[asterisk-users] res_pgsql re-connect on db failure?
We are using PostgreSQL real-time connector (res_config_pgsql) with Asterisk 1.6.2.15. From time to time, we need to reset our PostgreSQL server, causing all active DB connections to close. While other packages in our system re-connect gracefully when this happens, Asterisk appears to not bother trying. It instead goes into an endless loop complaining that the connection has closed. Question -- is there any option I might be missing, to tell Asterisk to try re-connecting to PostgreSQL if the existing connection fails? Thank you, Bryan -- _ -- 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] Realtime SIP, multiple AX servers question
On Jan 4, 2011, at 12:26 PM, Tilghman Lesher wrote: It wasn't designed to do this. While you can have the same sippeers table for multiple servers, you really should have a separate sipregs table for each backend server. The reason why is that some mappings depend implicitly on the host to which it was registered. For example, if a phone is behind a NAT, then the external port is dependent upon the same host responding. If a different host tries to communicate to that external port, some NAT devices will not route the packet properly. This is especially true for SIP over TCP, but it's also true for UDP packets. (Routing packets back through a NAT without verifying the sending IP is a security risk.) Probably more appropriate for your case is to use DUNDi to coordinate your machines as to which server presents holds the registration for any specific phone. We have one table which is serving both purposes (peers and reg). When we want to route a call to an ATA, we first look up that ATA's regserver in that table, and then construct a SIP URI based upon that regserver address. In that way, we route the call through the server to which the ATA is currently registered. So I guess we're covered already in the scenario you describe. It seems like not a great design to have to have a private sipregs table for every server in our pool, especially given that the pool will grow (or maybe shrink) over time. Is that really the recommended design? I haven't seen any articles describing that setup for RealTime in a multi-server environment. Thank you, Bryan -- _ -- 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] Realtime SIP, multiple AX servers question
Thanks Olle. Do you suppose I am the first Asterisk user to discover this behavior? I would find that hard to believe that I'm the first person to notice... Your idea for how to deal with sounds reasonable.. Thank you, Bryan On Jan 4, 2011, at 12:18 AM, Olle E. Johansson wrote: 3 jan 2011 kl. 00.26 skrev Bryan Field-Elliot: Normally, no matter which Asterisk server an ATA connects to, we get our database fields filled out correctly, such as regseconds, lastms, ipadr, etc. However, with some ATA's we are experiencing a problem as follows: 1. ATA reaches its re-registration timeout, which we typically configure to be 60 minutes. 2. ATA re-queries DNS SRV record, and ends up re-registering with a different AX server than it was on previously. 3. The new AX server updates the Realtime DB fields (regseconds, lastms, etc). 4. The old AX server, after a few more minutes, notices that the ATA has vanished, and therefore clears out these same fields. Oh, that's an interesting observation. Depending on how you see it, it's a bug or a feature request. Code-wise what you could do is that Asterisk could retrieve the information from realtime. If the sysname is not the same as the systems, it let the information be. If it's the same sysname, then erase the information. /O -- _ -- 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] Realtime SIP, multiple AX servers question
We have several Asterisk servers (1.6.2.15) all configured for Realtime, all backed by the same database. The Asterisk servers are all listed under DNS SRV records, and SIP ATAs find us this way. Normally, no matter which Asterisk server an ATA connects to, we get our database fields filled out correctly, such as regseconds, lastms, ipadr, etc. However, with some ATA's we are experiencing a problem as follows: 1. ATA reaches its re-registration timeout, which we typically configure to be 60 minutes. 2. ATA re-queries DNS SRV record, and ends up re-registering with a different AX server than it was on previously. 3. The new AX server updates the Realtime DB fields (regseconds, lastms, etc). 4. The old AX server, after a few more minutes, notices that the ATA has vanished, and therefore clears out these same fields. Is there any way to fix this problem? We need to know if ATA's go offline, but, we don't want them caught in this endless loop where our multiple AX servers are out-guessing eachother and overwriting valid data in the database. Our realtime options in sip.conf are as follows: rtcachefriends=yes rtsavesysname=yes ;rtautoclear=yes ;ignoreregexpire=yes Because we are using rtsavesysname, a perfect solution seems like it might be along the lines of If an ATA disappears, empty out the RT database fields ONLY if it's last regserver was this one. Is this possible? Thank you, Bryan -- _ -- 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] internal_timing not working (re: SIP silence suppression)
We are trying to get Asterisk to behave correctly when our SIP clients have Silence Suppression turn on, but are not having any luck. Basically, there are several apps in Asterisk which won't send any audio to the SIP client, unless the SIP client itself sends audio to Asterisk (which it won't do if Silence Suppression is enabled and the caller is quiet). We think we've done everything right in terms of setup, but obviously we're missing something. Can anyone please advise? Here are the relevant data points: - Asterisk version is 1.6.1 (r200516M) - Dahdi (version 2.2.0 rc2) is installed and running. Linux module is loaded. - Command-line tool dahdi_test returns +99% accuracy - res_timing_dahdi.so is loaded in Asterisk modules.conf file - internal_timing is set to yes in Asterisk asterisk.conf file - Asterisk CLI command timing test reports: Attempting to test a timer with 50 ticks per second. Using the 'DAHDI' timing module for this test. It has been 1019 milliseconds, and we got 51 timer ticks Based upon my understanding of things, all of the above points to go with respect to proper support of SIP Silence Suppression in the client, but we're just not seeing it. Any help would be appreciated, Thank you, Bryan ___ -- 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] SIP Silence Suppression?
