Re: [asterisk-users] phones behind nat
On 9/11/15 12:59 PM, Jerry Geis wrote: I have a setup where I have polycom phones in an office, behind firewall, going out to a server located elsewhere. I have set nat=force_rport,comedia for my phones. so if I call OUT to my cell I get audio both ways and the call is fine. My issue is if I call phone to phone in the office the phone doesnt even ring. The CLI shows I'm calling the correct extension like SIP/524. I am using asterisk 11.19.0 Is there another setting to correctly to this type of calling? Thanks, Jerry There are usually two issues that can cause this behaviour. One is that you do not have a correct "localnet" definition and the other is that you have directmedia=yes on your sip.conf. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez +52 (55)9116-91161 -- _ -- 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] phones behind nat
I have a setup where I have polycom phones in an office, behind firewall, going out to a server located elsewhere. I have set nat=force_rport,comedia for my phones. so if I call OUT to my cell I get audio both ways and the call is fine. My issue is if I call phone to phone in the office the phone doesnt even ring. The CLI shows I'm calling the correct extension like SIP/524. I am using asterisk 11.19.0 Is there another setting to correctly to this type of calling? Thanks, Jerry -- _ -- 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] phones behind nat
On Fri, 11 Sep 2015, Jerry Geis wrote: My issue is if I call phone to phone in the office the phone doesnt even ring. The CLI shows I'm calling the correct extension like SIP/524. The lack of sufficient connectivity to signal ringing suggests taking a peek with wireshark may be fruitful as well as reviewing the configuration of the endpoints. Can you check the web page on the phones to confirm the IP addresses and netmasks are as expected? (What you think your configuration does is less important than what the phone thinks it does.) I recently broke a lot of things between my office and my home when I decided to split 192.168.0.0 with a 255.255.255.128 net mask. I also recently broke a working configuration by running Asterisk and OpenSIPS on 5060. The phones would ring but could not answer. Lost a lot of time until I started confirming the really basic stuff and entered 'sudo netstat -a -n -p | grep 5060' -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST -- _ -- 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] Phones don't stop ringing when queue is answered
A few things I would try- Change WaitExten to Wait(2) Change Queue(queue_level_1,rtnC,18) to Queue(queue_level_1,rtnC,,,18) Add an Answer() after the first Wait(2) -- _ -- 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] Phones don't stop ringing when queue is answered
James, The WaitExten()s just provide a pause between the two Queue() calls to let the first group of phones finish ringing. In this example I am ringing the same group (queue_level_1) twice, however in a real-world scenario I would ring queue_level_1 and then ring queue_level_2 which each have a different list of phones. Thanks, Andrew - Original Message - From: James Thomas jthomas...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, May 7, 2015 10:20:10 AM Subject: Re: [asterisk-users] Phones don't stop ringing when queue is answered What purpose do the WaitExten()s serve here? Are you really allowing the caller to connect to different extensions in the test-queue context? Have you tried without the WaitExten()s? -- _ -- 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
Re: [asterisk-users] Phones don't stop ringing when queue is answered
What purpose do the WaitExten()s serve here? Are you really allowing the caller to connect to different extensions in the test-queue context? Have you tried without the WaitExten()s? -- _ -- 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] Phones flashing but not ringing
We have been using Asterisk SLA for a while with Cisco SPA series phones. Once in a while the phones flash, but not ring when a call comes in. We can pick it up and talk to the caller even if that's the case. This is pretty random (might not happen for couple of weeks). The quick solution is to restart Asterisk which we are trying to avoid. What might cause this? Thanks, Matt -- _ -- 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] Phones flashing but not ringing
Have you tried restarting the phone instead of Asterisk? I don't think that Asterisk sends separate commands to the bell and to the call LED. Since the LED is flashing, it is likely that the SIP INVITE signal from Asterisk is ok. Also the ring tone normally does not come from Asterisk itself. jg -- _ -- 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] Phones flashing but not ringing
Have you tried restarting the phone instead of Asterisk? I don't think that Asterisk sends separate commands to the bell and to the call LED. Since the LED is flashing, it is likely that the SIP INVITE signal from Asterisk is ok. Also the ring tone normally does not come from Asterisk itself. We tried it and it doesn't help. It's not one phone, multiple phones do it at the same time. I think it's related to Asterisk SLA - maybe device states get messed up. Matt -- _ -- 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] Phones flapping with * and Sonicwall.
Hi all, I've got a bunch of Polycom 301's at a site that sits behind a Sonicwall firewall. What I'm seeing is that the phones are constantly becoming unavailable, followed shortly by becoming available again. The phones register just fine and sound great on out-bound calls. The phones are configured NAT=yes and type=friend, as I do with all my Polycoms. A sniffer trace indicates that Asterisk is sending an OPTIONS request to the phones but no reply is being sent... most of the time. I'm thinking it's a firewall/NAT timeout issue. Has anyone seen this? Has anyone fixed it? Any ideas, otherwise? TIA. -- Take care and have fun, Mike Diehl. -- _ -- 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] Phones flapping with * and Sonicwall.
On 10/13/2011 06:21 PM, Mike Diehl wrote: I'm thinking it's a firewall/NAT timeout issue. Has anyone seen this? Has anyone fixed it? Any ideas, otherwise? Did you try turning off the SIP ALG on the Sonicwall? Regards, Patrick -- _ -- 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] Phones flapping with * and Sonicwall.
Have you tried adding 'qualify=no' in the peer definition? Are all the natted phones using port 5060 as their SIP port? I don't know how many a bunch is but if it's not too many you might try having each phone bind to a different port for its SIP signaling, sometimes that is helpful with strict firewalls. e.g. Mary x102 use port 50102 John x114 use port 50114 This port would need to be set at the endpoint itself either manually or via the provisioning server, setting it in Asterisk has no effect. Just an idea. Luke -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Diehl Sent: Thursday, October 13, 2011 12:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Phones flapping with * and Sonicwall. Hi all, I've got a bunch of Polycom 301's at a site that sits behind a Sonicwall firewall. What I'm seeing is that the phones are constantly becoming unavailable, followed shortly by becoming available again. The phones register just fine and sound great on out-bound calls. The phones are configured NAT=yes and type=friend, as I do with all my Polycoms. A sniffer trace indicates that Asterisk is sending an OPTIONS request to the phones but no reply is being sent... most of the time. I'm thinking it's a firewall/NAT timeout issue. Has anyone seen this? Has anyone fixed it? Any ideas, otherwise? TIA. -- Take care and have fun, Mike Diehl. -- _ -- 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] Phones flapping with * and Sonicwall.
Looks like we fixed it. The NAT session defaulted to 30 seconds...! When we increased it to 2 minutes, the problem went away! Thank you for your time! Mike. On Thursday 13 October 2011 1:00:49 pm Luke Hamburg wrote: Have you tried adding 'qualify=no' in the peer definition? Are all the natted phones using port 5060 as their SIP port? I don't know how many a bunch is but if it's not too many you might try having each phone bind to a different port for its SIP signaling, sometimes that is helpful with strict firewalls. e.g. Mary x102 use port 50102 John x114 use port 50114 This port would need to be set at the endpoint itself either manually or via the provisioning server, setting it in Asterisk has no effect. Just an idea. Luke -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Diehl Sent: Thursday, October 13, 2011 12:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Phones flapping with * and Sonicwall. Hi all, I've got a bunch of Polycom 301's at a site that sits behind a Sonicwall firewall. What I'm seeing is that the phones are constantly becoming unavailable, followed shortly by becoming available again. The phones register just fine and sound great on out-bound calls. The phones are configured NAT=yes and type=friend, as I do with all my Polycoms. A sniffer trace indicates that Asterisk is sending an OPTIONS request to the phones but no reply is being sent... most of the time. I'm thinking it's a firewall/NAT timeout issue. Has anyone seen this? Has anyone fixed it? Any ideas, otherwise? TIA. -- Take care and have fun, Mike Diehl. -- _ -- 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] Phones don't stop ringing
On Wed, Nov 10, 2010 at 8:52 AM, Paulo Santos paulo.r.san...@sapo.ptwrote: Hello list, I'm having some issues with some phones that don't stop ringing after the call is answered somewhere else. Basically, a call comes, enters a queue and all the phones in the queue ring. The problem is that when the call is answered, some phones don't stop ringing. What version are you using? I'm having a similar problem with 1.8.0. Yesterday I was expecting a call at home and I modified my dialplan to ring my work number via GoogleVoice along with the other phones in the house. When the call came through I answered it at work but my wife said all the phones in the house continued to ring, apparently until the call was completed. I haven't done any debugging on it yet. -- Chris -- _ -- 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] Phones don't stop ringing
Hello list, I'm having some issues with some phones that don't stop ringing after the call is answered somewhere else. Basically, a call comes, enters a queue and all the phones in the queue ring. The problem is that when the call is answered, some phones don't stop ringing. I don't know if it is a configuration file, but I don't think so. queues.conf, sip.conf and extensions.conf: http://pastebin.com/8TTHpk4Z I've also captured a moment when this occurred: http://b.imagehost.org/0630/sip_flow.png The green one is the one that didn't stop ringing. The phone sends the first 180 Ringing _after_ the call is answered. This can be a network issue or a buggy firmware on the phones, but either way, shouldn't Asterisk send a CANCEL to an INVITE even if the phone didn't send 180 Ringing? Thanks in advance. Best regards, Paulo Santos -- _ -- 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] Phones slow to ring
It worked! I['ll have to figure out how to add the dial string to the phone. Thanks a bunch for your help On Thu, Nov 4, 2010 at 9:04 PM, Mark Phillips g7...@g7ltt.com wrote: I would second that. If you don't set a dial string in your handset then it waits for N seconds before submitting the call. Pressing # will force an immediate dial. Mark On 11/04/2010 07:19 PM, Cary Fitch wrote: Watch the console as you dial. Dial the number and “#”. The ring should be “instant”. Or if not, look at the console and report what you see. Cary Fitch *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *jy *Sent:* Thursday, November 04, 2010 5:32 PM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] Phones slow to ring I am new to asterisk and using it for a research project. Have set up an server (version 1.6.2.6) and 2 SIP phones (Linksys spa901) which are registering fine with the server. They are able to call one another, however, the problem is it takes roughly 8-10 seconds for the called phone to ring. I have a really simple dialplan using only 4 digit extensions and have turned off callerid. Both phones are on the same subnet and I have enabled nat and keepalive. Does anyone have an idea what could be wrong here or idea on how to debug this problem? Thanks, John -- /\/\ark Phillips -- _ -- 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] Phones slow to ring
I am new to asterisk and using it for a research project. Have set up an server (version 1.6.2.6) and 2 SIP phones (Linksys spa901) which are registering fine with the server. They are able to call one another, however, the problem is it takes roughly 8-10 seconds for the called phone to ring. I have a really simple dialplan using only 4 digit extensions and have turned off callerid. Both phones are on the same subnet and I have enabled nat and keepalive. Does anyone have an idea what could be wrong here or idea on how to debug this problem? Thanks, John -- _ -- 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] Phones slow to ring
Watch the console as you dial. Dial the number and #. The ring should be instant. Or if not, look at the console and report what you see. Cary Fitch _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jy Sent: Thursday, November 04, 2010 5:32 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Phones slow to ring I am new to asterisk and using it for a research project. Have set up an server (version 1.6.2.6) and 2 SIP phones (Linksys spa901) which are registering fine with the server. They are able to call one another, however, the problem is it takes roughly 8-10 seconds for the called phone to ring. I have a really simple dialplan using only 4 digit extensions and have turned off callerid. Both phones are on the same subnet and I have enabled nat and keepalive. Does anyone have an idea what could be wrong here or idea on how to debug this problem? Thanks, John -- _ -- 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] Phones slow to ring
I would second that. If you don't set a dial string in your handset then it waits for N seconds before submitting the call. Pressing # will force an immediate dial. Mark On 11/04/2010 07:19 PM, Cary Fitch wrote: Watch the console as you dial. Dial the number and “#”. The ring should be “instant”. Or if not, look at the console and report what you see. Cary Fitch *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *jy *Sent:* Thursday, November 04, 2010 5:32 PM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] Phones slow to ring I am new to asterisk and using it for a research project. Have set up an server (version 1.6.2.6) and 2 SIP phones (Linksys spa901) which are registering fine with the server. They are able to call one another, however, the problem is it takes roughly 8-10 seconds for the called phone to ring. I have a really simple dialplan using only 4 digit extensions and have turned off callerid. Both phones are on the same subnet and I have enabled nat and keepalive. Does anyone have an idea what could be wrong here or idea on how to debug this problem? Thanks, John -- /\/\ark Phillips -- _ -- 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] Phones won't stop ringing
Quoting Jason Aarons (US) jason.aar...@us.didata.com: I'm experiencing runaway ringing too, can we make this a class action against someone? Strangely enough, I have experienced this on what is a small domestic system - when a call is answered, sometimes other SIP phones (softphones only tried so far) keep ringing when the call is answered on a Zap or IAX phone - I think I have also had it happen when the answering phone was a SIP phone. It happens just occasionally, but can be a bit of a nuisance. Zap phones have never rung on and I can't remember it happening with an IAX phone. -- Phil Reynolds mail: phil-aster...@tinsleyviaduct.com Web: http://www.tinsleyviaduct.com/phil/ Waltham 66, Emley Moor 69, Droitwich 79, Windows 95 This message was sent using IMP, the Internet Messaging Program. -- _ -- 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] Phones won't stop ringing
On Wed, Mar 10, 2010 at 09:27:46PM -0600, Chris Owen wrote: On Mar 10, 2010, at 9:05 PM, Warren Selby wrote: On Wed, Mar 10, 2010 at 8:27 PM, Chris Owen ow...@hubris.net wrote: This normally works fine but occasionally when someone picks up the call other phones don't seem to realize the call has been answered and will continue to ring. On at least once occasion I saw a call that went to voicemail and all the phones continued to ring. When this happens the phones will continue to ring forever. The only way to stop them from ringing is to pickup the handset at which time they realize there is no call and reset. What kind of phones? All Aastra 6755i I don't have any useful insight here. Thus the obvious thing to sugget is that you try to provide a SIP-level trace of such an event (sip debug peer PHONE_NAME) -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- 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] Phones won't stop ringing
