[asterisk-users] input request: progzone and zaptel hangup
Hi I noticed that my system has three sets of data regarding telephony behaviour in different parts of the world: 1. libtonezone , part of zaptel, and the data is from the source file zaptel/zonedata.c . Zaptel seems to use it for generating some tones. 2. /etc/asterisk/indications.conf . Asterisk uses it to play tones of busy, congestion, dialtone, etc. (PlayTones). The format is pratically equivalent to zonedata.c . 3. main/dsp.c has an aliases table with 5 countries that are aliases to three behaviours (North-America, Costa-Rica and the UK). I noticed that the busy tone listed for the UK in zonedata.c is different from the one listed as uk in dsp.c . The busydetect code in Asterisk's chan_zap uses those aliases for progzone. I noticed that for Israel (at least in Bezeq) progzone=uk seems to detect hangups. Anybody needed to change the dsp.c data to get the hangup detected? What about other countries? E.g: Singapure? -- 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] Possibility to catch DTMF when 2 users are in a conversation
Antoine Fressancourt wrote: I will sum up the results of my investigations : - When canreinvite is set to yes, I manage to make a video call between the 2 parties, when I emit a DTMF signal, it triggers the playback of a sound clip correctly, but I can't playback a video clip. What's the format of the video clip? I don't think Asterisk supports all formats. And, shouldn't it be canreinvite='no'? - When canreinvite is set to no, The DTMF I emit is not detected by Asterisk, although I see the SIP INFO message in the SIP debug messages of Asterisk. Should be canreinvite='yes'. This might be a bug. On the other hand, in your case, even if Asterisk did detected the messages. Without being in the media path, it still won't be able to playback video to the endpoint. Leo ___ --Bandwidth 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] Service Level Compliance
[EMAIL PROTECTED] wrote: The issues we would like to resolve are the following: 1) We can ping our originating SIP providers. However, that doesn't guaratee us that we can receive calls from them. In several occasions, some of our SIP providers have had routing (SIP) problems and when we dial any of the DIDs, they would not even hit our box. The call would simply die somewhere in their network or their providers' networks. How can we proactively confirm that they are actually routing calls to us? We thought we could probably dial out through any of our other providers so the call comes in via a different provider and maybe hit an AGI script. This script could update a MySQL table with a timestamp of the last successful test. We could then take the data from that table and bring it to Nagios and/or Cacti. Is there a better approach? 2) We can test Asterisk responsiveness by doing something like 'asterisk -rx show uptime' and parse the results. We can also connect to MySQL and execute a test query. However, how can we verify that Asterisk is actually talking to MySQL and that it's connection hasn't died? 3) As stated above, we can test the responsiveness of asterisk. However, we have noticed in, at least, one occasion, that even though asterisk seems to be responsive, it would not accept or place any calls. Somehow it's call engine was locked and we had to restart asterisk. How can we verify that asterisk is actually capable of receiving and placing calls? Solve all of 1 - 3 with a couple of scripts plus AGI. You have a test asterisk server using either voip or a local pstn line (or bri/pri) which calls the customers number. I'd try to use a PSTN line since this is the method most of your customers' customers will use to call. When the call arrives to your asterisk box A, write the date/time or something, to a text file (which is read and reported to your monitoring server), then pass the call into an AGI which does some mysql lookups (similar to a normal call would), and write the result of these lookups into another text file (which again is read and compared to the result you should get, and reported back to the monitoring system - assuming some sort of static result can be obtained). Finally, pass the call to Asterisk B, which places the call into the queue. Finally, when the agent answers the call, they hear a recording from your original test box (ie, the originator of the call) which simply loops with Please press 1 to acknowledge, or 0 if you can't understand/hear this properly, the originator is simply waiting for the dtmf (with some configurable timeout), if you get answered and get a 1 or 0, then return that to your monitoring server, either via text file or directly. If you timeout, also report that as needed (possible queue problem, or perhaps the agent took too long to answer, you decide on the timeout value and the alerting to send back). 4) We have no Digium boards and all kernels are 2.6 or above, so we end us using ztdummy, if needed. The client's agents are in a different country and the average lantency is around 250ms. Most of the time, call quality is good. However, there are a few situations where people complaint about echo. Is there a way to measure or improve this? I know it has been a topic discussed at lenght and if we could probably script a way to measure some sort of a MOS value, that would be great. Any ideas? If you have no PSTN interfaces, ie, everything is voip, then you can't be responsible for echo (neither introducing it, nor removing it). If you are responsible for the echo that your voip provider is leaving in the call, then you are pretty much stuffed. You can't really measure it (AFAIK), but you could perhaps ask your provider for an SLA on echo, and see what they have to say about it. Then just copy what they say into your own 5) Anything else that you could think of we could measure to make sure all components are working? The best method to ensure that the whole end-to-end system is working, is by testing the entire end-to-end system. You input is greatly appreciated it. I promise that whatever solution is best recommended and scriptable, we will post our development and working solutions for the community to benefit from. While this would be nice, and could probably be used as a guideline for other people, it should be mostly useless to anyone else, since their end-to-end system will be very different (assuming you really are testing your system properly). Just my 0.02c worth. Regards, Adam -- Adam Goryachev Website Managers Ph: +61 2 8304 [EMAIL PROTECTED] Fax: +61 2 8304 0001www.websitemanagers.com.au ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:
[asterisk-users] Didn't get a frame from channel
Using tdm400. While transfering a call from outside to another extensions, while this outside call is waiting with music, the another extension call hangs up suddenly, and the call is back to the outside call suddenly. Wathcing logs: Jan 15 13:32:44 DEBUG[30148] res_musiconhold.c: Read 462 bytes of audio while expecting 640 Jan 15 13:32:55 DEBUG[27850] channel.c: Didn't get a frame from channel: SIP/219-081d4d60 Jan 15 13:32:55 DEBUG[27850] channel.c: Bridge stops bridging channels SIP/219-081d4d60 and Zap/1-1 Jan 15 13:32:55 DEBUG[27850] chan_zap.c: Hangup: channel: 1 index = 0, normal = 16, callwait = -1, thirdcall = -1 Jan 15 13:32:55 DEBUG[27850] chan_zap.c: disabled echo cancellation on channel 1 Jan 15 13:32:55 DEBUG[27850] chan_zap.c: Set option TDD MODE, value: OFF(0) on Zap/1-1 Jan 15 13:32:55 DEBUG[27850] chan_zap.c: Updated conferencing on 1, with 0 conference users 15 13:32:55 VERBOSE[27850] logger.c: -- Hungup 'Zap/1-1' Jan 15 13:32:55 DEBUG[27850] app_dial.c: Exiting with DIALSTATUS=ANSWER. Jan 15 13:32:55 VERBOSE[27850] logger.c: == Spawn extension This may be the cause: Didn't get a frame from channel... I googled. It is recommended to disable busydetect, but no solution. Any ideas? ___ --Bandwidth 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] Recording queue calls after an xfer?
I implemented something on these lines for unattended transfer. Basically what I did was storing the call-id in an inheritable diaplan variable and then starting a new mixmonitor on the transferred extension. Hope this helps, l. On Mon, 15 Jan 2007 22:45:49 +0100, Jay Moore [EMAIL PROTECTED] wrote: I have a problem where my recorded queue calls stop recording once the call is transferred to a different extension. Is there some additional parameter I need to set so recording continues? Is it even possible to do this? Thanks, Jay -- Loway Research - Home of QueueMetrics http://queuemetrics.loway.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] zaptel hardware detection with genzaptelconf
Hi Someone complained in #asterisk-gui that the Zaptel hardware detection did not work. It seems that zapscan.bin did not work properly, as no /etc/asterisk/zapscan.conf was generated. Solution: detect zaptel hardware with genzaptelconf, and generate zapscan.conf from it. genzaptelconf grep '^fx[os][kl]s=' /etc/zaptel.conf \ | sed 's/\(fx[os]\)..=\(.*\)/[\2]\nport=\1\n/' /etc/asterisk/zapscan.conf # this needs to be run only once: echo '#include zapata-channels.conf' /etc/asterisk/zapata.conf Any need for genzaptelconf to generate zapspan.conf ? Note that genzaptelconf tried to add hints (in the form of comments) to its generated zaptel.conf. E.g: # astbanktype: output fxoks=9 for Astribank output ports, and: # termtype: nt bchan=10-11 dchan=12 for a ISDN span that is NT (vs. TE. Saner default is to assume TE). I don't like using a separate file, because it goes against the logic of the asterisk GUI of using valid configuration data. One may edit zaptel.conf manually and not bother updating zapscan.conf . -- 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] Possibility to catch DTMF when 2 users are in a conversation
2007/1/16, Leo Ann Boon [EMAIL PROTECTED]: Antoine Fressancourt wrote: I will sum up the results of my investigations : - When canreinvite is set to yes, I manage to make a video call between the 2 parties, when I emit a DTMF signal, it triggers the playback of a sound clip correctly, but I can't playback a video clip. What's the format of the video clip? I don't think Asterisk supports all formats. And, shouldn't it be canreinvite='no'? My video is tested against my Asterisk and softphones, by doing a plain old call to a Playback extension playing this video. - When canreinvite is set to no, The DTMF I emit is not detected by Asterisk, although I see the SIP INFO message in the SIP debug messages of Asterisk. Should be canreinvite='yes'. This might be a bug. On the other hand, in your case, even if Asterisk did detected the messages. Without being in the media path, it still won't be able to playback video to the endpoint. According to me, that's the point. I don't really mind having canreinvite set to yes or no for now. If canreinvite is set to no, then Asterisk can't inject video in the ongoing session. If canreinvite is set to yes, then the fact that Asterisk is not in the media path should not be a problem as it can perform a reinvitation to enter this path. So the problem remains. Thank you for your answers. Antoine ___ --Bandwidth 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] OT: Quad-band cellphones with wifi stable sip support
On 15 Jan 2007, at 06:01, Tomer Horn wrote: Hello, I am looking to purchase a new quad-band cellphone and I'm looking for one with WiFi and enough CPU power for stable SIP calls. I was wondering if anyone here can share his experience and recommend on a good cellphone. Any chance there is such a phone with even good WiFi profiles management or am I asking for too much now? :-) The nokia e60 is ok. (Much better than the original zyxel wifi phone) I found the configuration a struggle, but part of that was my fault, I couldn't see the difference between upper and lower case 'w's in the default nokia font! I was pleasantly surprised by it, but it is still a first generation solution, to be given to early adopters and technophiles. I lived with it for a week, and the only thing I can't cope with is that it isn't a clamshell. Tim. Tim Panton www.mexuar.net www.westhawk.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] To 1.4 or not
Hello Gents, following on this discussion, anyone particularly have one view or the other about 1.4 and the voicemail and meetme enhancements (supposedly) it has? We're not in production yet, I've tested 1.2 up until 1.2.13 in the lab as well as 1.4b3, since none of them got a real hammering Its hard to tell at the moment if one is more stable than the other. Also, since I don't use it for anything BUT voicemail and meetme, would a lot of instabilities in the PBX side of things affect me? They shouldn't but who knows. Any comments and/or advice would be appreciated :) \R ___ --Bandwidth 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] connecting 2 asterisk servers through OpenVPN
There is no problem with the CPU utilization, it is around 40%, I will not be able to try this without the VPN, maybe I should try another VPN solution like OpenSwan, or PPTP. Why do you think that IAX will make a difference than SIP? On 1/16/07, Gordon Henderson [EMAIL PROTECTED] wrote: On Mon, 15 Jan 2007, O.Kamal wrote: I am trying to connect 2 asterisk servers through OpenVPN, the VPN should carry 16 channel, however when active channels reached 4 concurrent channels, the connection became unstable, with a very high latency (around 900ms), the internet bandwidth is 1Mbps on each server, I have upgraded the bandwidth to double it, but still have exactly the same problem. Any tips or recommendations on such setup? No real answers, but questions that might help ... Have you tried it without using OpenVPN? Just port-forward the SIP RTP ports, if you need to and give it a go. I am using SIP and G729 between the 2 servers, openVPN using UDP with no compression. Why not IAX? Are your openVPN end-points up to it? Doing high-grade encryption in software might challenge some slower processors - are the VPN endpoints the asterisk boxes themselves? 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 ___ --Bandwidth 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] ENUMLOOKUP debug
Hello,
After upgrading to 1.4 my ENUMLOOKUP returns nothing. Even with new
format.
