Re: [asterisk-users] Low cost BRI gateway
On 2012-03-13 18:38, Chris Bagnall wrote: Greetings list, I'm trying to source a very basic ISDN BRI - SIP gateway. Unfortunately, everything I've seen seems to want to do lots of other things - registering handsets, IVRs, voicemail, etc. I only want it to present an ISDN BRI as a SIP account - I have an asterisk server for the other stuff. :-) In any other environment I'd just use one of the Digium ISDN PCIe cards, but in this case the ISDN lines come into one building and the asterisk servers are in the other building across the road, and there's no copper link between them. Any suggestions gratefully received. http://www.switchvoice.com/ or http://www.teles.de/en/products-and-solutions/access-gateways/voip-gateways/voipbox-bri/ or http://www.patton.com/ Regards Hans -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] how to debug udp daemon of asterisk
Hi, I'm using a packet interception module for modifying udp packets coming to asterisk sip port. now my packet modification application successfully forwards the packet but somehow there is no response from asterisk. it may be that the modifications destroyed sanity of the sip packet so asterisk is just ignoring it. or there may be problem inside my modification rule that is sending to a wrong socket. I want single out the problem. to do that i need to find the activity happening in the udp daemon runs under the hood of asterisk. so far i've straced my modification application which reports a successful sendmsg(). as its udp datagram it cant say anything about destination. i've straced asterisk using strace asterisk -dd . it waits in read() syscall. i've tried putting garbage data by netcat(nc -uvv asterisk_host 7160, where 7160 is the sip port i've defined in sip.conf) on the listening port of asterisk. but the read() does not returns. read() should return at whatever is available in udp socket. because packet's sanity check is left for upper layer code. so how should i go about debugging it? Thanks in advance. signature.asc Description: This is a digitally signed message part -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI_SPANCONFIG failed on span 1: No such device or address (6)
Original-Nachricht Datum: Tue, 13 Mar 2012 22:40:27 +0200 Von: Tzafrir Cohen tzafrir.co...@xorcom.com An: asterisk-users@lists.digium.com Betreff: Re: [asterisk-users] DAHDI_SPANCONFIG failed on span 1: No such device or address (6) On Tue, Mar 13, 2012 at 05:10:14PM +0100, rama...@gmx.de wrote: Original-Nachricht Datum: Tue, 13 Mar 2012 17:26:43 +0200 Von: Tzafrir Cohen tzafrir.co...@xorcom.com An: asterisk-users@lists.digium.com Betreff: Re: [asterisk-users] DAHDI_SPANCONFIG failed on span 1: No such device or address (6) On Tue, Mar 13, 2012 at 03:30:45PM +0100, rama...@gmx.de wrote: Hi all, I have problems starting dahdi. dahdi_cfg -vvv allwasy comes back with: DAHDI Tools Version - 2.2.1.1 DAHDI Version: 2.3.0.1 Echo Canceller(s): Configuration == SPAN 1: CCS/ AMI Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Echo Canceler: kb1) (Slaves: 01) Channel 02: Clear channel (Default) (Echo Canceler: kb1) (Slaves: 02) kb1? Why not mg2 (or OSLEC or whatever)? OSLEC ist planned in the future. I fist have to find where to get the sources and howto compile / load them. Hints appreciated. First wanted to get basics running. Channel 03: Hardware assisted D-channel (Default) (Echo