Re: [asterisk-users] res_rtp_asterisk.so problem with minimal (ish) chan-sip based Asterisk
> The config sorcery wizard is implemented by the res_sorcery_config.so module Yup, that fixed it, modules.conf now starts with [modules] autoload=no load => res_sorcery_config.so load => res_pjproject.so load => res_rtp_asterisk.so ; Thanks! -- Andreas Sikkema -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] res_rtp_asterisk.so problem with minimal (ish) chan-sip based Asterisk
Hi, For years I've been running a minimal (ish) SIP based Asterisk with the modules based on chan-sip. For various reasons unrelated to Asterisk the machine the latest incarnation of this configuration has been updated to Debian Buster and thus to Asterisk 16. Since this upgrade I have a dependency problem related to res_rtp_asterisk.so. So the old config was: [modules] autoload=no load => res_rtp_asterisk.so load => res_http_websocket.so load => chan_local.so load => codec_ulaw.so load => codec_alaw.so load => pbx_config.so load => chan_sip.so load => app_dial.so load => func_callerid.so load => func_cut.so load => func_logic.so [global] Since Asterisk 16 (Debian Buster version) I have a dependency problem, where res_rtp_asterisk.so is dependent on res_pjproject.so: When I try to make a call: [Dec 22 22:00:55] ERROR[6093][C-0001]: rtp_engine.c:474 ast_rtp_instance_new: No RTP engine was found. Do you have one loaded? mng*CLI> module load res_rtp_asterisk Unable to load module res_rtp_asterisk Command 'module load res_rtp_asterisk' failed. [Dec 22 22:03:39] ERROR[28261]: loader.c:170 module_load_error: res_rtp_asterisk loaded before dependency res_pjproject! mng*CLI> module load res_pjproject Unable to load module res_pjproject Command 'module load res_pjproject' failed. [Dec 22 22:04:04] ERROR[28261]: sorcery.c:840 __ast_sorcery_insert_wizard_mapping: Wizard 'config' could not be applied to object type 'log_mappings' as it was not found [Dec 22 22:04:04] WARNING[28261]: res_pjproject.c:665 load_module: Failed to register pjproject log_mappings object with sorcery I haven't been able to find what I need to do to get "Wizard 'config'" to be applied or why I need it. Googling for this phrase suggested I created an empty config file for pjproject but this also didn't resolve this problem. I am sure I must have missed something, can someone point me in the correct direction? Thanks! -- Andreas Sikkema -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP call drops after 32 seconds, but only when....
> but as soon as I configure another sip registration on another server, > outgoing > calls drop after 32 seconds. Are both your servers behind the same NAT router? -- Andreas Sikkema -- _ -- 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] Suspicious routers
Darryl, > I've seen and suspected this before, and changing the old cheap routers > has generally fixed this, but I'm wondering if anyone else has seen this > before, and if there are other routers I need to worry about. I don't > yet have an automated way to test routers for this, but I'm seriously > thinking about coming up with something. Most of the cheaper NAT implementations appear to assume that there's ever only just one client on the LAN side sending traffic from port A to a server port on the WAN side. For TCP this assumption is a nice hack with not too much risk, for UDP applications which send traffic from a well known port to a well known port, this is killing. I've added a full chapter on this problem in our manual that gets sent to customers, which basically says to reconfigure the SIP clients to all use a different source port for SIP traffic. This should be applicable to most UDP based protocols. I think this is valid for most routers below a certain price point ($250?), perhaps those running Linux might not be affected. -- Andreas Sikkema -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Into queue the caller doesn't hear the ringing
On 17/05/14 08:44 , Danilo Dionisi wrote:
> [latina_open]
> exten => s,1,Verbose( ** FILIALE DI LATINA APERTA.)
> same => n(menu),Background(risponditore-filiali/Latina/ivr)
> same => n,Waitexten(5)
> [latina_close]
> exten => s,1,Verbose( ** FILIALE DI LATINA CHIUSA.)
> same => n,Queue(coda_ivr-latina_close-${QUEUE},r) /_--> the ringing works!_
> /same => n,Hangup()*
In latina_open you first play a WAV file, and then go to the queue, in
latina_close you go directly to the queue. That probably explains the
difference between the two situations.
I've never played much with the Queue command so I don't know if there's
a flag that is needed to add ringing from the Queue command or that a
simple additional Ringing command in latina_open might help.