I've tried commands:
SET(foo=${ENUMLOOKUP(+13015611020loligo.com)});
${ENUMLOOKUP(+13015611020,ALL,c,,e164.org)};
Could I turn debug of ENUMLOOKUP on?
Thanks.
Michael
___
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[asterisk-users] J1/INS1500 and the Redirect Number
Hi everyone! I'm wondering if anyone on the list had the opportunity to work with an NTT INS1500 ISDN PRI service before. You see, in Japan, if you receive a call that was just forwarded by another number, the call presentation not only includes the caller (ANI) and your number (DNIS), it will also usually include the forwarding number (REDIRECT). Does anybody know how to extract this field on Asterisk? For reference, you can look at http://www.ntt-east.co.jp/ISDN/tech/spec/espec/3-5/content_3.html. This is the full specification of INS 1500 signalling. Any assistance would be very much appreciated. Thanks again! Sincerely, 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
[asterisk-users] command like break ore exit in the dialpan
Hi i have a similar dialplan: exten = 99,1,Gotoif(?2:3) exten = 99,2,Meetme(100) exten = 99,3,Meetme(100|options) i'd like to do something like: exten = 99,1,Gotoif(?2:4) exten = 99,2,Meetme(100) exten = 99,4, ... exit ... exten = 99,3,Meetme(100|options) How can i exit the dialplan? I won't use meetme twice! Thanks nik ___ --Bandwidth 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] IAX Channels language
Hello everybody, I have a small problem: I've set language=it in iax.conf, but MeetMe conferences still play en files. I see from the CLI that the Playback app is called with language=en parameter. From the sources of app_meetme I see that it takes the language from the channel, so I think this is a IAX problem. Can anybody help me? Asterisk version 1.2.13. TIA, -- Andrea Spadaccini Multimedia Technologies Institute s.r.l. ___ --Bandwidth 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] Disallowing unauthorized calls to Cisco Polycom phones
Hello, I would like the IP phones to not accespt SIP requests (like INVITE) from any device other than its proxy. Snom phones ignore this while Cisco Polycom accepts the call. Any idea what to do to disable it? Thanks! __Yehavi: ___ --Bandwidth 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: command like break ore exit in the dialpan
exten = 99,4,Hangup ? nik600 wrote: Hi i have a similar dialplan: exten = 99,1,Gotoif(?2:3) exten = 99,2,Meetme(100) exten = 99,3,Meetme(100|options) i'd like to do something like: exten = 99,1,Gotoif(?2:4) exten = 99,2,Meetme(100) exten = 99,4, ... exit ... exten = 99,3,Meetme(100|options) How can i exit the dialplan? I won't use meetme twice! Thanks nik ___ --Bandwidth 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] connecting 2 asterisk servers through OpenVPN
IAX allows trunking. With 16 channels open between 2 asterisk servers that should recover over 400kbs compared to SIP. I believe OpenVPN can be forced into a no-encryption test mode. If you can do that temporarily it is easier to judge the vpn overhead. O.Kamal wrote: There is no problem with the CPU utilization, it is around 40%, I will not be able to try this without the VPN, maybe I should try another VPN solution like OpenSwan, or PPTP. Why do you think that IAX will make a difference than SIP? On 1/16/07, *Gordon Henderson* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: On Mon, 15 Jan 2007, O.Kamal wrote: I am trying to connect 2 asterisk servers through OpenVPN, the VPN should carry 16 channel, however when active channels reached 4 concurrent channels, the connection became unstable, with a very high latency (around 900ms), the internet bandwidth is 1Mbps on each server, I have upgraded the bandwidth to double it, but still have exactly the same problem. Any tips or recommendations on such setup? No real answers, but questions that might help ... Have you tried it without using OpenVPN? Just port-forward the SIP RTP ports, if you need to and give it a go. I am using SIP and G729 between the 2 servers, openVPN using UDP with no compression. Why not IAX? Are your openVPN end-points up to it? Doing high-grade encryption in software might challenge some slower processors - are the VPN endpoints the asterisk boxes themselves? Gordon ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma A102d and Asterisk on Debian 3.1.
On Jan 11, 2007, at 11:08 PM, Jarek Jarzebowski wrote: Dnia 31-12-2006 o 18:05:00 Jarek Jarzebowski [EMAIL PROTECTED] napisał(a): Dnia 31-12-2006 o 17:39:10 Tzafrir Cohen [EMAIL PROTECTED] napisał(a): On Sun, Dec 31, 2006 at 05:08:26PM +0100, Jarek Jarzebowski wrote: Dnia 31-12-2006 o 16:17:19 Tzafrir Cohen [EMAIL PROTECTED] napisał(a): On Sun, Dec 31, 2006 at 03:59:14PM +0100, Jarek Jarzebowski wrote: Hi All, is anybody using Sangoma A102d card with Asterisk on Debian 3.1? I configure and install Sangoma wanpipe step by step based on Sangoma Wiki and manuals but can not get success results. I suppose that it may be some Debian specific case. AFAIK, that procedure has been tested on Debian Sarge before. What specific problems you have? I use wanpipe-2.3.4-3. I run ./Setup install. After standard 2 frist question (answer 'y') I got: Please specify absolute path name of your linux directory (Press Enter for Default: /lib/modules/2.4.27-3-686-smp/ build) I press Enter. And got: Upgrading WANPIPE kernel documentation ...Done. Upgrading WANPIPE kernel headers ...Done. Upgrading WANPIPE kernel drivers ...Done. cp: cannot stat `drivers/net/wan/Makefile': No such file or directory grep: drivers/net/wan/Makefile: No such file or directory Updating T1/E1 in /lib/modules/2.4.27-3-686-smp/build/drivers/net/wan/Makefile ./Setup: line 895: drivers/net/wan/Makefile.nex: No such file or directory cat: drivers/net/wan/Makefile: No such file or directory drivers/net/wan/Makefile does not exist in the kernel-headers package of 2.4 (e.g: your 2.4.27-3-686-smp) . It does seem to exist in the kernel-headers packagers of 2.6 . So one thing to try: use kernel 2.6: apt-get install kernel-image-2.6-686-smp kernel-headers-2.6-686- smp and reboot to that kernel. Note, however that this is just one educated guess of me. OK. I will try that and give an answer. I install wanpipe on Debian (kernel 2.6.8). Wanrouter seems to work OK. But I am affraid that I do something wrong with asterisk config. On asterisk I can see: *CLI pri show span 1 Primary D-channel: 16 Status: Provisioned, Down, Active Switchtype: EuroISDN Type: CPE Window Length: 0/7 Sentrej: 0 SolicitFbit: 0 Retrans: 0 Busy: 0 Overlap Dial: 0 T200 Timer: 1000 T203 Timer: 1 T305 Timer: 3 T308 Timer: 4000 T313 Timer: 4000 N200 Counter: 3 *CLI pri show span 2 Primary D-channel: 47 Status: Provisioned, Down, Active Switchtype: EuroISDN Type: CPE Window Length: 0/7 Sentrej: 0 SolicitFbit: 0 Retrans: 0 Busy: 0 Overlap Dial: 0 T200 Timer: 1000 T203 Timer: 1 T305 Timer: 3 T308 Timer: 4000 T313 Timer: 4000 N200 Counter: 3 *CLI pri show intense debug span 1 [ 00 01 7f ] Unnumbered frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 M3: 3 P/F: 1 M2: 3 11: 3 [ SABME (set asynchronous balanced mode extended) ] 0 bytes of data Sending Set Asynchronous Balanced Mode Extended [ 00 01 7f ] Unnumbered frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 M3: 3 P/F: 1 M2: 3 11: 3 [ SABME (set asynchronous balanced mode extended) ] 0 bytes of data It looks to me like some config misunderstunding between A102d and Aterisk. I am total newbe to sangoma cards, Digium like Tor2 cards have no such a problem. Looking forward to any ideas... Regards, -- Jarek Hi Jarek! Did you find any solution to this problem? I have the exact same problem with a Sangoma A102d card on debian 3.1, 2.6.19 and wanpipe 2.3.4-4. I've followed several different guides, including the one on sangoma's wiki. When I try to make a call out, I get this error: Jan 16 13:17:28] WARNING[18084]: app_dial.c:1081 dial_exec_full: Unable to create channel of type 'ZAP' (cause 34 - Circuit/channel congestion) Also got the same SABME errors as you do. Best regards, Erik Haider Forsén ___ --Bandwidth 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] Stumped with Dial - $50 for answer
On Monday 15 January 2007 6:21 pm, Anselm Martin Hoffmeister wrote: could you verify or negate that adding the T option makes it work? That or transfer=no in iax.conf for hte user/peer entries involved. I never thought of IAX2 transfers here, for some reason I thought that Asterisk was terminating the call to TDM itself (one of the two ends). I wouldn't try transfer=mediaonly at this point; remove the transfer capability altogether. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom phone locks up, send sip busy messages
I have a soundpoint 501 phone that has locked up twice now. You can make a call but when a call is sent to it, it responds with sip busy messages. You get the same message when the phone is in do not disturb. I reset to defaults the first time and it worked for a week or so and then stopped. The incoming calls are ringing three phones ( dial(sip/1sip/2sip/3 ), often two of them are in do not disturb. I read that I wasn't supposed to register these phone and have them set as static hosts. The interesting thing is that the phone displays missed calls every time asterisk tries to send it a call. So instead of ringing you see the counter fly off the chart. Can anyone give me some insite. Jordan Novak ___ --Bandwidth 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] TDM2400 Hardware Echo Cancel
Good Day List, I'm having some issues with echo cancel on my Asterisk box, and have done extensive reading and have gained some useful pointers from this list but have a couple of hopefully fairly simple questions. The Asterisk box is connected via 20 FXO ports on a TDM2400 with the Hardware echo cancel module. Echo cancel almost works, but the users hear what they describe as a 'crackle' coming back when they talk. I want to tune to echo canceller, but am unsure if any of the options provided have any effect on the hardware module. Do the settings such as echocancel and echotraining in Zapata.conf affect the hardware module? Would I be better removing the hardware module and tuning the software echo canceller? The asterisk box is currently running 1.2.13, with zaptel 1.2. Would you advise upgrading to the newer Zaptel drivers? I don't want to upgrade Asterisk itself just yet. Any help or pointers to documentation regarding the hardware echo cancel module would be greatly appreciated, Thanks, Adam Sharples ___ --Bandwidth 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] OT: Quad-band cellphones with wifi stable sip support
Pode. On 16 Jan 2007, at 10:21 , Tim Panton wrote: On 15 Jan 2007, at 06:01, Tomer Horn wrote: Hello, I am looking to purchase a new quad-band cellphone and I'm looking for one with WiFi and enough CPU power for stable SIP calls. I was wondering if anyone here can share his experience and recommend on a good cellphone. Any chance there is such a phone with even good WiFi profiles management or am I asking for too much now? :-) The nokia e60 is ok. (Much better than the original zyxel wifi phone) I found the configuration a struggle, but part of that was my fault, I couldn't see the difference between upper and lower case 'w's in the default nokia font! I was pleasantly surprised by it, but it is still a first generation solution, to be given to early adopters and technophiles. I lived with it for a week, and the only thing I can't cope with is that it isn't a clamshell. Tim. Tim Panton www.mexuar.net www.westhawk.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] spa942 and asterisk 1.2
currently using 1.2.14 and zaptel 1.2.12 i'm using mfc/r2 so i can't move to 1.4 with sip jitter control and improved jitter control in zaptel 1.4. my problem is excessive jitter using linksys spa942. when i set canreinvite=no, which forces rtp to pass through *, quality is horrible. clicking sounds, pauses, etc. but when omitted or canreinvite=yes, sip to sip calls are ok. now, the problem comes to zap calls, i have a te110p using unicall mfc/r2, since rtp passes through *, quality is again awful. just wanted to ask the list whether somebody out there had experience or had used linksys spa942 before. did you experience this phenomenon? how can you go around the zaptel jitter? obviously i tried using jitterbuffer=40 in zaptel.conf and/or even in unicall.conf to no avail. ___ --Bandwidth 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] Queue cmd option 'i'