We're having an issue that isn't easily googleable so I thought I might might try here. We have several customers who want all their extensions to ring on incoming calls. Frankly I think it is craziness to ring 11 extensions all at once but that is how they want it. We're doing this by creating an incoming route that goes to a hunt list containing all the extensions. This normally works fine but occasionally when someone picks up the call other phones don't seem to realize the call has been answered and will continue to ring. On at least once occasion I saw a call that went to voicemail and all the phones continued to ring. When this happens the phones will continue to ring forever. The only way to stop them from ringing is to pickup the handset at which time they realize there is no call and reset. I'm pretty sure the underlying cause of this problem is funkiness in their network but it just seems to happen too easily and then once it stops it won't stop.Even if this is caused by network issues is there anything I can do to mitigate the problem. Just seems wrong that the phones would continue to ring forever. Chris -- _ -- 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] Phones won't stop ringing
On Wed, Mar 10, 2010 at 8:27 PM, Chris Owen ow...@hubris.net wrote: This normally works fine but occasionally when someone picks up the call other phones don't seem to realize the call has been answered and will continue to ring. On at least once occasion I saw a call that went to voicemail and all the phones continued to ring. When this happens the phones will continue to ring forever. The only way to stop them from ringing is to pickup the handset at which time they realize there is no call and reset. What kind of phones? -- Thanks, --Warren Selby http://www.selbytech.com -- _ -- 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] Phones won't stop ringing
Chris- Sounds like the Toyota bug has migrated to Asterisk... it's mutated into runaway ringing :-) -Jeff Sorry for my attempt at levity; just couldn't help it plus I'm sure Digium guys will know how to resolve. We're having an issue that isn't easily googleable so I thought I might might try here. We have several customers who want all their extensions to ring on incoming calls. Frankly I think it is craziness to ring 11 extensions all at once but that is how they want it. We're doing this by creating an incoming route that goes to a hunt list containing all the extensions. This normally works fine but occasionally when someone picks up the call other phones don't seem to realize the call has been answered and will continue to ring. On at least once occasion I saw a call that went to voicemail and all the phones continued to ring. When this happens the phones will continue to ring forever. The only way to stop them from ringing is to pickup the handset at which time they realize there is no call and reset. I'm pretty sure the underlying cause of this problem is funkiness in their network but it just seems to happen too easily and then once it stops it won't stop.Even if this is caused by network issues is there anything I can do to mitigate the problem. Just seems wrong that the phones would continue to ring forever. Chris -- _ -- 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
Re: [asterisk-users] Phones won't stop ringing
On Mar 10, 2010, at 9:05 PM, Warren Selby wrote: On Wed, Mar 10, 2010 at 8:27 PM, Chris Owen ow...@hubris.net wrote: This normally works fine but occasionally when someone picks up the call other phones don't seem to realize the call has been answered and will continue to ring. On at least once occasion I saw a call that went to voicemail and all the phones continued to ring. When this happens the phones will continue to ring forever. The only way to stop them from ringing is to pickup the handset at which time they realize there is no call and reset. What kind of phones? All Aastra 6755i Chris - Chris Owen - Garden City (620) 275-1900 - Lottery (noun): President - Wichita (316) 858-3000 -A stupidity tax Hubris Communications Inc www.hubris.net - -- _ -- 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] Phones won't stop ringing
I'm experiencing runaway ringing too, can we make this a class action against someone? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff Brower Sent: Wednesday, March 10, 2010 10:20 PM To: Chris Owen Cc: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Phones won't stop ringing Chris- Sounds like the Toyota bug has migrated to Asterisk... it's mutated into runaway ringing :-) -Jeff Sorry for my attempt at levity; just couldn't help it plus I'm sure Digium guys will know how to resolve. We're having an issue that isn't easily googleable so I thought I might might try here. We have several customers who want all their extensions to ring on incoming calls. Frankly I think it is craziness to ring 11 extensions all at once but that is how they want it. We're doing this by creating an incoming route that goes to a hunt list containing all the extensions. This normally works fine but occasionally when someone picks up the call other phones don't seem to realize the call has been answered and will continue to ring. On at least once occasion I saw a call that went to voicemail and all the phones continued to ring. When this happens the phones will continue to ring forever. The only way to stop them from ringing is to pickup the handset at which time they realize there is no call and reset. I'm pretty sure the underlying cause of this problem is funkiness in their network but it just seems to happen too easily and then once it stops it won't stop.Even if this is caused by network issues is there anything I can do to mitigate the problem. Just seems wrong that the phones would continue to ring forever. Chris -- _ -- 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 - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. -- _ -- 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] Phones won't stop ringing
On Mar 10, 2010, at 10:27 PM, Chris Owen wrote: This normally works fine but occasionally when someone picks up the call other phones don't seem to realize the call has been answered and will continue to ring. On at least once occasion I saw a call that went to voicemail and all the phones continued to ring. When this happens the phones will continue to ring forever. The only way to stop them from ringing is to pickup the handset at which time they realize there is no call and reset. What kind of phones? All Aastra 6755i I've been seeing this lately on Cisco 7940, seems to happen on two of the three at a location I deal with. They worked fine for years and then all of a sudden this just started happening. Rebooting the phone will cure it for a period of time, but it always comes back, and always to the same two phones (although not always at the same time). I don't think anything changed when it started happening, but I can't say for sure. It may also happen on a Polycom at that location as well, reports on that one have been sketchy, so I can't be sure it really is versus they are hearing a 2nd call ringing and just think the phone is stuck ringing. (I do know for a fact it happens with the Cisco and is not simply a 2nd call). I had figured it was the old version of Asterisk I'm running and the fact that the server has had several power failures so who knows the health of the machine and install. But if it is happening to others, my assumption may be wrong. -chris www.mythtech.net -- _ -- 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] Phones dropping registration, but asterisk thinks phones are still registered
On Thu, Jun 4, 2009 at 11:08 AM, James Lamannajlama...@gmail.com wrote: Hi, I have a serious problem with Asterisk 1.4.18. Every so often, usually after Asterisk has been running for a few days consistently, phones start dropping registrations. However, when this happens, doing a sip show peer on those extensions shows them as OK. Therefore, I have no way to tell this problem is happening until customers start calling. The only way to fix it is to completely restart Asterisk. Has anyone experienced this? This is a serious problem. I've poured over the logs while and after this happens and there is nothing in the logs that would suggest there is a problem. This is a production server, so I can't just upgrade Asterisk to the latest 1.4 version. I know people have suggested upgrading the server, but I'm not in a position to do that right now. However, I believe there is a symptom. When I do a sip show peer on an affected phone, the expire time is NEGATIVE. I think this might be contributing to the problem, and why Asterisk thinks the phone is still registered. Thanks. -- James ___ -- 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] Phones dropping registration, but asterisk thinks phones are still registered
Why can't you just do a daily/weekly cron to restart when convenient in off/slow hours for local time. Is your business constantly on-line 24/7? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James Lamanna Sent: Monday, June 29, 2009 3:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Phones dropping registration,but asterisk thinks phones are still registered On Thu, Jun 4, 2009 at 11:08 AM, James Lamannajlama...@gmail.com wrote: Hi, I have a serious problem with Asterisk 1.4.18. Every so often, usually after Asterisk has been running for a few days consistently, phones start dropping registrations. However, when this happens, doing a sip show peer on those extensions shows them as OK. Therefore, I have no way to tell this problem is happening until customers start calling. The only way to fix it is to completely restart Asterisk. Has anyone experienced this? This is a serious problem. I've poured over the logs while and after this happens and there is nothing in the logs that would suggest there is a problem. This is a production server, so I can't just upgrade Asterisk to the latest 1.4 version. I know people have suggested upgrading the server, but I'm not in a position to do that right now. However, I believe there is a symptom. When I do a sip show peer on an affected phone, the expire time is NEGATIVE. I think this might be contributing to the problem, and why Asterisk thinks the phone is still registered. Thanks. -- James ___ -- 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 ___ -- 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] Phones dropping registration, but asterisk thinks phones are still registered
On Mon, Jun 29, 2009 at 4:23 PM, James Lamanna jlama...@gmail.com wrote: On Thu, Jun 4, 2009 at 11:08 AM, James Lamannajlama...@gmail.com wrote: Hi, I have a serious problem with Asterisk 1.4.18. Every so often, usually after Asterisk has been running for a few days consistently, phones start dropping registrations. However, when this happens, doing a sip show peer on those extensions shows them as OK. Therefore, I have no way to tell this problem is happening until customers start calling. The only way to fix it is to completely restart Asterisk. Has anyone experienced this? Yes, I've experienced the same thing. Not sure right now what Asterisk version I'm using, prob the latest in the Ubuntu 8.04 repos. Just my 2c, fwiw. -- Christopher Stamper Email: christopherstam...@gmail.com Web: http://tinyurl.com/2ooncg gTalk: http://tinyurl.com/6e359r Skype: cdstamper ___ -- 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] Phones dropping registration, but asterisk thinks phones are still registered
Why can't you just do a daily/weekly cron to restart when convenient in off/slow hours for local time. Is your business constantly on-line 24/7? I have tried that. Unfortunately restart when convenient doesn't always seem to actually restart asterisk, presumably because there are stuck calls or something. Very annoying as well. -- James On Thu, Jun 4, 2009 at 11:08 AM, James Lamannajlama...@gmail.com wrote: Hi, I have a serious problem with Asterisk 1.4.18. Every so often, usually after Asterisk has been running for a few days consistently, phones start dropping registrations. However, when this happens, doing a sip show peer on those extensions shows them as OK. Therefore, I have no way to tell this problem is happening until customers start calling. The only way to fix it is to completely restart Asterisk. Has anyone experienced this? This is a serious problem. I've poured over the logs while and after this happens and there is nothing in the logs that would suggest there is a problem. This is a production server, so I can't just upgrade Asterisk to the latest 1.4 version. I know people have suggested upgrading the server, but I'm not in a position to do that right now. However, I believe there is a symptom. When I do a sip show peer on an affected phone, the expire time is NEGATIVE. I think this might be contributing to the problem, and why Asterisk thinks the phone is still registered. ___ -- 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] Phones dropping registration, but asterisk thinks phones are still registered
How many phones are concerned ? ___ -- 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] Phones dropping registration, but asterisk thinks phones are still registered
Oliver wrote: How many phones are concerned ? The box currently has about 380 active phone registrations. Thanks. Please CC me directly as well because I'm on digest mode. -- James ___ -- 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] Phones dropping registration, but asterisk thinks phones are still registered
Hi, I have a serious problem with Asterisk 1.4.18. Every so often, usually after Asterisk has been running for a few days consistently, phones start dropping registrations. However, when this happens, doing a sip show peer on those extensions shows them as OK. Therefore, I have no way to tell this problem is happening until customers start calling. The only way to fix it is to completely restart Asterisk. Has anyone experienced this? This is a serious problem. I've poured over the logs while and after this happens and there is nothing in the logs that would suggest there is a problem. This is a production server, so I can't just upgrade Asterisk to the latest 1.4 version. Thanks. -- James ___ -- 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] Phones dropping registration, but asterisk thinks phones are still registered
Not a real solution, but why don't you just set up a cron job to issue asterisk -rx restart when convenient once a day? This will restart asterisk on the first zero load opportunity. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James Lamanna Sent: Thursday, June 04, 2009 1:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Phones dropping registration,but asterisk thinks phones are still registered Hi, I have a serious problem with Asterisk 1.4.18. Every so often, usually after Asterisk has been running for a few days consistently, phones start dropping registrations. However, when this happens, doing a sip show peer on those extensions shows them as OK. Therefore, I have no way to tell this problem is happening until customers start calling. The only way to fix it is to completely restart Asterisk. Has anyone experienced this? This is a serious problem. I've poured over the logs while and after this happens and there is nothing in the logs that would suggest there is a problem. This is a production server, so I can't just upgrade Asterisk to the latest 1.4 version. Thanks. -- James ___ -- 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 ___ -- 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] Phones dropping registration, but asterisk thinks phones are still registered
Hi! I have a serious problem with Asterisk 1.4.18. Every so often, usually after Asterisk has been running for a few days consistently, phones start dropping registrations. However, when this happens, doing a sip show peer on those extensions shows them as OK. Please check if this is related (maybe, maybe not): SIP stops working with multiple REGISTER statements in sip.conf Registration fails if multiple peers are specified in sip.conf https://issues.asterisk.org/view.php?id=15139 Philipp ___ -- 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] Phones dropping registration, but asterisk thinks phones are still registered
There were some serious issues with some of the earlier 1.4.x Asterisk releases. You say it's a production server and can't upgrade because of that. That is the one reason why you should upgrade. There are security risks with certain versions and some serious bugs that were fixed. While I can't say that the problem with go away with an upgrade, you'll get better support if you are running a more recent version. On 06/04/2009 01:08 PM, James Lamanna wrote: Hi, I have a serious problem with Asterisk 1.4.18. Every so often, usually after Asterisk has been running for a few days consistently, phones start dropping registrations. However, when this happens, doing a sip show peer on those extensions shows them as OK. Therefore, I have no way to tell this problem is happening until customers start calling. The only way to fix it is to completely restart Asterisk. Has anyone experienced this? This is a serious problem. I've poured over the logs while and after this happens and there is nothing in the logs that would suggest there is a problem. This is a production server, so I can't just upgrade Asterisk to the latest 1.4 version. ___ -- 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] Phones lose contact
On Oct 17, 2008, at 5:14 PM, Paul Douglas Franklin wrote: When off site, our IP phones lose contact after a few minutes of inactivity. They no longer receive calls, though they can call out. Asterisk acts as if it is ringing the phone, but the phone does not ring. The phones are behind a NAT/firewall. What is the most reasonable solution? qualify=yes ___ -- 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] phones start ringing randomly with Grandstream GXW-40XX - solution!
