.brendata.co.uk ___ --Bandwidth 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] Changing the messages that are played when a user is unavailable/busy
Hi, I have my custom sounds which should be played instead of the default ones when a user is busy or unavailable: The person at extension XXX is not available right now, please Of course I can simply replace the files, but the problem is my implementation shouldn't (MUST NOT) mention the extension. My files say something like The person you are trying to reach is not available right now. If you want to contact this person on his cellphone, press 1, if you want to leave a voice message, press 2 or wait. As you can see the extension is not mentioned, so simply replacing the files would probably cause something weird. Where do I define what message are/aren't played in this case? Thank you. ___ --Bandwidth 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] getting at ${CALLERIDNUM}
On 6/8/07, Matthew Pease [EMAIL PROTECTED] wrote: when will it be out? Soon... it's going through the copyediting process right now. I can't give any more specific timeframe than that, as I don't know how long it'll take to get through the entire process, but if I had to make a wild guess I'd say probably somewhere around August or September but hopefully sooner. -Jared ___ --Bandwidth 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 problem...
On Fri, Jun 08, 2007 at 02:52:39PM +0100, Carlos Jerónimo wrote: Hi, i got Ubuntu 6.06 installed and theres a problem with asterisk. I've sucessfully installed it with the command: #apt-get install asterisk Then after installing FreePBX i get this error when restarting asterisk: [EMAIL PROTECTED]:/home/hernandezz# asterisk -rvv Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) After looking at the logs i noticed the problem could be the module format_mp3.so not being loaded because not exists in my PC. in modules.conf i comment the line load = format_mp3.so and now it's works.This module is necessary? It can be used for playing mp3 files. It is part of asterisk-addons. You may need to reinstall / upgrade asterisk-addons for the current version of Asterisk. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Q931 Error with H323
Dovid, Please provide a simple network diagram for members of this list. Q931 cause 44 error is a layer 3 ISDN error (Requested circuit/channel not available) most likely mapped backwards from PRI T1 interworking. John Treble Ottawa, Canada From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dovid B Sent: June 7, 2007 5:14 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Q931 Error with H323 Hi List, I am having issues sending calls to my carrier who is using a Nextone switch to handle the session and a Cisco box for the RTP stream. He said that he keeps seeing a Q931 cause 44 error which he said he never received before. All of his other clients are able to get through so its not something on his end. Does anyone know what could be causing this ? I am sending the call over G729 with faststart and h245tunneling enabled. Thanks a lot. Dovid ___ --Bandwidth 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] Not getting CID Name from PRI
Having a problem w/ not getting CID name from a PRI. CID Name appears in the PRI debug, but even after a Wait(4) it still appears after the phone is ringing. Here is the relevant info from my PRI debug output. Line 4 is a NoOp showing me trying to echo Name and Number. Line 6 dials the extension, and you can see callerid name get presented on line 29. Again, there is a Wait(4) before the NoOp on line 3. 1 -- Processing IE 30 (cs0, Progress Indicator) 2 -- Processing IE 108 (cs0, Calling Party Number) 3 -- Processing IE 112 (cs0, Called Party Number) 4 -- Executing NoOp(Zap/1-1, Name: Num: 9515551212) in new stack 5 -- Executing Macro(Zap/1-1, stdexten|6448|SIP/6448) in new stack 6 -- Executing Dial(Zap/1-1, SIP/6448|20) in new stack 7 -- Called 6448 8 -- SIP/6448-08945b40 is ringing 9 Protocol Discriminator: Q.931 (8) len=10 10 Call Ref: len= 2 (reference 195/0xC3) (Terminator) 11 Message type: CALL PROCEEDING (2) 12 [18 03 a9 83 81] 13 Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 14 ChanSel: Reserved 15 Ext: 1 Coding: 0 Number Specified Channel Type: 3 16 Ext: 1 Channel: 1 ] 17 Extension Changed 6448 new state Ringing for Notify User 7799 18 Extension Changed 6448 new state Ringing for Notify User 6452 19 Protocol Discriminator: Q.931 (8) len=9 20 Call Ref: len= 2 (reference 195/0xC3) (Terminator) 21 Message type: ALERTING (1) 22 [1e 02 81 88] 23 Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) 24 Ext: 1 Progress Description: Inband information or appropriate pattern now available. (8) ] 25 Protocol Discriminator: Q.931 (8) len=36 26 Call Ref: len= 2 (reference 195/0xC3) (Originator) 27 Message type: FACILITY (98) 28 [1c 1d 9f 8b 01 00 a1 17 02 01 01 02 01 00 80 0f 4c 45 57 4f 4e 2c 52 59 41 4e 20 20 20 20 20] 29 Facility (len=31, codeset=0) [ 0x9f, 0x8b, 0x01, 0x00, 0xa1, 0x17, 0x02, 0x01, 0x01, 0x02, 0x01, 0x00, 0x80, 0x0f, 'LASTNAME', 0x2c, 'FIRSTNAME', 0x20, 0x20, 0x20, 0x20, 0x20 ] 30 -- Processing IE 28 (cs0, Facility) 31 Handle Q.932 ROSE Invoke component 32 Protocol Discriminator: Q.931 (8) len=9 33 Call Ref: len= 2 (reference 195/0xC3) (Originator) 34 Message type: DISCONNECT (69) -- Kyle Sexton ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UPDATE... 2 down, 1 to go (3 questions - variables, upgrading, and IRC)
On Fri, 8 Jun 2007, Jared Smith wrote: On 6/7/07, Nick Seraphin [EMAIL PROTECTED] wrote: Still need an answer to this one. I wrote a response yesterday, but it looks like it didn't come through for some reason. The answer is to use the DumpChan() application and watch the CLI when it's called. I interpreted this question as how do I see the variables for this channel using the CLI? -- show channel foo. Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --Bandwidth 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] Hot GXP-2000
This is off topic for Asterisk but I need a suggestion. I have a customer (travel agency) that has recently begun complaining that their GXP-2000 phones are getting very hot, they say that around mid day the handset is so hot that it can burn your ear. These phones are in constant use and right now the weather in Mexico is hot. Anyone know why the handset would get so hot? Only the phones assigned to sales get this way and the others in the office do not have this problem. They are all connected to the same Linksys PoE switch for power. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ --Bandwidth 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] Changing the messages that are played when a user is unavailable/busy
