[asterisk-users] packet counts: twice as high on one leg?
Hi all, I have two phones that I've been comparing (different manufacturers). To debug call quality issues on one of them, I've been using calls from the phone to our main DID, so 3 SIP sessions exist (phone asterisk then asterisk provider, and the providerasterisk for the DID). The bad phone shows roughly twice the number of packets on the phoneasterisk session as on the other two sessions. The good phone shows roughly equal packet counts on each of the 3 sessions. I've used asterisk server MTUs of 1440 and 1500, but it makes no difference. Is this double-packet-count a clue, a problem, or a red herring? The packet counts are shown using sip show channelstats. - Mike -- _ -- 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] packet counts: twice as high on one leg?
On 06/20/2013 11:56 AM, jg wrote: Have you checked whether the same codecs, or codecs with the same bandwidth requirements, are used? Yes, it is ulaw everywhere. - Mike -- _ -- 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] packet counts: twice as high on one leg?
On 06/20/2013 11:56 AM, jg wrote: Have you checked whether the same codecs, or codecs with the same bandwidth requirements, are used? Here's an example of a simple outgoing call. Everything is ulaw. The 192.x.x.x phone has roughly twice the packet count of the provider session. The lost packet count is nonsensical on one session. Sigh. - Mike steerpike*CLI sip show channelstats Peer Call ID Duration Recv: Pack Lost ( %) Jitter Send: Pack Lost ( %) Jitter 209.217.98.130 0c15efc03f2 00:01:03 003069 104829 (97.16%) 0. 003040 00 ( 0.00%) 0.0002 192.168.0.36 qY0p292XeDl 00:01:03 006121 00 ( 0.00%) 0. 006096 00 ( 0.00%) 0.0001 2 active SIP channels steerpike*CLI sip show channels Peer User/ANR Call ID Format Hold Last MessageExpiry Peer 209.217.98.130 6139419467 0c15efc03f243c7 (ulaw) No Tx: ACK6136866675 192.168.0.36 mjc_office qY0p292XeDlPcLk (ulaw) No Rx: ACKmjc_office steerpike*CLI core show version Asterisk 11.4.0 built by root @ steerpike.avtechpulse.com on a x86_64 running Linux on 2013-06-19 12:10:47 UTC -- _ -- 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] packet counts: twice as high on one leg?
Packet count is one thing, transferred data is another one. If one phone uses smaller UDP packages, then the packet count should increase in reciprocally. I have read some comments on the net that smaller packages are preferable because lost packages have less impact on the hearable audio. Aha. I overlooked that some phones had ulaw:10 in sip.conf, instead of the standard ulaw:20. That explains the packet count difference. It seems my call quality issues are coming from something else. - Mike -- _ -- 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] changing ringtones to a group of phones
Hi all, I've been modifying the ringtone on a group of Snom phones like this, depending on certain dial-plan conditions: Exten = s,1,SIPAddHeader(Alert-Info: http://192.168.0.200/tel_ring01.wav) exten = s,n,Dial(SIP/mjc_officeSIP/mjc_homeSIP/mjc_labSIP/mjc_server,20,trj) Now, I'm migrating slowly to Digium D70 phones, which have a different Alert-Info syntax (and different ringtone names). How can I dial a group of phones simultaneously, say half Snom and half Digium, with different sip alert-info headers? - Mike -- _ -- 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] changing ringtones to a group of phones
On 05/03/2013 01:22 PM, jg wrote: Maybe using a LOCAL channel could help. One ext. for Snom with Snom header, another for Digium with Digium header, then simultaneously call both local channels, which then call the appropriate phones. Thanks, that might work! - Mike -- _ -- 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] Digium D70 visual voicemail - won't play
Hi all, I'm trying out a Digium D70 phone with Asterisk 11. My voicemail messages are listed in the visual voicemail app on the phone, but they do not successfully play back. The correct duration is shown, but the progress bar just jumps back to zero when I press the Play softbutton. I can hear my messages fine if I manually dial into my voicemail extension. I have format=wav49 in voicemail.conf. Is that a problem format for the D70? - Mike D70 Current Firmware Version: 1_3_0_2_54153 Asterisk 11.3.0 -- _ -- 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] Door Contacts via Asterisk?
On 11/15/2010 01:35 PM, Cassius Smith wrote: Hi all, I have had (what I consider) an odd request. The installation I'm working on now is an office on a multi-floor building. They 're looking for some kind of solution with the phone system to provide door control. We are a non-profit so of course I'm looking for something VERY inexpensive. I'm sure /someone/ has done something like this. I'd appreciate any ideas. Well, I use Asterisk to call a Perl AGI script which drives a serial port DTR line high (using Device::SerialPort and Asterisk::AGI) for 20 seconds. The serial port line drives a transistor, which drives a large relay, which applies power to the (pre-existing) door solenoid. Fairly trivial if you know how, hard otherwise :-) - Mike -- _ -- 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] ADSL Load Balancing
On 11/03/2010 03:49 AM, Gordon Henderson wrote: I've got a client with two ADSL connections for redundancy. Is it possible to set up asterisk to connect to one SIP provider using both adsl connections and load balance between the two connections? Or to use one connection as the main one, and automatically fail over if the first connection drops? Or does this kind of thing need a serious network switch? I use the SG560 (http://www.snapgear.com/index.cfm?skey=1557) to do this. It handles two WAN connections (going to your ADSL modems). I set the routing policies so that VOIP goes on one link by default, and everything else on the other. If one link goes down, everything will be routed on the remaining link. (Unfortunately, it doesn't seem to revert to the default state after the downed link recovers, so I have to add some reboot-modems-after-recovery scripts in a cron job to make things recovery in an ideal way.) I think you can do the same with the Cisco RV016, which is cheaper, but the documentation is poor. - Mike -- _ -- 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] moving a bridged call to a conference room
I'm just figuring out conferencing. I have a super-simple setup with one room: exten = 600,1,Answer exten = 600,2,ConfBridge(1234,c|M|s) exten = 600,3,hangup If two people want to take their (bridged) call to the conference room, the local user has to do a transfer (to 600), and then dial 600 themselves. Is there an easier way to transfer both ends of a bridged call to the conference room? - Mike -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] audio cuts out during IVR
Hi all, I'm running 1.6.2.0-rc6, and I'm running into a problem: sometimes the audio vanishes in the middle of listening to an IVR background prompt. This happens with both analog (Digium card) and IAX2 incoming calls. The prompts are stored in ulaw format (and the IAX2 calls use ulaw). The asterisk console claims that the IVR prompts are proceeding in the expected fashion, but I can't hear anything. The logs don't report anything interesting. Has anyone seen anything like this? Suggestions? - Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] snapgear/mcafee sg560 rebooting
Hi all, Does anyone else use the SG560 firewall with Asterisk? I do, and it normally works great, except when it randomly reboots. Has anyone else experienced this annoyance? Did you fix it? - Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] audio cuts out during IVR