On 1/15/07, Douglas Garstang [EMAIL PROTECTED] wrote: -Original Message- From: BJ Weschke [mailto:[EMAIL PROTECTED] Sent: Monday, January 15, 2007 3:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Queue cmd option 'i' On 1/15/07, James Fromm [EMAIL PROTECTED] wrote: Using Asterisk 1.4, on the console 'show application queue' mentions an option 'i' that should ignore call forward requests from queue members and do nothing when they are requested. Does this work? My assumption is that the member whose next according to the queue strategy should get the call even if they have forwarding enabled on their SIP device. The forwarding should be ignored. Using Queue(customerservice|i) causes Asterisk to crash when sending the call to the member with forwarding enabled on their SIP device. Am I misinterpreting what this option does? You're not misinterpreting. If it crashes, please file a bug at bugs.digium.com. Thanks. I wonder how this could actually work? If Asterisk sends an INVITE to a phone, and the phone responds with 'Moved Temporarily', and Asterisk sends the INVITE again, isn't the phone just going to send 'Moved Temporarily' again? It doesn't send the Invite again, and it doesn't send a new call request (might be INVITE, might be whatever other channel tech is at the requested forwarded exten). -- Bird's The Word Technologies, Inc. http://www.btwtech.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] SPA-941 (and others ) Transmit Sound Quality
ron, i recently fixed the poor quality with our spa942 using canreinvite=yes have you found out the problem with your spa941? i can't get around the problem of poor quality audio when the rtp pass through * [EMAIL PROTECTED] wrote: Hello, This is not exactly an Asterisk question, but I was encouraged to seek advice here anyway. The kindness of the * open source community is legendary :) I am getting going with an Asterisk 1.2 box, and I'm having trouble getting good quality transmit sound using handsets with VoIP phones. I'm primarily trying to focus on SPA-941, but also experimenting with Aastra 9113i and Uniden UIP1868. I do not at this time have any PSTN cards in the box to provide hardware timing. The use case is calling from the SIP phones (which are extensions registered with the * 1.2 box) to a VoIP termination service which routes the call to a PSTN number. Everything sounds great on the SIP phone, but the sound on the other end of the line is distant and missing bass, most especially so on the SPA-941 (which is the phone we really want to use). If I use the default handset mic gain value of 0db, the sound is so loud for the other person they have to hold the phone away from their ear. If I set it to -6db, it is still too quiet. The Aastra 9113i sounds a little better, and the Uniden 5.4 GHz Cordless sounds actually very good, so I'm pretty sure my network setup is capable of transmitting good sound. Using the speaker-phone on the SPA-941 sounds significantly better than using the handset. But we need the handset to also sound good. I've tried different providers etc. and always come back to the phone. I'm using G711u codec in all cases and silence suppresion is off. I saw a previous thread that mentioned changing the RTP from .03 to .02, however the post was regarding a MeetMe issue. I tried anyway, and it introduced an echo on the line. I've seen many rave reviews regarding the sound quality on the SPA-941, so I'm wondering if maybe I got a bum handset? Would anyone be willing to receive/place a call to tell me if it sounds the way its supposed to or if there is indeed a problem? All suggestions/recommendations greatly appreciated. Much thanks, -- Ron [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth 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] HowTo Config Asterisk and SS7
Hello Asterisk, Can anyone help or put some light on, how can I configure Asterisk to work with SS7? What do I need, in terms of Hardware and Software? Regards, Neel ___ --Bandwidth 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] IAX Trunk timing
I have read that an IAX trunk requires a timing device. What wasn't clear to me was whether it is like TDM ie 1 timing device for the trunk, or if each end requires a timing device. I have a zaptel card in one server; do I have to have one in the second server in order to do an IAX trunk? I set up a trunk and so far calls can be made one way, but not the other. It is probably just not configured correctly, but I just wanted to make sure as I can't seem to find any reason at the moment. Thanks, Andy ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX Trunk timing
If the other server doesnt have any hardware device that can act as timer. then just compile zaptel and modprobe ztdummy .. This kernel module should act as timing source i think . ( it works with meetme ) . On 16/01/07, Andy Hester [EMAIL PROTECTED] wrote: I have read that an IAX trunk requires a timing device. What wasn't clear to me was whether it is like TDM ie 1 timing device for the trunk, or if each end requires a timing device. I have a zaptel card in one server; do I have to have one in the second server in order to do an IAX trunk? I set up a trunk and so far calls can be made one way, but not the other. It is probably just not configured correctly, but I just wanted to make sure as I can't seem to find any reason at the moment. Thanks, Andy ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth 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] IAX2 softphones can't (won't?) use PRI trunks....
I have call center PCs that switch between an IBEAM SIP softphone and a NEBU IAX softphone (for reasons that aren't germane here). The SIP softphones work fine, but the IAX softphones get a fast busy unless I give them an IAX trunk to use, instead of the PRI trunks that all the other phones are using. I am using Asterisk 1.2.3. svn rev 47264. I've appended a sample call trace. The call fails through all the configured PRI trunks to the IAX trunk with a CHANUNAVAIL error, whilst the SIP phones are actively calling out on those same PRI trunks. The numbers dialed are 10 digits with no prefix. I am hopeful that someone will recognize the issue and give me a pointer on where to look for the problem. - Registered IAX2 '4414' (AUTHENTICATED) at 192.168.1.102:4569 -- Accepting AUTHENTICATED call from 192.168.1.102: requested format = alaw, requested prefs = (), actual format = ulaw, host prefs = (ulaw|alaw|gsm), priority = mine -- Executing Set(IAX2/4414-6, EMERGENCYROUTE=YES) in new stack -- Executing Macro(IAX2/4414-6, dialout-trunk|4|xxxnnn||) in new stack -- Executing GotoIf(IAX2/4414-6, 1?3:2) in new stack -- Goto (macro-dialout-trunk,s,3) -- Executing Macro(IAX2/4414-6, user-callerid) in new stack -- Executing GotoIf(IAX2/4414-6, 0?report) in new stack -- Executing GotoIf(IAX2/4414-6, 0?start) in new stack -- Executing Set(IAX2/4414-6, REALCALLERIDNUM=4414) in new stack -- Executing NoOp(IAX2/4414-6, REALCALLERIDNUM is 4414) in new stack -- Executing Set(IAX2/4414-6, AMPUSER=4414) in new stack -- Executing Set(IAX2/4414-6, AMPUSERCIDNAME=User32-IAX) in new stack -- Executing GotoIf(IAX2/4414-6, 0?report) in new stack -- Executing Set(IAX2/4414-6, CALLERID(all)=User32-IAX 4414) in new stack -- Executing NoOp(IAX2/4414-6, Using CallerID User32-IAX 4414) in new stack -- Executing Macro(IAX2/4414-6, record-enable|4414|OUT) in new stack -- Executing GotoIf(IAX2/4414-6, 0 0?2:4) in new stack -- Goto (macro-record-enable,s,4) -- Executing AGI(IAX2/4414-6, recordingcheck|20070115-121440|1168881280.2233) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck recordingcheck|20070115-121440|1168881280.2233: Outbound recording not enabled -- AGI Script recordingcheck completed, returning 0 -- Executing NoOp(IAX2/4414-6, No recording needed) in new stack -- Executing Macro(IAX2/4414-6, outbound-callerid|4) in new stack -- Executing GotoIf(IAX2/4414-6, 1?start) in new stack -- Goto (macro-outbound-callerid,s,3) -- Executing NoOp(IAX2/4414-6, REALCALLERIDNUM is 4414) in new stack -- Executing Set(IAX2/4414-6, USEROUTCID=Business Name xxx-nnn-) in new stack -- Executing Set(IAX2/4414-6, EMERGENCYCID=) in new stack -- Executing Set(IAX2/4414-6, TRUNKOUTCID=Business Name xxx-nnn-) in new stack -- Executing GotoIf(IAX2/4414-6, 0?trunkcid) in new stack -- Executing GotoIf(IAX2/4414-6, 1?trunkcid) in new stack -- Goto (macro-outbound-callerid,s,11) -- Executing GotoIf(IAX2/4414-6, 0?usercid) in new stack -- Executing Set(IAX2/4414-6, CALLERID(all)=Business Name xxx-nnn-) in new stack -- Executing GotoIf(IAX2/4414-6, 0?report) in new stack -- Executing Set(IAX2/4414-6, CALLERID(all)=Business Name xxx-nnn-) in new stack -- Executing NoOp(IAX2/4414-6, CallerID set to Business Name xxx-nnn-) in new stack -- Executing Set(IAX2/4414-6, GROUP()=OUT_4) in new stack -- Executing GotoIf(IAX2/4414-6, 0?108) in new stack -- Executing Set(IAX2/4414-6, DIAL_NUMBER=xxxnnn) in new stack -- Executing Set(IAX2/4414-6, DIAL_TRUNK=4) in new stack -- Executing AGI(IAX2/4414-6, fixlocalprefix) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix -- AGI Script fixlocalprefix completed, returning 0 -- Executing Set(IAX2/4414-6, OUTNUM=xxxnnn) in new stack -- Executing Set(IAX2/4414-6, custom=ZAP/g0) in new stack -- Executing GotoIf(IAX2/4414-6, 0?16) in new stack -- Executing Dial(IAX2/4414-6, ZAP/g0/xxxnnn|120|r) in new stack == Everyone is busy/congested at this time (1:0/0/1) -- Executing Goto(IAX2/4414-6, s-CHANUNAVAIL|1) in new stack -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1) -- Executing NoOp(IAX2/4414-6, Dial failed due to CHANUNAVAIL) in new stack___ --Bandwidth 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] Stumped with Dial - $50 for answer
its notransfer=yes in iax.conf not transfer=no :) On 16/01/07, Andrew Kohlsmith [EMAIL PROTECTED] wrote: On Monday 15 January 2007 6:21 pm, Anselm Martin Hoffmeister wrote: could you verify or negate that adding the T option makes it work? That or transfer=no in iax.conf for hte user/peer entries involved. I never thought of IAX2 transfers here, for some reason I thought that Asterisk was terminating the call to TDM itself (one of the two ends). I wouldn't try transfer=mediaonly at this point; remove the transfer capability altogether. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stumped with Dial - $50 for answer
On Tuesday 16 January 2007 10:09 am, Vicky wrote: its notransfer=yes in iax.conf not transfer=no :) Ahh yes. force consistency in the CLI where it doesn't necessarily belong, but use idiotic variable names in the config files. :-) -A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] command like break ore exit in the dialpan
I don't know if I understand you correctly but you could place a Goto or a Hangup there: exten = 99,1,Gotoif(?2:4) exten = 99,2,Meetme(100) exten = 99,3,Goto or Hangup exten = 99,4,Meetme(100|options) -Original Message- From: nik600 [mailto:[EMAIL PROTECTED] Sent: Tuesday, January 16, 2007 1:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] command like break ore exit in the dialpan Hi i have a similar dialplan: exten = 99,1,Gotoif(?2:3) exten = 99,2,Meetme(100) exten = 99,3,Meetme(100|options) i'd like to do something like: exten = 99,1,Gotoif(?2:4) exten = 99,2,Meetme(100) exten = 99,4, ... exit ... exten = 99,3,Meetme(100|options) How can i exit the dialplan? I won't use meetme twice! Thanks nik ___ --Bandwidth 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] To 1.4 or not
I can only say that we use 1.4 in production (150 sip phones, sangoma and te410p cards, 75 agents 25+ queues). We do have a segfault now and then, but as I mentioned in a previous post, I think that the causes of that have been fixed in the 1.4 svn branch already. Julian. RR wrote: Hello Gents, following on this discussion, anyone particularly have one view or the other about 1.4 and the voicemail and meetme enhancements (supposedly) it has? We're not in production yet, I've tested 1.2 up until 1.2.13 in the lab as well as 1.4b3, since none of them got a real hammering Its hard to tell at the moment if one is more stable than the other. Also, since I don't use it for anything BUT voicemail and meetme, would a lot of instabilities in the PBX side of things affect me? They shouldn't but who knows. Any comments and/or advice would be appreciated :) \R ___ --Bandwidth 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] Stumped with Dial - $50 for answer