--
Andreas Sikkema
--
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Re: [asterisk-users] Movistar sip Mexico
On 20/11/13 20:32 , Damian Gonzalez wrote: > I have a problem with movistar in Mexico with a sip calls. Movistar send > to me T38 and G729 in the INVITE and they say that I have to ignore T38 > and use G729 in the voice call. I have had the same problem with a carrier, where some calls we receive from them have an image and an audio stream in the initial INVITE, even though the call is intended to use the audio stream. Responding back accepting T.38 will fail and *all* other options trying to reject the T.38 using known SIP supported methods will also fail. The *only* option is to just ignore the image stream, which is not allowed by the current set of SIP RFCs... Asterisk used to ignore the image stream, but since the 1.8(?) timeframe its behaviour has changed more towards standards compliance in this area. And now we're between a rock and a hard place. The only way out that I could find is to put something in front of Asterisk that just removes the image stream from initial INVITEs when received from the carrier. (OpenSIPS has this nice method called remove_stream() since a couple of versions) Complaining about this didn't help, "Asterisk is not certified because Open Source", was basically their answer. -- Andreas Sikkema -- _ -- 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] Asterisk sends the INTERNAL IP address of my equipments to my SIP friends?!?
On 6/13/13 16:20 , Matthew J. Roth wrote: > It's hard to be certain without seeing a full SIP trace, but I think the > INVITE > with the internal IP is actually a re-INVITE that Asterisk is sending to > establish a native bridge between the SIP friend and the SIP gateway to PSTN > converter. It's actually pretty easy. If an INVITE message has a tag parameter in both To and From headers, it's a re-INVITE. If the To header doesn't have a tag parameter, it's an initial INVITE. -- Andreas Sikkema -- _ -- 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] Asterisk 11- Answer with [m=image 0 udptl t38] and Call Drop
Matthew, > A part of me wonders is if you're really running into the issue > described on ASTERISK-20908 [1]. Do you mind trying the patch on there > to see if it helps? The problem is that some customers of mine (erm ours) who are running an Asterisk based sot PBX are having some issues that are remarkably similar to what is described here. I can unfortunately only confirm that not every SIP client handles a null port reply or a 488 Not Acceptable Here message as we would normally expect. It's a complicated scenario involving a large carrier with legacy hardware that I don't have access to to do a quick test :-( -- Andreas Sikkema -- _ -- 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] Asterisk 11- Answer with [m=image 0 udptl t38] and Call Drop
On 1/18/13 13:24 , Matthew Jordan wrote: > 1) Contact your carrier and ask why they are rejecting the 200 OK. > > 2) Assuming they won't change their behaviour, find out what they want > in a response that declines an image media format. Without knowing what > your carrier thinks the SDP should look like, any modifications you make > to Asterisk will be guesses. I ran into a similar problem this week. There's a number of SIP implementations (either legacy or not good enough) that don't handle a zero port denial of a media stream quite right. In the past these implementations would have worked fine when the called party would have just ignored the offending media stream, instead of sending an explicit deny. -- Andreas Sikkema -- _ -- 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] Recommended T.38 settings for receiving faxes from Cisco AS5350XM
Hi, What are the recommended T.38 settings for sending/receiving faxes from Cisco AS5350XM gateways? The chan_sip.conf file has a remark about what Cisco is doing wrong and says that the values received from the gateway should be overridden, but doesn't say what settings to use for maximum success. Can anyone give me some suggestions? I don't know much about T.38 and I've been told I have to solve this... Thanks! -- Andreas -- _ -- 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] USB FXS device
On 11/1/12 15:08 , Jeff LaCoursiere wrote: > An ATA that does OpenVPN would be just as welcome. > If Xorcom would come out with a 1 or 2 port FXS device I would be all > set! How about it Tzafrir? I don't know if these are available in the US (presuming you are from that area in te world), but there are DSL modem/router suppliers who sell an Ethernet version of their product complete with internal ATA. I know some of them have VPN functionality, so that might work around some of your limitations. They are not Raspberry Pi priced, but not *that* much more. Developing something yourself from the ground up might be more expensive in the short/mid term I am thinking along the lines of a Draytek Vigor2110Vn, the wifi and routing capabilities are not used, but the SIP, VPN and FXS should be able to fit your needs. And with the possibility for automatic provisioning using an ACS.. -- Andreas Sikkema -- _ -- 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] Differences between PBX and SBC
> That's my question...the sbc provides security over trunking, right? The > same can do Asterisk or a Proxy..isn't? Does an SBC can provide any kind of > add-value to an Asterisk deployment? A PBX provides functionality to users. An SBC *can* secure a PBX against the outside world, but that is configuration dependent. The more powerful the SBC, the more configuration it requires to make things work, let alone secure whatever it is supposed to protect. An SBC is in essence a B2BUA, looking quote a lot like a really simple pass through Asterisk configuration. -- Andreas -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to receive SMS ?