Canceler: none) (Slaves: 03) 3 channels to configure. DAHDI_SPANCONFIG failed on span 1: No such device or address (6) What's the output of lsdahdi ? lsdahdi returns nothing, while lspci 00:0d.0 Network controller: Cologne Chip Designs GmbH ISDN network controller [HFC-PCI] (rev 02 and dahdi_hardware pci::00:0d.0 zaphfc- 1397:2bd0 HFC-S ISDN BRI card I would have expected dahdi_hardware would be closer related to all dahdi commands. '-' means that no module is actually loaded. Do you have zaphfc ? If so: try: modprobe zaphfc I searched the internet but could not yet find a solution. I already tried to exchange the zaphfc drivers as suggested, but they did not compile. I actually did not find a new(er) tutorial how to build an Asterisk with a HFC-S card. https://gitorious.org/dahdi-extra/dahdi-linux-extra Well, mainly useful for producing patches and such). Thanks, but here I don't have any experience how / where to attach these patches. Where can I find more info about it ? Any suggestions / hints / tutorials / links welcome. Do I need some special drivers in the kernel ? Modprobe ? Anything else special I need ? Details: I run Debian 6.0.4 with a fresh 2.6.35 Kernel, most drivers compiled as modules. http://updates.xorcom.com/pkg-voip/ has some Squeeze backports. Though they're not kept up-to-date all the time. They include zaphfc. Same here, I don't have any experience how / where to attach these patches. Where can I find more info about it ? The -source is already patched. There is actually one big patch there (with the external drivers) and as it mentions - it is taken from the git repo I mentioned. I installed the newest Asterisk 10.2.0-rc2 and dahdi_sources and compiled them, but same happened with aptitude install dahdi on older kernel. download Asterisk - ./configure - make - make install - make samples download dahdi.tar.bz2 - make - make install dahdi_hardware: driver should be 'zaphfc' but is actually 'hfcpci' pci::00:0d.0 zaphfc+ 1397:2bd0 HFC-S ISDN BRI card You should probably blacklist hfcpci. echo 'blacklist hfcpci' /etc/modprobe.d/WHATEVER.conf Removed and blacklisted with echo 'blacklist hfcpci' /etc/modprobe.d/blacklist.conf Already blacklisted for unknown reasons: blacklist hfcmulti blacklist hfc4s8s_l1 blacklist wcb4xxp Don't I need all /one of these modules ? Those three are for a different card, so they are irrelevant (harmless, though) I blacklisted hfcpci and rebooted. Now dahdi_hardware pci::00:0d.0 zaphfc+ 1397:2bd0 HFC-S ISDN BRI card shows that zaphfc is loaded. Also lsdahdi ### Span 1: ZTHFC1 HFC-S PCI A ISDN card 0 [TE] (MASTER) AMI/CCS 1 BRIClear (In use) (SWEC: OSLEC) 2 BRIClear (In use) (SWEC: OSLEC) 3 BRIHardware-assisted HDLC shows output and dahdi_cfg - DAHDI Tools Version - 2.2.1.1 DAHDI Version: 2.3.0.1 Echo Canceller(s): MG2, OSLEC Configuration == SPAN 1: CCS/ AMI Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Echo Canceler: oslec) (Slaves: 01) Channel 02: Clear channel (Default) (Echo Canceler: oslec) (Slaves: 02) Channel 03: Hardware assisted D-channel (Default) (Echo Canceler: none)
Re: [asterisk-users] Getting Mac Address on connected IP phones
Am 14.03.2012 00:34, schrieb James Sharp: ping + arp isn't going to work if they're on a different VLAN. I believe this will work: Too complicated. Just have a look on the switch(es) the phones are connected to. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] libpri error??