Thought i would share this so it doesnt annoy others as much as it did me :) If you recently installed a GXW 40XX and your extensions start ringing magically now (ringing for no reason, pick it up its a clear tone) you need to check the Disable send MWI in your gateway. apparently certain old phones do not like the MWI signal and treat it like a ring tone. -- Faraz R Khan Chief Architect Emergen Consulting Pvt Ltd www.emergen.biz ___ -- 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] Phones Not Registering
Hi, I have a voip platform that has a SIP server where about 450 sipura phones adaptors register. On two occassions some phones (which were previously working) have refused to register with certain IPs but when I change the IP the phones register. The failing IP can the work after two days. A trace from the server shows that the phone is sending a registration signal to the server that the server is also sending back the same but its not getting to the phone. What could be the cause of this? Thanks, Edwin - Get easy, one-click access to your favorites. Make Yahoo! your homepage.___ --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] Phones Not Registering
The reason could be bad routing, IPs used by multiple devices.. n so on... Edwin Kariuki wrote: Hi, I have a voip platform that has a SIP server where about 450 sipura phones adaptors register. On two occassions some phones (which were previously working) have refused to register with certain IPs but when I change the IP the phones register. The failing IP can the work after two days. A trace from the server shows that the phone is sending a registration signal to the server that the server is also sending back the same but its not getting to the phone. What could be the cause of this? Thanks, Edwin Get easy, one-click access to your favorites. Make Yahoo! your homepage. http://us.rd.yahoo.com/evt=51443/*http://www.yahoo.com/r/hs ___ --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 EMAIL DISCLAIMER : This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. Any unauthorised distribution or copying is strictly prohibited. If you receive this transmission in error, please notify the sender by reply email and then destroy the message. Opinions, conclusions and other information in this message that do not relate to official business of Mascon shall be understood to be neither given nor endorsed by Mascon. Any information contained in this email, when addressed to Mascon clients is subject to the terms and conditions in governing client contract. Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we can not guarantee that any email or attachment is free from computer viruses and you are strongly advised to undertake your own anti-virus precautions. Mascon grants no warranties regarding performance, use or quality of any e-mail or attachment and undertakes no liability for loss or damage, howsoever caused. ___ --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] Phones Not Registering
On Thu, 22 Nov 2007, Edwin Kariuki wrote: Hi, I have a voip platform that has a SIP server where about 450 sipura phones adaptors register. On two occassions some phones (which were previously working) have refused to register with certain IPs but when I change the IP the phones register. The failing IP can the work after two days. A trace from the server shows that the phone is sending a registration signal to the server that the server is also sending back the same but its not getting to the phone. What could be the cause of this? What kind of voip platform? What SIP server? Out over the Internet or local lan or a VPN? Going to need a lot more information... Brett ___ --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] Phones Not Registering
Brett Crapser [EMAIL PROTECTED] wrote: On Thu, 22 Nov 2007, Edwin Kariuki wrote: Hi, I have a voip platform that has a SIP server where about 450 sipura phones adaptors register. On two occassions some phones (which were previously working) have refused to register with certain IPs but when I change the IP the phones register. The failing IP can the work after two days. A trace from the server shows that the phone is sending a registration signal to the server that the server is also sending back the same but its not getting to the phone. What could be the cause of this? What kind of voip platform? What SIP server? Out over the Internet or local lan or a VPN? Going to need a lot more information... Brett The platform run on Linux asterisk. Devices register over the internet. ___ --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 - Never miss a thing. Make Yahoo your homepage.___ --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] Phones Not Registering
On Thu, 22 Nov 2007, Edwin Kariuki wrote: Brett Crapser [EMAIL PROTECTED] wrote: On Thu, 22 Nov 2007, Edwin Kariuki wrote: Hi, I have a voip platform that has a SIP server where about 450 sipura phones adaptors register. On two occassions some phones (which were previously working) have refused to register with certain IPs but when I change the IP the phones register. The failing IP can the work after two days. A trace from the server shows that the phone is sending a registration signal to the server that the server is also sending back the same but its not getting to the phone. What could be the cause of this? What kind of voip platform? What SIP server? Out over the Internet or local lan or a VPN? Going to need a lot more information... Brett The platform run on Linux asterisk. Devices register over the internet. Some random thoughts: Too many NAT sessions for the router to track? Running out of ARP table space on the Linux box if not going via NAT? Gordon ___ --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] Phones fail to ring
I have commented out the zapateller line now. The problem persists. I wonder if there is a problem with the tones generated by some cell phones when choosing an extension. At this point the problem seems to come from cell phones only. My wife, for instance was pressing send after choosing an extension. I caught that in CLI. But one other assures me that he is not doing so. They say it sounds to them as though it is ringing. No sound here This problem is intermittent BTW. Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lee Jenkins Sent: Tuesday, May 22, 2007 9:58 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Phones fail to ring Jim Suber wrote: I am somewhat confused. I have the incoming (s) context playing a greeting and callers choose one of two extensions (100, or 101) To the caller it ALWAYS sounds as though the phone is ringing. However, sometimes it is not actually ringing the phones The listext.wav file suggests extensions 100 or 101 exten = s,1,Zapateller(nocallerid) exten = s,2,Answer() exten = s,3,Background(listext) exten = i,1,PlayBack(pbx-invalid) exten = i,2,Goto(incoming,s,1) exten = t,1,PlayBack(vm-goodbye) exten = t,2,Hangup() Try putting the Answer() first. See if that makes a difference. -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth 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] Phones fail to ring
I am somewhat confused. I have the incoming (s) context playing a greeting and callers choose one of two extensions (100, or 101) To the caller it ALWAYS sounds as though the phone is ringing. However, sometimes it is not actually ringing the phones The listext.wav file suggests extensions 100 or 101 exten = s,1,Zapateller(nocallerid) exten = s,2,Answer() exten = s,3,Background(listext) exten = i,1,PlayBack(pbx-invalid) exten = i,2,Goto(incoming,s,1) exten = t,1,PlayBack(vm-goodbye) exten = t,2,Hangup() Thanks in advance Jim ___ --Bandwidth 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] Phones fail to ring
Jim Suber wrote: I am somewhat confused. I have the incoming (s) context playing a greeting and callers choose one of two extensions (100, or 101) To the caller it ALWAYS sounds as though the phone is ringing. However, sometimes it is not actually ringing the phones The listext.wav file suggests extensions 100 or 101 exten = s,1,Zapateller(nocallerid) exten = s,2,Answer() exten = s,3,Background(listext) exten = i,1,PlayBack(pbx-invalid) exten = i,2,Goto(incoming,s,1) exten = t,1,PlayBack(vm-goodbye) exten = t,2,Hangup() Try putting the Answer() first. See if that makes a difference. -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Phones working with 1.2.17, not with 1.4.2
Hello, I've got various phones (mostly SPA-922) behind NAT registered to Asterisk. I've set nat=yes and canreinvite=no, and everything seemed to work great with 1.2.17. After upgrading to 1.4.2 using users.conf and macro-stdexten my spa-922 can't call other extensions. -- Executing [EMAIL PROTECTED]:1] Macro(SIP/22-b72006f0, stdexten|23| SIP/23) in new stack -- Executing [EMAIL PROTECTED]:1] Dial(SIP/22-b72006f0, SIP/23) in new stack -- Called 23 [Apr 18 12:29:16] NOTICE[3831]: chan_sip.c:2757 auto_congest: Auto-congesting SIP/23-081db528 -- SIP/23-081db528 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) == Auto fallthrough, channel 'SIP/22-b72006f0' status is 'CONGESTION' Debugging SIP messages seems that the called exten is not replying to invites, but it registers correctly. Other phones (Siemens C450 IP) seem to be able to call other extensions: -- Executing [EMAIL PROTECTED]:1] Macro(SIP/27-b72020e0, stdexten|22| SIP/22) in new stack -- Executing [EMAIL PROTECTED]:1] Dial(SIP/27-b72020e0, SIP/22) in new stack -- Called 22 -- SIP/22-081de4f8 is ringing Phones configuration is unaltered. What could it be? thanks Luca ___ --Bandwidth 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] phones can make outgoing calls but no incoming
Hello, What's your sip.conf and extensions.conf? Regards On 1/12/07, kevin bergner [EMAIL PROTECTED] wrote: i am having a problem where the phones are registered and can make outgoing calls but all incoming calls go directly to voicemail and do not ring any of the phones any ideas? -- kevin ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] phones can make outgoing calls but no incoming
i am having a problem where the phones are registered and can make outgoing calls but all incoming calls go directly to voicemail and do not ring any of the phones any ideas? -- kevin ___ --Bandwidth 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] Phones cutting out.....again - PLEASE HELP!!!
Hate to drag this one back up, butit's happening again. Overview of architecture: Dell poweredge 2850, running fedora core 4, asterisk 1.2.7.1, zaptel 1.2.5, and sangoma wanpipe 2.3.4 drivers. T1 interface card is the sangoma a104d with onboard echo can. Server is located in our data center and connected directly to our cisco 6513 core switch, so we have almost zero latency. The office having the issues is located several miles away and is connected via a 10Mbit fiber pipe, also low latency. Ping times between remote office and here are well under 10ms. T1's are robbed-bit, EM wink signalling --- (this may be cause, but want your input). Server load is averaging around 20%, plenty of memory, disk space, and bandwidth available. No QOS running on network. ulaw is the primary codec. Server is stable, and there are no extraneous services running, save mysql and httpd. Even running a processor intensive query doesn't trigger the droputs, they happen randomly. Phones are a mix of Eyebeam 1.5.5 and Eyebeam 1.10 3010n. Both types of phones are experiencing cutting out of the signal, mainly in the Rx stream, but occassional in the Tx stream as well. The cutting out was NOT occurring last night, and the phone server is being rebooted nightly. Nothing has changed AT ALL, and the problem has started occurring again. If I don't do ANYTHING at all today, there is a 50% chance that this will NOT occur tomorrow. In other words, SOMETHING is causing our phones to drop out, but whatever changes I make seem to have no effect. The problem will start and stop seeminly at it's own whim. --- Things I have tried: 1. changed from ulaw to gsm as primary codec - no change 2. disabled hardware echo can on A104D - no change 3. moved from asterisk 1.2.4 to 1.2.7.1, recompiled both several times - no change 4. have played with gain settings a bit, doesn't seem to make much difference --- At this point, i am nearing the end of my rope - i have rebuilt this machine three times now, and have recompiled the system at least a dozen times. We have gone from Digium hardware to Sangoma harware and back again. I have changed every conceivable setting on the phones to no avail. The problem will randomly disappear, only to come back a few days later. I can make a change, it seems to have an effect, then we're back to the same old thing again. I am in dire need of ANY help anyone can offer, this has been going on in some form for almost three months. Thanks for reading, Wes ___ --Bandwidth 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] Phones cutting out.....again - PLEASE HELP!!!