Timothy Parez wrote: Hi, I have my custom sounds which should be played instead of the default ones when a user is busy or unavailable: The person at extension XXX is not available right now, please Of course I can simply replace the files, but the problem is my implementation shouldn't (MUST NOT) mention the extension. My files say something like The person you are trying to reach is not available right now. If you want to contact this person on his cellphone, press 1, if you want to leave a voice message, press 2 or wait. As you can see the extension is not mentioned, so simply replacing the files would probably cause something weird. Where do I define what message are/aren't played in this case? Log into your mailbox. Press 0, then press the option listed to record your unavail and busy greetings. ___ --Bandwidth 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] Best Codec
I know that g729 is the king-all, but I want to know what the rest of the professional are using out there. g729 has a cost involved, so does the cost really offset the performance? Or is it better to go with g711 to start off? I'm wary of using g711 of public broadband networks. Although theoretically the bandwidth should be there, many consumer ISPs appear to be rather overloaded at certain times of day, so that 256k theoretical (235k real-world) can easily drop to between 100 and 150kbps. Throw in even a small amount of light web browsing and you'll start to get g711 packet loss. g729 is good for speech but very poor for music on hold. I've found speex to be a reasonable compromise - voice quality is similar to g729 (subjectively it's very difficult to tell the difference), but MoH quality is noticeably better. Of course, if your endpoints are SIP hardphones or ATAs you may be forced to use g729 simply because so few hardware devices support speex. It'd be interesting to see some comparisons or comments from people using g726 as this does seem to be supported by quite a few hardware devices. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited For full contact details visit http://www.minotaur.it/chris.html This email is made from 100% recycled electrons ___ --Bandwidth 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] No/unknown event '0' on timer
Hey guys, I'm currently running Asterisk 1.2.18 Under Mandriva Linux. Three Facilities are hooked together via IAX2 (Trunked) over a OpenVPN connection on a 10mbit (uplink/downlink) internet connection. I was parked for around thirty seconds at a remote facility. All of a sudden, the call drops. The log entry was: Jun 8 11:34:14 NOTICE[10458]: channel.c:1918 ast_read: No/unknown event '0' on timer for 'IAX2/asterisk.bc-4'? Jun 8 11:34:14 NOTICE[10458]: chan_iax2.c:3167 iax2_read: I should never be called! Hanging up. -- Stopped music on hold on IAX2/asterisk.bc-4 Any ideas to what the cause? I found an entry that related the Festival on bugs.digium that didn't really relate to this. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation 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] Console duplicate output problem
Barton Fisher wrote: Anybody have an answer? TIA This is really strange. Every message to the (VGA) console is written twice to the screen, but not on the SSH connection. Any clues how to stop this behavior? -- Executing BackGround(Zap/216-1, custom/3566/91_|m|) in new stack -- Executing BackGround(Zap/216-1, custom/3566/91_|m|) in new stack Stop running in graphics mode. ___ --Bandwidth 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] Best Codec
Yes. In fact it's around 32kbps, for a high duration call. MRTG statistics. Are you using G729A ou B? (VAD can reduce the usage). Att, Ricardo Martins. Henry Cobb escreveu: On 6/7/07, Ricardo Martins [EMAIL PROTECTED] wrote: We use G.729. Consumes only 35kbps of bandwidth and has a level 4 (from 0 to 5) of voice quality. We still have very poor public data networks here in Brazil that makes G.711 a very high bandwith consunption codec for us. 35kbps sounds very large. We only use 20 kbps untrunked and 13-15 kbps when using IAX trunks. Have you verified this bandwidth usage? -HJC ___ --Bandwidth 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] agi with java?
Lenz wrote: Hello Matthew, Java is not a great solution for AGIs because they are script you should fire up and terminate very fast, while the overhead of launching a JVM, loading all classes, etc, is pretty large. Also, you don't want multiple JVMs in parallel loading everything multiple times. This is not to say AGI is not feasible: you should look for FastAGI, where * connects to an external server with resident proceses, and that suits the Java model much better. Of course, if all you want to do is lookup the callerid on mysql or something just as trivial, go for a 20-line Perl script. We have found that generally speaking, running the FastAGI server on the same machine as Asterisk yields better performance than launching separate exe processes through the dial plan. Completely anecdotal of course. This is careful research conducted over our entire 5 customer base... -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Bridged PRI calls - processor involvement?
It probably did but we run in updates every week and nobody can state exactly when the problem started only a few weeks ago - not very helpful. I can see that when I hear the issue the iowait time is high on the processor. Steve -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Fredrickson Sent: 08 June 2007 15:43 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Bridged PRI calls - processor involvement? Did it accompany an update you made? If you can find out what version the problem started occurring, that would help in fixing the problem. Matthew Fredrickson Software/Hardware Engineer Digium, Inc. On Jun 8, 2007, at 2:59 AM, Steve Hanselman wrote: The setup. Asterisk is on a 3G Zeon Dell 2850 running Fedora Core 5/6 (all yum updates applied), the TE410 lives on it's own interrupt. Asterisk sits between our telco and a PRI enabled PBX. These are the relevant versions installed: Linux: 2.6.20-1.2316.fc5smp Zaptel: 1:1.4.2.1-34.fc5 Asterisk: 1:1.4.0-34.fc5.at Libpri: 1:1.4.0-16.fc5.at Wildcard details: Found TE4XXP at base address fe3ffc00, remapped to f88bec00 TE4XXP version c01a016a, burst OFF, slip debug: OFF Octasic optimized! FALC version: 0005, Board ID: 00 Reg 0: 0x377bb400 Reg 1: 0x377bb000 Reg 2: 0x Reg 3: 0x Reg 4: 0x0001 Reg 5: 0x Reg 6: 0xc01a016a Reg 7: 0x1f00 Reg 8: 0x Reg 9: 0x00ff Reg 10: 0x004a TTE4XXP: Launching card: 0 TE4XXP: Setting up global serial parameters Found a Wildcard: Wildcard TE410P (3rd Gen) TE4XXP: Span 1 configured for CCS/HDB3/CRC4 TE4XXP: Span 2 configured for CCS/HDB3/CRC4 The problem: At random points during calls we lose 1-3 seconds of speech (both ways both callee and caller), this can be replicated (or at least a very good approximation!) by generating a high level of interrupt/cpu activity (for instance copying data from a USB caddy as we tried the other day in an attempt to reproduce this more reliably). The calls are bridged PRI:PRI calls, no VOIP involvement. This was not a problem until approx 3-4 weeks ago, but I can't tie it down to an exact date. Steve Interrupt sharing is not a problem anymore with those cards. What version of zaptel did you try installing? Can you explain more about your problems? Also, your configuration and setup would help out as well. --- Matthew Fredrickson Digium, Inc. The information contained in this email is intended for the personal and confidential