On 11/24/2009 02:14 PM, David Backeberg wrote: The asterisk console claims that the IVR prompts are proceeding in the expected fashion, but I can't hear anything. Are you playing with the system clock? ... dramatic ntp changes? No, that shouldn't be happening. But I'll keep it in mind while debugging... - Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] snapgear/mcafee sg560 rebooting
On 11/24/2009 01:19 PM, Dr. Michael J. Chudobiak wrote: Does anyone else use the SG560 firewall with Asterisk? I do, and it normally works great, except when it randomly reboots. Has anyone else experienced this annoyance? Did you fix it? Oops, never mind. The SG560 was fine. The AC power to it wasn't! - Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] audio cuts out during IVR
On 11/24/2009 02:14 PM, David Backeberg wrote: I'm running 1.6.2.0-rc6, and I'm running into a problem: sometimes the audio vanishes in the middle of listening to an IVR background prompt. Are you playing with the system clock? Actually, setting the internal_timing option seems to have fixed the problem. https://issues.asterisk.org/view.php?id=15932 - Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] my kernel is dazed and confused
On 11/12/2009 09:31 AM, Dr. Michael J. Chudobiak wrote: Nov 12 08:54:27 steerpike kernel: Uhhuh. NMI received for unknown reason a0 on CPU 0. Nov 12 08:54:27 steerpike kernel: You have some hardware problem, likely on the PCI bus. Nov 12 08:54:27 steerpike kernel: Dazed and confused, but trying to continue Would my Digium TDM410P cause an NMI, or is my computer failing? It seemed to be a problem with the on-board network adapter's driver. Sticking in a PCI NIC (which uses a different driver), and using it instead of the on-board one, seems to have fixed the problem. Thanks to all for the various suggestions! - Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] my kernel is dazed and confused
Nov 12 08:54:27 steerpike kernel: Uhhuh. NMI received for unknown reason a0 on CPU 0. Nov 12 08:54:27 steerpike kernel: You have some hardware problem, likely on the PCI bus. Nov 12 08:54:27 steerpike kernel: Dazed and confused, but trying to continue Would my Digium TDM410P cause an NMI, or is my computer failing? - Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] my kernel is dazed and confused
On 11/12/2009 09:42 AM, Francesco Peeters wrote: Dr. Michael J. Chudobiak wrote: Nov 12 08:54:27 steerpike kernel: Uhhuh. NMI received for unknown reason a0 on CPU 0. Nov 12 08:54:27 steerpike kernel: You have some hardware problem, likely on the PCI bus. Nov 12 08:54:27 steerpike kernel: Dazed and confused, but trying to continue Would my Digium TDM410P cause an NMI, or is my computer failing? - Mike Googling for the error seems to indicate a possible kernel bug... Are you using Ubuntu or Debian?... I'm using Fedora 11, kernel 2.6.30.8-64.fc11.x86_64. - Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best Firewall Suggestions?
On 10/14/2009 01:29 PM, David Wathen wrote: Hi Myles, Thanks to you and everyone else that has responded. I've really learned a lot. pFSense and IPCop sounds let best so far for LINUX based firewalls. I'm also wondering if anyone has any suggestions for a standalone firewall appliance like my Linksys WRT54G except one better suited for a small business and that NAT works well with VOIP. I use the Secure Computing SG560 (which which recommended by my VOIP provider), and it works very well with IAX2. I haven't tried SIP. Avoid SonicWall. I had bad experiences with their products and VOIP. - Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ITSP's no longer supporting IAX?
OCG Technical Support wrote: After a variety of connectivity problems, my itsp (Unlimitel.ca) blamed the problem on the IAX protocol. They told me that as of Asterisk 1.4 the IAX protocol went downhill and many carriers (like VoicePulse) are discontinuing support for IAX. Is this correct? We are all heading for SIP? I use IAX with unlimitel.ca on Asterisk 1.6, and I haven't had any issues at all. The choice of router/NAT is critical though. Unlimitel recommended the SnapGear 560 to me, and it eliminated all the issues I was having with IAX going through my Sonicwall devices. Just another datapoint for you... - Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ITSP's no longer supporting IAX?
The choice of router/NAT is critical though. Unlimitel recommended the SnapGear 560 to me, and it eliminated all the issues I was having with IAX going through my Sonicwall devices. Just another datapoint for you... Just curious. Since IAX only uses ONE port, do you have any idea what the technical reason behind a specific router would be critical? Well, with a Sonicwall TZ170, you had to enabled Firewall VOIP Enable consistent NAT, which was not the default setting. Then, you had to figure out that Firewall Advanced Default UDP Connection Timeout defaulted to 30 seconds, less than the normal Asterisk 60 second registration timeout. Then, for some reason, the TZ170 would simply lose packets. A fraction of calls would end almost immediately after they started, with Asterisk reporting a raw hangup error and INVAL packets, suggesting that some IAX2 packets were being lost, mis-ordered, or mis-translated. Anyway, the Sonicwall TZ170 was totally unreliable for IAX2 connections. They caused me a lot of grief. Avoid them like the plague. The Snapgear 560 just works, which I appreciate very much! - Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sonicwall potentially causing long ping times toSIP phones
Craig Van Ham wrote: I had weird issues when using a Sonicwall, gave up. Same here, avoid them! I use the SnapGear SG560 now. - Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] debugging hints in 1.6
Hi, I use hints to drive the LEDs on my snom phones, something like: exten = 601,1,Dial(SIP/mjc_officeSIP/mjc_homeSIP/mjc_labSIP/mjc_serverSIP/mjc_library,20,trj) exten = 601,2,Voicemail([EMAIL PROTECTED],u) exten = 601,102,Voicemail([EMAIL PROTECTED],u) exten = 601,hint,SIP/mjc_officeSIP/mjc_homeSIP/mjc_labSIP/mjc_serverSIP/mjc_library Sometimes asterisk gets confused, though, and reports my extension as in-use, even though no channels are active. Dialing something makes the hint report inactive - the states are inverted, in other words. How can I debug that? Can I display the individual states of each sip line (SIP/mjc_office, SIP/mjc_home, etc) comprising the hint? Has anyone seen similar behavior? - Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] debugging hints in 1.6
601,hint,SIP/mjc_officeSIP/mjc_homeSIP/mjc_labSIP/mjc_serverSIP/mjc_library Sometimes asterisk gets confused, though, and reports my extension as in-use, even though no channels are active. Dialing something makes the hint report inactive - the states are inverted, in other words. How can I debug that? Can I display the individual states of each sip line (SIP/mjc_office, SIP/mjc_home, etc) comprising the hint? sip show subscriptions Hmm, I'll see if that gives me any clues... core show hints That just tells me what the LED is showing (which I can already see), with no further insights as to why it's wrong. I don't suppose there is some sort of core show devicestates command, or something similar? - Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] debugging hints in 1.6
Philipp Kempgen wrote: Hmm, I'll see if that gives me any clues... Or you could try 'sip show inuse'. Thanks, Philipp! I never noticed that command; I'm sure it will be very handy for debugging. - Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] rebooting snoms in 1.6
With Asterisk 1.4 I could use commands like: /usr/sbin/asterisk -rx sip notify reboot-snom mjc_home to reboot a snom phone. Now, with 1.6, when I try that, I get: Unable to find notify type 'reboot-snom' Command 'sip notify reboot-snom mjc_home' failed. Do I need to add some magic to sip_notify.conf? I haven't quite figured out how to make it work. Found it. I needed: ; Untested - from Snom docs [reboot-snom] Event=reboot Content-Length=0 - Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] rebooting snoms in 1.6