-Original Message- Thanks for reading, Wes ___ Please reply with the output of the following: lspci -vv lspci -vv | grep IRQ lspci cat /proc/interrupts Thank you. Andrew ___ --Bandwidth 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] Phones cutting out.....again - PLEASE HELP!!!
snip Server load is averaging around 20%, plenty of memory, disk space, and bandwidth available. No QOS running on network. ulaw is the primary codec. Server is stable, and there are no extraneous services running, save mysql and httpd. Even running a processor intensive query doesn't trigger the droputs, they happen randomly. snip You mention that the remote office is fiber connected, but don't identify what equipment is used to at the ends of the fiber. How many people are in the remote office, what is their work process/habits, do they also use this circuit for internet access? The key to my questions is that I suspect you do need at least a minimal QoS implimentation. A quick check on the circuit utilization when the issue occurs can confirm this, or eliminate it. Dan ___ --Bandwidth 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] Phones cutting out.....again - PLEASE HELP!!!
Here ya go: lspci -vv --- 00:00.0 Host bridge: Intel Corporation E7520 Memory Controller Hub (rev 09) Subsystem: Dell: Unknown device 016d Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr+ Stepping- SERR+ FastB2B- Status: Cap+ 66Mhz- UDF- FastB2B+ ParErr- DEVSEL=fast TAbort- TAbort- MAbort- SERR- PERR- Latency: 0 Capabilities: [40] Vendor Specific Information 00:02.0 PCI bridge: Intel Corporation E7525/E7520/E7320 PCI Express Port A (rev 09) (prog-if 00 [Normal decode]) Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr+ Stepping- SERR- FastB2B- Status: Cap+ 66Mhz- UDF- FastB2B- ParErr- DEVSEL=fast TAbort- TAbort- MAbort- SERR- PERR- Latency: 0, Cache Line Size 10 Bus: primary=00, secondary=01, subordinate=03, sec-latency=0 Memory behind bridge: dfd0-dfff Prefetchable memory behind bridge: d800-d800 Secondary status: 66Mhz- FastB2B- ParErr- DEVSEL=fast TAbort- TAbort- MAbort+ SERR+ PERR- BridgeCtl: Parity+ SERR+ NoISA+ VGA- MAbort- Reset- FastB2B- Capabilities: [50] Power Management version 2 Flags: PMEClk- DSI+ D1- D2- AuxCurrent=0mA PME(D0+,D1-,D2-,D3hot+,D3cold+) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Capabilities: [58] Message Signalled Interrupts: 64bit- Queue=0/1 Enable- Address: fee0 Data: Capabilities: [64] Express Root Port (Slot-) IRQ 0 Device: Supported: MaxPayload 256 bytes, PhantFunc 0, ExtTag- Device: Latency L0s 64ns, L1 1us Device: Errors: Correctable+ Non-Fatal+ Fatal+ Unsupported- Device: RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop- Device: MaxPayload 256 bytes, MaxReadReq 128 bytes Link: Supported Speed 2.5Gb/s, Width x8, ASPM L0s, Port 2 Link: Latency L0s 4us, L1 unlimited Link: ASPM Disabled RCB 64 bytes CommClk- ExtSynch- Link: Speed 2.5Gb/s, Width x8 Root: Correctable- Non-Fatal- Fatal- PME- 00:04.0 PCI bridge: Intel Corporation E7525/E7520 PCI Express Port B (rev 09) (prog-if 00 [Normal decode]) Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr+ Stepping- SERR- FastB2B- Status: Cap+ 66Mhz- UDF- FastB2B- ParErr- DEVSEL=fast TAbort- TAbort- MAbort- SERR- PERR- Latency: 0, Cache Line Size 10 Bus: primary=00, secondary=04, subordinate=04, sec-latency=0 Secondary status: 66Mhz- FastB2B- ParErr- DEVSEL=fast TAbort- TAbort- MAbort+ SERR- PERR- BridgeCtl: Parity+ SERR+ NoISA+ VGA- MAbort- Reset- FastB2B- Capabilities: [50] Power Management version 2 Flags: PMEClk- DSI+ D1- D2- AuxCurrent=0mA PME(D0+,D1-,D2-,D3hot+,D3cold+) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Capabilities: [58] Message Signalled Interrupts: 64bit- Queue=0/1 Enable- Address: fee0 Data: Capabilities: [64] Express Root Port (Slot-) IRQ 0 Device: Supported: MaxPayload 256 bytes, PhantFunc 0, ExtTag- Device: Latency L0s 64ns, L1 1us Device: Errors: Correctable+ Non-Fatal+ Fatal+ Unsupported- Device: RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop- Device: MaxPayload 128 bytes, MaxReadReq 128 bytes Link: Supported Speed 2.5Gb/s, Width x8, ASPM L0s, Port 4 Link: Latency L0s 4us, L1 unlimited Link: ASPM Disabled RCB 64 bytes Disabled CommClk- ExtSynch- Link: Speed 2.5Gb/s, Width x8 Root: Correctable- Non-Fatal- Fatal- PME- 00:05.0 PCI bridge: Intel Corporation E7520 PCI Express Port B1 (rev 09) (prog-if 00 [Normal decode]) Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr+ Stepping- SERR- FastB2B- Status: Cap+ 66Mhz- UDF- FastB2B- ParErr- DEVSEL=fast TAbort- TAbort- MAbort- SERR- PERR- Latency: 0, Cache Line Size 10 Bus: primary=00, secondary=05, subordinate=07, sec-latency=0 I/O behind bridge: e000-efff Memory behind bridge: dfa0-dfcf Secondary status: 66Mhz- FastB2B- ParErr- DEVSEL=fast TAbort- TAbort- MAbort+ SERR+ PERR- BridgeCtl: Parity+ SERR+ NoISA+ VGA- MAbort- Reset- FastB2B- Capabilities: [50] Power Management version 2 Flags: PMEClk- DSI+ D1- D2- AuxCurrent=0mA PME(D0+,D1-,D2-,D3hot+,D3cold+) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Capabilities: [58] Message Signalled Interrupts: 64bit- Queue=0/1 Enable- Address: fee0 Data: Capabilities: [64] Express Root Port (Slot-) IRQ 0 Device: Supported:
RE: [asterisk-users] Phones cutting out.....again - PLEASE HELP!!!
Your problem is intermittent. It is probably Network related as if you reboot that problems may or may not comeback. In addition to the lspci stuff requested. Have you checked your fiberlink. Is it possible that something or someone is saturating the link with Virus/Spy/PtP Ware??? SIP doesn't have a Jitter Buffer so it is sensitive to the traffic. You may want to try Olle's branch with the Jitter Buffer. See if that helps. I think you are fine on the Hardware side of things. It is also possible you got root-kitted. SNIP. ___ --Bandwidth 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] Phones cutting out.....again - PLEASE HELP!! !
So you need a divide and conquer strategy here: 1. Is it Asterisk or the WAN? This should be easy enough to test for. Do call dropouts happen in your datacentre? If not, your Asterisk install is good. My money's on the 10mbit WAN pipe, and that's what I would be focussing on. 2. If it's the WAN, is it a connectivity issue or a bandwidth issue? Do a continous ping from the remote location to your Asterisk server for a day. You should get NO packets dropped. If you are getting drops, it's a connectivity issue and you have to look at your SLA to see what your provider considers good. Otherwise, bandwidth issue. 3. If it's a bandwidth issue, is it your users doing things or is it a service that is eating bandwidth? If it's a service that is aggregated to a remote server, like email, then you can use bandwidth management tools like AstShape or good old tc to severely retard available bandwidth to the troublesome service. If it's your users, you have to determine what they are doing. Look at patterns: Does it happen every Tuesday afternoon when you know Bob from Accounting is running his reports? 4. Sounds like you are running Asterisk -- SIP -- 10mbit WAN -- SIP -- Phones - which probably is half the issue right there because of no jitterbuffer. Dig up an old P-3, stick in Trixbox, run it out to your remote location, and have your Eyebeam clients use *it* instead of your big Asterisk server for local connectivity. Then tie your P-3 to your big Asterisk server with IAX. Jitterbuffer + trunking = goodness and your P-3 won't choke under load if you avoid transcoding by using the same codec end-to-end. Yes it will blow having to maintain two dialplans. But IAX works frigging great. I use it to aggregate 30 remote locations over the *public* Internet to my big Asterisk server, and I never get complaints of dropouts, and in fact I use it extensively myself and IMO it sounds better* than the local CableCo's VoIP offering, which is a big POS. 5. Regardless of what it actually is, I would have some sort of traffic shaper at both ends of the WAN pipe. Again, dig up a couple of old P-2 or P-3's and stick in a bootable Monowall CD, change the default rules to allow all traffic through, but create a traffic shaping ruleset to give priority and bandwidth to 5060, 4569, 1-2 and dump everything else to a low priority queue. 6. I'd run GSM anyway (even though you tried it) because it would eliminate half your bandwidth consumption. Another variable eliminated. hth *By 'sounds better' I mean it sounds like a perfectly normal PSTN call, ALL THE TIME in s d of co s an ly s nd ng li e t hs -Original Message- From: whois wes [mailto:[EMAIL PROTECTED] Sent: Thursday, July 06, 2006 10:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Phones cutting out.again - PLEASE HELP!!! Hate to drag this one back up, butit's happening again. Overview of architecture: Dell poweredge 2850, running fedora core 4, asterisk 1.2.7.1, zaptel 1.2.5, and sangoma wanpipe 2.3.4 drivers. T1 interface card is the sangoma a104d with onboard echo can. Server is located in our data center and connected directly to our cisco 6513 core switch, so we have almost zero latency. The office having the issues is located several miles away and is connected via a 10Mbit fiber pipe, also low latency. Ping times between remote office and here are well under 10ms. T1's are robbed-bit, EM wink signalling --- (this may be cause, but want your input). Server load is averaging around 20%, plenty of memory, disk space, and bandwidth available. No QOS running on network. ulaw is the primary codec. Server is stable, and there are no extraneous services running, save mysql and httpd. Even running a processor intensive query doesn't trigger the droputs, they happen randomly. Phones are a mix of Eyebeam 1.5.5 and Eyebeam 1.10 3010n. Both types of phones are experiencing cutting out of the signal, mainly in the Rx stream, but occassional in the Tx stream as well. The cutting out was NOT occurring last night, and the phone server is being rebooted nightly. Nothing has changed AT ALL, and the problem has started occurring again. If I don't do ANYTHING at all today, there is a 50% chance that this will NOT occur tomorrow. In other words, SOMETHING is causing our phones to drop out, but whatever changes I make seem to have no effect. The problem will start and stop seeminly at it's own whim. --- Things I have tried: 1. changed from ulaw to gsm as primary codec - no change 2. disabled hardware echo can on A104D - no change 3. moved from asterisk 1.2.4 to 1.2.7.1, recompiled both several times - no change 4. have played with gain settings a bit, doesn't seem to make much difference --- At this point, i am nearing the end of my rope - i have rebuilt this machine three times now, and have recompiled the system at least a dozen times. We have gone from Digium hardware to Sangoma harware
Re: [asterisk-users] Phones cutting out.....again - PLEASE HELP!! !