use of the addressee only. It may also be privileged information. If you are not the intended recipient then you are hereby notified that you have received this document in error and that any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) Ltd Nevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UK Registered Office as above. Registered in England No. 2764339 See our current vacancies at www.brendata.co.uk ___ --Bandwidth 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 The information contained in this email is intended for the personal and confidential use of the addressee only. It may also be privileged information. If you are not the intended recipient then you are hereby notified that you have received this document in error and that any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) Ltd Nevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UK Registered Office as above. Registered in England No. 2764339 See our current vacancies at www.brendata.co.uk ___ --Bandwidth 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] choppy sound with playback, background, etc... but not with musiconhold
Hello, I have an asterisk 1.2.18 working fine, the only problem is that all applications that play audio, sound like tremolo or vibrato, but musiconhold plays fine. The same audio file (wav, mp3, ...) works fine with Musiconhold() but not with Playback() or Background()... If I move app_playback.so from this system to another asterisk, playback works fine... Do you know what is happening and how can I fix it? It's an only SIP system, no fxo/fxs cards. Thanks in advance. -- Paco Brufal[EMAIL PROTECTED] ServiTux Servicios Informáticos S.L. Tel. 966 160 600 / Fax. 966 160 601 ___ --Bandwidth 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] Hot GXP-2000
Carlos, We had this happen once here with a batch of phones received from Grandstream about a year ago now. Email Grandstream on it and they should know exactly what the problem is, I believe they ended up replacing the phones for us. Jessee Holmes Atacomm / Ataractic Corporation www.atacomm.com V: 1-877-700-VOIP [EMAIL PROTECTED] Looking for voice over IP products? Visit our VoIP store at http:// voipstore.atacomm.com/ On Jun 8, 2007, at 10:47 AM, Carlos Chavez wrote: This is off topic for Asterisk but I need a suggestion. I have a customer (travel agency) that has recently begun complaining that their GXP-2000 phones are getting very hot, they say that around mid day the handset is so hot that it can burn your ear. These phones are in constant use and right now the weather in Mexico is hot. Anyone know why the handset would get so hot? Only the phones assigned to sales get this way and the others in the office do not have this problem. They are all connected to the same Linksys PoE switch for power. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 ___ --Bandwidth 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] Best Codec
I'm wary of using g711 of public broadband networks. ... It'd be interesting to see some comparisons or comments from people using g726 as this does seem to be supported by quite a few hardware devices. We are using g711 pretty much exclusively for all residential customers in the US and it worked out well for us. For those with very slow DSL connections in rural areas (128 kbps up / 256 kbps down) we use 40 ms packets as 20 ms packets still used two ATM frames and hence the overhead was rather large. If that fails, we found g726-32 to be a good alternative. Voice quality is almost as good as g711 (a bit duller), music is acceptable. Transcoding overhead is low, many ATAs support it, and the bandwidth (with overhead) is about 40 kbit/sec. It's a good alternative, IMO. --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
[asterisk-users] Write to multiple databases as redundancy scheme
Hi, Can Asterisk write to multiple MySQL databases in different machines, at the same time, as a backup scheme? If it does, where can that be configured? In res_mysql.conf file? Does anyone ever made it? Regards, Ricardo. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call Hold event asterisk
Lee Jenkins wrote: sathish s wrote: i need to catch the call hold event from my asterisk-java program. Im using net.sf.asterisk.*; for communicating with asterisk server. I need to get the call hold status on my java program . I can able to get the music on hold status but i cannot able to get the call hold status. The events like 1. HoldEvent , 2.HoldedcallEvent 3. UnHold event are not getting fired when the call hold is happening . When the call is put in hold , i need to update the satus as CAll is in Hold. For this i need to catch the call hold event . How can i make this ...reply me Thanks in advance sathish. Does Asterisk AMI support even support these events? I'm using 1.2 and have not seen them. Does 1.4 fire these events? From app_dial.c: case AST_CONTROL_HOLD: if (option_verbose 2) ast_verbose(VERBOSE_PREFIX_3 Call on %s placed on hold\n, o-chan-name); ast_indicate(in, AST_CONTROL_HOLD); break; case AST_CONTROL_UNHOLD: if (option_verbose 2) ast_verbose(VERBOSE_PREFIX_3 Call on %s left from hold\n, o-chan-name); ast_indicate(in, AST_CONTROL_UNHOLD); break; As you can see no event calls to the manager are being made. ___ --Bandwidth 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] Changing the messages that are played when a user is unavailable/busy
On 6/8/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: Log into your mailbox. Press 0, then press the option listed to record your unavail and busy greetings. I'm no expert, so someone feel free to correct me if I'm wrong, but you should be able to make one or two recordings and then either copy or symlink that file in /var/spool/asterisk/voicemail/[context]/[user]/busy.[gsm|wav] and /var/spool/asterisk/voicemail/[context]/[user]/unavail.[gsm|wav] If those files exist, Asterisk will play them instead of the default The person at extension XXX is not available right now, please The only problem I forsee with this scenario is if the end-user goes in to their voicemail and creates a new unavailable or busy message. Surely you should be able to block them from over-writing those files though my making them read-only. Best of luck. -- Justin Moore aka wantmoore --- www.wantmoore.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] Re: Not getting CID Name from PRI