With Asterisk 1.4 I could use commands like: /usr/sbin/asterisk -rx sip notify reboot-snom mjc_home to reboot a snom phone. Now, with 1.6, when I try that, I get: Unable to find notify type 'reboot-snom' Command 'sip notify reboot-snom mjc_home' failed. Do I need to add some magic to sip_notify.conf? I haven't quite figured out how to make it work. - Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Chef-secretary scenario
I use Snoms. I know there's the feature. I just don't know how to use it, and there's so little documentation on the web.. Anyway, with see I meant that the secretary's phone would have one of the function keys on whenever the chef is on the phone (also when he picks it up, right before dialing). Until now I've only managed to make both phones blink on incoming calls. But that's not what I want and I could've done that with extension = 11,1,Dial(SIP/11SIP/12SIP/13...). This should be very easy. Use something like: exten = 602,1,Dial(SIP/boss_officeSIP/boss_home,20,trj) exten = 602,2,Voicemail([EMAIL PROTECTED]) exten = 602,102,Voicemail([EMAIL PROTECTED]) exten = 602,hint,SIP/boss_officeSIP/boss_home exten = 603,1,Dial(SIP/secretary,20,trj) exten = 603,2,Voicemail([EMAIL PROTECTED]) exten = 603,102,Voicemail([EMAIL PROTECTED]) exten = 603,hint,SIP/secretary and set snom function keys to extensions 602 and 603. (Some firmware versions say Destination instead of extension, I think.) - Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Snom 300 Echo
Brent Davidson wrote: I thought I had the echo out of the system, but it keeps coming back... What I'm being told is that when the users call out from their snom phones they hear their own voice. There's no delay, but it's extremely Does it happen on all-digital calls (e.g., intercom between two Snom phones on the same LAN)? If it only happens on analog calls, I would buy an adapter card with real hardware echo cancellation. I use the Sangoma A20002d with my Snom 360s. I had a very hard time eliminating echo before getting the hardware echo canceller. It could be something else entirely, of course. - Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Lamps on Snom phones
Phil Knighton wrote: Been through lots of stuff in the forums, and as far as I can tell I have got the hints setup correctly and everything *should* be working fine. There must be something different within 1.4 that I'm missing? Yes, the metermaid format changed slightly. See the Parking Lot Status / Access from the Programmable Buttons / LEDs - Asterisk 1.4.x section of http://www.voip-info.org/wiki/view/Asterisk+phone+snom. - Mike ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX complaints? What are they?
randulo wrote: On Nov 30, 2007 1:40 PM, Steve Totaro [EMAIL PROTECTED] wrote: solved these issues. I think trunking (one of the main selling points of IAX due to less overhead) may be a common denominator. That does tend to explain why I've never experienced (or at least noticed) problems. I never trunk which is, as you state, another important advantage of IAX. I find the audio quality to be better on IAX - better jitter buffer! I don't trunk. - Mike ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Snom 360 lights not working on subscription
In order to get subscriptions working and the Snom 360 lights turns on, I have set everything just like all the pages in the net explain. So, I get subsciption working. I can list subscription on the asterisk and if I use the SIP trace function built in at the SNOM nad see NOTIFY messages and 200 OK responses. But I realized that content length = 0 in all messsages and there isn't any XML content in those Notify headers.. What we found is that even if you get the lights working, they go off after a few days. The BLF lights on the Snom 360s work for me (Asterisk 1.4, Snom 6.5.12 firmware), but I reboot them nightly. I have noticed that the Snom BLFs can stop working if the network is busy for a long period of time (i.e., longer than the re-registration period), like during system-wide backups and yum-upgrades. To avoid this problem, I have a cron job reboot the Snoms nightly after scheduled backups/upgrades. I'm not sure if this is a network congestion issue or a server CPU overload issue, or something else. Anyway, this arrangement does seem to be pretty reliable. To reboot a Snom: http://www.voip-info.org/wiki/view/Asterisk+phone+snom#RebootingaSNOM360320. Hope this helps. - Mike ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma Wanpipe installation problems
Rory Campbell-Lange wrote: I'm trying to install wanpipe on my new 2.6.21-2-amd64 core 2 duo machine (Debian's amd64 works with EMT64 too) to run Asterisk. I'm getting compilation errors when trying to install the wanpipe utilities. Sangoma says that 2.6.21/22 is not supported yet, just 2.6.20. They're working on it. - Mike ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] les.net losing DID's
What we tend to do with people who require out-of-area calling ability is grab a toll free DID from a bit of a bigger or more stable provider. Here in Ontario, Canada, we've had great success with Unlimitel for providing toll free DIDs. I have run across that name before as well - anyone else have any experience wth them ? (I am in ontario as well) We use them (Unlimitel) here in Ottawa. They are small, but they are stable and responsive. Outages occur occasionally (every few months), but they are dealt with rapidly and a detailed email usually explains what went wrong. I'm not really aware of anyone else in the area who handles Asterisk well. - Mike ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] uptime script?
Can someone point me to an agi script that will read back the asterisk uptime, if such a thing exists? - Mike ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DID providers in Toronto
I've had a good ongoing experience using http://www.unlimitel.ca. They are responsive and reliable. Ditto here - Unlimitel is small but reliable and supportive. - Mike ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] playing wav49/gsm files on linux?
How can I play wav49 or gsm voicemail files on FC6? Nothing seems to play them (hxplay, xine, mplayer, etc). I think I have all the normal codec packages installed. I can play regular wav files, but they're too big. - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] playing wav49/gsm files on linux?
Gordon Henderson wrote: On Mon, 5 Feb 2007, Dr. Michael J. Chudobiak wrote: How can I play wav49 or gsm voicemail files on FC6? Nothing seems to play them (hxplay, xine, mplayer, etc). I think I have all the normal codec packages installed. Have you got 'sox' installed? It comes with a command-line 'play' application which, er, plays audio files... Thanks, Gordon! That's exactly what I needed. - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] playing wav49/gsm files on linux?
Derek Whitten wrote: switch voicemail to .ogg format voicemail.conf: format=ogg but you can't actually do that, can you? WARNING[9933]: file.c:984 ast_writefile: No such format 'ogg' mp3 would be better, but it doesn't work either. WARNING[9879]: file.c:984 ast_writefile: No such format 'mp3' - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Snom side car annoyance
J. Oquendo wrote: Andrew Latham wrote: you are asking about Shared line apperance or hints. Look at this http://www.voip-info.org/wiki/view/snom+360 Been there done that page. Nothing worth noting in there. Do the line appearances work on the 12 non-sidecar buttons? - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] anyone using metermaid / parked call BLF?