Hello list, I have a client who's taking intermittent errors on their PRI. The server is configured with one PRI from the TELCO, and two PRI connecting to their Iwatsu ADIX legacy system. The odd thing is, the system can run for days, weeks or months without a reported error and then just bomb. The only thing that fixes it is stopping and starting the services. It then runs well for a random time period. I've upgraded and downgraded software, even tried new hardware and CentOS 5.x. none of these changes have made a difference. Now here's the crazy thing, this system ran fine for a couple years BEFORE a change in the local provider(from Level 3 to CenturyLink). CL claims that they do nothing different from Level 3 but we noticed right away that we had to adjust the pridialplan to get outbound to work. So much for that. I figured there was a problem with the circuit. So we worked with Sangoma and the Telco to troubleshoot the problem. After a lot of ordeal, Sangoma cleared the Telco and said the problem is most likely in libpri. I upgraded libpri to SVN release 2279 and that seemed to be the fix we needed. We had a couple errors but figured they were in the dialplan, made some adjustments and it ran clean. We noticed a few errors on the Telco PRI so I upgraded to libpri 2283. This time everything ran so clean, we thought our problems were behind us. Unfortunately for us, this morning everything went haywire. The Iwatsu could not make internal or outbound calls via the PRI. SIP users on the asterisk server could not call out and I could not call in. The message was All circuits busy. A quick restart and everything is back. The TELCO swears it's not their problem and upgrading libpri to SVN have seemed to help. I'm hoping someone here can provide some insight. I have an IDSN pcap from this morning and the relevant log file located here: http://www.sayso.net/031412/8841.pcap http://www.sayso.net/031412/asterisk.log I just realized that debugging was not set in asterisk so this is probably not enough information to get started. What should I set debug level to next time? I will do that and turn on debugging on span 1. Anyway, if there's anything that can be done now Here's my /etc/dahdi/system.conf == #autogenerated by /usr/sbin/wancfg_dahdi do not hand edit #autogenrated on 2012-02-29 #Dahdi Channels Configurations #For detailed Dahdi options, view /etc/dahdi/system.conf.bak loadzone=us defaultzone=us #Sangoma A104 port 1 [slot:4 bus:9 span:1] wanpipe1 span=1,1,0,esf,b8zs bchan=1-23 echocanceller=HWEC,1-23 hardhdlc=24 #Sangoma A104 port 2 [slot:4 bus:9 span:2] wanpipe2 span=2,2,0,esf,b8zs bchan=25-47 echocanceller=HWEC,25-47 hardhdlc=48 #Sangoma A104 port 3 [slot:4 bus:9 span:3] wanpipe3 span=3,3,0,esf,b8zs bchan=49-71 echocanceller=HWEC,49-71 hardhdlc=72 #Sangoma A104 port 4 [slot:4 bus:9 span:4] wanpipe4 span=4,4,0,esf,b8zs bchan=73-95 echocanceller=HWEC,73-95 hardhdlc=96 == /etc/asterisk/chan_dahdi.conf == ;autogenerated by /usr/sbin/wancfg_dahdi do not hand edit ;autogenrated on 2012-02-29 ;Dahdi Channels Configurations ;For detailed Dahdi options, view /etc/asterisk/chan_dahdi.conf.bak [trunkgroups] [channels] context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no ; Required for Embarq / CenturyTel pridialplan=unknown prilocaldialplan=local priindication=outofband priexclusive=no ;Sangoma A104 port 1 [slot:4 bus:9 span:1] wanpipe1 switchtype=national context=from-pstn group=0 echocancel=yes signalling=pri_cpe channel =1-23 ;Sangoma A104 port 2 [slot:4 bus:9 span:2] wanpipe2 switchtype=national context=from-internal group=1 echocancel=yes signalling=pri_net channel =25-47 ;Sangoma A104 port 3 [slot:4 bus:9 span:3] wanpipe3 switchtype=national context=from-internal group=2 echocancel=yes signalling=pri_net channel =49-71 ;Sangoma A104 port 4 [slot:4 bus:9 span:4] wanpipe4 switchtype=national context=from-internal group=3 echocancel=yes signalling=pri_net channel =73-95 == /etc/wanpipe/wanpipe1.conf == # # WANPIPE1 Configuration File # # # Date: Wed Dec 6 20:29:03 UTC 2006 # # Note: This file was generated automatically # by /usr/local/sbin/setup-sangoma program. # # If you want to edit this file, it is # recommended that you use wancfg program # to do so.
Re: [asterisk-users] Polycom CX3000 IP with Asterisk?
I have a customer that has a CX3000 IP that was designed for MS Lync. Anyone know if these can run as standard SIP so we can use it with Asterisk? Thanks Bryant -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom CX3000 IP with Asterisk?