On 6/8/07, Kyle Sexton [EMAIL PROTECTED] wrote: Having a problem w/ not getting CID name from a PRI. CID Name appears in the PRI debug, but even after a Wait(4) it still appears after the phone is ringing. Here is the relevant info from my PRI debug output. Line 4 is a NoOp showing me trying to echo Name and Number. Line 6 dials the extension, and you can see callerid name get presented on line 29. Again, there is a Wait(4) before the NoOp on line 3. 1 -- Processing IE 30 (cs0, Progress Indicator) 2 -- Processing IE 108 (cs0, Calling Party Number) 3 -- Processing IE 112 (cs0, Called Party Number) 4 -- Executing NoOp(Zap/1-1, Name: Num: 9515551212) in new stack 5 -- Executing Macro(Zap/1-1, stdexten|6448|SIP/6448) in new stack 6 -- Executing Dial(Zap/1-1, SIP/6448|20) in new stack 7 -- Called 6448 8 -- SIP/6448-08945b40 is ringing 9 Protocol Discriminator: Q.931 (8) len=10 10 Call Ref: len= 2 (reference 195/0xC3) (Terminator) 11 Message type: CALL PROCEEDING (2) 12 [18 03 a9 83 81] 13 Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 14 ChanSel: Reserved 15 Ext: 1 Coding: 0 Number Specified Channel Type: 3 16 Ext: 1 Channel: 1 ] 17 Extension Changed 6448 new state Ringing for Notify User 7799 18 Extension Changed 6448 new state Ringing for Notify User 6452 19 Protocol Discriminator: Q.931 (8) len=9 20 Call Ref: len= 2 (reference 195/0xC3) (Terminator) 21 Message type: ALERTING (1) 22 [1e 02 81 88] 23 Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) 24 Ext: 1 Progress Description: Inband information or appropriate pattern now available. (8) ] 25 Protocol Discriminator: Q.931 (8) len=36 26 Call Ref: len= 2 (reference 195/0xC3) (Originator) 27 Message type: FACILITY (98) 28 [1c 1d 9f 8b 01 00 a1 17 02 01 01 02 01 00 80 0f 4c 45 57 4f 4e 2c 52 59 41 4e 20 20 20 20 20] 29 Facility (len=31, codeset=0) [ 0x9f, 0x8b, 0x01, 0x00, 0xa1, 0x17, 0x02, 0x01, 0x01, 0x02, 0x01, 0x00, 0x80, 0x0f, 'LASTNAME', 0x2c, 'FIRSTNAME', 0x20, 0x20, 0x20, 0x20, 0x20 ] 30 -- Processing IE 28 (cs0, Facility) 31 Handle Q.932 ROSE Invoke component 32 Protocol Discriminator: Q.931 (8) len=9 33 Call Ref: len= 2 (reference 195/0xC3) (Originator) 34 Message type: DISCONNECT (69) -- Kyle Sexton Solved it, I had the diaplan going straight to Dial(), put an Answer() in and CallerID Name now works. -- Kyle Sexton ___ --Bandwidth 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] Write to multiple databases as redundancy scheme
MySQL has its own ways of doing this kind of thing. Take a look at the documentation http://dev.mysql.com/doc/refman/5.0/en/replication.html on MySQL's website related to replication. Bobby _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Friday, June 08, 2007 11:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Write to multiple databases as redundancy scheme Hi, Can Asterisk write to multiple MySQL databases in different machines, at the same time, as a backup scheme? If it does, where can that be configured? In res_mysql.conf file? Does anyone ever made it? Regards, Ricardo. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Write to multiple databases as redundancy scheme
It would be better to let MySQL handle that - use the built-in replication facilities. It's easy to setup. -- Chris Mason (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth 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] Write to multiple databases as redundancy scheme
Why would you do this why put the overhead inside asterisk when mysql has perfectly good replication mechanisms built in? On Jun 8, 2007, at 12:44 PM, [EMAIL PROTECTED] wrote: Hi, Can Asterisk write to multiple MySQL databases in different machines, at the same time, as a backup scheme? If it does, where can that be configured? In res_mysql.conf file? Does anyone ever made it? Regards, Ricardo. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- Bryan Laird, Sr. Manager CM Operations -+- Cablemodems are the gateway to the Internet. The Internet is a gateway to some things that are better left un-seen. ___ --Bandwidth 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] Write to multiple databases as redundancy scheme
[EMAIL PROTECTED] wrote: Hi, Can Asterisk write to multiple MySQL databases in different machines, at the same time, as a backup scheme? If it does, where can that be configured? In res_mysql.conf file? Not that I'm aware of, but you can setup MySQL to mirror the data to a slave database. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation 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] Write to multiple databases as redundancy scheme
Justin Moore wrote: On 6/8/07, Chris Mason (Lists) [EMAIL PROTECTED] wrote: It would be better to let MySQL handle that - use the built-in replication facilities. It's easy to setup. That's a great idea for backup purposes, but if the OP is wanting true redundancy, that won't help much. What happens then when the primary box fails? CDR not written to the primary can't be replicated... This is when you would use a mysql cluster and a VIP. ___ --Bandwidth 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] agi with java?
Java is not a great solution for AGIs because they are script you should fire up and terminate very fast, while the overhead of launching a JVM, loading all classes, etc, is pretty large. Also, you don't want multiple JVMs in parallel loading everything multiple times. How about writing your AGIs in JSP? In that case, you could leave a VM (+Tomcat) running, and the AGIs would simply use the existing in-memory VM. Java does actually scale much better than many other environments once you take away the cost of instantiating the VM for each execution. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited For full contact details visit http://www.minotaur.it/chris.html This email is made from 100% recycled electrons ___ --Bandwidth 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] Console duplicate output problem
Eric ManxPower Wieling wrote: This is really strange. Every message to the (VGA) console is written twice to the screen, but not on the SSH connection. Any clues how to stop this behavior? Stop running in graphics mode. OK, that's a great clue, but can you tell me how to disable now? Bart ___ --Bandwidth 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] Write to multiple databases as redundancy scheme
On 6/8/07, Chris Mason (Lists) [EMAIL PROTECTED] wrote: It would be better to let MySQL handle that - use the built-in replication facilities. It's easy to setup. That's a great idea for backup purposes, but if the OP is wanting true redundancy, that won't help much. What happens then when the primary box fails? CDR not written to the primary can't be replicated... -- Justin Moore aka wantmoore --- www.wantmoore.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] getting at ${CALLERIDNUM}
On Thu, Jun 07, 2007 at 04:52:31PM -0400, Jared Smith wrote: On 6/7/07, Matthew Pease [EMAIL PROTECTED] wrote: I'm having awesome fun with Asterisk voicepulse connect together. So cool. I'm glad you're having fun! I'm trying to have the caller id read back to me.Do I need to do something to have this sent across in the sip.conf? Or is there something I need to do somewhere to enable the reading of this data? If you're using Asterisk 1.4, the syntax has changed: The 1.4 syntax works in 1.2 as well - I was getting warnings about deprecated stuff and managed to work it all out of my dialplan. That means my current dialplan should work just as well in 1.4 - though things may start to change then, I can handle that at the time. It's usually worth testing things out watching for messages of this nature. -- Phil Reynolds o mail: [EMAIL PROTECTED] |L_ \ / Web: http://www.tinsleyviaduct.com/phil/ (_)- \/ Waltham 66, Emley Moor 69, Droitwich 79, Windows 95 ___ --Bandwidth 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] Bridged PRI calls - processor involvement?