I'm using 1.2.9.1, with the metermaid patches to show parking spot status on Snom BLF lights. I see from http://www.asterisk.org/node/97 that the metermaid code has changed substantially since 1.2.9.1. Is anyone successfully using the new metermaid functionality in 1.4.x? Did anyone get back to you on this? Did the Metermaid functionality get written into 1.4? I'd love to know if anyone ever replied to you privately. Jeronimo, No, I never heard back from anyone. I've cc'd this to asterisk-users - maybe someone is familiar with metermaid/1.4 now... - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] anyone using metermaid / parked call BLF?
Hi all, I'm using 1.2.9.1, with the metermaid patches to show parking spot status on Snom BLF lights. I see from http://www.asterisk.org/node/97 that the metermaid code has changed substantially since 1.2.9.1. Is anyone successfully using the new metermaid functionality in 1.4.x? I'd like to hear any good/bad experiences before I attempt even a test upgrade... - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk behind Sonicwall firewall
Barry Fawthrop wrote: Hi all Anyone using a sonicwall firewall ? I have been and then suddenly it drops UDP packets because SIP is no longer on port 5060 but some random assigned port ? Why ? Which Sonicwall model? Some (like the TZ170) have special VOIP settings, like Enable consistent NAT and Enable SIP Transformations. Check those; they work well with SIP. If you don't have one of these newer models, please see http://www.voip-info.org/wiki-IAX, in the NAT Issues section. It deals with IAX2, but the issues are same for SIP UDP. The Sonicwall UDP-connection-memory timeout may be VERY short - 30 seconds by default on some! It is adjustable in some firmware versions. I use the TZ170, but with IAX2 rather than SIP. - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2 snom 360 MWI
[EMAIL PROTECTED] wrote: Just upgraded my * box to 1.2 and don't seem to be able to get MWI working. Worked with my previous installation. My conf files are the same ( except for a few 1.2 changes ). I've tried: In sip.conf fromuser=Anyname fromdomain=my * ip vmexten=7000 Are you missing something like [EMAIL PROTECTED],password in sip.conf? Also, when I press the message key to get my voicemail my phone just calls it's extension so in effect I call myself. In the past on polycom phones I What do you have set in the Snom login preferences, in the mailbox dialog? 7000, or something else? - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dropped call question - Maximum retries exceeded on transmission
Kohler, Jeffrey wrote: I am encountering an intermittent issue where some of my calls are being dropped. Most of the calls that are made are successful. However, some calls will be dropped after having been connected for some time. Each time a call gets dropped, I get output similar to the following in the Asterisk console: ... Does anyone have any suggestions? I honestly don't know where to start investigating this issue, so if anyone has any ideas they would be greatly appreciated. Jeffrey, That's all a bit vague (how long before it drops, what protocol, are there firewalls, etc...), but my first guess would be a firewall NAT timeout. See the NAT Issues section at http://www.voip-info.org/wiki-IAX for example (it discusses IAX rather than SIP, but you get an idea of the issues). - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Asterisk hangs up after 10-15 minutes when SIPPhone is on mute
Mike wrote: Thanks Tony. Its possible that the phone stops sending RTP stream (but it certainly is receiving some!). How do I get Asterisk to stop caring whether it receives RTP or not? Yes there is a NAT between the phone the the Internet. The Asterisk server doesn't have NAT though. My Sonicwall NAT/firewall has a 15 minute default inactivity timeout for TCP NAT connections, which is suggestive (it can be increased, though). If the signaling vanishes in one direction, maybe your NAT device is forgetting about the connection. - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [RESOLVED] One way audion on Sangoma
The sangoma has hardware echo cancel ? If so it makes sence, because the settings in zapata.conf are for the software echo cancel, and that should be disabled for all interfaces that have hardware echo can. No, that is incorrect. From http://wiki.sangoma.com/wanpipe-asterisk-configure: The Wanpipe TDM driver enables HW Echo Cancellation only on channels that have active calls: It waits for zaptel to enable echo cancellation after the call has been established. Therefore, Echo Cancellation option MUST be enabled in /etc/asterisk/zapata.conf. - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Metermaid - Parking Slot
David Gagnon wrote: Finally, in the trunk all the states of my device are broken. If I downgrade to 1.2.10, everything is fine. The device get busy and ringing. But in the current trunk Asterisk SVN-trunk-r40632M none of my hints works. Anyone could confim this bugs ? David, I haven't heard of anyone using the metermaid function in the svn trunk. I haven't even seen any documentation for it - I guess its buried in the source code :-( According to bug 5779, oej extensively rewrote everything for svn trunk... better open a bug report. - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VOIP phone for Receptionist use
- Ability for the phone to ring when the receptionist is on one call and a second or third call is incoming. (this has been the biggest frustration up to now. When a second call comes, there is no tone that heard on the IP500. Perhaps I am missing a setting?) The Snom 360 can certainly do this - you can have a muted ringer, or just visual indication, or you can turn it off entirely. - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma Stops Receiving Calls
Alex Robar wrote: Hi all, I have a Sangoma A200 card with hardware echo cancellation. The card has 12 ports (10 of which are active; All FXO). Twice on this particular card I've seen all ports simply stop receiving incoming calls. There is no other indication of this, however. I am able to place outgoing calls just fine, and call other extensions without issue. When someone calls in, the line just rings and rings, with no indication that the card even sees the calls. I'm not even sure where to begin looking into this. Could anyone give me some pointers as to what I might need to be looking for? I'll be giving Sangoma tech support a call, but if anyone has any debugging pointers, they would be much appreciated. Alex, Does it work if you disable the hardware echo cancellation? I had an A20002D that started to fail after a month or too of normal operation - it would answer PSTN calls, but the callers couldn't hear me, although I heard them. Disabling the HWEC cancellation made things work, but the echo was intolerable. My vendor (Telephonyware) replaced the card (after I tried it in another computer, running another kernel, and testing the original server with a spare A20002D, and cleaning the FXO and HWEC module sockets), and the replacement has worked great since then. - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Metermaid phone compatibility
shadowym wrote: Yes, I am using the 1.2.7.1 patch on 1.2.9.1. It seemed to work fine. Still curious if anyone has this working on an Aastra phone? I can't get it to work but someone in the bug.digium.com list said they had it working on an Aastra phone. Maybe I am missing something. I tried just about everything I can think of. How does metermaid work? Is it using devicestate() similar to what the bristuff patch does or is it a different mechanism. What does the phone need to support in order for this to work. As far as I know, Aastra phones only support SIP device monitoring for BLF with the current firmware. The metermaid-1.2.7.1.txt patch uses devicestate (AST_DEVICE_NOT_INUSE and AST_DEVICE_INUSE) and SIP subscribe/notify messages. If you can use hints to monitor the status of normal lines, then it should work for the parking slots too. See the Parking Lot Status / Access from the Programmable Buttons / LEDs section at http://www.voip-info.org/wiki/view/Asterisk+phone+snom for the procedure for setting it up with Snom 360s. Maybe it will help with your Aastra too... (The trunk code has something different apparently. I'm not sure where that is documented.) - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Remote employees using Polycom 501 lose ability to receive incoming calls after few minutes.