notransfer has been deprecated in 1.4 in favor of transfer ast_log(LOG_NOTICE, The option 'notransfer' is deprecated in favor of 'transfer' which has options 'yes', 'no', and 'mediaonly'\n); - Andrew Kohlsmith [EMAIL PROTECTED] wrote: On Tuesday 16 January 2007 10:09 am, Vicky wrote: its notransfer=yes in iax.conf not transfer=no :) Ahh yes. force consistency in the CLI where it doesn't necessarily belong, but use idiotic variable names in the config files. :-) -A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Jason Parker Digium ___ --Bandwidth 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] Stumped with Dial - $50 for answer
On Tuesday 16 January 2007 11:58 am, Jason Parker wrote: notransfer has been deprecated in 1.4 in favor of transfer ast_log(LOG_NOTICE, The option 'notransfer' is deprecated in favor of 'transfer' which has options 'yes', 'no', and 'mediaonly'\n); Sure, make an ass out of me, or rather Vicky eggs me on so I do it to myself. :-) I'm very glad to see that change. :-) -A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX Trunk timing
You need a timing device on both ends. Zoa Vicky wrote: If the other server doesnt have any hardware device that can act as timer. then just compile zaptel and modprobe ztdummy .. This kernel module should act as timing source i think . ( it works with meetme ) . On 16/01/07, *Andy Hester* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I have read that an IAX trunk requires a timing device. What wasn't clear to me was whether it is like TDM ie 1 timing device for the trunk, or if each end requires a timing device. I have a zaptel card in one server; do I have to have one in the second server in order to do an IAX trunk? I set up a trunk and so far calls can be made one way, but not the other. It is probably just not configured correctly, but I just wanted to make sure as I can't seem to find any reason at the moment. Thanks, Andy ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to detect long calls
We have been running an Asterisk box with 1.2.9.1 on it since August in a call center environment. We use the Asterisk box as an IVR and then pass the calls on to a Nortel Option 11C. Today we found in our long distance bill two calls that lasted a VERY long time. One was 58 hours and another was 38 DAYS!!! Nortel does not show this call being that long. Obviously the person that called in didn't hold the line for 58 days so somehow between Asterisk and MCI the call got stuck open and didn't hang up on the network. My question is two parts, part one, has anyone heard of anything like this where a call doesn't hang up properly and seems stuck in the system. Part two is there anyway to monitor in Asterisk the length of all active calls and then if a call lasts longer then, say one hour, we could send off a text message or warning. Any ideas or comments would be helpful Thanks _ Kevin Savoy Business Unit Telecom Analyst 2218 4th Ave W Williston, ND 58801 Ph: 701-774-4023 Fax: 701-774-2901 http://www.novo1.com http://www.novo1.com/ Novo 1 is a service mark of Novo 1, Inc image001.gif Description: image001.gif ___ --Bandwidth 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, SpanDSP and RXFax
Hey All, I am attempting to get the RXFax app working and having a hell of a time of it. I am hoping that some of you fine folks can help me out. I have installed Asterisk v1.2.14, SpanDSP v0.0.2pre26 and app_rxfax. All compiled and installed fine. When I attempt to call the extension I have created for receiving fax's then I get the following error once just as the rxfax application is invoked: Jan 16 09:29:11 NOTICE[5414]: channel.c:1950 ast_read: Dropping incompatible voice frame on SIP/192.168.2.250-b3203b30 of format slin since our native format has changed to ulaw Strange thing is that after that error Asterisk will sit and wait for the FAX to complete: Executing Application: (rxfax) Options: (/tmp/11689649465416/fax.tiff) When the fax is completed then the sending fax machine always says that the fax was sent successfully, but Asterisk errors out of the rxfax application and never writes the fax.tiff file. Has anyone seen this behavior before? Any help that you could provide would be very much appreciated. Thanks in advance! Darren Nay [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to detect long calls
Savoy, Kevin - Williston, ND wrote: We have been running an Asterisk box with 1.2.9.1 on it since August in a call center environment. We use the Asterisk box as an IVR and then pass the calls on to a Nortel Option 11C. Today we found in our long distance bill two calls that lasted a VERY long time. One was 58 hours and another was 38 DAYS!!! Nortel does not show this call being that long. Obviously the person that called in didn’t hold the line for 58 days so somehow between Asterisk and MCI the call got stuck open and didn’t hang up on the network. My question is two parts, part one, has anyone heard of anything like this where a call doesn’t hang up properly and seems “stuck” in the system. Part two is there anyway to monitor in Asterisk the length of all active calls and then if a call lasts longer then, say one hour, we could send off a text message or warning. Hi , similiar thing happend to me. Try looking at the L() optin in Dial. I define a max call time, say few hours, then warn every x seconds, then cut the call. -- thanks, Yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- 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
[asterisk-users] MP3player distortion with Asterisk 1.4
I upgraded my Asterisk configuration to 1.4.0 yesterday, when I was adding a TDM400P to have two PSTN connections to my analog phone lines. Adding the phone lines was a success, however, I now notice the MP3Player audio sounds horrible (incomprehensible). The only changes that I have made to the environment were the upgrade from 1.2 to 1.4. Any thoughts? ___ --Bandwidth 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] IAX Trunk timing
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Zoa Sent: Tuesday, January 16, 2007 11:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] IAX Trunk timing You need a timing device on both ends. Zoa But ztdummy should suffice yes? Andy ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RE: Asterisk, SpanDSP and RXFax
Hey All, Nevermind this question. I figured out that my problem was that I needed to downgrade my libtiff library to v3.7.1. My OS had installed 3.8.2 during system install and apparently spandsp doesn't like that version. It's all working perfectly now. Thanks in any case! Darren Nay From: Darren Nay Sent: Tuesday, January 16, 2007 10:25 AM To: 'asterisk-users@lists.digium.com' Subject: Asterisk, SpanDSP and RXFax Hey All, I am attempting to get the RXFax app working and having a hell of a time of it. I am hoping that some of you fine folks can help me out. I have installed Asterisk v1.2.14, SpanDSP v0.0.2pre26 and app_rxfax. All compiled and installed fine. When I attempt to call the extension I have created for receiving fax's then I get the following error once just as the rxfax application is invoked: Jan 16 09:29:11 NOTICE[5414]: channel.c:1950 ast_read: Dropping incompatible voice frame on SIP/192.168.2.250-b3203b30 of format slin since our native format has changed to ulaw Strange thing is that after that error Asterisk will sit and wait for the FAX to complete: Executing Application: (rxfax) Options: (/tmp/11689649465416/fax.tiff) When the fax is completed then the sending fax machine always says that the fax was sent successfully, but Asterisk errors out of the rxfax application and never writes the fax.tiff file. Has anyone seen this behavior before? Any help that you could provide would be very much appreciated. Thanks in advance! Darren Nay [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2 softphones can't (won't?) use PRI trunks....
On 16 Jan 2007, at 13:46, Patrick W. Foster wrote: I am hopeful that someone will recognize the issue and give me a pointer on where to look for the problem. - Registered IAX2 '4414' (AUTHENTICATED) at 192.168.1.102:4569 -- Accepting AUTHENTICATED call from 192.168.1.102: requested format = alaw, requested prefs = (), actual format = ulaw, host prefs = (ulaw|alaw|gsm), priority = mine Looks like a codec negotiation issue. The softphones are saying alaw (only), but your pri trunk is ulaw. Try enabling ulaw on the softphones. Tim Panton www.mexuar.net www.westhawk.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] IAX Trunk timing
On 12:06, Tue 16 Jan 07, Andy Hester wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Zoa Sent: Tuesday, January 16, 2007 11:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] IAX Trunk timing You need a timing device on both ends. Zoa But ztdummy should suffice yes? yes -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM2400 Hardware Echo Cancel
Adam Sharples wrote: I want to tune to echo canceller, but am unsure if any of the options provided have any effect on the hardware module. Do the settings such as echocancel and echotraining in Zapata.conf affect the hardware module? No. Hardware echo cancelers on Digium cards are either 'on' or 'off', there are no tuning parameters. ___ --Bandwidth 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] HowTo Config Asterisk and SS7
Though I haven't really tried it out myself, one option I've seen would be to use the same set of A100-series cards (the same one also being used for T1/E1/J1) from Sangoma to handle the physical port and their commercial SS7 gateway software for the out-of-band signalling. However, the SS7 Gateway is not really cheap and has to be setup by Sangoma personnel themselves. The SS7 Gateway will set you back around US$5K. Anyone else have an alternative. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nitesh Divecha Sent: Tuesday, January 16, 2007 11:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] HowTo Config Asterisk and SS7 Hello Asterisk, Can anyone help or put some light on, how can I configure Asterisk to work with SS7? What do I need, in terms of Hardware and Software? Regards, Neel ___ --Bandwidth 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] Stumped with Dial - $50 for answer - close
its notransfer=yes in iax.conf not transfer=no :) this is getting close! however, it takes about SEVEN seconds after the called party hangs up before the next priority is executed - same as with the T option. as contrast to h option, when called party hits asterisk, the next priority is almost immediate. the seven second delay makes the application very difficult to use. Expecting? Get great news right away with email Auto-Check. Try the Yahoo! Mail Beta. http://advision.webevents.yahoo.com/mailbeta/newmail_tools.html ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] How to detect long calls
You should: Set(TIMEOUT(absolute)=14400) When the call is received - this will set the maximum limit of a call and asterisk will force hang-up when the limit is reached. 14400 seconds = 4 hours, which for our purposes is longer then any call we expect. Even if you double-it or set it to several days some limit is better then nothing. When we found the same problem we had a call that was stuck open for 20 days. The call was stuck in a conference and was sending the on-hold music, which is what kept it open. Hope that helps. Cullin J. Wible -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of yusuf Sent: Tuesday, January 16, 2007 12:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to detect long calls Savoy, Kevin - Williston, ND wrote: We have been running an Asterisk box with 1.2.9.1 on it since August in a call center environment. We use the Asterisk box as an IVR and then pass the calls on to a Nortel Option 11C. Today we found in our long distance bill two calls that lasted a VERY long time. One was 58 hours and another was 38 DAYS!!! Nortel does not show this call being that long. Obviously the person that called in didn't hold the line for 58 days so somehow between Asterisk and MCI the call got stuck open and didn't hang up on the network. My question is two parts, part one, has anyone heard of anything like this where a call doesn't hang up properly and seems stuck in the system. Part two is there anyway to monitor in Asterisk the length of all active calls and then if a call lasts longer then, say one hour, we could send off a text message or warning. Hi , similiar thing happend to me. Try looking at the L() optin in Dial. I define a max call time, say few hours, then warn every x seconds, then cut the call. -- thanks, Yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- 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 ___ --Bandwidth 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] force ulaw passthrough if call from modem extension?