iowait time? I'm not familiar with that. Where are you seeing that? Also, is it a reproducible problem? --- Matthew Fredrickson Software Engineer Digium, Inc. On Jun 8, 2007, at 11:23 AM, Steve Hanselman wrote: It probably did but we run in updates every week and nobody can state exactly when the problem started only a few weeks ago - not very helpful. I can see that when I hear the issue the iowait time is high on the processor. Steve -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Fredrickson Sent: 08 June 2007 15:43 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Bridged PRI calls - processor involvement? Did it accompany an update you made? If you can find out what version the problem started occurring, that would help in fixing the problem. Matthew Fredrickson Software/Hardware Engineer Digium, Inc. On Jun 8, 2007, at 2:59 AM, Steve Hanselman wrote: The setup. Asterisk is on a 3G Zeon Dell 2850 running Fedora Core 5/6 (all yum updates applied), the TE410 lives on it's own interrupt. Asterisk sits between our telco and a PRI enabled PBX. These are the relevant versions installed: Linux: 2.6.20-1.2316.fc5smp Zaptel: 1:1.4.2.1-34.fc5 Asterisk: 1:1.4.0-34.fc5.at Libpri: 1:1.4.0-16.fc5.at Wildcard details: Found TE4XXP at base address fe3ffc00, remapped to f88bec00 TE4XXP version c01a016a, burst OFF, slip debug: OFF Octasic optimized! FALC version: 0005, Board ID: 00 Reg 0: 0x377bb400 Reg 1: 0x377bb000 Reg 2: 0x Reg 3: 0x Reg 4: 0x0001 Reg 5: 0x Reg 6: 0xc01a016a Reg 7: 0x1f00 Reg 8: 0x Reg 9: 0x00ff Reg 10: 0x004a TTE4XXP: Launching card: 0 TE4XXP: Setting up global serial parameters Found a Wildcard: Wildcard TE410P (3rd Gen) TE4XXP: Span 1 configured for CCS/HDB3/CRC4 TE4XXP: Span 2 configured for CCS/HDB3/CRC4 The problem: At random points during calls we lose 1-3 seconds of speech (both ways both callee and caller), this can be replicated (or at least a very good approximation!) by generating a high level of interrupt/cpu activity (for instance copying data from a USB caddy as we tried the other day in an attempt to reproduce this more reliably). The calls are bridged PRI:PRI calls, no VOIP involvement. This was not a problem until approx 3-4 weeks ago, but I can't tie it down to an exact date. Steve Interrupt sharing is not a problem anymore with those cards. What version of zaptel did you try installing? Can you explain more about your problems? Also, your configuration and setup would help out as well. --- Matthew Fredrickson Digium, Inc. The information contained in this email is intended for the personal and confidential use of the addressee only. It may also be privileged information. If you are not the intended recipient then you are hereby notified that you have received this document in error and that any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) Ltd Nevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UK Registered Office as above. Registered in England No. 2764339 See our current vacancies at www.brendata.co.uk ___ --Bandwidth 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 The information contained in this email is intended for the personal and confidential use of the addressee only. It may also be privileged information. If you are not the intended recipient then you are hereby notified that you have received this document in error and that any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) Ltd Nevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UK Registered Office as above. Registered in England No. 2764339 See our current vacancies at www.brendata.co.uk ___ --Bandwidth 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] Changing the messages that are played when a user is unavailable/busy
Timothy Parez wrote:
I have my custom sounds which should be played instead of the default
ones when a user is busy or unavailable:
The person at extension XXX is not available right now, please
Of course I can simply replace the files, but the problem is my
implementation shouldn't (MUST NOT) mention the extension.
My files say something like
The person you are trying to reach is not available right now.
If you want to contact this person on his cellphone, press 1,
if you want to leave a voice message, press 2 or wait.
As you can see the extension is not mentioned, so simply
replacing the files would probably cause something weird.
Where do I define what message are/aren't played in this case?
Use the s (=skip) option to VoiceMail().
Playback(myfile);
Playback(vm-intro);
VoiceMail(${EXTEN},s);
Philipp
--
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's use IT to solve problems and not to create new ones.
Asterisk? - http://www.das-asterisk-buch.de
Geschäftsführer: Stefan Wintermeyer
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[asterisk-users] Polycom phone registration problem
Hi, One of my users is in trouble with his polycom phone hooked to an asterisk server. The phone works fine for a few days, and then disappears from the registered sip peers in asterisk. The user is able to place outbound phone calls, but can't receive incoming calls until the network plug is unplugged/plugged. Working line XXYYZZAA24/XXYYZZAA24 10.50.5.186 D A 5060 OK (12 ms) Non working line (sip show peers) XXYYZZAA24/XXYYZZAA24 (Unspecified) D A 5060 OK (12 ms) Do you guys have any clue about this issue ? Thanks Laurent ___ --Bandwidth 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] Need help on Text entry for asterisk through touchpad
Hi, I need to build Text entry application by using asterisk. I already tried this with spandsp application along with app_dtmftotext.c file, it was not working because of some version problem. Is there any way of building the text entry application through touch pad. Regards K.Rajesh. _ Post your 2nd hand stuff for free on Yello.www.yello.in http://www.yello.in/home.php?m_source=hotmailtagutm_medium=textlinkutm_content=inutm_campaign=may ___ --Bandwidth 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 Hold event asterisk
On Fri, 2007-06-08 at 11:12 -0600, Anthony Francis wrote: Lee Jenkins wrote: sathish s wrote: i need to catch the call hold event from my asterisk-java program. Im using net.sf.asterisk.*; for communicating with asterisk server. I need to get the call hold status on my java program . I can able to get the music on hold status but i cannot able to get the call hold status. The events like 1. HoldEvent , 2.HoldedcallEvent 3. UnHold event are not getting fired when the call hold is happening . When the call is put in hold , i need to update the satus as CAll is in Hold. For this i need to catch the call hold event . How can i make this ...reply me Thanks in advance sathish. Does Asterisk AMI support even support these events? I'm using 1.2 and have not seen them. Does 1.4 fire these events? From app_dial.c: case AST_CONTROL_HOLD: if (option_verbose 2) ast_verbose(VERBOSE_PREFIX_3 Call on %s placed on hold\n, o-chan-name); ast_indicate(in, AST_CONTROL_HOLD); break; case AST_CONTROL_UNHOLD: if (option_verbose 2) ast_verbose(VERBOSE_PREFIX_3 Call on %s left from hold\n, o-chan-name); ast_indicate(in, AST_CONTROL_UNHOLD); break; Nope, as you can see, those events, and perhaps several hundreds of others (oh, ok, maybe dozens of others) have not yet been implemented, and may never be, unless someone, or a group of someones, want it bad enough to code it up and get it in. And the other end of the equation is that, Asterisk may end up so bloated with manager traffic, that it clogs everything up. Every now and then, I've been updating the gastman stuff to keep it working with new events. Perhaps we could have event types be something configurable, to keep down unwanted event traffic on manager communication channels, so we don't send unwanted events to recipients who just toss them away. murf As you can see no event calls to the manager are being made. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users smime.p7s Description: S/MIME cryptographic signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Can Asterisk RAS?