Von L. wrote: plugged in. They work immediately after being plugged in, but they lose the ability shortly thereafter. They can always make outbound calls, but only to real phone numbers, not extensions. They each have NAT routers, and I have triple checked that they have opened/forwarded the correct ports, basically 5060-3 UDP. Once they See the NAT Issues section at http://www.voip-info.org/wiki/view/IAX. (The page is for IAX2, but the NAT issues are relevant for UDP ISP ports too). Basically, some NAT routers forget UDP mappings after a VERY short time (like 30 seconds). Took me a while to figure that out. - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Remote employees using Polycom 501 lose ability to receive incoming calls after few minutes.
Von L. wrote: plugged in. They work immediately after being plugged in, but they lose the ability shortly thereafter. They can always make outbound calls, but only to real phone numbers, not extensions. They each have NAT routers, and I have triple checked that they have opened/forwarded the correct ports, basically 5060-3 UDP. Once they See the NAT Issues section at http://www.voip-info.org/wiki/view/IAX. (The page is for IAX2, but the NAT issues are relevant for UDP SIP ports too). Basically, some NAT routers forget UDP mappings after a VERY short time (like 30 seconds). Took me a while to figure that out. - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 360 with Firmware 6.1?
Koopmann, Jan-Peter wrote: Hi, Has anybody experience with Snom360 and Firmware 6.X with Asterisk 1.2.X? I am currently using Firmware 5.5 without serious problems but wanted to make sure 6.X will work as well (including subscription etc.) Use the very latest - 6.2.1. It seems quite good. Earlier versions (including 6.2.0) had problems. - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] call quality statistics?
Is it possible to set up some sort of call-quality statistics reporting/logging for IAX2 calls? Something that can keep track of dropped packet / jitter trends? (I know iax2 show channels shows this info for active calls.) Suggestions appreciated! - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 360 doesn't register after reboot
Mimmus wrote: Hi, I'm trying my new Snom 360 phone (6.2 firmware) and I'm seeing that it doesn't register with the Asterisk 1.2.9.1 server after a reboot. I need to click Re-register in the web interface. I think that was fixed in 6.2.1. See http://www.snom.com/wiki/index.php/Beta_Firmware and http://www.voip-info.org/wiki/view/snom+360 - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fun with Echo
Brian Swan wrote: I've spent the last week or so troubleshooting echo problems at my Wife's business, and I've been able to clear up about 99% of the echo, but there is still a little residual echo that I can't seem to tweak out. The users describe it as buzzing or crackling, but what it sounds like to me is a slight echo, but just of one syllable of the word. I've followed the numerous suggestions in the mailing list archives which is what has enabled me to get this far. After trying all the echo cancelers, and all the settings on each I settled on: ... If anyone has any suggestions, I'd sure appreciate it! Consider getting a Sangoma A200D (http://www.sangoma.com/datasheets/p_a200-specs) with the optional hardware echo canceller module. It just works for echo cancellation; no tweaks required. It takes a while to figure out how to install it, but once it's working it's great! - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom firmwares suck --additional datapoint to consider
I looked long and hard at the LAN and it was basically narrowed down to the switches. In this smaller install, several cheapo Dlink ($30) switches de-aggregate a Cisco Catalyst switch. What I noticed was that any phone plugged direcly into the Catalyst did *not* lock up or reboot. Any phone plugged into the crap switches experienced the lockup. So now we are down to the cheap switches themselves. We are nuking the Dlink switches and replacing them with 3com workgroup switches, same as what we use in the large install to good effect, and I fully expect the problem to dissapear. So does anyone have any theories as to what the technical difference between a good switch and a bad or cheapo switch actually is? Lower latency? Better grounding? More cowbell? - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom firmwares suck --additional datapoint to consider
Blaming the 3com switch is very likely to be the wrong root cause. High probability the 3com was not configured properly for the phone. Just curious - what configuration issues did you have in mind? - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX2 + port translation
Hi all, I'm having trouble with incoming IAX2 calls on 1.2.7.1 - mostly they work, but sometimes the caller just gets dead air or disconnects. IAX2 debugs show HANGUP and INVALID codes in these cases, rather than a proper RINGING transaction. My firewall is doing NAT, and changing the source port from 4569 to something else - my IAX2 provider suggested this might be a problem. Is it? Should this work: steerpike*CLI iax2 show registry Host UsernamePerceived Refresh State 64.26.157.230:45698886708729 64.26.155.62:14353 60 Reg 64.26.157.230:45696134827945 64.26.155.62:14353 60 Reg 64.26.157.230:45696136866597 64.26.155.62:14353 60 Reg 64.26.157.230:45696136866675 64.26.155.62:14353 60 Reg There are four DIDs, and all are registered to an odd port (14353). Is this OK? (I am using a Sonicwall TZ170 with Enable Consistent NAT on). - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2 + port translation
If memory serves me properly what you are showing looks correct. You server is registering to your provider on port 4569 as it should. Their server is seeing you register from 64.26.155.62 and using the prt 14353 which is the port that your firewall has given that outgoing connection. Possibly that the firewall is removing that connection port after some time and your provider cannot get back to your box? Try setting the reregistration time lower than 60 and see if it helps. Hmm, it looks like I have to edit channels/chan_iax2.c to lower the registration timeout - I'm trying 15 seconds, and we'll see if that makes a difference. (You have to override the provider's requested timeout of 60 seconds). Does anyone have any idea what the IP/port PAT pair timeouts are for the Sonicwall TZ170? I see that someone had a similar problem (PAT timeouts, on an unknown device) here: http://lists.digium.com/pipermail/asterisk-dev/2005-February/009341.html - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] jitterbuffer causes flaky IAX2 incoming connections?
There isn't quite enough info in that log to tell what is going on. What you have above is part of 2 separate conversations. You have the tail end of a successful registration with 70.87.18.51 and the HANGUP of a call with 64.26.157.230 which your asterisk seems to be confused about. Could you try it again, and make sure you include the NEW message that starts the call which fails ? (assuming that is that there was a NEW !) Tim, There was no NEW. Some IAX2 messages just aren't reaching me, I think. I think that the real problem is a short timeout (maybe 60 seconds?) in my hardware firewall (Sonicwall TZ170) for UDP address:port pairs in the NAT/PAT translation memory. I've hacked the chan_iax2.c code to force a 15 second registration refresh time, instead of 60 seconds, and so far things have worked much better (i.e., the registration is like a keep-alive for the PAT translation pairs). I'll keep the list posted ... - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sangoma A200 4 port FXO card suddenly stopped answer on channels 2, 3, 4
Mike Garey wrote: It turns out that the Sangoma card had suddently decided to stop answering on channels 2,3 and 4, so if someone was using channel 1, then no other calls would be picked up. We could, however, make outgoing calls. I tried restarting Asterisk and it didn't make a difference. I then tried restarting the Wanrouter and it started working again. Has anyone else run into this problem? Do you have the optional echo canceler? The echo canceler on my A20002D died after two months, resulting in erratic one-way audio. Sangoma sent a replacement after I presented my debugging efforts to my vendor (Telephonyware). The replacement works fine. Try re-seating the FXO option card in the main card. The optional echo canceler card can also be unscrewed and re-installed. Anyway, call your vendor about the fall-through problem. The disconnect problems that you mentioned are the same for any FXO card - see http://www.asteriskguru.com/tutorials/resolving_hangup_detection_problems_fxo_tdm_voicemail.html.) I use minmessage=5 maxsilence=3 silencethreshold=128 in voicemail.conf. Seems to work reasonably well. - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] jitterbuffer causes flaky IAX2 incoming connections?