I have Teliax trunk set to ulaw and g729 and I have a modem/fax extension from a sipura forced to ulaw. When the call goes out through Teliax IAX trunk, asterisk transcodes to g729. Is there a way to tell asterisk not to transcode calls from/to a specific extension? I'm running asterisk 1.2.4 and that extension is for my home alarm/dish network and fax calls. Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Bootcamp in Pacific Northwest (Vancouver, BC)
Greetings, The District of North Vancouver, a municipal government in BC, Canada, is hosting a Digium instructed Asterisk Bootcamp at our training center from February 5th-9th, 2007. Primarily arranged to provide training to some of our staff, there is space available for others to avail of this opportunity to obtain Asterisk bootcamp training in the Pacific Northwest. Space on the course can be booked via the Digium web site at http://www.digium.com/en/training/locator/enroll/46. Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp ___ --Bandwidth 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] Outbound IVR for Asterisk
Hi, Someone knows an Open Source solution that can handle Outbound IVR for asterisk?. What I'm looking it a pre-preprogrammed a telephone call that reach a Person and start making an Interview over the telephone. Specifically I want to call all my customers exactly one hour after the service has been performed and ask some questions in an IVR, also the results of the Interview I will need them on a Database (MySQL) Best, Alex Never miss an email again! Yahoo! Toolbar alerts you the instant new Mail arrives. http://tools.search.yahoo.com/toolbar/features/mail/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to detect long calls
I had the same problem last year, at the time for some reason Timeout
instruction wouldn't trigger, so, just to be sure not to have to pay
for another longdistance call, I did the following, (following
someone's advise in here)
/usr/sbin/asterisk -rx show channels concise |awk -F : '($11 1500)
{print /usr/sbin/asterisk -rx \soft hangup $1 \} '|sh
this will hangup any call longer than 1500 seconds, or what ever
value you choose
hope it helps you somehow
;-)
Manrique
Cullin J. Wible escribió:
You should:
Set(TIMEOUT(absolute)=14400)
When the call is received - this will set the maximum limit of a call and
asterisk will force hang-up when the limit is reached.
14400 seconds = 4 hours, which for our purposes is longer then any call we
expect. Even if you double-it or set it to several days some limit is better
then nothing.
When we found the same problem we had a call that was stuck open for 20
days. The call was stuck in a conference and was sending the on-hold music,
which is what kept it open.
Hope that helps.
Cullin J. Wible
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of yusuf
Sent: Tuesday, January 16, 2007 12:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to detect long calls
Savoy, Kevin - Williston, ND wrote:
We have been running an Asterisk box with 1.2.9.1 on it since August
in a call center environment. We use the Asterisk box as an IVR and
then pass the calls on to a Nortel Option 11C. Today we found in our
long distance bill two calls that lasted a VERY long time. One was 58
hours and another was 38 DAYS!!!
Nortel does not show this call being that long. Obviously the person
that called in didn't hold the line for 58 days so somehow between
Asterisk and MCI the call got stuck open and didn't hang up on the
network.
My question is two parts, part one, has anyone heard of anything like
this where a call doesn't hang up properly and seems stuck in the
system. Part two is there anyway to monitor in Asterisk the length of
all active calls and then if a call lasts longer then, say one hour,
we could send off a text message or warning.
Hi ,
similiar thing happend to me. Try looking at the L() optin in Dial. I
define a max call time, say few hours, then warn every x seconds, then cut
the call.
--
thanks,
Yusuf
--
This message has been scanned for viruses and dangerous content by
MailScanner, and is believed to be clean.
--
**
Manrique Feoli
R D Director
[EMAIL PROTECTED]
Kínetos Software
www.kinetos.com
408-538-2113
**
___
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Re: [asterisk-users] Stumped with Dial - $50 for answer - close
On Tuesday 16 January 2007 2:31 pm, chester c young wrote: however, it takes about SEVEN seconds after the called party hangs up before the next priority is executed - same as with the T option. What kind of last leg are these calls? to POTS (even CAS T1) or PRI? as contrast to h option, when called party hits asterisk, the next priority is almost immediate. This is because Asterisk knows you want a hangup. My hunch is that you're terminating to POTS instead of PRI, and that is how long it takes for your telco provider to supply CPD signaling on the analog interface. I know Bell Canada is about that long. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Outbound IVR for Asterisk
Am Dienstag, den 16.01.2007, 12:01 -0800 schrieb Alejandro Duplat: Hi, Someone knows an Open Source solution that can handle Outbound IVR for asterisk?. What I'm looking it a pre-preprogrammed a telephone call that reach a Person and start making an Interview over the telephone. Specifically I want to call all my customers exactly one hour after the service has been performed and ask some questions in an IVR, also the results of the Interview I will need them on a Database (MySQL) If you are ready to write the extensions yourself (plus database logic), you can use .call files for that purpose. Have the one end (the later originator of the call be the customer, so that this customer will be run through the dialplan - where your IVR can work as usual. Sorry I do not know pre-fabricated solutions for that, neither commercial apps. BR Anselm ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.2.14 and Audiocodes Mediant 1000
I sent this yesterday, but saw zero traffic, so I think it got lost in the ether, so I'm sending again. I'm having an issue using Asterisk 1.2.14 and an Audiocodes Mediant 1000 ISDN gateway. For the most part, everything is working except for attended transfers. When I do an attended transfer, and complete the transfer before the 3rd party answers, the PSTN side hears dead air until the PSTN party answers or the transfer goes to voicemail. This happens regardless of whether I use the phone to do the transfer, or *2 to have Asterisk do it. Originally, it was actually disconnecting the call, but I fixed that by telling it not to disconnect on a broken connection, however that fact makes me think something is not quite right. Anyone else have experience with the Mediant gateways? Thanks, James ___ --Bandwidth 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] Ring tone too loud on IAX channel
Hi, We are using MozIAX as a softphone with a USB headset and are making outbound calls using IAX with ulaw encoding to our voip provider. We're running asterisk 1.4 Users are complaining that the ring tone generated by asterisk is much louder than the voice call once connected. They are having to turn the volume down to avoid being deafened by the ring tone, but then have an unacceptably low volume for the voice call. Can anyone suggest what might be the problem here, or steps I could take to address it? Thanks, Russell. ___ --Bandwidth 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] HowTo Config Asterisk and SS7
Gary Mensenares wrote: Though I haven't really tried it out myself, one option I've seen would be to use the same set of A100-series cards (the same one also being used for T1/E1/J1) from Sangoma to handle the physical port and their commercial SS7 gateway software for the out-of-band signalling. However, the SS7 Gateway is not really cheap and has to be setup by Sangoma personnel themselves. The SS7 Gateway will set you back around US$5K. Anyone else have an alternative. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nitesh Divecha Sent: Tuesday, January 16, 2007 11:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] HowTo Config Asterisk and SS7 Hello Asterisk, Can anyone help or put some light on, how can I configure Asterisk to work with SS7? What do I need, in terms of Hardware and Software? Regards, Neel ___ --Bandwidth 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 Thanks Gary, So can I use Digium T1/E1 card and interconnect using SS7? Please has anyone done installation with SS7 using Digium T1/E1 card? Regards, Neel ___ --Bandwidth 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 startup is slow
Hi, Having moved from asterisk-1.2.8 to 1.2.14, i've noticed that startup is much slower. In other words, if I say asterisk -R they type stop now, it takes on the order of 7 seconds instead of 1 second. The old asterisk startup printed out something like 650 lines, whereas the new one prints out lsomething like 720 lines. Have other people been seeing this? Is there something I have done wrong? As far as I know, both were built with make all make install (i.e. no compile time options being set differently). tia Mark ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Help with DISA
Hi, I'm trying to configure Asterisk and DISA. Asterisk is working, but I cannot have DISA dialing out. This is a snippet of my extensions.conf: [internal] exten = 1003,1,DISA(no-password|outgoing2) [outgoing2] exten = 1003,1,Playback(beep.gsm) exten = 1005,1,Playback(beep.gsm) My understanding is that if I dial the extension 1003, I should then be redirected to the context outgoing2 , and from there I will be able to dial extensions 1003 or 1005. If I can manage to sort out this dummy example I suppose I will be then able to solve the rest. Looking in the asterisk screen when I call, I see: Executing DISA(SIP/1001-c5bf, no-password|outgoing2) in new stack After that, nothing happens. I can't hear dialtones from Asterisk, but that's related to the fact that I do not have a sound card on the server. I have similar results if I set up a password. I can arrive to the new context, but cannot dial anything. Any suggestions? Thanks in advance, Andres ___ --Bandwidth 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] Polycom IP601 - some hints working, not others?
Hi, I've got an Asterisk setup including a TDM2400 for analog trunks extensions plus two IP501s an IP601 (all firmware 1.6.7 as supplied). The initial buddy / hint setup was fairly straightforward, but I have a strange problem in that some extensions don't show any status indication. Asterisk (V 1.2.13) CLI report for 'show hints' seems to indicate that the hints are set up correctly, but the phones are just not attempting to monitor certain extensions:- -= Registered Asterisk Dial Plan Hints =- 304 : ZAP/4 State:Idle Watchers 0 303 : ZAP/3 State:Idle Watchers 0 302 : ZAP/2 State:Idle Watchers 0 301 : ZAP/1 State:Idle Watchers 0 210 : ZAP/16State:Idle Watchers 0 209 : ZAP/15State:Idle Watchers 0 208 : ZAP/14State:Idle Watchers 3 207 : ZAP/13State:Idle Watchers 3 206 : ZAP/12State:Idle Watchers 3 205 : ZAP/11State:Idle Watchers 0 204 : ZAP/10State:Idle Watchers 3 203 : ZAP/9 State:Idle Watchers 0 202 : SIP/202 State:Idle Watchers 3 201 : SIP/201 State:Idle Watchers 3 200 : SIP/200 State:Idle Watchers 3 The (mac)-directory.xml files have all the extensions in, in identical format, but the phones simply don't seem to be subscribing to certain 'buddys' to show the status. I've tried deleting directory entries at both an IP501 and the IP601 and re-creating them, with without rebooting, but with no effect. All entries have buddy watch enabled. The list of working / not working indications is consistent across reboots of both the phones and the Asterisk PC. Any ideas appreciated, Robert Jenkins. ___ --Bandwidth 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] force ulaw passthrough if call from modem extension?