On 2/18/12 00:17 , Gilles wrote: > I'd also be interested in learning from anyone who uses a GSM gateway > to TX/RX text messages with Asterisk and SIP clients. We're using a GSM gateway to send SMS messages from our network monitoring system. Once you dig through some chipset specs it was suprisingly easy to start sending SMS messages. While we didn't investigate receiving messages fully we did one quick test and that was easy enough. You just need some daemon to monitor the gateway to see if it has received a message and pass it on to Asterisk, sending the other way around is not that different. -- Andreas Sikkema -- _ -- 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] Cordless SIP phone
On 1/23/12 4:39 PM, eherr wrote: > Where I want to put the new on is outside the range. > > I thought SIP cordless phones would be better on the range. If you want to extend the range of a DECT basestation you can use repeaters, but you then lose DECT encryption and you can only add up to 6 repeaters around one basestation. Extending your range beyond that requires a proper DECT network and brings you into a whole new cost level. But that can go up to 256x12 handsets and 256x8 (IIRC) simultaneous calls... -- Andreas Sikkema -- _ -- 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] Sporadic one way audio problem
> deny=0.0.0.0/0.0.0.0 > permit=XXX.XXX.X.X/29 > permit=192.168.1.0/24 Are you sure your provider *always* sends data from this /29? Maybe you have this in your iptables as well and sometimes audio is received from outside this /29 and therefore blocked? -- Andreas Sikkema -- _ -- 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] dialplan problem : not including context
On 1/13/12 2:32 PM, Jonas Kellens wrote: > So the context TrunkAccounts is not included. > > Do you know why ? Does reloading the dialplan (dialplan reload) give any useful output relating to these two contexts? -- Andreas Sikkema -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi not installed and application's details is missing in Asterisk
> [root@haddock8-astrx dahdi-linux-complete-2.5.0.2+2.5.0.2]# make all > make -C linux all > make[1]: Entering directory > `/usr/src/dahdi-linux-complete-2.5.0.2+2.5.0.2/linux' > make -C drivers/dahdi/firmware firmware-loaders > make[2]: Entering directory > `/usr/src/dahdi-linux-complete-2.5.0.2+2.5.0.2/linux/drivers/dahdi/firmware' > make[2]: Leaving directory > `/usr/src/dahdi-linux-complete-2.5.0.2+2.5.0.2/linux/drivers/dahdi/firmware' > You do not appear to have the sources for the 2.6.18-194.11.1.el5 kernel > installed. > make[1]: *** [modules] Error 1 > make[1]: Leaving directory > `/usr/src/dahdi-linux-complete-2.5.0.2+2.5.0.2/linux' > make: *** [all] Error 2 > > this is the information of installed kernel. > > [root@haddock8-astrx dahdi-linux-complete-2.5.0.2+2.5.0.2]# rpm -qa|grep > kernel > kernel-xen-devel-2.6.18-274.12.1.el5 > kernel-debug-devel-2.6.18-274.12.1.el5 > kernel-debug-2.6.18-274.12.1.el5 > kernel-devel-2.6.18-274.12.1.el5 > kernel-doc-2.6.18-274.12.1.el5 > kernel-2.6.18-274.12.1.el5 > kernel-2.6.18-194.11.1.el5 > kernel-headers-2.6.18-274.12.1.el5 > kernel-xen-2.6.18-274.12.1.el5 You have headers installed for the kernel version 2.6.18-274.12.1.el5 but the DAHDI build is looking for kernel headers for 2.6.18-194.11.1.el5. Either install those kernel headers or reboot to kernel version 2.6.18-274.12.1.el5 and try again. -- Andreas -- _ -- 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] No rtpmap codec info in 200 OK
On 12/18/11 12:55 AM, William Scott wrote: > Notice there is no "rtpmap:18 G729/8000" in the reply. > > The call continues fine. > > Is it right that there is no codec info in the reply and the call continues? The value for 18 is defined in some RFC as being G729/8000 so there's no real need to redefine it in the SDP, especially not since the answering party already knows that the initiating party also uses the same value. -- Andreas Sikkema -- _ -- 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] CallerID inconsistently presented through ISDN/cellular networks
On 10/11/11 8:10 PM, Olivier wrote: > I'll start a test session in a couple of minutes and report here. > > The strangest things is this inconsistency: I can imagine million of > reasons why a number is not presented but I can't think of any > explaining why it would change in a couple of hours. Inconsistent configuration over multiple routes probably. I know I have one route (the default actually) to a number of destinations where I am 100% percent able to send redirected number information, but another route just will not pass it on to the destination. So normally calls to these destinations have nice caller id as if A was calling C (at least that's what C sees in their display) but every now and then I flow over to the alternative route and the information is lost, C doesn't see A, but B. Nothing I can do about it, been fighting over it for ages but I just doesn't seem to be able to make it work. -- Andreas Sikkema -- _ -- 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] Cisco AS5400XM