I am trying to set up somthing so I can dial into my asterisk box, and have it behave as a modem bank. Is there anything like that already, or am I going to have to write my own. I checked googls and found no leads, but thought I would ask here before I tried writing my own, just to make sure I wasnot reinventing the wheel. Thank you in advance for any responses. -Chris ___ --Bandwidth 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] Write to multiple databases as redundancy scheme
UltraMonkey (www.ultramonkey.com) and MySQL Cluster (http://dev.mysql.com/doc/refman/5.1/en/mysql-cluster.html) It works a charm. - Brad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Justin Moore Sent: Friday, June 08, 2007 2:13 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Write to multiple databases as redundancy scheme On 6/8/07, Chris Mason (Lists) [EMAIL PROTECTED] wrote: It would be better to let MySQL handle that - use the built-in replication facilities. It's easy to setup. That's a great idea for backup purposes, but if the OP is wanting true redundancy, that won't help much. What happens then when the primary box fails? CDR not written to the primary can't be replicated... -- Justin Moore aka wantmoore --- www.wantmoore.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 The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy 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] Log interpretation
Hi, Are there any decent (commercial or free) LOG parsers for A*k. Its *really* hard to debug issues involving multiple calls (eg meetme) when all of the messages are interlaced with each other. There must be an easier way. (A*K 1.2.18) Adrian ___ --Bandwidth 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: custom cdr fields and cdr_mysql, howto?
On 6/7/07, JR Richardson [EMAIL PROTECTED] wrote: Hi All, http://www.voip-info.org/wiki/index.php?page=Asterisk+func+cdr Under example: exten = s,2,Set(CDR(MyFavoriteBand)=Foo Fighters) exten = s,3,Set(CDR(MyFavoriteSong)=Hero) and under description: -userfield: The channel's user specified field. -any custom value that you wish to store. My question is how do you setup more custom fields in the cdr and be able to write to them through cdr_addon_mysql.so? I know how to add columns to the MySQL cdr table, but nothing is populated when I add something to them. I'm wondering is there a patch to allow this? I'm running 1.2.9 and addons 1.2.3. I found mantis patch 9424, add more userfields to addon-mysql and asterisk-func-cdr. I had to make it compatible for 1.2 but got it working, very nice. Works great, jut what I needed. Thanks ZX81. JR -- JR Richardson Engineering for the Masses ___ --Bandwidth 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: Write to multiple databases as redundancy scheme
Can Asterisk write to multiple MySQL databases in different machines, at the same time, as a backup scheme? If it does, where can that be configured? In res_mysql.conf file? No, you cannot write to 2 different mysql servers with res_mysql. Just use MySQL replication as an alternative. Easy to setup. Asterisk writes to the Master database and the Master replicates changes to slave databases for backup. JR -- JR Richardson Engineering for the Masses ___ --Bandwidth 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] Write to multiple databases as redundancy scheme
On 6/8/07, Justin Moore [EMAIL PROTECTED] wrote: On 6/8/07, Chris Mason (Lists) [EMAIL PROTECTED] wrote: It would be better to let MySQL handle that - use the built-in replication facilities. It's easy to setup. That's a great idea for backup purposes, but if the OP is wanting true redundancy, that won't help much. What happens then when the primary box fails? CDR not written to the primary can't be replicated... If it's only for the unlikely event that a DB server is unavailable, why not have it log the CDR in text and in MySQL? If the DB server is unavailable, the records could be parsed from the text file and the database updated. Of course, if you had to do this more than once or twice, it would get a bit annoying, I'm sure. But then again, write the script to do it, and use it to populate the other databases? Dunno, just thinking out loud here. I've written a few parsers, and the format appears to be easy to parse. It really wouldn't be too big of a deal. The hardest part will be kicking it off (I'd use cron), parsing the file (my personal preference would be perl or PHP), updating the database, and making sure you don't insert duplicates. I think I would use the UniqueID as they key, and then just use INSERT statements. You may need an IGNORE in it to allow it to keep going, even when there are duplicates. It's been a while since I wrote something to update a DB where I was unsure of the data hygiene. ___ --Bandwidth 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 problem...
Hi, tahnks for your answer. but i haved install with command apt-get install asterisk, but i don't have package asterisk-addons. if i download asterisk-addons by digium site, run well with asterisk debian pakages?? thanks 2007/6/8, Tzafrir Cohen [EMAIL PROTECTED]: On Fri, Jun 08, 2007 at 02:52:39PM +0100, Carlos Jerónimo wrote: Hi, i got Ubuntu 6.06 installed and theres a problem with asterisk. I've sucessfully installed it with the command: #apt-get install asterisk Then after installing FreePBX i get this error when restarting asterisk: [EMAIL PROTECTED]:/home/hernandezz# asterisk -rvv Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) After looking at the logs i noticed the problem could be the module format_mp3.so not being loaded because not exists in my PC. in modules.conf i comment the line load = format_mp3.so and now it's works.This module is necessary? It can be used for playing mp3 files. It is part of asterisk-addons. You may need to reinstall / upgrade asterisk-addons for the current version of Asterisk. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Carlos Jerónimo ___ --Bandwidth 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] Replacing SX-2000 Centigram Voicemail with Asterisk?
We have a customer with an obsolete Centigram voicemail system who would like to replace it with Asterisk. Any one with experience doing this or information on the signalling and trunking used to connect the Mitel SX-2000 to the Centigram server? -- George Pajari (dCAP), netVOICE communications 604 484 VOIP(8647) x102 www.netvoice.ca www.ip-centrex.ca www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102) ___ --Bandwidth 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 Asterisk RAS?