On 16 Jan 2007, at 19:56, Victor Perez wrote: I have Teliax trunk set to ulaw and g729 and I have a modem/fax extension from a sipura forced to ulaw. When the call goes out through Teliax IAX trunk, asterisk transcodes to g729. Is there a way to tell asterisk not to transcode calls from/to a specific extension? try creating a separate (duplicate) entry in iax.conf for the teliax connection disallow 729 on that trunk and use it for fax/alarm calls. Tim Panton www.mexuar.net www.westhawk.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] Ring tone too loud on IAX channel
On 16 Jan 2007, at 20:33, Russell Horn wrote: Hi, We are using MozIAX as a softphone with a USB headset and are making outbound calls using IAX with ulaw encoding to our voip provider. We're running asterisk 1.4 Users are complaining that the ring tone generated by asterisk is much louder than the voice call once connected. They are having to turn the volume down to avoid being deafened by the ring tone, but then have an unacceptably low volume for the voice call. Can anyone suggest what might be the problem here, or steps I could take to address it? There are 2 places the ring tone could be generated, either in asterisk - then sent to the softphone as audio media, or alternatively in the softphone itself - in response to asterisk sending a 'ringing' message. Tim Panton www.mexuar.net www.westhawk.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] Audiocodes GPL
I have some Audiocodes units which appear to be running Linux, according to the unit's own System Log kern.warn Linux version 2.4.21openrg-rmk1 #2 Wed Aug 30 17:05:29 IDT 2006 However my contact at Audiocodes claims otherwise On 12/4/06, Yaniv Nizan [EMAIL PROTECTED] wrote: I doubt that we are running Linux on the MP-202. Perhaps there is a reference to the OS on the PC that configures the device So a few questions: 1) Does anyone know if the older Audiocodes devices (such as the multiport gateways) run Linux as well? 2) What does one go about doing to correct GPL violations? Perhaps someone has a generic legal letter that can be used in these situations? ___ --Bandwidth 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] Practical limit on dial prefixes for a route
I'm aware of Cingular being GSM. We're standardizing on Sprint since Cingular is less than optimal around here. Even with LNP, knowing the NPA-NXX would nail probably 90%+ of our people. The ones that are on LNP could be added as 10 digit LCR. From a technical standpoint, can * handle over 1000+ prefixes on a route? EKG -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Novack Sent: Monday, January 15, 2007 9:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Practical limit on dial prefixes for a route Eric Germann wrote: Colleagues, We're in the process of standardizing on Sprint PCS and Cingular phones on a national basis (~ 50 properties, 1000's of lines). I manage an Asterisk install at one location. I've been looking at the Multitech CellFinder CDMA for Sprint as a dial backup solution. Basically, it's a CDMA to POTS gateway, tied to a PCS account. We would see it as a trunk line and I would like to do LCR and route out the CellFinder line(s)^ all PCS calls, since we have free PCS to PCS. Two comments: Cingular is GSM, Sprint is CDMA With LNP , NPA-NXX isn't enough information to determine free on network calling Since wireline to wireless LNP, the NPA assignments are no longer locked to a specific carrier. John Novack ___ --Bandwidth 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] Diff. Btn TE405P and TE410P
Hello Asterisk, Please can anyone explain what is the difference between TE405P and TE410P? According to the data sheet, the difference I see is the PCI voltage. TE405P use only 5.0 volt PCI slot and TE410P use only 3.3 volt PCI. Regards, Nitesh ___ --Bandwidth 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] Practical limit on dial prefixes for a route
On Tue, 16 Jan 2007, Eric Germann wrote: I'm aware of Cingular being GSM. We're standardizing on Sprint since Cingular is less than optimal around here. Where is here? Planet Earth? Down here (San Diego), Cingular advertises they have the fewest dropped calls. I think they pulled a Clinton and re-defined dropped to mean when the customer dropped the handset -- probably out of surprise from connecting the call :) 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
Re: [asterisk-users] Asterisk HA
2007/1/11, Ale [EMAIL PROTECTED]: Ciao, Enrico Pasqualotto wrote: Is better ultramonkey, dundi or SER proxy in front of * server? You can also consider Hartbeat + rsync, or simply pfsync + rsync ;) The problem with Asterisk HA, is mainly the lost of calls when failover occurs. This is because all traffic pass through Asterisk always. In order to solve this, you could use SER + Asterisk + OpenSER. That way, you'll only lose calls that are going outside your network, but calls inside will remain. -- Diego Quintana a.k.a. RouterMaN Ingeniería de las Telecomunicaciones PUCP Linux Registered User #382615 - http://counter.li.org/ SIP # 1-747-633-6676 Ext. 1011 FWD # 764839 Ext. 1011 http://routerman.blogsome.com http://planeta.debianperu.org ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: How to detect long calls
KS == Savoy, Kevin - Williston, ND [EMAIL PROTECTED] writes: KS We have been running an Asterisk box with 1.2.9.1 on it since KS August in a call center environment. We use the Asterisk box as an KS IVR and then pass the calls on to a Nortel Option 11C. Today we KS found in our long distance bill two calls that lasted a VERY long KS time. One was 58 hours and another was 38 DAYS!!! There has been some excellent suggestions in this thread. I just want to add one. Sometimes a SIP packet can get lost or a phone rebooted without closing a call properly. Then the call will just stay open forever. You can solve that with rtptimeout=3600 or something similar in sip.conf. Obviously it only works when the rtp stream is actually going through Asterisk, and it will also kill the call if a Snom phone turns the microphone off for an hour (Snom phones do silence suppression unconditionally when you press the mute button). Still, even with those limitations it works nicely for us. At least as a stopgap until Asterisk gets TCP-support for SIP. /Benny ___ --Bandwidth 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] Diff. Btn TE405P and TE410P
Hi Nitesh - Please can anyone explain what is the difference between TE405P and TE410P? According to the data sheet, the difference I see is the PCI voltage. TE405P use only 5.0 volt PCI slot and TE410P use only 3.3 volt PCI. That is the only difference. - Noah ___ --Bandwidth 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] Audiocodes GPL
Andrew Joakimsen wrote: 2) What does one go about doing to correct GPL violations? Perhaps someone has a generic legal letter that can be used in these situations? Only a copyright holder whose code is being used outside the terms of the GPL can pursue action against the violator. ___ --Bandwidth 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] Polycom IP601 - some hints working, not others?
Are all of the sip phones in the same context? -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Robert Jenkins Sent: Tuesday, January 16, 2007 1:44 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Polycom IP601 - some hints working, not others? Hi, I've got an Asterisk setup including a TDM2400 for analog trunks extensions plus two IP501s an IP601 (all firmware 1.6.7 as supplied). The initial buddy / hint setup was fairly straightforward, but I have a strange problem in that some extensions don't show any status indication. Asterisk (V 1.2.13) CLI report for 'show hints' seems to indicate that the hints are set up correctly, but the phones are just not attempting to monitor certain extensions:- -= Registered Asterisk Dial Plan Hints =- 304 : ZAP/4 State:Idle Watchers 0 303 : ZAP/3 State:Idle Watchers 0 302 : ZAP/2 State:Idle Watchers 0 301 : ZAP/1 State:Idle Watchers 0 210 : ZAP/16State:Idle Watchers 0 209 : ZAP/15State:Idle Watchers 0 208 : ZAP/14State:Idle Watchers 3 207 : ZAP/13State:Idle Watchers 3 206 : ZAP/12State:Idle Watchers 3 205 : ZAP/11State:Idle Watchers 0 204 : ZAP/10State:Idle Watchers 3 203 : ZAP/9 State:Idle Watchers 0 202 : SIP/202 State:Idle Watchers 3 201 : SIP/201 State:Idle Watchers 3 200 : SIP/200 State:Idle Watchers 3 The (mac)-directory.xml files have all the extensions in, in identical format, but the phones simply don't seem to be subscribing to certain 'buddys' to show the status. I've tried deleting directory entries at both an IP501 and the IP601 and re-creating them, with without rebooting, but with no effect. All entries have buddy watch enabled. The list of working / not working indications is consistent across reboots of both the phones and the Asterisk PC. Any ideas appreciated, Robert Jenkins. ___ --Bandwidth 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] Stumped with Dial - $50 for answer - closed!
the answer sucks, but is apparently correct. imho Andrew Kohlsmith is The Man, although there was someone in Germany who emailed about the T option which actually works about as well - please email me. Andrew Kohlsmith please email me. Will pay paypal if that's ok. --- Andrew Kohlsmith [EMAIL PROTECTED] wrote: On Tuesday 16 January 2007 2:31 pm, chester c young wrote: however, it takes about SEVEN seconds after the called party hangs up before the next priority is executed - same as with the T option. What kind of last leg are these calls? to POTS (even CAS T1) or PRI? as contrast to h option, when called party hits asterisk, the next priority is almost immediate. This is because Asterisk knows you want a hangup. My hunch is that you're terminating to POTS instead of PRI, and that is how long it takes for your telco provider to supply CPD signaling on the analog interface. I know Bell Canada is about that long. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Need Mail bonding? Go to the Yahoo! Mail QA for great tips from Yahoo! Answers users. http://answers.yahoo.com/dir/?link=listsid=396546091 ___ --Bandwidth 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] Audiocodes GPL
Andrew Joakimsen wrote: I have some Audiocodes units which appear to be running Linux, according to the unit's own System Log kern.warn Linux version 2.4.21openrg-rmk1 #2 Wed Aug 30 17:05:29 IDT 2006 Googling turns up: http://www.jungo.com/openrg/openrg.html OpenRG is a Linux based device platform. So, Audiocodes probably licensed it from Jungo. Just because the unit runs Linux, doesn't necessarily imply that there's a GPL violation. Leo ___ --Bandwidth 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] Polycom IP601 - some hints working, not others?
Hi, Yes, there are just the three Polycoms (200 - 202), the rest of the system is analog. The Polycoms always 'see' each other, the problem is with them seeing some Zap channels. Although the 501s don't have the display of the 601 plus sidecar, from Asterisks point of view the 'watchers' count is always 3 or 0, so all the Polycom phones appear to be behaving identically, all or none show status, in respect of each contact. (In the 'Show Hints' list, the lines with 'Watchers 0' are the extensions that don't show status). Robert Jenkins. -Original Message- From: Damon Estep [mailto:[EMAIL PROTECTED] Sent: 16 January 2007 22:46 To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Polycom IP601 - some hints working, not others? Are all of the sip phones in the same context? -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Robert Jenkins Sent: Tuesday, January 16, 2007 1:44 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Polycom IP601 - some hints working, not others? Hi, I've got an Asterisk setup including a TDM2400 for analog trunks extensions plus two IP501s an IP601 (all firmware 1.6.7 as supplied). The initial buddy / hint setup was fairly straightforward, but I have a strange problem in that some extensions don't show any status indication. Asterisk (V 1.2.13) CLI report for 'show hints' seems to indicate that the hints are set up correctly, but the phones are just not attempting to monitor certain extensions:- -= Registered Asterisk Dial Plan Hints =- 304: ZAP/4 State:Idle Watchers 0 303: ZAP/3 State:Idle Watchers 0 302: ZAP/2 State:Idle Watchers 0 301: ZAP/1 State:Idle Watchers 0 210: ZAP/16State:Idle Watchers 0 209: ZAP/15State:Idle Watchers 0 208: ZAP/14State:Idle Watchers 3 207: ZAP/13State:Idle Watchers 3 206: ZAP/12State:Idle Watchers 3 205: ZAP/11State:Idle Watchers 0 204: ZAP/10State:Idle Watchers 3 203: ZAP/9 State:Idle Watchers 0 202: SIP/202 State:Idle Watchers 3 201: SIP/201 State:Idle Watchers 3 200: SIP/200 State:Idle Watchers 3 The (mac)-directory.xml files have all the extensions in, in identical format, but the phones simply don't seem to be subscribing to certain 'buddys' to show the status. I've tried deleting directory entries at both an IP501 and the IP601 and re-creating them, with without rebooting, but with no effect. All entries have buddy watch enabled. The list of working / not working indications is consistent across reboots of both the phones and the Asterisk PC. Any ideas appreciated, Robert Jenkins. ___ --Bandwidth 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] Audiocodes GPL
Am Mittwoch, den 17.01.2007, 07:38 +0800 schrieb Leo Ann Boon: Andrew Joakimsen wrote: I have some Audiocodes units which appear to be running Linux, according to the unit's own System Log kern.warn Linux version 2.4.21openrg-rmk1 #2 Wed Aug 30 17:05:29 IDT 2006 Googling turns up: http://www.jungo.com/openrg/openrg.html OpenRG is a Linux based device platform. So, Audiocodes probably licensed it from Jungo. Just because the unit runs Linux, doesn't necessarily imply that there's a GPL violation. Surely not. Linux is intented to be used in proprietary hardware, applications et cetera. But, if I am not mistaken, if a device uses any GPL'd software, this must be clearly stated by the vendor, a copy of the GPL must be handed along with the device and you have the right to obtain a copy of all open source source code files involved in the project, for a marginal charge. Outright denial of the usage of Linux in such a device seems to not comply with that. If you intend to pursue this, you could try to find information on www.gpl-violations.org (and no, this is not an organisation that helps to violate the GPL ;-) BR Anselm ___ --Bandwidth 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] Audiocodes Mediant 1000, Polycom, and no ringback on transfer
On Mon, 2007-01-15 at 15:26 -0600, David Gomillion wrote: I don't think you can do that. Here's why: on the Polycom's, the Transfer button doesn't reappear until the transferree picks up the phone. Unless something changed in the firmware recently. But, if you're completing it before the 3rd party answers, it's not an attended transfer. I found it all depends on the dialplan, and the sip.cfg for the phone. If you call Answer() before Dial(), it will allow it. There is also a setting in sip.cfg for the phones, voIpProt.SIP.allowTransferOnProceeding that I think allows that as well. I should have mentioned in my original post, but MOH works just fine. When I complete the transfer, the MOH stops, and that's when the dead air starts. Anyone else have any suggestions? Thanks, James ___ --Bandwidth 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] Audiocodes GPL
There is nothing in the GPL that prohibits you from selling the software (RedHat Software). There is also nothing stops a sales person from denying it. They must provide a copy of the GPL and they must give you the source code and related modifications if you ask (not sure if you have). There are other terms and depending on who you ask, lots of interpretation. If you truly believe there is a violation (which I doubt) you should contact the Free Software Foundation - they wrote the license. Cullin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anselm Martin Hoffmeister Sent: Tuesday, January 16, 2007 6:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Audiocodes GPL Am Mittwoch, den 17.01.2007, 07:38 +0800 schrieb Leo Ann Boon: Andrew Joakimsen wrote: I have some Audiocodes units which appear to be running Linux, according to the unit's own System Log kern.warn Linux version 2.4.21openrg-rmk1 #2 Wed Aug 30 17:05:29 IDT 2006 Googling turns up: http://www.jungo.com/openrg/openrg.html OpenRG is a Linux based device platform. So, Audiocodes probably licensed it from Jungo. Just because the unit runs Linux, doesn't necessarily imply that there's a GPL violation. Surely not. Linux is intented to be used in proprietary hardware, applications et cetera. But, if I am not mistaken, if a device uses any GPL'd software, this must be clearly stated by the vendor, a copy of the GPL must be handed along with the device and you have the right to obtain a copy of all open source source code files involved in the project, for a marginal charge. Outright denial of the usage of Linux in such a device seems to not comply with that. If you intend to pursue this, you could try to find information on www.gpl-violations.org (and no, this is not an organisation that helps to violate the GPL ;-) BR Anselm ___ --Bandwidth 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] Stumped with Dial - $50 for answer - closed!