We're using Asterisk 1.6.1. When our SIP clients have silence suppression turned on, it's a problem for many apps. Is there a workaround for this in Asterisk? Other than turning silence suppression off in the SIP client, is there anything I can do on the Asterisk side to make things work again? Basically, Asterisk will often not send any audio to the client, until it receives an audio packet from the client, which is not going to happen when the client itself is silent (and when it has silence suppression enabled). I know this is an old problem for Asterisk, but I would be surprised to learn that after so long, it's still unresolved. Thank you, Bryan ___ -- 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] SIP qualify time - best practices?
For the typical home user who has a SIP ATA behind (usually) a Linksys home router/firewall, what's the best practice qualify= time we should be running on the server, to keep the home user's NAT happy? The default, 2 seconds, is way too short (generates too much net traffic). I am wondering how high we can go, and still make the majority of our customers' home nat's happy. 1 minute? 2 minutes? 10 minutes? Thanks for opinions, Bryan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk 1.2.9 cli -x doesn't flush?
We have a script which executes asterisk -n -r -x . periodically against the running server, to check the status of a few things, and pipe the output to a file. With prior versions of Asterisk this worked fine, but having just upgraded to 1.2.9, we are finding that if the output is lengthy, then Asterisk seems to terminate before fully flushing stdout. Is this a known bug, is there a patch, etc? Thank you in advance, Bryan ___ --Bandwidth 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] Sending ANI to TDM40B FXS?
We are using TDM40B's to connect some devices to Asterisk which depend on caller information arriving as ANI, rather than as Caller ID. I am unsure if the TDM40B supports this in the first place, and if so, I am unsure how to configure it so. I've searched the wiki but couldn't find anything. Can someone please confirm whether or not this is possible? As a fallback I could reconfigure my FXS analog device to use Caller ID but I would rather leave it as ANI if possible. Thank you, Bryan ___ --Bandwidth 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] RTP ports in use grows and grows...
We've been seeing a pattern over the last couple of weeks with our Asterisk servers (1.0.9). The number of ports in use (RTP) seems to grow by two every minute or so. Eventually the server will run out of allowable files open and crash. We are resetting the server once per day to prevent this from occurring. Running lsof shows the end of the list like this: asterisk 26733 astx 1794u IPv4 183341654 UDP 192.168.1.123:22790 asterisk 26733 astx 1795u IPv4 183341655 UDP *:22791 asterisk 26733 astx 1796u IPv4 183312190 UDP 192.168.1.123:24170 asterisk 26733 astx 1797u IPv4 183312191 UDP *:24171 asterisk 26733 astx 1798u IPv4 183335059 UDP 192.168.1.123:21132 asterisk 26733 astx 1799u IPv4 183335060 UDP *:21133 asterisk 26733 astx 1802u IPv4 183342252 UDP 192.168.1.123:20358 asterisk 26733 astx 1803u IPv4 183342253 UDP *:20359 and the list grows and grows (slowly but steadily). We have only around 600 SIP clients on this particular server so something is clearly amiss. My questions are: 1. Are there any known file handle leaks in rtp.c or sip.c? 2. If the number of ports open eventually reaches the port boundaries I have configured in rtp.c (e.g. start at 2, end at 25000), will Asterisk crash, or it will it start to recycle those ports? 3. I have set ulimit -n to a higher number but this doesn't seem like a solution, only a temporary band-aid. Thank you, Bryan ___ --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
[Asterisk-Users] Received packet with bad UDP checksum - whats the real problem?