On 6/8/07, Christopher Dobbs [EMAIL PROTECTED] wrote: I am trying to set up somthing so I can dial into my asterisk box, and have it behave as a modem bank. Is there anything like that already, or am I going to have to write my own. I checked googls and found no leads, but thought I would ask here before I tried writing my own, just to make sure I wasnot reinventing the wheel. You may want to check out the ZapRAS() dialplan application. I know it's there, and it's supposed to do some sort of RAS stuff, but I've never tried it out. -Jared ___ --Bandwidth 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 Asterisk RAS?
The IAXMODEM might get you half way there...but if you want to connected it to a windows box (which I assume is why you use the RAS acronym), you'll have to look for remote serial port software. -MD- -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christopher Dobbs Sent: Friday, June 08, 2007 2:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Can Asterisk RAS? I am trying to set up somthing so I can dial into my asterisk box, and have it behave as a modem bank. Is there anything like that already, or am I going to have to write my own. I checked googls and found no leads, but thought I would ask here before I tried writing my own, just to make sure I wasnot reinventing the wheel. Thank you in advance for any responses. -Chris ___ --Bandwidth 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 problem...
On Fri, Jun 08, 2007 at 08:24:45PM +0100, Carlos Jerónimo wrote: Hi, tahnks for your answer. but i haved install with command apt-get install asterisk, but i don't have package asterisk-addons. if i download asterisk-addons by digium site, run well with asterisk debian pakages?? Not exactly. It will need some adaptations for it to build vs. the asterisk-dev package, I guess. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] choppy sound with playback, background, etc... but not with musiconhold
On Fri, 8 Jun 2007, Paco Brufal wrote: Hello, I have an asterisk 1.2.18 working fine, the only problem is that all applications that play audio, sound like tremolo or vibrato, but musiconhold plays fine. The same audio file (wav, mp3, ...) works fine with Musiconhold() but not with Playback() or Background()... If I move app_playback.so from this system to another asterisk, playback works fine... Do you know what is happening and how can I fix it? It's an only SIP system, no fxo/fxs cards. Do you have ztdummy loaded? Gordon ___ --Bandwidth 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] Bad Echo between SIP calls
Hi, We have a PRI connection when its was on test networks we had echo problems withoutside line. So I bought a TE212P card resolve the echo problem. Which did to an extent. Its using asterisk 1.2.18 RHEL4-Update 4. But now when we are live, there is a terrible echo between 2 SIP calls. If I call the same extension from outside the voice is clear. I am not sure whats the problem. Also there's slight echo when calling Digium support. Totally lost Digium says we need to remove the echo module to resolve SIP echo problems. Then ? the heck we pay for... Has anyone come through this issue. -- Deepak - Yahoo! Answers - Get better answers from someone who knows. Tryit 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] Unicall/R2 for Asterisk 1.4 Available for TESTING
Moy: I have working an Asterisk 1.4.4 with Unicall rn MFR2. The only problem i have is the RxFAX application, that broke every time... With and error in the linking to the spandsp library. If i have time this weekend i will review to fix the app, Thanks. On 6/4/07, Tobias Wolf [EMAIL PROTECTED] wrote: Humberto Figuera schrieb: HI Tobias, look in www.soft-switch.org/unicall/unicall/index.html ;p Thank you. Not very complete but it has given me an idea what to think of unicall. ___ --Bandwidth 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] Bad Echo between SIP calls
On Sat, 9 Jun 2007, Deepak Naidu wrote: But now when we are live, there is a terrible echo between 2 SIP calls. If I call the same extension from outside the voice is clear. My impression is that the transcoding that takes place between two purely software SIP calls never goes through the TE212P card. There are probably echo cancellation options you can enable that are relevant to software channels. I distantly recall there even being some stuff youc an uncomment in the source. Totally lost Digium says we need to remove the echo module to resolve SIP echo problems. Then ? the heck we pay for... Not sure why Digium would say that. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth 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] FW: Delivery Status Notification(Failure)
[EMAIL PROTECTED], you are email bombing me, please fix your blackberry! Thanks, Steve Totaro http://www.asteriskhelpdesk.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Friday, June 08, 2007 7:38 PM To: Steve Totaro Subject: Delivery Status Notification(Failure) Your message: To: [EMAIL PROTECTED] Subject: RE: [asterisk-users] Centos kernel source Sent Date: 25:28 + has not been delivered to the recipient's BlackBerry Handheld. The returned error status is GENERAL_ERROR ATT08383.txt Description: ATT08383.txt ---BeginMessage--- ---End Message--- ___ --Bandwidth 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 1.4 with Unicall
I have a small call center running with Asterisk 1.4.4 and Unicall. Everything seems to be working but twice now we had to reset the server because all lines stopped working. You can see users dialing in and reaching the queue but the agents never get the call and the lines are not released. I saw that there is a new Zaptel driver which fixes a racing condition with a TE110P card which is what we are using. Could this be the problem? I also keep getting the following messages: [Jun 8 18:36:02] NOTICE[16202]: chan_unicall.c:2401 unicall_indicate: unicall_indicate 16 [Jun 8 18:36:02] NOTICE[16350]: chan_unicall.c:2401 unicall_indicate: unicall_indicate -1 [Jun 8 18:36:02] WARNING[16202]: chan_unicall.c:2449 unicall_indicate: Don't know how to set condition 16 on channel UniCall/1-1 [Jun 8 18:36:02] NOTICE[16202]: chan_unicall.c:2401 unicall_indicate: unicall_indicate 16 [Jun 8 18:36:02] WARNING[16202]: chan_unicall.c:2449 unicall_indicate: Don't know how to set condition 16 on channel UniCall/1-1 [Jun 8 18:36:02] NOTICE[16350]: chan_unicall.c:2401 unicall_indicate: unicall_indicate 17 [Jun 8 18:36:02] WARNING[16350]: chan_unicall.c:2449 unicall_indicate: Don't know how to set condition 17 on channel UniCall/14-1 What do they mean? I've never seen them under Asterisk 1.2 -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ --Bandwidth 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] Changing the messages that are played when a user is unavailable/busy