On 10/6/11 11:25 PM, Kyle Sexton wrote: > I'm looking at the Cisco AS5400XM to convert some incoming T1s to SIP > signaling. Has anyone had any experience with these devices? The > feature cards that Cisco sells can be a little confusing. I'm > thinking something like below is what I need. > > (1) AS5400XM, AS5400XM Starter Kit (inc Chassis, MB, Def Mem) > (1) AS54-AC-RPS-PWR, AS5400 AC Redundant Power Supply > (1) AS54-DFC-8CT1, AS5400 OCTAL T1/PRI DFC Card > (2) AS54-DFC-108NP, AS5400 108 Voice/Universal Port Feature Card > > Any thoughts would be appreciated. Thanks. I've used them in the past and still use the little brother (AS5350XM). I have no experience with T1s, but I used them to convert EuroISDN E1s to SIP. They were very stable (I don't think I've ever seen one crash) but can be a pain when you want to set them up. These machines were originally designed as modembanks for internet access so the default config has an interface for every B channel. That is a pain to browse through the configuration. Grouping them solves this. Make sure you understand how to route calls using dialpeers, and make sure you understand this before putting them in service. These are very, very capable machines with lots of useful configuration options. Make sure you buy enough DSP channels to cover all simultaneous calls that need transcoding, we generally bought enough DSP cards so we could transcode all simultaneous calls. If you add it all up we were actually buying more DSP channels than E1 channels were available, for some reason Cisco designed the machine like this, perhaps to cover for slow call teardowns occupying DSPs too long. -- Andreas Sikkema -- _ -- 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] Beep file with Record
On 10/5/11 9:50 AM, Jeroen Eeuwes wrote: > Hi Arjan, > >> I also try to place the voicefile in the directory /var/lib/asterisk/sounds/ >> and /var/lib/asterisk/sounds/applications/ of but without any success. > > Just for double-checking, but what directory is listed as the > astdatadir in asterisk.conf? And if that still doesn't give a clue where Asterisk is looking for beep, monitor the asterisk executable with strace and grep its output for beep and that should point to your problem. You might need a little while to figure strace out but that is the way to be absolutely sure what Asterisk is trying to do. Everything else is just guessing. -- Andreas Sikkema -- _ -- 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] Re : Re : Re : Direct RTP with Asterisk
On 6/20/11 5:19 PM, Lyle Giese wrote: > That's why other free providers don't use SIP phones, but build > their own client software. Real SIP providers fix this for their customers. -- Andreas Sikkema -- _ -- 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] standalone PRI-to-SIP converter
On 5/27/11 6:33 PM, Gordon Henderson wrote: > Personally I'd avoid Patton. No-one has a clue how to configure them. > I've struggled for the past couple of days and have given up and they're > being sent back to be replaced by Mediatrix boxes. Then you're asking the wrong people. It is totally possible to get a Patton to be configured correctly. Since PRI is much easier to configure than a BRI interface (PtP, PtMP?) it shouldn't be that hard. The problem with these very powerful VoIP to ISDN gateways is that they have lots of things to configure, some more intuitive than others. If you're using real hardware, be prepared to spend real time and effort into configuring them. The webinterfaces on Patton or Audiocodes gateways are miles better than the CLI on a Cisco AS5350 or the CLI on an Acme Packet SBC. The bad rep Patton and Audiocodes seem to have is probably related to them using the same software for a simple 2xFXO port gateway as those for 4xISDN BRI or 4x ISDN PRI. -- Andreas Sikkema -- _ -- 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] Remove "name" part of SIP From header
On 5/4/11 7:10 PM, John Hablitzel wrote: > exten => xxx,n,Set(CALLERID(name)=) I'd either leave the name alone or do te following (haven't had the need for removing it): exten => xxx,n,Set(CALLERID(name)="") -- Andreas Sikkema -- _ -- 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] anybody out there sucessfully using gnugk?