I've been having problems with incoming IAX2 calls - some work, but a large fraction are answered with dead air or disconnects from my IAX provider. Disabling the jitterbuffer seems to eliminate the problem (so far)! Has anyone else seen this? I'm using 1.2.6, but I'm not sure what my provider is using. A snippet of the a failed incoming call IAX2 debug is attached below (with jitterbuffer on). Note the HANGUP and INVAL codes. - Mike Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: REGACK Timestamp: 00087ms SCall: 00235 DCall: 3 [70.87.18.51:4569] USERNAME: avtech DATE TIME : 2006-05-25 09:26:46 REFRESH : 60 APPARENT ADDRES : IPV4 64.26.155.62:14353 Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00087ms SCall: 3 DCall: 00235 [70.87.18.51:4569] Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 000 Type: IAX Subclass: HANGUP Timestamp: 04016ms SCall: 00379 DCall: 0 [64.26.157.230:4569] CAUSE CODE : 0 Tx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: INVAL Timestamp: 0ms SCall: 0 DCall: 00379 [64.26.157.230:4569] steerpike*CLI ___ --Bandwidth 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] jitterbuffer causes flaky IAX2 incoming connections?
Dr. Michael J. Chudobiak wrote: Disabling the jitterbuffer seems to eliminate the problem (so far)! Has anyone else seen this? I'm using 1.2.6, but I'm not sure what my provider is using. Oops, the problem still happens without the jitterbuffer - so something else is causing it. Any ideas? - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Are my expectations too high?
I'm getting a slight echo...sometimes...it varies from call-to-call, but the biggest problem I have is a constant hiss in the background. Again, this varies from call-to-call. I know my SIP phones are fine as SIP-to-SIP calls on my LAN work perfectly. I only have problems going out to the PSTN. Get an FXO card with hardware echo cancellation. I use the Sangoma A20002D (four FXO ports with echo cancellation). It definitely costs more, but the hardware echo cancellation makes a huge difference in call quality! Software echo cancellation doesn't really work... I don't know what to suggest about your hiss problem though. - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Are my expectations too high?
Derek Lee-Wo wrote: With this card, would you say your audio quality is identical to that of an analog phone connected directly to the PSTN? I'm trying to understand if I should expect some audio degradation when going through Asterisk. In my experience, this card provides the sames quality that our old Nortel Norstar PBX provided using the same lines. However, our PSTN lines are not particularly good (heavily filtered, low volume). We find that local SIP calls through a nearby provider have MUCH better audio quality (clarity, lack of noise) than our PSTN lines. However, long-distance SIP calls suffer from choppiness, so we use the PSTN for them. Unfortunately, it takes a LOT of experimenting to find the perfect mix. - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Are my expectations too high?
While I agree that the Sangoma cards are good, your statement that software echo cancellation doesn't really work is ... incorrect. Software echo cancel works very well if it's done correctly, if your audio levels are where the canceller's sweet spot is, and the tail is not longer than the longest tail that the canceller's designed to work with. Most people don't attempt to set their stuff up correctly and they achieve poor results and blame it on the software. Well, I couldn't make the software echo canceler totally eliminate the echo. It did reduce it, but not enough. I tried enabling/disabling various algorithms, tweaking gains, thresholds, taps, etc, etc, etc... The software approach is great in theory, but the hardware echo canceler just works, without configuring anything - in my experience, anyway. I'm curious though: did you find that the software approach totally eliminated echo in typical situations? - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 320 Shared line + speed dial
Just after some info on the Snom 320 before I got out an buy some... I'm looking to use the shared line feature and hints with * so that i can monitor the activity of other users, but I'm not sure If this also turns the programmable buttons into a speed dial for quick transfers etc (or if it can be done). Ideally, I just want the users to be able to see the state of other users and be able to transfer to that user by using the programmable buttons... Is this possible with this phone ?? Set the buttons to Destination mode, and set up corresponding hints in extensions.conf. Then the LED shows the user's status and hitting transfer+button will transfer a call to that user. Just hitting the button will dial that user's extension. See SNOM SUBSCRIBE/NOTIFY support for monitoring extension states at http://www.voip-info.org/wiki/view/Asterisk+phone+snom. - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom firmwares suck
Remco Barende wrote: Most people seem quite positive about Snom phones, I cannot share this opinion. The displays are dying quite often, and firmware is buggy. I have tried every firmware from 4.5 up to 5.x and 6.04 but keep having problems with phones locking up or rebooting during an ongoing conversation. REALLY annoying for a phone that is advertised / targeted as a business class phone Remco, I have a dozen Snom 360s. One had a defective LCD that would become garbled after time. Snom support quickly confirmed that this was a known issue, and my vendor (voipsupply) quickly sent a replacement. I've never seen any lockups or reboots. I reboot the phones each night at midnight, just to be safe - try doing that to see if it reduces problems. I've posted a sample perl cron script at http://www.voip-info.org/wiki/view/Asterisk+phone+snom to show how. I use shielded ethernet cables (STP) everywhere too. Try that - good grounding may be beneficial. It can't hurt, anyway. Snom support is pretty responsive. Try emailing [EMAIL PROTECTED]; they have fixed some issues for me (for example, the clock was showing the wrong time due to daylight savings time problems). Try using a Grandstream GXP-2000 phone, and you'll see why people like the Snoms :-) Hope this helps - let us know if anything makes a difference! - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom firmwares suck
Old English saying A bad workman always blames his tools I don't think that's fair... these are very complicated phones, made in China for very low prices. Problems do occur with them. Some Snom LCDs do have problems. There are firmware glitches, though I've only run into minor ones. Overall though, they are very good phones. - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Non automated call parking
Steven wrote: Make extensions that can hold a call. (like a 701) Make this extension hintable for use in button programming. If I am on a call and hit a non-lit button, it parks it there. If I am not on a call and push the lit button, I connect to the park. I suppose that if you are on a call and hit a lit button, it should either not be processed, or should join as a three way call. (either logic is justifiable) These park extensions should still be callable so analog, etc. extensions can also connect to them. Steven, That arrangement would be great, but right now the closest existing method is the metermaid patch at http://bugs.digium.com/view.php?id=5779, and it looks like that won't even make it into 1.4. Sigh. (oej: people need this patch!) I created a bounty two years ago at http://www.voip-info.org/wiki/view/Asterisk+bounty+snom+call+park for an arrangement like you describe, but there was no interest, so I dropped sponsorship of it. You can take over the bounty if you like. I'm using the metermaid patch on 1.2.6, and it works very nicely with my Snom 360s. (Press a 700 button to park, and observe the 701-7xx button LEDs to see parking slot status). - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sangoma A200D problem
I've been having problems with my A20002D lately - callers from the PSTN don't hear me when I answer, but I hear them. Disabling echo cancellation in zapata.conf brings the audio (and echo) back. This used to work fine, until two days ago. Well, just to complete my own thread, this seems to be a probable hardware defect and Sangoma is sending a replacement. I had to live with software echo cancellation for a day or two - shudder - it's amazing how much better the hardware echo cancellation is! - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sangoma A200D problem
Hi all, I've been having problems with my A20002D lately - callers from the PSTN don't hear me when I answer, but I hear them. Disabling echo cancellation in zapata.conf brings the audio (and echo) back. This used to work fine, until two days ago. The only weird thing in the logs is this: May 12 07:42:53 steerpike wan_ecd: wp1ec: The H100 slave has lost its framing on the bus! May 12 07:42:53 steerpike wan_ecd: wp1ec: The CT_C8_A clock behavior does not conform to the H.100 spec! Is this a problem? (The server is an HP ProLiant DL140 G2). Has anyone else seen this type of erratic problem? I've tried re-seating the A20002D card, the FXO plug-ins, and the echo-canceller plug-ins. Right now it works ... we'll see if it stays that way. - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Speex fans?