Am Dienstag, den 16.01.2007, 15:04 -0800 schrieb chester c young: the answer sucks, but is apparently correct. If your application involves the caller (e.g. an employee of your company) to rate the call he just did, or to enter any data to a mysql database over the phone right after the call, you could use the H option (neither T nor h, then) and tell your phone personell about it: After the call finished, press * and answer the questions the computer reads out to you. That way, Asterisk would (expectedly) stay in the Audio path and even find out that the call ended if your employee did not *g* - and your employees could cut those 7 second delays. Your IVR for aprés-call interaction should skip the first digit if it happens to be an * though, because it could happen that Asterisk sees the far end hangup just a blink before the user hits the * key. imho Andrew Kohlsmith is The Man, although there was someone in Germany who emailed about the T option which actually works about as well - please email me. Andrew Kohlsmith please email me. Will pay paypal if that's ok. If you mean me (being in Germany and all that), and if you intend to hand out any money to me (which is not absolutely clear from that statement), please donate to openvpn.org - they accept paypal :-). It is one of the many open source projects whose software I use regularly and have no time ressources (let us not talk about skills :-) to contribute to. BR Anselm ___ --Bandwidth 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] Polycom IP601 - some hints working, not others?
You are trying to subscribe to a non SIP channel? Not sure that can be done...never tried. -Original Message- From: Robert Jenkins [mailto:[EMAIL PROTECTED] Sent: Tuesday, January 16, 2007 4:43 PM To: Damon Estep; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Polycom IP601 - some hints working, not others? Hi, Yes, there are just the three Polycoms (200 - 202), the rest of the system is analog. The Polycoms always 'see' each other, the problem is with them seeing some Zap channels. Although the 501s don't have the display of the 601 plus sidecar, from Asterisks point of view the 'watchers' count is always 3 or 0, so all the Polycom phones appear to be behaving identically, all or none show status, in respect of each contact. (In the 'Show Hints' list, the lines with 'Watchers 0' are the extensions that don't show status). Robert Jenkins. -Original Message- From: Damon Estep [mailto:[EMAIL PROTECTED] Sent: 16 January 2007 22:46 To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Polycom IP601 - some hints working, not others? Are all of the sip phones in the same context? -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Robert Jenkins Sent: Tuesday, January 16, 2007 1:44 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Polycom IP601 - some hints working, not others? Hi, I've got an Asterisk setup including a TDM2400 for analog trunks extensions plus two IP501s an IP601 (all firmware 1.6.7 as supplied). The initial buddy / hint setup was fairly straightforward, but I have a strange problem in that some extensions don't show any status indication. Asterisk (V 1.2.13) CLI report for 'show hints' seems to indicate that the hints are set up correctly, but the phones are just not attempting to monitor certain extensions:- -= Registered Asterisk Dial Plan Hints =- 304: ZAP/4 State:Idle Watchers 0 303: ZAP/3 State:Idle Watchers 0 302: ZAP/2 State:Idle Watchers 0 301: ZAP/1 State:Idle Watchers 0 210: ZAP/16State:Idle Watchers 0 209: ZAP/15State:Idle Watchers 0 208: ZAP/14State:Idle Watchers 3 207: ZAP/13State:Idle Watchers 3 206: ZAP/12State:Idle Watchers 3 205: ZAP/11State:Idle Watchers 0 204: ZAP/10State:Idle Watchers 3 203: ZAP/9 State:Idle Watchers 0 202: SIP/202 State:Idle Watchers 3 201: SIP/201 State:Idle Watchers 3 200: SIP/200 State:Idle Watchers 3 The (mac)-directory.xml files have all the extensions in, in identical format, but the phones simply don't seem to be subscribing to certain 'buddys' to show the status. I've tried deleting directory entries at both an IP501 and the IP601 and re-creating them, with without rebooting, but with no effect. All entries have buddy watch enabled. The list of working / not working indications is consistent across reboots of both the phones and the Asterisk PC. Any ideas appreciated, Robert Jenkins. ___ --Bandwidth 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] ERROR[2453]: chan_zap.c:8142 zt_pri_error: !! Unexpected Channel selection 3
Hi guys, I did an upgrade on one asterisk from 1.2.14 to 1.4.0, after this, all calls originated from PBX trunked with asterisk through TE110 board i receive this message: [Jan 16 21:19:42] ERROR[2453]: chan_zap.c:8142 zt_pri_error: !! Unexpected Channel selection 3 the call was completed and two ends talks normaly, the only incovenient is that message. Anybody know why this message appear ? Thanks. -- Frederico Madeira [EMAIL PROTECTED] www.madeira.eng.br ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] How to detect long calls
show channels will display all calls including Duration and BridgeTo. You can check the BridgeTo column to determine if one call leg is still attached to the other. If that fails, you can also check the duration for hung calls. To automate, there are a number of approaches. I personally suggest looking into writing/deploying an Asterisk manager. _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Savoy, Kevin - Williston, ND Sent: Wednesday, January 17, 2007 2:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] How to detect long calls We have been running an Asterisk box with 1.2.9.1 on it since August in a call center environment. We use the Asterisk box as an IVR and then pass the calls on to a Nortel Option 11C. Today we found in our long distance bill two calls that lasted a VERY long time. One was 58 hours and another was 38 DAYS!!! Nortel does not show this call being that long. Obviously the person that called in didn't hold the line for 58 days so somehow between Asterisk and MCI the call got stuck open and didn't hang up on the network. My question is two parts, part one, has anyone heard of anything like this where a call doesn't hang up properly and seems stuck in the system. Part two is there anyway to monitor in Asterisk the length of all active calls and then if a call lasts longer then, say one hour, we could send off a text message or warning. Any ideas or comments would be helpful Thanks _ Kevin Savoy Business Unit Telecom Analyst 2218 4th Ave W Williston, ND 58801 Ph: 701-774-4023 Fax: 701-774-2901 http://www.novo1.com/ http://www.novo1.com Novo 1 is a service mark of Novo 1, Inc image001.gif Description: GIF image ___ --Bandwidth 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] Absolute Timeout or Dial Limit option???
I need a method of limiting the duration of calls when RTP media does NOT travel through Asterisk. I know that the Dial() command limit option L requires Asterisk to carry the media, but what about Set(TIMEOUT(absolute)=XX)? Are there any other apps/options that might work for this? Thanks! David ___ --Bandwidth 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] prompt for send a message not played in VM main, HOWTO resolve
All, Just came across the prompt #3 from inside the top menu of VM in latest stable. Allison does not announce the prompt, but if you know it is there, you can press 3 successfully follow the prompts from there to send your message to other users on the system. But, of course, obviously, I am asking: how do I resolve the situation whereby the users are not hearing this prompt? (since most nearly all users will never know that this is here) (I sure hope my googling didnt miss this one) Thanks very much. Most appreciated. Jason Sjobeck ___ --Bandwidth 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] Really Big Queues
Hi, How do you folks handle really large queues (350+ simultaneous callers) in your Asterisk PBXes? We're going to be bringing in around 16 PRIs' worth of inbound callers, doing skills-based routing, and queuing them up for approximately 200 agents. What's the best way to handle all of these callers? We want to record the calls and we'll probably use the ramdisk method that has been discussed on this list. Here's some ideas that I'm considering: Idea #1: Use servers with (2) Digium 4-port PRI cards, running Asterisk, as media gateways. From here, send calls to 2 or more Asterisk queue servers. For each incoming call, run an AGI on the media gateways that determines which queue server is least loaded. Send this incoming call to the queue server over an IAX2 trunk. The problem with this method is that the queues are not unified; if one queue server suddenly has available agents, queued callers on the other queue server cannot be (easily?) transfered to the server with available agents. Also, running an AGI for each incoming call is lame and slow. Idea #2: Use 3com VCX V7122 media gateways to terminate the PRIs and send the calls to a load balanced pair of SER proxies. These proxies will somehow keep track of the state of the Asterisk queue servers and redirect the incoming calls to the least loaded (most available) queue server. The problem with this method is that, by using SIP, we'll probably see higher interrupt load on the Asterisk queue servers. Additionally, I'm not a SER expert yet and I have no idea how to get SER to monitor the state of the Asterisk queue servers. As with Idea #1, the queues are also not unified, which sucks. Idea #3: ??? (profit!) Do you fine folks have any ideas or suggestions? thanks, 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] How to detect long calls
Hi guys, Look my example: pabx*CLI sip show channels Peer User/ANRCall ID Seq (Tx/Rx) Form Hold Last Message 64.71.xx.xx322121226ee03b46000 00103/15992 unkn No (d) Rx: BYE 64.71.xx.xx0113941735 57344d766af 00103/0 unkn No Tx: INVITE 64.71.xx.xx0113677599 5456e05e17d 00103/0 unkn No Tx: INVITE 64.71.xx.xx0113388754 3fe71d9114a 00103/0 unkn No Tx: INVITE 64.71.xx.xx0113388754 75c54f392c3 00103/0 unkn No Tx: INVITE 64.71.xx.xx0113677599 22fe2ae1237 00103/0 unkn No Tx: INVITE 64.71.xx.xx0823241639 3b99e044545 00103/0 unkn No Tx: INVITE 64.71.xx.xx0823231223 4345657f406 00103/0 unkn No Tx: INVITE 64.71.xx.xx0823327211 5516645b4b7 00103/0 unkn No Tx: INVITE 64.71.xx.xx0823336651 5692acca779 00103/0 unkn No Tx: INVITE 64.71.xx.xx0823235526 14b7d28729f 00103/0 unkn No Tx: INVITE 64.71.xx.xx0793246319 3fe706487f1 00103/0 unkn No Tx: INVITE 64.71.xx.xx0613364414 13ea2109500 00103/0 unkn No Tx: INVITE 64.71.xx.xx0613364414 531f94b42c4 00103/0 unkn No Tx: INVITE 14 active SIP channels I can confirm that when i run this command, no one was in the office. What is this status ? Where can i see duration of this calls ? How can i kill them ? Thanks. Fred Em Ter, 2007-01-16 às 11:08 -0600, Savoy, Kevin - Williston, ND escreveu: We have been running an Asterisk box with 1.2.9.1 on it since August in a call center environment. We use the Asterisk box as an IVR and then pass the calls on to a Nortel Option 11C. Today we found in our long distance bill two calls that lasted a VERY long time. One was 58 hours and another was 38 DAYS!!! Nortel does not show this call being that long. Obviously the person that called in didn’t hold the line for 58 days so somehow between Asterisk and MCI the call got stuck open and didn’t hang up on the network. My question is two parts, part one, has anyone heard of anything like this where a call doesn’t hang up properly and seems “stuck” in the system. Part two is there anyway to monitor in Asterisk the length of all active calls and then if a call lasts longer then, say one hour, we could send off a text message or warning. Any ideas or comments would be helpful Thanks _ Kevin Savoy Business Unit Telecom Analyst 2218 4th Ave W Williston, ND 58801 Ph: 701-774-4023 Fax: 701-774-2901 http://www.novo1.com Novo 1 is a service mark of Novo 1, Inc ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users image001.gif Description: image001.gif signature.asc Description: Esta é uma parte de mensagem assinada digitalmente ___ --Bandwidth 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] Stumped with Dial - $50 for answer - closed!