On 4/28/11 10:30 PM, Danny Nicholas wrote: > Hi List, > I have a client that wants me to replace their existing H323 > gateway. I am able to get ooh323 and h323 to work fine in a native > environment, but the whole thing goes to heck when I have to cross networks. > Gnugk seems to be the answer to this, but I can't seem to get it to work > right. Any ideas? It's been years since I used GNUGk, but I'd check the mailinglist at http://www.gnugk.org/ The core developers have always been very helpful to me. -- Andreas Sikkema -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to create distortion, echo, and chopping sound in a SIP trunk?
On 4/28/11 5:25 PM, Bruce B wrote: > Is there any easy way to simulate a distorted SIP line temporarily for > testing? Build a Linux based router and use netem/tc to mess around with the routed traffic. You can insert packetloss, jitter, etc and have it be reproducable. -- Andreas Sikkema -- _ -- 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] Asterisk as a Condo door opener/intercom
On 4/12/11 1:21 AM, Don Kelly wrote: > Continuing top posting... > > The same argument could be made for any commercial solution. Why use > Asterisk when we could throw $4,000 at our problem for a commercial > solution? > > I'd like to have a solution that would have the features you suggest for > $400. What part of the system isn't working? The "route calls to the appartment" part? That could be replaced by Asterisk with enough (analogue?) ports to serve the front door and appartments using existing wiring. If the door part also needs replacing because it is proprietary to the old system, you could use a SIP dooropener/intercom, but these are generally expensive, starting around EUR800/$800? or so and probably need an expension for apartment buttons. And you'd need to run new wiring to the door and perhaps change the lock. And then there's the apartments, it could get *very* expensive when you need to replace wiring and the "phones" in each apartment to something VoIP like. -- Andreas Sikkema -- _ -- 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] SIP register and contact header
On 4/4/11 5:13 PM, Jonas Kellens wrote: > I define SIP registrations as follow in sip.conf : > register => number:passwd@sip-server > > example : > register => 33:mypass@ip_sip_server > But apparently the SIP 'contact' header in the SIP REGISTER looks like > this : > /Contact: / Change your register line into this: register => 33:mypass@ip_sip_server/33 -- Andreas Sikkema -- _ -- 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] doorphone?
On 3/9/11 6:35 AM, Tóth Csaba wrote: > could anybody suggest a usable doorphone and magnetic door opener > "hardphone" system for me, please? Of course should be connectable to > asterisk. I am in the EU, should be available here. I don't have direct Asterisk exerience, but when I tested http://robin.nl/en/products/robin-compact-sip/ it worked flawlessly; I don't have a doubt it will work with Asterisk. -- Andreas Sikkema -- _ -- 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] 1.8.3 - IAX - echo - jitterbuffer
On 3/7/11 11:15 PM, sean darcy wrote: > I'm using iaxagent on a Droid X to connect by IAX to 1.8.3 at the > office. 1.8.3 has sip phones. The audio is fine on the Droid X side. On > the office side, they hear an echo of _their_ speech, not mine. We tried an Android SIP client last week and it had *huge* "echo" issues. The Android client operated by a colleague sitting next to me calling a normal SIP phone which I answered. When I started talking I heard myself twice. First the soft direct echo trough the air and then a really loud, very good quality second echo. Muting the microphone on the Android side did not solve the really loud second echo which suggests to me there might be something of a loop in the operating system looping the audio back. The phone is a Sony Ericsson Experia model, no idea what the exact type is. -- Andreas Sikkema -- _ -- 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] Occasional robotic sound while call in progress
On 1/18/11 12:01 AM, Michelle Dupuis wrote: > We have an application that plays a variety of sound files on one leg of > a call (generated by a call file). We've been told that the party > listening to the audio files intermittantly hears "robotic" sounding > audio (on/off during the same call). > > Anyone have ideas on cause? These calls are on an internal network > (lots of network bandwidth), and from a server running 99% idle. Hm I have heard/seen these kind of complaints and in my experience they occur with _very_ low amounts of packet loss. The codec gets confused and can't output the proper audio, just a slightly incorrect version of it. Packet loss like this at the start of a call, which could be caused by some form of NAT traversal via a media proxy where media is only sent both ways when audio has been received from both endpoints, is not unheard of. Network bandwidth is not a very good indicator of the quality of your network Make sure you know if there's packet loss on individual links (managed switches FTW), what the jitter is end to end, etc. -- Andreas Sikkema -- _ -- 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] Sound quality issue
> I am sure there are RTP packets losses somewhere, except RTP debug in > the asterisk CLI, how can i determine where the problem come from ? If it is possible to make a network trace in a Wireshark compatible format, Wireshark can parse all the SIP and RTP messaging and give you lots of statistics, including packet loss, jitter, etc. Check the Wireshark site (http://www.wireshark.org/) for more information. -- Andreas Sikkema -- _ -- 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] Do I need a sip proxy?