Hi all, I've been testing various codecs to eliminate choppiness that I sometimes get on my Asterisk IAX2 DSL provider (Exgn) connections, and Speex seems to work the best, so far - but Speex seems oddly unpopular. Can anyone share their experiences with Speex (good and bad)? Is anyone using it in a production environment? I like the variable bit rate and packet loss concealment features... - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Sangoma A200D problem
Last time I had this problem was following a unclean powerdown and the solution was: - Kill Asterisk - Stop wanpipe - cd /etc/wanpipe/wan_ec - In there there should be 2 files: wan_ec_pid wan_ec_socket= - Delete those files - Perform a reboot of your system Andre, Thanks for the tip, it's food for thought - but I actually don't have those two files! Do you still have wan_ec_pid and wan_ec_socket files with the latest drivers (wanpipe beta 2.3.4)? - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] echo in Snom 360 phones
Hi all, One of my users reports frequently hearing echo on her Snom 360 phone, even while talking to other Snom phones (via Asterisk) on the same LAN (i.e., all-digital low-latency connection). I can never reproduce it though, and swapping the phone didn't help. Has anyone else seen mystery echo on Snom phones? Any suggestions for debugging? On my own Snom 360, I sometimes hear an echo for the first second or two, and then it goes away. I guess an echo cancellation circuit kicks in, inside the Snom. - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 360 Firmware 6.0 - Microbrowser
TWV wrote: By now, every Snom fan should have installed the 6.0 (beta) firmware :-) See http://www.snom.com/wiki/index.php/Beta_Firmware I had to revert back to 5.5, because 6.0 kept garbling my LCD screen (the screen would become unreadable). You might want to wait for 6.0.1 :-) - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] simple wav ringtones?
Can anyone suggest a good source of simple-but-distinctive wav ringtones for a business environment, to use on Snom phones? The built-in Bellcore tones are hard to distinguish, to my ear. I want variations of ring, ring, not Madonna or Eminem :-) - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400 FXO module not answering or dialing out.
Hi Mike, may I ask where you purchsded your A200 card from? I managed to get one of the pre-production cards from Sangoma back in November, however there are some bugs with it and I am unable to flash the firmware or run latest drivers with it. Sure, I got it at: http://www.telephonyware.com/telephonyware/tw00274.html So far I've done only basic tests. I haven't tried flashing any firmware, or the hardware echo canceller. However, I am using the latest drivers, and calls seem to go through just fine. I was impressed that they supplied both a full-height and half-height card mounting bracket. I ended up using the optional half-height bracket in my server, for space reasons. Beware that the card is quite thick, so it can be tricky to install in 1U servers! The installation procedure is pretty well documented at http://sangoma.editme.com. I had to hack my init scripts to make things start gracefully, but that's not a big deal. - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] best MTU?
Hi all, I have several locations, each connected by a Sonicwall VPN through PPPOE DSL, with Snom 360 phones. I've found that I have to tweak the Asterisk server MTU (inside one of the firewalls) to get everything to work just right. Set the server MTU too low, and the Snom phones don't communicate correctly anymore. Too high and the phones work, but the server can't access the web (for yum updates, for example). So I've settled on an MTU of 1448. Has anyone else struggled with this? What did you settle on? - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400 FXO module not answering or dialing out.
Hadley Rich wrote: Hi all, I have hit a wall configuring a TDM400, I have set these up before without issue but today I just can't seem to figure out what I am doing wrong. I couldn't make TDM400/FXO work on my 1.2.5 Asterisk either. It wouldn't answer calls, for unknown reasons. I gave up and replaced it with a Sangoma A200 card, which works just fine. - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] An FXO version of IAXy?
D-Link has a 4 port FXO device on their site. http://www.dlink.com/products/?sec=2pid=451 Apparently it hasn't shipped yet and costs $500.00 I've been testing a AudioCodes MP104-FXO-C3S (around $1000) 4-port FXO box. It works, but the number of configuration options are staggering, complex, and inter-related, and the documentation support just aren't good enough to make installation easy. The D-link DVG-3004S is pretty much impossible to get. There is also the Mediatrix 1104 (also around $500), but it is reputed to be hard to configure (no web interface - just snmp!). Slapping a Sangoma A200 into a computer (and configuring it through Zaptel/Asterisk) is much, much simpler than trying to make the appliance gateways work, at least in my experience. - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Analog POTS line - Rhino FXO Channel Bank - No Hangup
[EMAIL PROTECTED] wrote: If so, is there a way to detect the hangup? Check out http://www.asteriskguru.com/tutorials/resolving_hangup_detection_problems_fxo_tdm_voicemail.html for some possible clues. - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] can't get TDM400P to answer
Hi all, I can't figure out why my TDM400P (with one FXO plugin) won't answer any calls. There are no messages in the Asterisk console when a call is placed to the FXO line from the PSTN. Any suggestions would be most appreciated. The wctdm and zaptel modules are loaded: [EMAIL PROTECTED] asterisk]# lsmod | grep wc wctdm 37952 0 zaptel189700 1 wctdm The green LED on the input connector is lit. ztcfg says: ... Channel map: Channel 04: FXS Kewlstart (Default) (Slaves: 04) 1 channels configured. My zaptel.conf is: fxsks=4 loadzone=us defaultzone=us and zapata.conf is: [channels] group=1 context=tdm400p-inbound signalling=fxs_ks echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=1.5 txgain=1.5 immediate=no busydetect=no callprogress=no musiconhold=default usecallerid=yes callerid=asreceived channel = 4 The relevant sections of extensions.conf are: [tdm400p-inbound] exten = s,1,Ringing() exten = s,2,Goto(MainMenu,s,1) exten = s,3,Hangup; ... [MainMenu] exten = s,1,Wait,3 ; ring for 1 second exten = s,2,Answer ; answer exten = s,3,Background,welcome etc... The TDM400P power connector is attached, even though it isn't supposed to be required for FXO modules. Any ideas? - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2 + Sonicwall
OK apart of my beleive that sonicwall is a piece of crap (personal), try to do a port forwarding for the IAX port (4569) Saul, Why do you consider Sonicwalls to be crap? Aside from this odd issue (which is fixed by using an obscure setting) they've been rock solid for me, for years. - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX2 + Sonicwall
Hi all, I currently have an Asterisk test server behind a TZ170 Sonicwall firewall / NAT box, with several DIDs. I've found that inbound IAX2 calls don't work reliably (i.e., I get a busy tone) unless I enable Use Consistent NAT in the Sonicwall. This feature is poorly documented by Sonicwall, so I thought I'd pass it along. Has anyone else run into this, or figured out the rationale for it? - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2 + Sonicwall
I've found that inbound IAX2 calls don't work reliably (i.e., I get a busy tone) unless I enable Use Consistent NAT in the Sonicwall. This feature is poorly documented by Sonicwall, so I thought I'd pass it along. I've used the iaxcomm softphone and a snom 200 behind serveral different sonicwalls over the past year or so without any problem. The sonicwall should not be a problem for iax calls at all. I think the problem occurs when an Asterisk server inside the firewall tries to register multiple DIDs with one IAX2 provider outside the firewall. The Asterisk server worked fine when it was connected outside the firewall. The Sonicwall TZ170s do handle SIP transformations very nicely, though, if your Asterisk server is outside the firewall. - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] pstn to asterisk, DVG-3004S, MP104?