Thanks for the quick responses everyone. To answer some of the questions posed: The main traffic going over this pipe is voice, with a small amount of web traffic as well. There are 60 total users, 5 of which access anything other than what is on their LAN up there. In any case, we are not saturating the pipe, and our telco put some sort of filters on the Optiman switches on each side to eliminate any jitter (or so they say). Prior to the filter being installed, we had our main application server for that location located down here - when the issue started (out of the blue, nothing really triggered it, and our bandwidth didn't change or spike) we moved that server to the remote location. So, before we even had the issue, we were using WAY more bandwidth, almost 8Mbit at times...we're averaging around 2-3 now, and it rarely spikes above that. Also, when I connect to the server locally (the server is in the room next to me, in other words, and i have 1 Gbit of bandwidth all the way to the back of the server, I still get call dropouts. In other words, completely bypassing the fiber pipe results in the same problem. For that reason alone, I don't think it's the WAN (although I agree with what all of you said in regards to QOS, etc, it's just not up to me to implement that, even though it's been suggested numerous times). However, this IS the only server (of 8 total, all in the same rack and connected to the telco via the same DS3) that is having the issue, which DOES point to it being the WAN, as that is our ONLY remote location. See why I'm frustrated? I do like the idea of putting a local box up there and using an IAX trunk over the pipe, and will see about getting that implemented. GSM was already shot down as 'too low-quality' - we'd rather up the pipe to 20Mbit than go with a lower quality codec. Sorry that I forgot to mention some of this in my initial post, and hopefully the above info will shed a bit more light on my confusion. Thank you all again for replying so quickly, and if you have any other suggestions, please let me know. Wes On 7/6/06, Colin Anderson [EMAIL PROTECTED] wrote: So you need a divide and conquer strategy here: 1. Is it Asterisk or the WAN? This should be easy enough to test for. Do call dropouts happen in your datacentre? If not, your Asterisk install is good. My money's on the 10mbit WAN pipe, and that's what I would be focussing on. 2. If it's the WAN, is it a connectivity issue or a bandwidth issue? Do a continous ping from the remote location to your Asterisk server for a day. You should get NO packets dropped. If you are getting drops, it's a connectivity issue and you have to look at your SLA to see what your provider considers good. Otherwise, bandwidth issue. 3. If it's a bandwidth issue, is it your users doing things or is it a service that is eating bandwidth? If it's a service that is aggregated to a remote server, like email, then you can use bandwidth management tools like AstShape or good old tc to severely retard available bandwidth to the troublesome service. If it's your users, you have to determine what they are doing. Look at patterns: Does it happen every Tuesday afternoon when you know Bob from Accounting is running his reports? 4. Sounds like you are running Asterisk -- SIP -- 10mbit WAN -- SIP -- Phones - which probably is half the issue right there because of no jitterbuffer. Dig up an old P-3, stick in Trixbox, run it out to your remote location, and have your Eyebeam clients use *it* instead of your big Asterisk server for local connectivity. Then tie your P-3 to your big Asterisk server with IAX. Jitterbuffer + trunking = goodness and your P-3 won't choke under load if you avoid transcoding by using the same codec end-to-end. Yes it will blow having to maintain two dialplans. But IAX works frigging great. I use it to aggregate 30 remote locations over the *public* Internet to my big Asterisk server, and I never get complaints of dropouts, and in fact I use it extensively myself and IMO it sounds better* than the local CableCo's VoIP offering, which is a big POS. 5. Regardless of what it actually is, I would have some sort of traffic shaper at both ends of the WAN pipe. Again, dig up a couple of old P-2 or P-3's and stick in a bootable Monowall CD, change the default rules to allow all traffic through, but create a traffic shaping ruleset to give priority and bandwidth to 5060, 4569, 1-2 and dump everything else to a low priority queue. 6. I'd run GSM anyway (even though you tried it) because it would eliminate half your bandwidth consumption. Another variable eliminated. hth *By 'sounds better' I mean it sounds like a perfectly normal PSTN call, ALL THE TIME in s d of co s an ly s nd ng li e t hs -Original Message- From: whois wes [mailto:[EMAIL PROTECTED] Sent: Thursday, July 06, 2006 10:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Phones cutting out
RE: [asterisk-users] Phones cutting out.....again - PLEASE HELP!! !
Also, when I connect to the server locally (the server is in the room next to me, in other words, and i have 1 Gbit of bandwidth all the way to the back of the server, I still get call dropouts. However, this IS the only server (of 8 total, all in the same rack and connected to the telco via the same DS3) that is having the issue, which DOES point to it being the WAN, as that is our ONLY remote location. So perhaps what you are seeing is two or more subtle issues with the same symptom, so subjectively it looks like the *same* issue. 1. Definitely try the remote IAX box to rule out bandwidth starvation. 2. Definitely try the ping test to rule out connectivity. 3. You have to figure out what the problem is with your big Asterisk box. There should be no reason why you are getting dropouts on the local LAN. What is the output of zttest? Is it good? Does zttool indicate IRQ misses? If it's OK, then your hardware - T1 setup is good, so you have ruled out your Asterisk box. It is also a worthwhile excercise to rule out the onboard ethernet card in the Dell. In fact, whenever I do a new box, I automatically disable the onboard LAN and replace it with an add-in 3com or Intel. It is also a worthwhile excercise to user setpci to change the latency of the cards in the Dell so that your Zap boards can grab the bus as much as possible. 4. The thing that is common in all scenarios is the EyeBeam client itself. Any soft phone is subject to the strengths and weaknesses of the audio chipset in the PC, with issues to consider like latency, audio threshold before it starts the TX, and duplex settings. Because troubleshooting these variables is often as hard as troubleshooting an entire Asterisk install, I would never run a soft-phone and expect people to use it productively. What happens when you put in a real phone? If you don't have a hardphone, maybe try something else like the Snom soft-phone. In the end, this is all about eliminating variables as much as possible, and this will determine your decision matrix of things to try. The first matrix will be the most difficult to implement because you have a whole wack of stuff to eliminate, but they will get smaller and smaller as you eliminate variables and eventually you will only have 2 or 3 variables to test for, and then you are golden. OT: I find it useful to make painstaking notes or keep a spreadsheet of test results when going through a troubleshooting process like this. Often, referring back to the spreadsheet gives me valuable insight into a problem. I read this book, and I got shivers down my spine because it's like these guys got into my brain and stole (what I thought) was an original problem-solving idea of mine: http://www.transcendstrategy.com/html/index.php?module=htmlpagesfunc=displa ypid=7 Every person that troubleshoots a complex system should read this book (disclaimer: I just read it, I have nothing to do with these guys) good luck ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Phones cutting out.....again - PLEASE HELP!! !
Colin, Very good points, and you are right, I need to start tracking what has been done. A bit of history - this server was very unstable when running Digium hardware - every day or two, it would kernel panic and lock up, requiring a manual reboot. The other servers had issues as well, and ALL of the stability problems were solved when we moved to Sangoma cards about 6 weeks ago...this problem started a few weeks after that migration. On point #3, you mention a few things - we have NEVER gotten zttest to show 100% on ANY of our boxes, which is one reason we migrated from Digium to Sangoma. For a while, we did try running a third party NIC in the box to help with the stability issues. Once we moved to Sangoma, we went back to the onboard, and when we started having audio issues, we did try putting the third party NIC back in, to no avail. In regards to eyebeam, the rest of our company is using it as well. We're a call center, and our reps are actually more productive with the softphone than with the hardphones (we've tried both). The sister division of our remote location is set up almost identically in terms of dialplan, T1 config, desktop software package, softphone, etc, and they have ZERO issues. The two divisions are literally mirrors of one another, and the only difference between the two is that one office is remote. We also have hardphones in use by the managers at the remote location, and they also experience the issue. I do see what you're saying, though, about it possibly being two smaller issues...I hope it's not, though - that much harder to pin down. Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Phones cutting out.....again - PLEASE HELP!!!
Do you have tetheral network analyser installed on server, that can be a good start, look at the analyses of the graphs. Also try pinging the CPE's and see if there is any latency. Do you also have the abilty to check the upstreams signals? -- Original message -- From: "whois wes" [EMAIL PROTECTED] Hate to drag this one back up, butit's happening again. Overview of architecture: Dell poweredge 2850, running fedora core 4, asterisk 1.2.7.1, zaptel 1.2.5, and sangoma wanpipe 2.3.4 drivers. T1 interface card is the sangoma a104d with onboard echo can. Server is located in our data center and connected directly to our cisco 6513 core switch, so we have almost zero latency. The office having the issues is located several miles away and is connected via a 10Mbit fiber pipe, also low latency. Ping times between remote office and here are well under 10ms. T1's are robbed-bit, EM wink signalling --- (this may be cause, but want your input). & gt; Server load is averaging around 20%, plenty of memory, disk space, and bandwidth available. No QOS running on network. ulaw is the primary codec. Server is stable, and there are no extraneous services running, save mysql and httpd. Even running a processor intensive query doesn't trigger the droputs, they happen randomly. Phones are a mix of Eyebeam 1.5.5 and Eyebeam 1.10 3010n. Both types of phones are experiencing cutting out of the signal, mainly in the Rx stream, but occassional in the Tx stream as well. The cutting out was NOT occurring last night, and the phone server is being rebooted nightly. Nothing has changed AT ALL, and the problem has started occurring again. If I don't do ANYTHING at all today, there is a 50% chance that this will NOT occur tomorrow. In other words, SOMETHING is causing our phones to drop out, but whatever changes I m ake se em to have no effect. The problem will start and stop seeminly at it's own whim. --- Things I have tried: 1. changed from ulaw to gsm as primary codec - no change 2. disabled hardware echo can on A104D - no change 3. moved from asterisk 1.2.4 to 1.2.7.1, recompiled both several times - no change 4. have played with gain settings a bit, doesn't seem to make much difference --- At this point, i am nearing the end of my rope - i have rebuilt this machine three times now, and have recompiled the system at least a dozen times. We have gone from Digium hardware to Sangoma harware and back again. I have changed every conceivable setting on the phones to no avail. The problem will randomly disappear, only to come back a few days later. I can make a change, it seems to have an effect, then we're back to the same o ld thing again. I am in dire need of ANY help anyone can offer, this has been going on in some form for almost three months. Thanks for reading, Wes ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Phones cutting out.....again - PLEASE HELP!!!
Did you try set autofallthrough=no. We have the same problem when using 1.2.9.1 (we are using A104d with IBM x306). So we downgraded to 1.2.6 and set autofallthrough=no. The call drop problem seems fixing. But we have IVR DTMF recognition and queue not assign call to static agents (Local channel) problem if I don't reboot the server more than 2 days. Now I am trying downgrade A104d F/W from v20 to v18 (don't know what is new in v20 and v19) and Asterisk 1.2.6 to 1.2.4. BTW, we are using a lot of group pickup, call transfer in queue, not sure if it is cause or not. Isaac Xiao ___ --Bandwidth 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] Phones that work well through NAT
On Tuesday 18 April 2006 21:06, Sean Garland wrote: So I have * box shorewall/linux NAT firewall internet - WRT54G with openwrt - IP500 I have 5060, 4569, and 1 through 2 forwarded to * box from internet. I have tried everything I can think of on the wrt to get it to work but it appears, looking at tcpdump that my phone is trying to get to the * box (I can get one way audio with port mapping in the WRT) using the 192.168.x.x address it has as its internal interface... Is there a way to force the IP500 to use the public IP of the * box for RTP? Then it should work... I think you need to do that on the Asterisk side, in sip.conf, setting externip or externhost and externrefresh may be what you want. Don't forget about localnet if you have SIP phones on the LAN too. (info taken from the sample sip.conf in asterisk) -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Phones that work well through NAT
So how do you get a Polycom phone to work with * over NAT? I can't seem to get it to work. If I forward ports, I can get one-way audio, but that’s it. Looking at a packet capture, it appears that my phone is trying to send data to the internal address of the * server, which is of course, not available from the private side of the NAT lan... I have a polycom soundpoint IP 500. Thanks Sean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Sunday, April 16, 2006 1:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Phones that work well through NAT I'm really not interested to look back, but IIRC, when using just one Polycom phone behind NAT we didn't have any problems, but when using more than one behind the same NAT that is when problems started, qualify=somethingbutno seemed to help it a bit, but didn't eliminate the problem. On 4/16/06, Andrew Kohlsmith [EMAIL PROTECTED] wrote: On Saturday 15 April 2006 22:37, C F wrote: That is until you run into problems, while they do work, I wouldn't say that Polycoms work EXEPTIONALLY well, Cisco, and SPA work *MUCH* better. Can you detail some problems? Just about any off-the-shelf router seems to work with these. There may be some cheap-ass broken routers you can get for $5 which will not work, but all of the brand-name stuff I've tried Just Works, which is why I say they work exceptionally well. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.385 / Virus Database: 268.4.1/313 - Release Date: 4/15/2006 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.385 / Virus Database: 268.4.3/317 - Release Date: 4/18/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Phones that work well through NAT