According to the specifications, it should connect with little difficulty. http://www.voipsupply.com/polycom-cx3000 From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bryant Zimmerman Sent: Wednesday, March 14, 2012 12:32 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Polycom CX3000 IP with Asterisk? I have a customer that has a CX3000 IP that was designed for MS Lync. Anyone know if these can run as standard SIP so we can use it with Asterisk? Thanks Bryant -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Normal ringing tone for the caller, while call waiting.
Hi , When I make a call to an extension, which is on another call, the called party (who is on call waiting) will get a BEEP sound. But the caller gets the normal ringing tone. Is there any way to have a different dialer tone for the Caller, which lets him know that the other person is on a call.. i.e. When A calls B, while B is already on a call with C, Is there a way to let A get a message that B is busy on another call. Thank you. Regards, Najim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] libpri error??
snip /etc/asterisk/chan_dahdi.conf == ;autogenerated by /usr/sbin/wancfg_dahdi do not hand edit ;autogenrated on 2012-02-29 ;Dahdi Channels Configurations ;For detailed Dahdi options, view /etc/asterisk/chan_dahdi.conf.bak [trunkgroups] [channels] context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no Try adding resetinterval = never here. The default is to reset any idle channels every 3600 seconds. Unfortunately, if a channel is being reset just as an incoming call arrives, there is a chance that the channel will get stuck in a resetting state and block any further use of that channel. A quick look at your traces indicates that your channels are already in use so an incoming call is cleared with a cause 44. This can happen if there is a call collision or the channel is stuck. I would recommend that you minimize call collisions by making outbound calls pick channels from the opposite end of the channel range as the network. This is usually by using uppercase G in the Dial(DAHDI/G1/number) to the first available channel from the upper end of the channel range. Richard ; Required for Embarq / CenturyTel pridialplan=unknown prilocaldialplan=local priindication=outofband priexclusive=no ;Sangoma A104 port 1 [slot:4 bus:9 span:1] wanpipe1 switchtype=national context=from-pstn group=0 echocancel=yes signalling=pri_cpe channel =1-23 ;Sangoma A104 port 2 [slot:4 bus:9 span:2] wanpipe2 switchtype=national context=from-internal group=1 echocancel=yes signalling=pri_net channel =25-47 ;Sangoma A104 port 3 [slot:4 bus:9 span:3] wanpipe3 switchtype=national context=from-internal group=2 echocancel=yes signalling=pri_net channel =49-71 ;Sangoma A104 port 4 [slot:4 bus:9 span:4] wanpipe4 switchtype=national context=from-internal group=3 echocancel=yes signalling=pri_net channel =73-95 == snip Installed Software == asterisk-1.4.43-1.C4.SC asterisk-addons-1.4.13-1.C4.LSE asterisk-core-sounds-en-wav-1.4.21-1.C4.SC asterisk-devel-1.4.43-1.C4.SC asterisk-extra-sounds-en-gsm-1.4.11-2.C4.LSE asterisk-libpri-2283-1svn.C4.SC asterisk-perl-1.01-1.C4.LSE dahdi-linux-2.6.0-2.6.9_103.plus.c4.LSE.1smp_3.C4.SC dahdi-tools-2.6.0-2.C4.SC iaxmodem-static-1.2.0-1.C4.SC kernel-smp-2.6.9-103.plus.c4.LSE.1 kernel-utils-2.4-23.el4 wanpipe-3.5.25-1.SC wanpipe-modules-3.5.25-kernel.2.6.9.103.plus.c4.LSE.1smp.dahdi.2.6.0_1.SC == -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] libpri error??