Hi all, I want to link three incoming Bell Canada centrex pstn lines (which currently go to an old norstar pbx) into asterisk. Can anyone suggest the most painless (i.e., just works) way to do this? Has anyone used the D-link DVG-3004S four-port FXO-to-sip adapter, or the twice-as-costly Audiocode MP104-FXO-C3S? I know the Digium TDM400P or Sangoma A200 are options too, but it's physically easier for me to use an FXO ethernet appliance (simpler wiring). I've had problems with hangup detection in some tests of different approaches... Suggestions appreciated! - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] parking slot lights - testers wanted
Hi all, The metermaid patch allows you to use the programmable buttons and LEDs on phones (like the Snom 2xx or 3xx) to view the status of parking slots and transfer to them. This should be really useful for small-office environments. Anyway, the patch seems to work with Snom phones (and hopefully others) now! The curious are encouraged to download the metermaid-v3.txt patch against v1.2.4 for testing and feedback! See http://bugs.digium.com/view.php?id=5779 for details. - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX provider recommendation wanted
Hi, Can someone recommend an IAX provider for US DIDs who will: 1) Accept Canadian credit cards (rules out Junction Networks!) 2) Can do local number porting (LNP) 3) Have great audio quality I tried Teliax, but the IAX audio quality was terrible - pops and clicks galore! The Teliax SIP quality was better, but still horrible compared to my Canadian DID IAX provider, Unlimitel.ca. - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] jitterbuffer and DTMF conflict?
I find that DTMF does not work reliably if jitterbuffer=on for certain IAX providers. For instance, DTMF tones are missed entirely about half the time when I dial into an exgn.net account. However, it always works fine for an unlimitel.ca account. Someone else has seen this too: http://bugs.digium.com/view.php?id=6011 Can anyone suggest a workaround (other than jitterbuffer=off)? - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] jitterbuffer and DTMF conflict?
Rich Adamson wrote: I find that DTMF does not work reliably if jitterbuffer=on for certain Can anyone suggest a workaround (other than jitterbuffer=off)? Might try turning off trunking (assuming you have it turned on) and test again. Seems a couple of parameters interact and probably has something to do with different versions of iax. Rich, I'm not sure if trunking is on by default, but I turned it off explicitly. No difference, sadly. - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX provider recommendation wanted
Martin Joseph wrote: snip 3) Have great audio quality This is somewhat a meaningless question, as the route from you to the call terminating service can make or break the quality. Sure, but some carriers have problems inside their own networks. I can optimize the routing to the provider as needed, but it doesn't matter if they aren't actively addressing support issues and their own connections. - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] metermaid patch
I'd like to be able to use my Snom 360 LEDs to view the status of parking slots, so I'm trying to install the metermaid patch (http://bugs.digium.com/view.php?id=5779). Can someone help an svn newbie figure out how to install this patch? I've done the following: svn checkout http://svn.digium.com/svn/asterisk/team/oej/metermaids cd metermaids make clean make but I get this error: asterisk.c: In function ‘main’: asterisk.c:2209: error: ‘option_dumpcore’ undeclared (first use in this function) asterisk.c:2209: error: (Each undeclared identifier is reported only once asterisk.c:2209: error: for each function it appears in.) make: *** [asterisk.o] Error 1 [EMAIL PROTECTED] metermaids]# What am I doing wrong? (I have 1.2.4 installed and working fine, from the tar.gz). - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk, SIP phone , NAT
nat=yes qualify=yes That works, but it works better if you use a NAT/firewall box that can do VOIP transformations automatically. The Sonicwall TZ170 can do this. It rewrites the packets auto-magically so things just work. The above parameters can be set to no then. It seems to work more reliably that way, in my experience. - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] snom 360 problem - only one call works after reboot
After rebooting, I can make one outgoing call successfully. Subsequent calls don't work - the 360 just seems to do nothing after pressing the OK button (but I can cancel the call, the phone isn't frozen). The Asterisk console shows the first call going through, but nothing appears for the subsequent calls, so they aren't even getting to Asterisk. Define an outbound proxy for your line. Dan, Thanks, but that wasn't the problem. I had to set RTP Encryption on the snom 360 to off. By default it is on. I have no idea why it causes a problem, but that is the solution! - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] snom 360 problem - only one call works after reboot
Hi, I updated the firmware in my Snom 360 from 4.3 to version 5.3.6 (and then back to 5.2), but I'm having a weird problem now: After rebooting, I can make one outgoing call successfully. Subsequent calls don't work - the 360 just seems to do nothing after pressing the OK button (but I can cancel the call, the phone isn't frozen). The Asterisk console shows the first call going through, but nothing appears for the subsequent calls, so they aren't even getting to Asterisk. Has anyone seen this? - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] call parking hint
Hi, Is it possible to use the hint priority to allow call parking slots to be monitored on (for example) Snom indicator lamps? How do you refer to the slots (i.e., what is the channel) in the hint? - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Phones-Receptionist Setup
I'm not surprised. Asterisk is a PBX, not a key system or a hybrid system. The kind of functionality that is being described here is one or both of those 'other' beasts. Now I'm not saying that this wouldn't be nice, or even a long term requirement if you really want to open the entire SME market, but it's not typical PBX behavior. Is there an open source key system, comparable to *? - Mike ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