Hi, > At least > that is my understanding of NAT. The provider should see me trying to > register from the same IP with multiple different ports (high number > ports; not talking about 5060 as this is outbound and not inbound) and > should be able to differentiate between SIP packets coming from various > servers. However, it seems to not happen. > > There is some sort of clash and only one of the servers shows registered > with the provider and other's trunks go down. I have noticed that > keeping one server works. What I have noticed with consumer grade NAT routers is that they seem to be optimized to only keep track of one single client that is allowed to connect to a server:port tuple on the outside. So if a SIP client on local ip_a and port 5060 on the inside of the router is talking to a server outside of the NAT at ip_s and port 5060 it works fine, but the minute a second client at local IP ip_b and port 5060 starts to talk to ip_s at port 5060 on the outside of the same NAT router all traffic from server_s is sent to ip_b and ip_a will receive nothing. With NAT entry timeouts and regular traffic from ip_a and ip_b you might see only one local client being reachable all the time or connectivity hopping from one to te other at regular intervals. There are NAT implementations that do not have this problem, but that might require a more expensive router or you can configure the SIP clients to all use different local ports. Perhaps this is a result of some sort of SIP ALG or a stupid basic NAT implementation to reduce code complexity on the router, but it is a nuisance either way. -- Andreas Sikkema -- _ -- 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] Is Asterisk ready for Prime-Time?
> Although this is a "users" list, I think it is more of a list > for Asterisk "resellers". I'd be interested in how many of you > are simply using Asterisk as your phone system and NOT selling > your services or an Asterisk based solution? I'm responsible (development, maintenance, support) for an Asterisk based VoIP platform providing a replacement for residential PSTN lines. So I'm technically just a user ;-) I've literally got _thousands_ of users and Asterisk is rock solid for us. -- Andreas Sikkema ___ -- 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] AEL - SQL and TIMEDIFF()
> context testsql {
> s => {
> MYSQL(Connect connid ${DBHOST} ${DBUSER} ${DBPASS} ${DB});
> MYSQL(Query resultid ${connid} SELECT
> TIMEDIFF(callend,callstart) FROM tblCall WHERE id=7);
> MYSQL(fetch fetchid ${resultid} temp);
> MYSQL(Disconnect ${connid});
> }
> }
>
>
> The error I'm getting is below:
> [Mar 6 08:59:35] WARNING[27116]: app_addon_sql_mysql.c:268
> aMYSQL_query: aMYSQL_query: mysql_query failed. Error: You
> have an error in your SQL syntax; check the manual that
> corresponds to your MySQL server version for the right syntax
> to use near ') FROM tblCall WHERE id=7' at line 1
I think the solution would be to escape the , with a backslash, so
your query would look like this:
SELECT TIMEDIFF(callend\,callstart) FROM tblCall WHERE id=7
Maybe even the brackets ()
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Re: [asterisk-users] Multiple contacts.
> I'm going to jump in here without reading everything... Yeah, me too ;-) > You > said you can log into your email. Yes, you can. You can > also login with a softphone on that same system at an > Internet Cafe, as long as you remember your login credentials > for your SIP device. As Steve said, the LAST device to > register will ring. If Asterisk was not used for registration purposes, but only for diallingplan "tricks" and leaving the registration parts to a SIP registrar/proxy like OpenSER. Then registering multiple contacts for the same username is, IIRC, more or less default behaviour. -- Andreas Sikkema ___ --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] Is there real benefits on a SMP machine forAsterisk?