On Tuesday 18 April 2006 09:57, Sean Garland wrote: So how do you get a Polycom phone to work with * over NAT? I can't seem to get it to work. If I forward ports, I can get one-way audio, but that’s it. Looking at a packet capture, it appears that my phone is trying to send data to the internal address of the * server, which is of course, not available from the private side of the NAT lan... I have a polycom soundpoint IP 500. You don't do anything to get it to work through NAT. If your * box is behind NAT you need to screw around a little, but for situations like this: * box --- [internet] --- [nat dsl router] --- IP501 all you do is set 'nat=yes' on the * box, in the IP501's peer setting. That's it. It even works with multiple IP501s behind the same NAT DSL router. If you have a stupid NAT box that closes ports off too quickly or plays too many games with the packets you may need some additional configuration (shorter registration expirations, etc.) but just buy a decent NAT box... WRT54Gs work just fine in their default configuration, for example. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Phones that work well through NAT
So I have * box shorewall/linux NAT firewall internet - WRT54G with openwrt - IP500 I have 5060, 4569, and 1 through 2 forwarded to * box from internet. I have tried everything I can think of on the wrt to get it to work but it appears, looking at tcpdump that my phone is trying to get to the * box (I can get one way audio with port mapping in the WRT) using the 192.168.x.x address it has as its internal interface... Is there a way to force the IP500 to use the public IP of the * box for RTP? Then it should work... Thanks Sean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: Tuesday, April 18, 2006 7:31 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Phones that work well through NAT On Tuesday 18 April 2006 09:57, Sean Garland wrote: So how do you get a Polycom phone to work with * over NAT? I can't seem to get it to work. If I forward ports, I can get one-way audio, but that’s it. Looking at a packet capture, it appears that my phone is trying to send data to the internal address of the * server, which is of course, not available from the private side of the NAT lan... I have a polycom soundpoint IP 500. You don't do anything to get it to work through NAT. If your * box is behind NAT you need to screw around a little, but for situations like this: * box --- [internet] --- [nat dsl router] --- IP501 all you do is set 'nat=yes' on the * box, in the IP501's peer setting. That's it. It even works with multiple IP501s behind the same NAT DSL router. If you have a stupid NAT box that closes ports off too quickly or plays too many games with the packets you may need some additional configuration (shorter registration expirations, etc.) but just buy a decent NAT box... WRT54Gs work just fine in their default configuration, for example. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.385 / Virus Database: 268.4.3/317 - Release Date: 4/18/2006 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.385 / Virus Database: 268.4.3/317 - Release Date: 4/18/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Phones that work well through NAT
jennyw wrote: We've been reasonably happy with Polycom SoundPoint phones, but we only have them installed on the LAN. I've read that they have problems working across NAT. So ... I guess I have a few questions. First, is there a way to get Polycoms to work well over NAT? If not, then are there phones of comparable voice quality that do work well over NAT? Without costing a lot more? It's not NAT that's the problem, it's the implementation of NAT that is the variable and causes the problems. I send Polycoms to our remote users and usually have no problems behind NAT, but one user had so much trouble we had to move the Polycom outside the firewall and use the passt hrough to connect the firewall. If you can proscribe a M0n0wall firewall box you can handle any NAT problems but if you are stuck using some funky $50 firewall/router, chances are you will have problems. -- Chris Mason NetConcepts -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Phones that work well through NAT
On Saturday 15 April 2006 22:37, C F wrote: That is until you run into problems, while they do work, I wouldn't say that Polycoms work EXEPTIONALLY well, Cisco, and SPA work *MUCH* better. Can you detail some problems? Just about any off-the-shelf router seems to work with these. There may be some cheap-ass broken routers you can get for $5 which will not work, but all of the brand-name stuff I've tried Just Works, which is why I say they work exceptionally well. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Phones that work well through NAT
What firewall was the problem user running? We have Polycoms behind Linux, Mikrotik, Linksys, Dlink, Netgear, etc all without any problems. Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Mason (Lists) Sent: Sunday, April 16, 2006 9:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Phones that work well through NAT jennyw wrote: We've been reasonably happy with Polycom SoundPoint phones, but we only have them installed on the LAN. I've read that they have problems working across NAT. So ... I guess I have a few questions. First, is there a way to get Polycoms to work well over NAT? If not, then are there phones of comparable voice quality that do work well over NAT? Without costing a lot more? It's not NAT that's the problem, it's the implementation of NAT that is the variable and causes the problems. I send Polycoms to our remote users and usually have no problems behind NAT, but one user had so much trouble we had to move the Polycom outside the firewall and use the passt hrough to connect the firewall. If you can proscribe a M0n0wall firewall box you can handle any NAT problems but if you are stuck using some funky $50 firewall/router, chances are you will have problems. -- Chris Mason NetConcepts -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Phones that work well through NAT
I'm really not interested to look back, but IIRC, when using just one Polycom phone behind NAT we didn't have any problems, but when using more than one behind the same NAT that is when problems started, qualify=somethingbutno seemed to help it a bit, but didn't eliminate the problem. On 4/16/06, Andrew Kohlsmith [EMAIL PROTECTED] wrote: On Saturday 15 April 2006 22:37, C F wrote: That is until you run into problems, while they do work, I wouldn't say that Polycoms work EXEPTIONALLY well, Cisco, and SPA work *MUCH* better. Can you detail some problems? Just about any off-the-shelf router seems to work with these. There may be some cheap-ass broken routers you can get for $5 which will not work, but all of the brand-name stuff I've tried Just Works, which is why I say they work exceptionally well. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Phones that work well through NAT
There are two approaches to get NAT working properly: - Use UDP and send and receive from the same port. This is extremly simple, however some phones do (by default) send and recieve from a different ports. Then you have to tell explicity no no, dont do that; use the same port. There are even phones that send and receive from different RTP ports. I would say they are extremly NAT unfriendly. And I don't know why a phone vendor would do that. Anyway, the IETF specs allow it. The problem with the UDP approach is the high keep-alive traffic (every 15-20 secs you must refresh it) and the number of buggy NAT implementations out there. I would say this approach works with 95 % of the equipment. - Use TCP/TLS and keep the TCP connection to the PBX open all the time. This reduces and amound of keep-alive traffic and works with almost anything on the market. Because a router that does not support https or MS Exchange traffic will have a real hard time in the market place! TLS has the advantage that smart routers cannot see the SIP traffic any more and mess around with it. For example, there is a vendor out there that does not understand the rport parameter in the Via and removes it (but leave the ; standing there)!!! Especially when there are relatively few user agents registered to the system (number of file descriptors), this approach is superior. AFAIK the next * version will support this approach; there are already systems available that support TCP and TLS. Just my two cents. Christian -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: Sunday, April 16, 2006 11:16 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Phones that work well through NAT On Saturday 15 April 2006 22:37, C F wrote: That is until you run into problems, while they do work, I wouldn't say that Polycoms work EXEPTIONALLY well, Cisco, and SPA work *MUCH* better. Can you detail some problems? Just about any off-the-shelf router seems to work with these. There may be some cheap-ass broken routers you can get for $5 which will not work, but all of the brand-name stuff I've tried Just Works, which is why I say they work exceptionally well. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Phones that work well through NAT
Hi, everyone, We've been reasonably happy with Polycom SoundPoint phones, but we only have them installed on the LAN. I've read that they have problems working across NAT. So ... I guess I have a few questions. First, is there a way to get Polycoms to work well over NAT? If not, then are there phones of comparable voice quality that do work well over NAT? Without costing a lot more? Thanks! Jen ___ --Bandwidth 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] Phones that work well through NAT
On Saturday 15 April 2006 21:12, jennyw wrote: We've been reasonably happy with Polycom SoundPoint phones, but we only have them installed on the LAN. I've read that they have problems working across NAT. So ... I guess I have a few questions. First, is there a way to get Polycoms to work well over NAT? If not, then are there phones of comparable voice quality that do work well over NAT? Without costing a lot more? Polycoms (the IP501s at any rate) work EXCEPTIONALLY well through NAT. It's as literally dead-simple as plug-and-go. No configuration on the phone, and all you want is a nat=yes in their sip.conf entry. That's it. Seriously. Olle's Symmetric RTP code is what makes it work so well. I have two IP501s behind a factory-default WRT54G on regular consumer ADSL hitting an Asterisk box on a real IP. The WRT54 has no configuration to reflect port-forwards and the only thing Asterisk has is nat=yes for those two extensions. It Just Works. And I'm still stunned by it. :-) Olle... Thank you once again for the symmetric RTP code in Asterisk. It's a godsend. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Phones that work well through NAT
On 4/15/06, Andrew Kohlsmith [EMAIL PROTECTED] wrote: On Saturday 15 April 2006 21:12, jennyw wrote: We've been reasonably happy with Polycom SoundPoint phones, but we only have them installed on the LAN. I've read that they have problems working across NAT. So ... I guess I have a few questions. First, is there a way to get Polycoms to work well over NAT? If not, then are there phones of comparable voice quality that do work well over NAT? Without costing a lot more? Polycoms (the IP501s at any rate) work EXCEPTIONALLY well through NAT. It's as literally dead-simple as plug-and-go. No configuration on the phone, and all you want is a nat=yes in their sip.conf entry. That's it. Seriously. That is until you run into problems, while they do work, I wouldn't say that Polycoms work EXEPTIONALLY well, Cisco, and SPA work *MUCH* better. Olle's Symmetric RTP code is what makes it work so well. I have two IP501s behind a factory-default WRT54G on regular consumer ADSL hitting an Asterisk box on a real IP. The WRT54 has no configuration to reflect port-forwards and the only thing Asterisk has is nat=yes for those two extensions. It Just Works. And I'm still stunned by it. :-) Olle... Thank you once again for the symmetric RTP code in Asterisk. It's a godsend. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Phones that work well through NAT
Polycoms (the IP501s at any rate) work EXCEPTIONALLY well through NAT. It's as literally dead-simple as plug-and-go. No configuration on the phone, and all you want is a nat=yes in their sip.conf entry. That's it. Seriously. In addition to nat=yes, I recommend adding qualify=yes for all phones behind NAT. Otherwise the router doing NAT may flush out the port mappings relative to your phone. The qualify essentially sends a keep-alive. We have Polycom IP500s and 501s and this works very well for them (one sitting right here on my desk). -Ron ___ --Bandwidth 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] Phones are all auto answering
Kind of like DND, but some phones seem ok. They all give the message even if it rings through that the person is on the phone even if they are not. Normally it says that when they are on the phone, and it says unavailable if they are not on the phone but never answer... Almost like astericks thinks all the phones are busy, at least by the recording it gives. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Tuesday, April 04, 2006 10:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Phones are all auto answering What phones you using? On 4/4/06, Christian Buchter [EMAIL PROTECTED] wrote: Strange, but all the phones when called immediately return a user is on the phone and the phone never rings. Anyone else ever experience this before? TIA _ This email has been scanned by MessageLabs on behalf of E-INS ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ This email has been scanned by MessageLabs on behalf of E-INS _ This email has been scanned by MessageLabs on behalf of E-INS ___ --Bandwidth 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] Phones are all auto answering
On Tue, 2006-04-04 at 10:44 -0400, Christian Buchter wrote: Strange, but all the phones when called immediately return a user is on the phone and the phone never rings. Anyone else ever experience this before? TIA Have the users managed to set DND on the phones? That would give the exact symptom. Rgds Pete ___ --Bandwidth 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] Phones are all auto answering
Snom 190s and 220s, it seems to happen intermittently but not sure why -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Tuesday, April 04, 2006 10:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Phones are all auto answering What phones you using? On 4/4/06, Christian Buchter [EMAIL PROTECTED] wrote: Strange, but all the phones when called immediately return a user is on the phone and the phone never rings. Anyone else ever experience this before? TIA _ This email has been scanned by MessageLabs on behalf of E-INS ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ This email has been scanned by MessageLabs on behalf of E-INS _ This email has been scanned by MessageLabs on behalf of E-INS ___ --Bandwidth 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] Phones are all auto answering