Try adding resetinterval = never here. The default is to reset any idle channels every 3600 seconds. Unfortunately, if a channel is being reset just as an incoming call arrives, there is a chance that the channel will get stuck in a resetting state and block any further use of that channel. Thanks, I'll give this a shot. A quick look at your traces indicates that your channels are already in use so an incoming call is cleared with a cause 44. This can happen if there is a call collision or the channel is stuck. I would recommend that you minimize call collisions by making outbound calls pick channels from the opposite end of the channel range as the network. This is usually by using uppercase G in the Dial(DAHDI/G1/number) to the first available channel from the upper end of the channel range. Our Telco PRI is set inbound and asterisk set descending so we should be good there. I'll double check the Iwatsu PRI's. Thank you, -- Andrew McRory Sayso Communications, Inc. 2850 Industrial Plaza Tallahassee, Florida 32301 Office) 850-224-5737 Mobile) 850-778-3206 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to show used wrong password
On 03/13/2012 11:06 PM, Dave Platt wrote: Ouch. That isn't going to be so easy to spot, then! You would have to guess a bunch of likely passwords, fake up a challenge with some known nonce, and compare the response against those you would expect with each of the various possible passwords. (You've already got the Source Code to do all this, of course.) You'll have to try the selective unplugging method instead . There may be a way to do this, even in the face of the nonce-and-hash security system. As I understand it: when a system (re)registers with a good password, what you'll typically see is: - A registration request from the client (with the client's ID in the SIP parameters) - A response from Asterisk, saying something on the order of Stale authentication. Try again. Here's a new nonce for you. - Another registration request from the same client, specifying the newly-issued nonce, and having a hash based on that nonce and the shared secret. - An OK response from Asterisk. When a system (re)registers, and has the wrong password/secret, the exchange will be different. - A registration request from the client (with the client's ID in the SIP parameters) - A response from Asterisk, saying something on the order of Stale authentication. Try again. Here's a new nonce for you. - Another registration request from the same client, specifying the newly-issued nonce, and having a hash based on that nonce and the shared secret. - A response from Asterisk, rejecting the second registration request with something like a bad digest error. So, if you examine all of the SIP protocol exchanges taking place, you should see a whole bunch of successful four-way handshakes (from clients that have the correct secrets), and an occasional four-way handshake failure (from the one client that has the wrong password in its configuration). You won't be able to tell what password the client is actually trying to use - that's the whole point of the nonce-and-hash approach - but you'll be able to identify its client name, and (unless the far end is using a NAT or proxy) its IP address. To pin down the actual location of the client, you'll either have to go there, or have someone at the remote site do some investigation and (possibly) packet tracing on the LAN. this will be of little use in this situation, the location is a shared office space/building in Vietnam and the local hands i have already checked our computers for soft phones, but quit possible some machines got swapped there or some local admin installed it somewhere for testing purposes... and the local hands i have, not really usefull explaining them to look up the meaning of packet tracing Or, I suppose one could simply use Asterisk to try to phone the device or softphone in question, at whatever address it called in from, and ask whoever answers the phone to disable it! this was my original idea yes, but how can i call it without it being registered? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Getting Remote UNIX connection disconnected
Hi, I'm getting the messages listed below after login to asterisk cli; -- Remote UNIX connection -- Remote UNIX connection disconnected Usually verbose level is set to 4, after setting to 2, I'm not getting these messages. Is there other way to stop these messages? because I'm getting very irritated and need to set verbose level at least 4. Further added, I also tried to stop and start asterisk service but still getting these messages. -- Regards, Ahmed Munir Chohan -- Regards, Ahmed Munir Chohan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Getting Remote UNIX connection disconnected
Those messages someone or something is running asterisk -r or similar. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ahmed Munir Sent: Wednesday, March 14, 2012 3:13 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Getting Remote UNIX connection disconnected Hi, I'm getting the messages listed below after login to asterisk cli; -- Remote UNIX connection -- Remote UNIX connection disconnected Usually verbose level is set to 4, after setting to 2, I'm not getting these messages. Is there other way to stop these messages? because I'm getting very irritated and need to set verbose level at least 4. Further added, I also tried to stop and start asterisk service but still getting these messages. -- Regards, Ahmed Munir Chohan -- Regards, Ahmed Munir Chohan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Getting Remote UNIX connection disconnected
Depending on your Asterisk version, add hideconnect = yes to asterisk.conf and restart. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ahmed Munir Sent: Wednesday, March 14, 2012 2:13 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Getting Remote UNIX connection disconnected Hi, I'm getting the messages listed below after login to asterisk cli; -- Remote UNIX connection -- Remote UNIX connection disconnected Usually verbose level is set to 4, after setting to 2, I'm not getting these messages. Is there other way to stop these messages? because I'm getting very irritated and need to set verbose level at least 4. Further added, I also tried to stop and start asterisk service but still getting these messages. -- Regards, Ahmed Munir Chohan -- Regards, Ahmed Munir Chohan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Grandstream GXP20XX BLF lamps stop working soon after a phone reboot.