> > The mistake people often seem to make is to assume that
> > loadavg == cpu usage.
>
> It is a good indication. Even a better indicaton to the ammount of
> threads ("processes") starved for CPU time.
On a quad core Linux machine it is possible to have a totally
unusable machine with a loadavg of 4 or a perfectly usable
machine with loadavg of 10.
If you look at loadavg alone you never get the "full" picture.
In the first case the CPU load on all cores is 100% while the
CPU load in case number two could be something around 25% on
each, but they are all doing very I/O intensive stuff.
I frequently see machines doing loads of over 4 with total CPU
load not above 100% (of 400% possible)
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Re: [asterisk-users] IVR and MySQL
> > One way to do it is working with app MYSQL(), where you > will put your sql as > > argumment. > > read more in > > http://www.voip-info.org/wiki/view/Asterisk+cmd+MYSQL > > That's possible, but i wouldn't recommend on large production system. > Using MySQL you would need to connect and disconnect all the time, and > it takes resources.. I would suggest to append that info to CDR > userfield (if you are storing your CDR in MySQL), and run periodically > some script that extracts them. Of course it's more complex, but that > would be my way. Using MYSQL() (or equivalent) heavily in the dialingplan is IMHO the nicest way of doing things like this. You can do lots of simultaneous calls before getting into trouble. Appending stuff to the CDR userfield is just plain ugly and asking for trouble (are you sure you can always separate the different values?). -- Andreas Sikkema ___ --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] Large dial plans and variables
> You're so right! > > I thought about having just a catchall _. extension in the > dialplan and doing everything else in a "real" language via AGI - > PHP, Perl, ... whichever you like. It would make the programming > part much easier as the scope of variables is just as you > expect it to be. Well, they're called macro's for a reason You guys are proposing adding functions or procedures. My first step in any macro would be to copy incoming variables, be it arguments or even asterisk defined stuff to local variables. But that is just me and my coding convention. -- Andreas Sikkema ___ --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] CDR changes in 1.4.3?
> I will, in the coming days, look at some of the extraneous CDR's that > are generated, and see what I can do to get rid of them. It's > not always > that simple. > If we ring a phone, for instance, and no-one answers it, is > that truly, > really, something that no-one will ever be, could ever be, interested > in? (just a fer-instance). I'd prefer more CDRs where I can decide if they are extraneous than that some "random" developer gets to decide what's interesting or not ;-) There are several other things I would like to see from CDRs... - a way to get the sip callid it's own column, instead of appending it to the userfield column (maybe as a "channel-callid" or something like that so other channels can use it as well?) - when a call takes longer than x seconds/minutes/hours a "yes, this call is still active" CDR - and maybe start/stop (a "continuation", see above) cdrs, instead of just one... -- Andreas Sikkema ___ --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] Calls in ulaw, not gsm as desired
> However, even once I reloaded the extensions, its still only > using ulaw. You didn't reload the sip config? Maybe that's your problem? -- Andreas Sikkema ___ --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] MySQL query from extensions?
> I also dropped the quotes on the dnis=${IVR-Exten}.
That's only allowed if the dnis column doesn't contain a string.