What phones you using? On 4/4/06, Christian Buchter [EMAIL PROTECTED] wrote: Strange, but all the phones when called immediately return a user is on the phone and the phone never rings. Anyone else ever experience this before? TIA _ This email has been scanned by MessageLabs on behalf of E-INS ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Phones are all auto answering
Strange, but all the phones when called immediately return a user is on the phone and the phone never rings. Anyone else ever experience this before? TIA _ This email has been scanned by MessageLabs on behalf of E-INS ___ --Bandwidth 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] Phones were working fine - Now there is noaudiowhen calling between extensions
C F wrote: Polycoms are not the best if you want a phone that works behind NAT. Do you mean in general? Or only if you are trying to interconnect multiple offices? Are Polycoms fine for just one office, if the entire office is behind a NAT device, and the phones are only being used for normal calling? Thanks, -- Charles ___ --Bandwidth 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] Phones were working fine - Now there is noaudiowhen calling between extensions
Are you using which version of Asterisk?? Did you check if you are facing the old audio bug on bridge calls that appeared ? http://asteriskvoip.blogspot.com/2006_01_01_asteriskvoip_archive.html Wednesday, January 25, 2006 Update: No audio - Update your Asterisk This morning we discovered a serious bug that stopped all bridged audio in our Asterisk servers. Mark found the problem and soon fixed it. If you get this problem today, please update your Asterisk server. A fix has been commited to the subversion repository for 1.2 as well as trunk. A fixed 1.2.3 release will be published on ftp.digium.com as soon as we can find a release engineer (consider the time zone problem). A big thank you to everyone in the IRC channel that helped us locate this issue and to Mark that fixed it so quickly. --- I hope it helps. Best regards, Marco Mouta On 3/22/06, Charles Marcus [EMAIL PROTECTED] wrote: C F wrote: Polycoms are not the best if you want a phone that works behind NAT. Do you mean in general? Or only if you are trying to interconnect multiple offices? Are Polycoms fine for just one office, if the entire office is behind a NAT device, and the phones are only being used for normal calling? Thanks, -- Charles ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Phones were working fine - Now there isnoaudiowhen calling between extensions
The reason I say they are not the best is because they have no ability to connect to a STUN server and they have no keep-alive mechanism for NAT. At VON, I was talking to a company called Ranch Networks who said they have a device that will solve NAT traversal issues with Asterisk and its phones. Here is a link to their product: http://www.ranchnetworks.com/pdfs/RN40_41_brief.pdf I am talking to them now to see exactly what this thing does. - Gabe - Original Message - From: Charles Marcus [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, March 22, 2006 4:43 AM Subject: Re: [Asterisk-Users] Phones were working fine - Now there isnoaudiowhen calling between extensions C F wrote: Polycoms are not the best if you want a phone that works behind NAT. Do you mean in general? Or only if you are trying to interconnect multiple offices? Are Polycoms fine for just one office, if the entire office is behind a NAT device, and the phones are only being used for normal calling? Thanks, -- Charles ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Phones were working fine - Now there is no audiowhen calling between extensions
On Mar 20, 2006, at 4:39 PM, Gabriel Afana wrote: I just did a little RTP debug and this is what it shows: == Spawn extension (304, 301, 1) exited non-zero on 'SIP/304-d9f8' -- Accepting AUTHENTICATED call from 216.152.244.81: requested format = ulaw, requested prefs = (), actual format = ulaw, host prefs = (), priority = mine -- Executing Dial(IAX2/to_80-1, SIP/301) in new stack -- Called 301 -- SIP/301-1fec is ringing -- SIP/301-1fec answered IAX2/to_80-1 Got RTP packet from 24.50.66.128:2228 (type 0, seq 16810, ts 344311448, len 160) Got RTP packet from 24.50.66.128:2228 (type 0, seq 16811, ts 344311608, len 160) Got RTP packet from 24.50.66.128:2228 (type 0, seq 16812, ts 344311768, len 160) Got RTP packet from 24.50.66.128:2228 (type 0, seq 16813, ts 344311928, len 160) Got RTP packet from 24.50.66.128:2228 (type 0, seq 16814, ts 344312088, len 160) Got RTP packet from 24.50.66.128:2228 (type 0, seq 16815, ts 344312248, len 160) Got RTP packet from 24.50.66.128:2228 (type 0, seq 16816, ts 344312408, len 160) Got RTP packet from 24.50.66.128:2228 (type 0, seq 16817, ts 344312568, len 160) Got RTP packet from 24.50.66.128:2228 (type 0, seq 16818, ts 344312728, len 160) Got RTP packet from 24.50.66.128:2228 (type 0, seq 16819, ts 344312888, len 160) Got RTP packet from 24.50.66.128:2228 (type 0, seq 16820, ts 344313048, len 160) Got RTP packet from 24.50.66.128:2228 (type 0, seq 16821, ts 344313208, len 160) Got RTP packet from 24.50.66.128:2228 (type 0, seq 16822, ts 344313368, len 160) .. that goes on for ever while the call is in progress. This is a call between phones that go between two * servers. If I make a call between phones both registered to the same asterisk server, this is my RTP stream: -- Executing Dial(SIP/304-c211, SIP/301|30|r) in new stack -- Called 301 -- SIP/301-b2c8 is ringing -- SIP/301-b2c8 answered SIP/304-c211 -- Attempting native bridge of SIP/304-c211 and SIP/301-b2c8 Got RTP packet from 24.50.66.128:2234 (type 0, seq 39729, ts -1972065425, len 160) Sent RTP packet to 24.50.66.128:2232 (type 0, seq 24819, ts 0, len 160) Got RTP packet from 24.50.66.128:2234 (type 0, seq 39730, ts -1972065265, len 160) Sent RTP packet to 24.50.66.128:2232 (type 0, seq 24820, ts 160, len 160) Got RTP packet from 24.50.66.128:2234 (type 0, seq 39731, ts -1972065105, len 160) Sent RTP packet to 24.50.66.128:2232 (type 0, seq 24821, ts 320, len 160) Got RTP packet from 24.50.66.128:2234 (type 0, seq 39732, ts -1972064945, len 160) Sent RTP packet to 24.50.66.128:2232 (type 0, seq 24822, ts 480, len 160) Got RTP packet from 24.50.66.128:2232 (type 0, seq 46694, ts 1105329892, len 160) Sent RTP packet to 24.50.66.128:2234 (type 0, seq 42690, ts 0, len 160) Got RTP packet from 24.50.66.128:2232 (type 0, seq 46695, ts 1105330052, len 160) Sent RTP packet to 24.50.66.128:2234 (type 0, seq 42691, ts 160, len 160) Got RTP packet from 24.50.66.128:2232 (type 0, seq 46696, ts 1105330212, len 160) Sent RTP packet to 24.50.66.128:2234 (type 0, seq 42692, ts 320, len 160) Got RTP packet from 24.50.66.128:2232 (type 0, seq 46697, ts 1105330372, len 160) Sent RTP packet to 24.50.66.128:2234 (type 0, seq 42693, ts 480, len 160) Got RTP packet from 24.50.66.128:2232 (type 0, seq 46698, ts 1105330532, len 160) Sent RTP packet to 24.50.66.128:2234 (type 0, seq 42694, ts 640, len 160) [THE END] Once I anser the call, the RTP string starts and then stops right where I put [THE END]. Did you try setting reinvite to no? Seems the native bridge is what's failing. Rethink your routing with regards native bridging (ie everybody is able to get through there nats and be identified? I don't really know, I am only trying to be helpful. Hope it's worth something. Marty ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Phones were working fine - Now there is noaudiowhen calling between extensions
Thanks for the response. Yes, canreinvite is set to no on all lines. After some testing, I was able to get sound between phones when they were both registered to the same server. Maybe the IAX trunk is messing something up. strange because it was working perfect last week and nothing changed! - Gabe Did you try setting reinvite to no? Seems the native bridge is what's failing. Rethink your routing with regards native bridging (ie everybody is able to get through there nats and be identified? I don't really know, I am only trying to be helpful. Hope it's worth something. Marty ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth 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] Phones were working fine - Now there is noaudiowhen calling between extensions
Polycoms are not the best if you want a phone that works behind NAT. On 3/21/06, Gabriel Afana [EMAIL PROTECTED] wrote: Thanks for the response. Yes, canreinvite is set to no on all lines. After some testing, I was able to get sound between phones when they were both registered to the same server. Maybe the IAX trunk is messing something up. strange because it was working perfect last week and nothing changed! - Gabe Did you try setting reinvite to no? Seems the native bridge is what's failing. Rethink your routing with regards native bridging (ie everybody is able to get through there nats and be identified? I don't really know, I am only trying to be helpful. Hope it's worth something. Marty ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Phones were working fine - Now there is noaudiowhen calling between extensions
On Tuesday 21 March 2006 10:55, C F wrote: Polycoms are not the best if you want a phone that works behind NAT. Are you kidding me? I used to think that anything SIP was a pain behind NAT until I turned on 'nat=yes' for the peer/user/friend entry in sip.conf and told the IP501 to register to the Asterisk box. (All defaults too, no special hyper-fast register interval or goofy Polycom configuration at all.) And even after that I wouldn't believe it until I had three of them behind a plain-jane WRT54G on standard Telco dynamic IP ADSL talking to my Asterisk box connected through third-party ADSL. Calls go out, calls come in, it's as if they're on the same LAN. Seriously: It Just Works. I keep popping into #asterisk-dev and thanking OEJ. I'm still not a huge fan of SIP but I have had *no* issues with Polycom IP501s behind NAT talking to an Asterisk box on a real IP. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Phones were working fine - Now there is noaudiowhen calling between extensions
I didn't say it doens't work, I said it's not the best, and if you want I'll repeat myslef, Polycoms are not the best behind NAT, Cisco, or SPAs are much better. Just because you didn't run into any problems doesn't mean that it works well with all NAT devices. On 3/21/06, Andrew Kohlsmith [EMAIL PROTECTED] wrote: On Tuesday 21 March 2006 10:55, C F wrote: Polycoms are not the best if you want a phone that works behind NAT. Are you kidding me? I used to think that anything SIP was a pain behind NAT until I turned on 'nat=yes' for the peer/user/friend entry in sip.conf and told the IP501 to register to the Asterisk box. (All defaults too, no special hyper-fast register interval or goofy Polycom configuration at all.) And even after that I wouldn't believe it until I had three of them behind a plain-jane WRT54G on standard Telco dynamic IP ADSL talking to my Asterisk box connected through third-party ADSL. Calls go out, calls come in, it's as if they're on the same LAN. Seriously: It Just Works. I keep popping into #asterisk-dev and thanking OEJ. I'm still not a huge fan of SIP but I have had *no* issues with Polycom IP501s behind NAT talking to an Asterisk box on a real IP. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Phones were working fine - Now there is noaudiowhen calling between extensions
Andrew Kohlsmith wrote: On Tuesday 21 March 2006 10:55, C F wrote: Polycoms are not the best if you want a phone that works behind NAT. Are you kidding me? I used to think that anything SIP was a pain behind NAT until I turned on 'nat=yes' for the peer/user/friend entry in sip.conf and told the IP501 to register to the Asterisk box. (All defaults too, no special hyper-fast register interval or goofy Polycom configuration at all.) And even after that I wouldn't believe it until I had three of them behind a plain-jane WRT54G on standard Telco dynamic IP ADSL talking to my Asterisk box connected through third-party ADSL. Calls go out, calls come in, it's as if they're on the same LAN. Seriously: It Just Works. I keep popping into #asterisk-dev and thanking OEJ. I'm still not a huge fan of SIP but I have had *no* issues with Polycom IP501s behind NAT talking to an Asterisk box on a real IP. Same here with IP600; they just work. :) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Phones were working fine - Now there is no audio when calling between extensions
Hey group, I have a Polycom 501 and a 301 together in my office. Each phone is registered to a different server. When I call one of the phones from the other, the other phone rings no problem (the calls are passed between servers via IAX). However, when I answer it, there is absolutely no audio in either direction. This just started happening today. It was working great before - I just plugged them in, got them registered and I was calling between phones no problem. Now I dont know what is happening and I cannot figure it out. It seems like a NAT issue, but I have qualify=yes, nat=yes and insecure=port,invite. - Gabe ___ --Bandwidth 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] Phones were working fine - Now there is no audiowhen calling between extensions