Any one having this problem. The Grandstream Firmware revision is 1.2.5.3. We have the registration time set to 5 minutes, and every time after a reboot, the BLF's will initially indicate the correct state, then stop working a few minutes later. The workaround has previously been to reboot twice. The 2nd time just after the BLFs' stop working. I have a patch for this, that's available at https://reviewboard.asterisk.org/r/1813/ Alec -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Rate sheet normalization
Markus unive...@truemetal.org writes: Does such a thing exist? How does a2billing do it? It should be pretty easy in an AGI. If you can afford a linear lookup per call, just grep through the array of prefixes to find the ones matching a particular call, then pick the cheapest from the results. If you need something faster than linear it gets tricky. It would be tempting to preprocess the list to say 5 digits, do a hash lookup on those, and then use the process above. /Benny -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 10 and AMR?
Hello, Is anyone aware of an AMR-NB codec for Asterisk 10? The old patch floating around for older versions of Asterisk doesn't seem to work anymore. Best regards, Jan Blom -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Rate sheet normalization
Am 15.03.2012 00:35, schrieb Benny Amorsen: Markusunive...@truemetal.org writes: Does such a thing exist? How does a2billing do it? It should be pretty easy in an AGI. If you can afford a linear lookup per call, just grep through the array of prefixes to find the ones matching a particular call, then pick the cheapest from the results. a2billing chooses the cheapest rate if there is more than 1 rate for a prefix that matches exactly. I'm not a programmer so I can't tell you how they do it internally. The call is simply passed to a2billing.php via DeadAGI. I know that the a2billing guys are currently working on fixing exactly the mentioned issue, but it's unknown when that will be done. I was just hoping there was some software out there that I can throw 10 different ratesheets at with different length of codes and it will output a single true LCR ratesheet cut down to the official worldwide codes. Thanks! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Normal ringing tone for the caller, while call waiting.
On Wednesday 14 Mar 2012, NaJIm wrote: When I make a call to an extension, which is on another call, the called party (who is on call waiting) will get a BEEP sound. But the caller gets the normal ringing tone. Is there any way to have a different dialer tone for the Caller, which lets him know that the other person is on a call.. i.e. When A calls B, while B is already on a call with C, Is there a way to let A get a message that B is busy on another call. If SIP, have a look at the busylevel and call-limit sip.conf parameters. Regards, -- Raj -- Raj Mathur || r...@kandalaya.org || GPG: http://otheronepercent.blogspot.com || http://kandalaya.org || CC68 It is the mind that moves || http://schizoid.in || D17F -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Rate sheet normalization
A2Billing doesn't do that. A2Billing in fact has a lot of shortcomings one of which is this exact issue. I would suggest running rate sheets against each other for finding true LCR and then only uploading the rates that are cheaper into the system. In most cases there are not such high differences but if there are then this is the only way. I know rate normalization talk comes up all the time on FreeSwitch Freenode channel and it probably does on OpenSIPs as well. Check there for some good advice. On Wed, Mar 14, 2012 at 7:35 PM, Benny Amorsen benny+use...@amorsen.dkwrote: Markus unive...@truemetal.org writes: Does such a thing exist? How does a2billing do it? It should be pretty easy in an AGI. If you can afford a linear lookup per call, just grep through the array of prefixes to find the ones matching a particular call, then pick the cheapest from the results. If you need something faster than linear it gets tricky. It would be tempting to preprocess the list to say 5 digits, do a hash lookup on those, and then use the process above. /Benny -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