We have a customer trying to dial through our server, and our server is throwing tons of these log messages: Jul 27 14:21:02 NOTICE[29210]: rtp.c:431 ast_rtp_read: RTP: Received packet with bad UDP checksum Is it pretty certain, that these are caused by a bad or misconfigured router along the path, or something else network-related? As opposed to the SIP hardware itself? The SIP ATA is the same model in use by many of our other customers, without any problems. Thank you, Bryan ___ 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] Load per server?
I'm trying to gauge the amount of overhead for idle users (NOT in the middle of a phone call) per user, per server. These are a combination of SIP and IAX2 clients, with qualify=yes. On, for example, a dual 2.4 Ghz Pentium server (with plenty of RAM), how many hundreds, or thousands (rough ballpark) of clients can be supported? Again, these are mostly idle, and I'm interested in a calculation of idle users (with qualify=yes), not # of simultaneous calls. Thank you for any ballparks anyone might provide, Bryan ___ 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] RFC2833 firewall problems? (16-byte UDP packets)
We are tracking the following situation: SIP client connects to our Asterisk server, and then connects to another SIP user. Re-invite is OFF, so Asterisk is in the middle of the whole conversation. When one SIP client sends DTMF tones, the SIP client uses RFC2833 to send the tones to the server. (This is correct). The server then sends RFC2833 tones out to the other SIP client. The problem is, the other SIP client is never receiving the RFC2833 packets. An ethereal trace on the same local network shows that the regular conversation UDP packets are coming through just fine (packet length: 172), but the RFC2833 packets are never received on the SIP client LAN (though they are sent by the server). RFC2833 UDP packets appear to be packet length: 16. Has anyone seeing this kind of behavior, perhaps from firewalls or otherwise? ___ 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] RFC2833 firewall problems? (16-byte UDP packets)
When watching from the Asterisk server (running tcpdump), the destination IP address and port number are the same for the conversation packets and the RFC2833 packets. But when watching from the SIP client's local LAN (another computer on the subnet, in promisc mode, running tcpdump), the RFC2833 packets (length 16) never arrive. Weird? On Thu, 2005-06-02 at 13:01 +1200, Matt Riddell wrote: Bryan Field-Elliot wrote: The problem is, the other SIP client is never receiving the RFC2833 packets. An ethereal trace on the same local network shows that the regular conversation UDP packets are coming through just fine (packet length: 172), but the RFC2833 packets are never received on the SIP client LAN (though they are sent by the server). RFC2833 UDP packets appear to be packet length: 16. Has anyone seeing this kind of behavior, perhaps from firewalls or otherwise? I don't suppose you see the packets go anywhere else? ___ 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] How to get in touch with sixTel?
If anybody here is a sixTel customer, can you share any tips tricks for getting in touch with anybody there? They are absurdly hard to get a hold of, particularly when you have a technical issue needing to be resolved. If anyone has any phone numbers other than their main 800 line, I'd sure appreciate it. Thank you, Bryan ___ 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] Transferring a call, IAX2-SIP, DTMF/RFC2833 doesn't work?
We are using Asterisk 1.0.7. We have this scenario: IAX2 user comes in to Asterisk, dials an extension, and transfers to a SIP user. The dial command is simple, looks like this: exten = 300,1,Dial(SIP/300) Extension 300 is a legacy PBX device operated by touchtones. The user (coming in over IAX2) is trying to drive this PBX using touchtones. But the trouble is, by the time the touchtones go out to the SIP extension, it's being sent as audio, not as RFC2833. An ethereal trace confirms this. DTMF is arriving to Asterisk (via IAX2) as OOB data, but no RFC2833 is going back out to the SIP device. The SIP device is configured to use RFC2833 wherever possible. This doesn't work because the DTMF is arriving really choppy-sounding, and the PBX doesn't recognize it. Am I doing something wrong? Help would be appreciated! P.S. When we go IAX2--Asterisk--IAX2, DTMF OOB is preserved correctly. This is just when we go IAX2--Asterisk--SIP. ___ 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] IAXy Hung, Power-cycle Required
On Wed, 2005-01-26 at 15:59 -0500, Paul Dugas wrote: I've got a single IAXy installed in a little office nearby and got a call from someone on site a finew mintues ago. Apparently they couldn't make a call on that extension. They'd pick up the phone and get nothing; no dial-tone. Has snyone else had trouble with these things sticking like this? Paul Yes - we are having the exact same problem with a portion of our IAXys in the field. In all cases the IAXys are behind simple SOHO firewalls like the Linksys. After an idle period - perhaps 1-3 days - they just stop working, in both directions, but a simple power cycle restores functionality. We have an open support incident with Digium but have not yet heard back. FWIW we have stopped selling deploying the IAXys until we have a resolution to the problem. Bryan ___ 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