I just did a little RTP debug and this is what it shows: == Spawn extension (304, 301, 1) exited non-zero on 'SIP/304-d9f8' -- Accepting AUTHENTICATED call from 216.152.244.81: requested format = ulaw, requested prefs = (), actual format = ulaw, host prefs = (), priority = mine -- Executing Dial(IAX2/to_80-1, SIP/301) in new stack -- Called 301 -- SIP/301-1fec is ringing -- SIP/301-1fec answered IAX2/to_80-1 Got RTP packet from 24.50.66.128:2228 (type 0, seq 16810, ts 344311448, len 160) Got RTP packet from 24.50.66.128:2228 (type 0, seq 16811, ts 344311608, len 160) Got RTP packet from 24.50.66.128:2228 (type 0, seq 16812, ts 344311768, len 160) Got RTP packet from 24.50.66.128:2228 (type 0, seq 16813, ts 344311928, len 160) Got RTP packet from 24.50.66.128:2228 (type 0, seq 16814, ts 344312088, len 160) Got RTP packet from 24.50.66.128:2228 (type 0, seq 16815, ts 344312248, len 160) Got RTP packet from 24.50.66.128:2228 (type 0, seq 16816, ts 344312408, len 160) Got RTP packet from 24.50.66.128:2228 (type 0, seq 16817, ts 344312568, len 160) Got RTP packet from 24.50.66.128:2228 (type 0, seq 16818, ts 344312728, len 160) Got RTP packet from 24.50.66.128:2228 (type 0, seq 16819, ts 344312888, len 160) Got RTP packet from 24.50.66.128:2228 (type 0, seq 16820, ts 344313048, len 160) Got RTP packet from 24.50.66.128:2228 (type 0, seq 16821, ts 344313208, len 160) Got RTP packet from 24.50.66.128:2228 (type 0, seq 16822, ts 344313368, len 160) .. that goes on for ever while the call is in progress. This is a call between phones that go between two * servers. If I make a call between phones both registered to the same asterisk server, this is my RTP stream: -- Executing Dial(SIP/304-c211, SIP/301|30|r) in new stack -- Called 301 -- SIP/301-b2c8 is ringing -- SIP/301-b2c8 answered SIP/304-c211 -- Attempting native bridge of SIP/304-c211 and SIP/301-b2c8 Got RTP packet from 24.50.66.128:2234 (type 0, seq 39729, ts -1972065425, len 160) Sent RTP packet to 24.50.66.128:2232 (type 0, seq 24819, ts 0, len 160) Got RTP packet from 24.50.66.128:2234 (type 0, seq 39730, ts -1972065265, len 160) Sent RTP packet to 24.50.66.128:2232 (type 0, seq 24820, ts 160, len 160) Got RTP packet from 24.50.66.128:2234 (type 0, seq 39731, ts -1972065105, len 160) Sent RTP packet to 24.50.66.128:2232 (type 0, seq 24821, ts 320, len 160) Got RTP packet from 24.50.66.128:2234 (type 0, seq 39732, ts -1972064945, len 160) Sent RTP packet to 24.50.66.128:2232 (type 0, seq 24822, ts 480, len 160) Got RTP packet from 24.50.66.128:2232 (type 0, seq 46694, ts 1105329892, len 160) Sent RTP packet to 24.50.66.128:2234 (type 0, seq 42690, ts 0, len 160) Got RTP packet from 24.50.66.128:2232 (type 0, seq 46695, ts 1105330052, len 160) Sent RTP packet to 24.50.66.128:2234 (type 0, seq 42691, ts 160, len 160) Got RTP packet from 24.50.66.128:2232 (type 0, seq 46696, ts 1105330212, len 160) Sent RTP packet to 24.50.66.128:2234 (type 0, seq 42692, ts 320, len 160) Got RTP packet from 24.50.66.128:2232 (type 0, seq 46697, ts 1105330372, len 160) Sent RTP packet to 24.50.66.128:2234 (type 0, seq 42693, ts 480, len 160) Got RTP packet from 24.50.66.128:2232 (type 0, seq 46698, ts 1105330532, len 160) Sent RTP packet to 24.50.66.128:2234 (type 0, seq 42694, ts 640, len 160) [THE END] Once I anser the call, the RTP string starts and then stops right where I put [THE END]. - Gabe Hey group, I have a Polycom 501 and a 301 together in my office. Each phone is registered to a different server. When I call one of the phones from the other, the other phone rings no problem (the calls are passed between servers via IAX). However, when I answer it, there is absolutely no audio in either direction. This just started happening today. It was working great before - I just plugged them in, got them registered and I was calling between phones no problem. Now I dont know what is happening and I cannot figure it out. It seems like a NAT issue, but I have qualify=yes, nat=yes and insecure=port,invite. - Gabe ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Phones were working fine - Now there is noaudiowhen calling between extensions
This is driving me nuts!! After unplugging all the phones, restarting the router and the modem, and reconfigurating my * boxes, I was finally able to communicate between both phones only when they were both registered to the same server. If I try to call between phones between two different servers trunked with IAX, there is no sound (but the call rings and connects perfectly). This was working last week *no problem*, all of a sudden its dead!!! Its killing me because it *was* working and now its not and I cannot figure out. - Gabe - Original Message - From: Gabriel Afana [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, March 20, 2006 4:39 PM Subject: Re: [Asterisk-Users] Phones were working fine - Now there is noaudiowhen calling between extensions I just did a little RTP debug and this is what it shows: == Spawn extension (304, 301, 1) exited non-zero on 'SIP/304-d9f8' -- Accepting AUTHENTICATED call from 216.152.244.81: requested format = ulaw, requested prefs = (), actual format = ulaw, host prefs = (), priority = mine -- Executing Dial(IAX2/to_80-1, SIP/301) in new stack -- Called 301 -- SIP/301-1fec is ringing -- SIP/301-1fec answered IAX2/to_80-1 Got RTP packet from 24.50.66.128:2228 (type 0, seq 16810, ts 344311448, len 160) Got RTP packet from 24.50.66.128:2228 (type 0, seq 16811, ts 344311608, len 160) Got RTP packet from 24.50.66.128:2228 (type 0, seq 16812, ts 344311768, len 160) Got RTP packet from 24.50.66.128:2228 (type 0, seq 16813, ts 344311928, len 160) Got RTP packet from 24.50.66.128:2228 (type 0, seq 16814, ts 344312088, len 160) Got RTP packet from 24.50.66.128:2228 (type 0, seq 16815, ts 344312248, len 160) Got RTP packet from 24.50.66.128:2228 (type 0, seq 16816, ts 344312408, len 160) Got RTP packet from 24.50.66.128:2228 (type 0, seq 16817, ts 344312568, len 160) Got RTP packet from 24.50.66.128:2228 (type 0, seq 16818, ts 344312728, len 160) Got RTP packet from 24.50.66.128:2228 (type 0, seq 16819, ts 344312888, len 160) Got RTP packet from 24.50.66.128:2228 (type 0, seq 16820, ts 344313048, len 160) Got RTP packet from 24.50.66.128:2228 (type 0, seq 16821, ts 344313208, len 160) Got RTP packet from 24.50.66.128:2228 (type 0, seq 16822, ts 344313368, len 160) .. that goes on for ever while the call is in progress. This is a call between phones that go between two * servers. If I make a call between phones both registered to the same asterisk server, this is my RTP stream: -- Executing Dial(SIP/304-c211, SIP/301|30|r) in new stack -- Called 301 -- SIP/301-b2c8 is ringing -- SIP/301-b2c8 answered SIP/304-c211 -- Attempting native bridge of SIP/304-c211 and SIP/301-b2c8 Got RTP packet from 24.50.66.128:2234 (type 0, seq 39729, ts -1972065425, len 160) Sent RTP packet to 24.50.66.128:2232 (type 0, seq 24819, ts 0, len 160) Got RTP packet from 24.50.66.128:2234 (type 0, seq 39730, ts -1972065265, len 160) Sent RTP packet to 24.50.66.128:2232 (type 0, seq 24820, ts 160, len 160) Got RTP packet from 24.50.66.128:2234 (type 0, seq 39731, ts -1972065105, len 160) Sent RTP packet to 24.50.66.128:2232 (type 0, seq 24821, ts 320, len 160) Got RTP packet from 24.50.66.128:2234 (type 0, seq 39732, ts -1972064945, len 160) Sent RTP packet to 24.50.66.128:2232 (type 0, seq 24822, ts 480, len 160) Got RTP packet from 24.50.66.128:2232 (type 0, seq 46694, ts 1105329892, len 160) Sent RTP packet to 24.50.66.128:2234 (type 0, seq 42690, ts 0, len 160) Got RTP packet from 24.50.66.128:2232 (type 0, seq 46695, ts 1105330052, len 160) Sent RTP packet to 24.50.66.128:2234 (type 0, seq 42691, ts 160, len 160) Got RTP packet from 24.50.66.128:2232 (type 0, seq 46696, ts 1105330212, len 160) Sent RTP packet to 24.50.66.128:2234 (type 0, seq 42692, ts 320, len 160) Got RTP packet from 24.50.66.128:2232 (type 0, seq 46697, ts 1105330372, len 160) Sent RTP packet to 24.50.66.128:2234 (type 0, seq 42693, ts 480, len 160) Got RTP packet from 24.50.66.128:2232 (type 0, seq 46698, ts 1105330532, len 160) Sent RTP packet to 24.50.66.128:2234 (type 0, seq 42694, ts 640, len 160) [THE END] Once I anser the call, the RTP string starts and then stops right where I put [THE END]. - Gabe Hey group, I have a Polycom 501 and a 301 together in my office. Each phone is registered to a different server. When I call one of the phones from the other, the other phone rings no problem (the calls are passed between servers via IAX). However, when I answer it, there is absolutely no audio in either direction. This just started happening today. It was working great before - I just plugged them in, got them registered and I was calling between phones no problem. Now I dont know what is happening and I cannot figure
[Asterisk-Users] Phones
Title: Phones Greetings, Has anyone had any success utilizing Asterisk with Digium or other cards to allow plugging/utilizing Partner/Lucent Phones? These are the phones that usually come with the Partner ACS PBX system? (Partner 8,16,24 phones, I think they are proprietary). Also is there any add on programs/plug-in for Asterisk that can be used as a 'call back' from a pool of numbers with an associated wave, mp3 or other recorded file? (Note I am referring to a 'War' Dialing type scenario it is so I can remind people of their appointments. This is not cold calling, nor would it be utilizing predictive dialing at least I do not think so on the predictive dialing part anyway). Thanks in advance. Je ___ --Bandwidth 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] Phones
Greetings, Has anyone had any success with Asterisk and Digium or other cards to allow plugging/utilizing Partner/Lucent Phones? These are the phones that usually come with the Partner ACS PBX system? (Partner 8,16,24 phones, I do not know if they are analog or digital but do think they are proprietary). Also is there any add on programs/plug-in for Asterisk that can be used as a 'call back' from a pool of numbers with an associated wave, mp3 or other recorded file? (Note I am referring to a 'War' Dialing type scenario and nope it is so I can remind people of their appointments. This is not cold calling, nor would it be utilizing predictive dialing at least I do not think so on the predictive dialing part anyway). Thanks in advance. Je ___ --Bandwidth 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] Phones no longer register - except one?
Hi I've got an interesting problem. A few days ago (maybe even a week or two) all my sip phones lost registrations with my asterisk box. All that is but one. The asterisk box is out on the internet, I have two phones at my location and 1 at another separate location. The only phone that remains registered is an Integrated Networks IN002 (or something like that). This is at my location. I also have a grandstream GXP-2000 that will not register. This is also at my location. I have tried xlite and sjphone (on my desktop and mobile phone (via wireless) respectivley) to test, These also fail to register. I have a Budgetone 102 at another location which also fails to register. There is nothing on the command line apart from the IN002 phone registering and talking to the * server. It dials in and out fine. The only thing I can see that mine and the remote location have in common is the ISP that provides DSL (plusnet), and I was wondering if they were limiting traffic as they have recently announced their own telephony service. But I doubt it and if that was the case then why does the IN002 register and not the budget tone? Its a crazy paranoid theory, but I can't think of anything else. The * is 1.0.9 and was working perfectly when I upgraded from a CVS version. Any ideas - I must be missing something obvious - but I've not changed anything since the upgrade. Any why one phone? Mark ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Phones Callwaiting enable by default?
I'm using Sipura SPA-841 and SPA-2000 phones and ATAs... Yes.. it's a good assumption that I'm using asterisk, since I posted to this list Umm *70 is there to turn call waiting on/off in the asterisk database. On Mar 31, 2005 7:57 PM, C F [EMAIL PROTECTED] wrote: What phones? are you using Avaya, or Toshiba? Since you are posting to this list I will guess you are using Asterisk, in which case I have no clue why *70 is there in the first place. Did you notice that the guy that went to the Doctor that his eye hurts when drinking coffee, refused to remove the spoon from the cup? On Thu, 31 Mar 2005 11:03:12 -0500, Matt [EMAIL PROTECTED] wrote: Hi, how can I get all the phones to enable call waiting by default instead of having to dial *70 on each one to activate call waiting? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Phones Callwaiting enable by default?
Hi, how can I get all the phones to enable call waiting by default instead of having to dial *70 on each one to activate call waiting? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Phones Callwaiting enable by default?
What phones? are you using Avaya, or Toshiba? Since you are posting to this list I will guess you are using Asterisk, in which case I have no clue why *70 is there in the first place. Did you notice that the guy that went to the Doctor that his eye hurts when drinking coffee, refused to remove the spoon from the cup? On Thu, 31 Mar 2005 11:03:12 -0500, Matt [EMAIL PROTECTED] wrote: Hi, how can I get all the phones to enable call waiting by default instead of having to dial *70 on each one to activate call waiting? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Phones for vitural office business
I am looking for IP phones that are suited to serviced office operation. The business is to answer calls for customers. The incoming lines are E1 and customers are allocated with DID. So the customers' phone can be answered with the customers' designed messages and instruction. This can be done easily with key phone system. It seems to be a problem for IP phone system. I read previous discussions on the pros and cons of key system and ip phone system. However, still need to offer an solution to the operation whereby when calls come in, the operators will be able to identify whose customer's call and answer for that customer accordingly. I have a look at Snom 220 with console. Does the line appearance function solve this problem if DID is assigned to difference line of the ip phone? Any suggestion will be welcome. David Kwok ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Phones for vitural office business
On Mon, 2005-02-21 at 13:25 +1100, dkwok wrote: I am looking for IP phones that are suited to serviced office operation. The business is to answer calls for customers. The incoming lines are E1 and customers are allocated with DID. So the customers' phone can be answered with the customers' designed messages and instruction. This can be done easily with key phone system. It seems to be a problem for IP phone system. I read previous discussions on the pros and cons of key system and ip phone system. However, still need to offer an solution to the operation whereby when calls come in, the operators will be able to identify whose customer's call and answer for that customer accordingly. IMHO, any phone with some lights/buttons on it will not be helpful, unless you will only ever have less than X different customers/DID's. IMHO, you need some sort of simple PC based app, and, if you think about it, you could do a lot more than any other key system. eg, When the call arrives, lookup the DID in the database, and retrieve the call script/announcement, then send that data to your client application along with the CND, and ring the phone. You know now what to say, and who is calling before you even answer the phone. Extra points if you also include some work to see that this person has called previously and left three messages in the last 20 minutes. (even if the calls are being answered by different people, you could still let them know that the previous messages have been sent, etc... If you would like someone to write this solution, contact me off-list, or see the wiki for asterisk consultants... Regards, Adam ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] phones with two ethernet ports
On Mon, 2005-01-03 at 14:34 -0700, Harry McGregor wrote: Try finding a VOIP Phone that does gigabit... Won't happen for a while. Cable is cheap when you look at the cost of running the cable. If you use two boxes, it will take virtually the same amount of time to run two as it does to run one. I ordered some special siamese cat5e cable to overcome this, the electrician told to pull the cables and left on his own, promptly spent all his time pulling them apart and running _one_ to each end point! You can't get the staff these days. -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