$25 to openvpn.org - thanks to Anselm Martin Hoffmeister --- Anselm Martin Hoffmeister [EMAIL PROTECTED] wrote: Am Dienstag, den 16.01.2007, 15:04 -0800 schrieb chester c young: the answer sucks, but is apparently correct. If your application involves the caller (e.g. an employee of your company) to rate the call he just did, or to enter any data to a mysql database over the phone right after the call, you could use the H option (neither T nor h, then) and tell your phone personell about it: After the call finished, press * and answer the questions the computer reads out to you. That way, Asterisk would (expectedly) stay in the Audio path and even find out that the call ended if your employee did not *g* - and your employees could cut those 7 second delays. Your IVR for aprés-call interaction should skip the first digit if it happens to be an * though, because it could happen that Asterisk sees the far end hangup just a blink before the user hits the * key. This is for volunteers calling other members of their organization, so need to keep everything low key and polite. A volunteer will call in, either by POT or SIP and will stay connected as Asterisk dials the number of the fellow member whom they've selected on a browser. The seven seconds is bad because that's a bit too long between calls - people tend to loose their concentration. Be a PS3 game guru. Get your game face on with the latest PS3 news and previews at Yahoo! Games. http://videogames.yahoo.com/platform?platform=120121 ___ --Bandwidth 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] Refreshing DNS lookups
Hi there The dnsmgr in Aterisk 1.4.0 seems not to work. I enabled DNS lookups in dnsmgr.conf but after reloading the conf files * never refreshes DNS lookups. Any ideas how to debug this issue? Thanks in advance Housi Mueller - Don't pick lemons. See all the new 2007 cars at Yahoo! Autos.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Realtime Voicemail Password Change Not Working
Hi All, I'm using asterisk 1.2.9.1 and mysql 3.23, asterisk add-ons 1.2.3. All seems to work normally with realtime voicemail, reads vmbox parameters from the db fine. When I try to change the password, asterisk operates normally, enter new password ok, re-enter new password ok, password has been changed There are no entries in the mysql.log setting the new password in the database. How can I isolate between asterisk, realtime driver, and mysql? Thanks. 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] Really Big Queues
On Tue, 16 Jan 2007, Christopher Snell wrote: Idea #1: Use servers with (2) Digium 4-port PRI cards, running Asterisk, as media gateways. From here, send calls to 2 or more Asterisk queue servers. For each incoming call, run an AGI on the media gateways that determines which queue server is least loaded. Send this incoming call to the queue server over an IAX2 trunk. The problem with this method is that the queues are not unified; if one queue server suddenly has available agents, queued callers on the other queue server cannot be (easily?) transfered to the server with available agents. Also, running an AGI for each incoming call is lame and slow. This is similar to what I am doing now. I have 3 1u's with a single Digium 4 port PRI card. Each server services 2 T1's. This configuration was based on management's tolerance for in-flight call revenue loss, not CPU capacity. The telco servers dial a single application server via IAX. I disagree with lame and slow. In my system, 3 AGI's are executed on the telco servers before the call is handed over to the application server and 1 more when the call is hung up. 8 AGI's are executed on the application server before the caller connects to the product of their choice and 2 more when the call is hung up. All of the boxes are 3 gHz Intel's. This system handles a load of about 15,000 calls a day with about 100 simultaneous callers and based on top could handle several times that load -- most of the boxes run about 80%+ idle. Just for grins, I wrote an AGI (in C, like all my AGI's) that just reads the AGI environment from STDIN, parses it, and stuff's the strings into a structure, just like any real AGI would and then exits. I find that a 3 gHz non-HT Intel can execute about 300 noop AGI's per second. For comparison, this same box can execute the noop application about 6,000 times per second. This is not to say that an application executes 20 times faster than an AGI, just that the invocation of an AGI is much more expensive than an application because it is a separate process and communicates with Asterisk via STDIN/STDOUT. Once the AGI is started, I believe the execution time of the meat of the AGI should be similar to that of an application. AGI's: ) don't crash Asterisk if you make a simple coding error. ) take effect as soon as you move them to the agi-bin directory. ) never require a reload or a restart. ) can be coded in a variety of languages by programmers with meager coding skills. ) are less complex because they only have to deal with a single thread of execution. ) are quicker to develop. AGI's rock and are appropriate for many applications. 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
Re: [asterisk-users] SPA-941 (and others ) Transmit Sound Quality
I too seem to have the same problem, dont know about poor quality but its certainly not loud enough, I have to put my mouth to the microphone, otherwise the other end reports they cannot hear me. This does however seem to do a good job to cancel out the background noise On 11/10/06, Ron Winograd [EMAIL PROTECTED] wrote: Hello, This is not exactly an Asterisk question, but I was encouraged to seek advice here anyway. The kindness of the * open source community is legendary :) I am getting going with an Asterisk 1.2 box, and I'm having trouble getting good quality transmit sound using handsets with VoIP phones. I'm primarily trying to focus on SPA-941, but also experimenting with Aastra 9113i and Uniden UIP1868. I do not at this time have any PSTN cards in the box to provide hardware timing. The use case is calling from the SIP phones (which are extensions registered with the * 1.2 box) to a VoIP termination service which routes the call to a PSTN number. Everything sounds great on the SIP phone, but the sound on the other end of the line is distant and missing bass, most especially so on the SPA-941 (which is the phone we really want to use). If I use the default handset mic gain value of 0db, the sound is so loud for the other person they have to hold the phone away from their ear. If I set it to -6db, it is still too quiet. The Aastra 9113i sounds a little better, and the Uniden 5.4 GHz Cordless sounds actually very good, so I'm pretty sure my network setup is capable of transmitting good sound. Using the speaker-phone on the SPA-941 sounds significantly better than using the handset. But we need the handset to also sound good. I've tried different providers etc. and always come back to the phone. I'm using G711u codec in all cases and silence suppresion is off. I saw a previous thread that mentioned changing the RTP from .03 to .02, however the post was regarding a MeetMe issue. I tried anyway, and it introduced an echo on the line. I've seen many rave reviews regarding the sound quality on the SPA-941, so I'm wondering if maybe I got a bum handset? Would anyone be willing to receive/place a call to tell me if it sounds the way its supposed to or if there is indeed a problem? All suggestions/recommendations greatly appreciated. Much thanks, -- Ron [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth 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] TDM404B VS TDM2401B
Hi List, any good comparison between TDM404B and TDM2401B . i'm not very happy with TDM404B voice quality, low volume and sometimes echo. I was wondering if any of you guys have good experience with TDM2401B. thanks!___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dell 860
Hi All, I'm having some troubles with my Dell 860 and TE110P card. Using Asterisk 1.2.14, Zaptel 1.2.12 and Libpri 1.2.4. I'm getting digital noise, like a half ring almost and other jitter. Here's the kicker it's only on the outside part of the call. Ie. if I rang you, you would here it but I don't and the opposite if you rang me you would hear it but I wouldn't. I have tried two different cards and different E1 lines still the same thing? I'm going to try the two port card soon but I don't think that will fix my problem. Is it just one dodgy server or are all 860's no good? Thanks for your help. Joel. ___ --Bandwidth 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] SPA-941 (and others ) Transmit Sound Quality
Andrew Joakimsen wrote: I too seem to have the same problem, dont know about poor quality but its certainly not loud enough, I have to put my mouth to the microphone, otherwise the other end reports they cannot hear me. This does however seem to do a good job to cancel out the background noise In the SIPura setup change the packet size from .3 to .2. ___ --Bandwidth 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: OT: Quad-band cellphones with wifi stable sip support
On 2007-01-14 22:01:44 -0800, Tomer Horn [EMAIL PROTECTED] said: Hello, I am looking to purchase a new quad-band cellphone and I'm looking for one with WiFi and enough CPU power for stable SIP calls. I was wondering if anyone here can share his experience and recommend on a good cellphone. Any chance there is such a phone with even good WiFi profiles management or am I asking for too much now? :-) The Nokia E60 meets your requirements on paper, but seems to be a firmware update or two away from reliability with the SIP thing. Marty ___ --Bandwidth 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] windows mobile 5 softphone for square screen devices
Guys, anybody has seen or is using some kind of softphone on any square screen device with WM5? Ive tried sjlabs one and xten for pocket pc and they do work on Wm5 but they are designed for standard screens, anybody using anything on square ones? ___ --Bandwidth 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] newbie asterisk 1.4 installation problem
Hello friends, I am trying to install asterisk 1.4. I am configuring it as follows:- ./configure --prefix=/home/vivek/downloads/install/asterisk/ But still while running 'make install', it tries to install it in /var/lib/asterisk/ and stops because of failing permissions. I have provided it a prefix, But it doesn't install it there. Can anybody tell me the solution for this. I dont want to install it in the default directories. I want it to be in my home directory. With warm regards. Vivek J. Joshi. [EMAIL PROTECTED] Trikon Electronics Pvt. Ltd. All science is either physics or stamp collecting. -- Ernest Rutherford ___ --Bandwidth 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] Using the SIPAddHeader Application
Hi, I'm trying to use the SIPAddHeader application to add a header containing to semicolon separated strings like this: exten = 12, 1, SIPAddHeader(X-TestHeader:a=test1;b=test2) But in the resulting INVITE message only the first part (X-TestHeader:a=test1) is added. Setting into quotation mark doesn't change anything. exten = 12, 1, SIPAddHeader(X-TestHeader:a=test1;b=test2) Do you have an idea how to achieve it? Thank you, Thomas ___ --Bandwidth 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] windows mobile 5 softphone for square screen devices
I've been trying the SJPhone with no luck. Where did you download the Xten version from? On 1/16/07, Anton Krall [EMAIL PROTECTED] wrote: Guys, anybody has seen or is using some kind of softphone on any square screen device with WM5? Ive tried sjlabs one and xten for pocket pc and they do work on Wm5 but they are designed for standard screens, anybody using anything on square ones? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mitchel Constantin Snap - A desktop user interface for Asterisk www.snapanumber.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