Thnx for your quick replies. I will try all of the above methods :-) On Fri, 2007-06-08 at 13:02 -0400, Justin Moore wrote: On 6/8/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: Log into your mailbox. Press 0, then press the option listed to record your unavail and busy greetings. I'm no expert, so someone feel free to correct me if I'm wrong, but you should be able to make one or two recordings and then either copy or symlink that file in /var/spool/asterisk/voicemail/[context]/[user]/busy.[gsm|wav] and /var/spool/asterisk/voicemail/[context]/[user]/unavail.[gsm|wav] If those files exist, Asterisk will play them instead of the default The person at extension XXX is not available right now, please The only problem I forsee with this scenario is if the end-user goes in to their voicemail and creates a new unavailable or busy message. Surely you should be able to block them from over-writing those files though my making them read-only. Best of luck. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP Transit problem
Hi! Hope someone can help me. I'm trying to pass SIP traffic from one asterisk to another through a third server. Here is the desired scenario: ServerA -- SIP -- ServerB -- SIP -- ServerC When a call is placed on a ServerA local, I can see that ServerB receives the call and dials ServerC. But ServerC says: Jun 8 09:38:32 NOTICE[3269] chan_sip.c: Failed to authenticate user asterisk sip:[EMAIL PROTECTED];tag=as15c8b5e0 However, when I change the configuration between ServerA and ServerB such that: ServerA -- IAX/2 -- ServerB -- SIP -- ServerC This works just fine. If I understand correctly, ServerA only needs to authenticate to ServerB. The fact that ServerB dials ServerC when both legs are SIP seems to indicate that there is no AUTH problem between A and B. And with the 2nd scenario, it proves that there is no auth issue between B and C. Am I missing something? Has anybody got a recipe for this? I'd appreciate any info. Thanks Jug Mensenares ___ --Bandwidth 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] Bad Echo between SIP calls
Ya, I have done that, below is zapata.conf. Also we had an TMP card with analog lines. SIP cals were great on them. now when we switched over. SIP calls have echo.. which shouldnt be at all. [channels] language=en #include zapata_additional.conf context=from-pstn switchtype=national pridialplan=national signalling=pri_cpe faxdetect=incoming usecallerid=yes echocancel=yes callerid=asreceived echocancelwhenbridged=no echotraining=128 ;rxgain=-3.0 ;txgain=-7.0 group=0 channel=1-23 -- Deepak Alex Balashov [EMAIL PROTECTED] wrote: On Sat, 9 Jun 2007, Deepak Naidu wrote: But now when we are live, there is a terrible echo between 2 SIP calls. If I call the same extension from outside the voice is clear. My impression is that the transcoding that takes place between two purely software SIP calls never goes through the TE212P card. There are probably echo cancellation options you can enable that are relevant to software channels. I distantly recall there even being some stuff youc an uncomment in the source. Totally lost Digium says we need to remove the echo module to resolve SIP echo problems. Then ? the heck we pay for... Not sure why Digium would say that. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - All New Yahoo! Mail Tired of unwanted email come-ons? Let our SpamGuard protect you.___ --Bandwidth 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] CDR accuracy
Hi all users, I has been joining this user list for about 1 year, and always has seen the successful story about the Asterisk act as IP PBX and even communication appliances solutions. And thank for this list to help each other and make everyone success. I also being inspired by this user-list and wish to start my implementation of Asterisk as IP PBX. However, billing is one of the main concern in the real life production server, I has been trying with my testing server and it show like the non-pro path of Asterisk csv file. For example; (Polycom phone) and using polycom build in function blind Transfer function. Exten SIP 200 call outsider (Mr.X) through Zap Channel, Talk .and then SIP 200 transfer the call (Mr.X) to SIP 300. The CSV billing shows, SIP 200 call Mr X and started and end as below; WSang 200,200,90124086376,200,SIP/WSang-08b52148,Zap/1-1,Dial,zap/1 /0124086376||WTt,2007-06-09 10:32:52,2007-06-09 10:32:56,2007-06-09 10:33:18,26,22,ANSWERED, Mr.X has spoken to SIP 300 for about 12sec 90124086376,90124086376,300,300,Zap/1-1,SIP/chan-08b57688,Hangu p,0.5,2007-06-09 10:33:33,,2007-06-09 10:33:33,0,0,NO ANSWER, Now, when come to billing, first I can bill SIP 200 for the period of conversation. However, how can I bill SIP 300 for the period of 12sec conversation? And where to prove that this call is being transfer by SIP 200 to SIP300. Since, we have so many experiences expert around the list, can some one help on this issues? Or do you all have such issues after implemented to your customer or your own use??? ___ --Bandwidth 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] Softphone for smartphone such as Nokia N90 / 93 / N95
looking for good sip softphone for wifi and 3G network. 1 are there any sip softphone ( with gsm/g723/G729 codec ) for smartphone such as Nokia N90 / 93 / N95 ? 2 are there any sip softphone ( with gsm/g723/G729 codec ) for Window mobile5 Or wm2003 ? 3 How is the sound quality of GSM /G723/729 codec on wifi/3G network? which codec is better for wift/3G ? Jackie ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UPDATE... 2 down, 1 to go (3 questions - variables, upgrading, and IRC)
On Fri, 8 Jun 2007, Jared Smith wrote: On 6/7/07, Nick Seraphin [EMAIL PROTECTED] wrote: Still need an answer to this one. I wrote a response yesterday, but it looks like it didn't come through for some reason. The answer is to use the DumpChan() application and watch the CLI when it's called. Yeah... Thanks... I got your first reply after I sent the second message. I guess the mail list server was backed up. Unfortunately, DumpChan didn't appear until 1.2, so I'm going to have to upgrade anyway. Thanks, -- Nick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UPDATE... 2 down, 1 to go (3 questions - variables, upgrading, and IRC)
On Fri, 8 Jun 2007, Steve Edwards wrote: On Fri, 8 Jun 2007, Jared Smith wrote: On 6/7/07, Nick Seraphin [EMAIL PROTECTED] wrote: Still need an answer to this one. I wrote a response yesterday, but it looks like it didn't come through for some reason. The answer is to use the DumpChan() application and watch the CLI when it's called. I interpreted this question as how do I see the variables for this channel using the CLI? -- show channel foo. That's true... that's exactly what I wanted. But DumpChan does provide the functionality too. The problem is, both DumpChan and the enhancement to show channel that you describe (listing the variables) weren't added until at least 1.2. My version, when you do a show channel whatever doesn't show any of the variables. I checked that long before I sent my first message. :-) Thanks, -- Nick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