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RE: [asterisk-users] Dell Servers
> I bought a Dell 2850 as a pbx server and it just sucks IMHO > > The stupid thing has only 3 pci slots and even with only 3 > pci slots Dell > managed to have a shared irq on every slot, 1 for the scsi > controller and > one for each nic We're using a couple of Dell 1850's and I couldn't be happier with them. But then I don't use any Digium, Sangoma or other cards. We're running 100% VoIP through them. -- Andreas Sikkema ___ --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] Match a Numer - then continue with dialplan
If you have no statuc stuff in your dialplan, how do you use the 'include =>' statement? We don't have users... we have companies. It's a hosted IPT service... and to make the problem even more insane, each company has multiple levels of organisational structure. Hardly, you're not required to use it ;-) You have heard of the s? I think in our hundreds of lines of dialplan, I think we have at most 10 lines that match extensions, the rest is all handled by s. What we have a lot is the following in extensions.conf include _.,1,Goto(example-outgoing,s,1) And then in example.conf: [example-outgoing] exten => s,1,DoSomeStuff exten => s,2,SetVar(outgoingCallerID=VALUE) exten => s,3,Goto(example_real_outgoing,s,1) [example_real_outgoing] exten => s,1,DoSomeMoreStuff exten => s,2,*SET outgoign caller id* exten => s,3,Dial() ___ --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] Match a Numer - then continue with dialplan
> Bzzt. In order to call SetVar, I have to match the extension > dialled. When that happens, there is NO WAY to continue > searching the dialplan after that point for another extension > to match. You can't use a generic extension and search a database table for $EXTEN <-> callerid relation and then set it? Your diallingplan is _so_ different to what we do, yet what you want to do is pretty much the same to what we do all the time. But our Asterisk boxes have _no_ sip CPE's registered to them and our diallingplan is littered with database lookups. We have no static stuff in our dialingplan. And we have quite a number of users. But no queues etc. -- Andreas SikkemaBBeyond Software EngineerPlaneetbaan 4 +31 (0)23 70743422132 HZ Hoofddorp ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Match a Numer - then continue with dialplan
> [snip]
> > >
> > > [coo1_CallStart]
> > > include => coo1_OnNet
> > > include => syst_OnNet
> > > include => syst_OffNet
> >
> > Instead of including your system-wide logic for offnet calling,
> > introduce a per-company offnet and include that instead:
> >
> > [coo1_CallStart]
> > include => coo1_OnNet
> > include => syst_OnNet
> > include => coo1_OffNet
> >
> > [coo1_OffNet]
> >
> > exten => _X.,1,Set(CALLERID(NUM)=3254000)
> > exten => _X.,2,Set(CALLERID(NUM)=Widgets Inc.)
> > exten => _X.,3,Goto(syst_OffNet,${EXTEN},1)
>
> Bradley, If I do this, then I can no longer continue with
> further extensions in my dialplan as Asterisk has already
> matched a number. I still need to check black/white lists,
> set pic codes and rate centers, 4 digit extensions etc within
> the company context. I just need to set the caller id and
> then move on. If I goto over to ${EXTEN} within syst_OffNet,
> I'd have to put ALL this logic within that extension, which
> would mean potentiall several hundred priorities. Asterisk
> really does need a way to match a number, execute some code,
> and then go back to looking for extensions.
Why not do something like this (in pseudo dialplan):
SetVar(outgoing_callerid=1234567)
Set(CALLERID(NUM)=${outgoing_callerid})
Dial(outgoing destination)
This will not screw up your extesnions matching, but you will
need to check that outgoing_callerid has been filled before setting
callerid (or make sure it is always filled with something sensible).
Check the variables page in the wiki on exact syntax ;-)
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RE: [asterisk-users] Glitches in sound every time that Asteriskreceives reINVITEs
> My Asterisk server is working fine, although every time that > in the middle of > any call there is a reinvite, the user hears a glitch. Why is > this happening? > How can I solve this problem? That's because a REINVITE is generally used to change from one codec to another. For some reason this involves stopping the existing audio, waiting a little while and then starting a new audio stream. So far this one of the reasons why I don't like reinvite... -- Andreas SikkemaBBeyond Software EngineerPlaneetbaan 4 +31 (0)23 70743422132 HZ Hoofddorp ___ --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] Why does it take at least 4 flipping days before asterisk tries to resolve a provider?
Remco, > Asterisk starts before the internet connection is up and dns > is working. > And then people say nightly asterisk restarts are not a good idea Why is your asterisk startup script running before networking has been setup? Asterisk has the same networking dependencies as apache, so I start it around the same time using the same priority as apache and as far as I know networking should work at that time or not at all, not somewhere in between. pebkac? -- Andreas SikkemaBBeyond Software EngineerPlaneetbaan 4 +31 (0)23 70743422132 HZ Hoofddorp ___ --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] E164 caller ID
> Question is, most of the caller IDs I see with examples > online are Bellcore > style caller IDs (only useful in the US and a couple of other > places) which > are 10-digit. > > My question is, will the standard set(CALLERID(num)=BLAH) work with > non-Bellcore caller ID strings? Is it expected to be able to > also handle E164 > numbers (which can be up to 15 digits) as well, or is there > another method for > that? Sure, no problem. As another reply said, it's just a number. -- Andreas SikkemaBBeyond Software EngineerPlaneetbaan 4 +31 (0)23 70743422132 HZ Hoofddorp ___ --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] E61
Simon, > That is incorrect. It works just fine through NAT providing: > > - The server is proxying RTP as it has no support for STUN etc. > - The NAT is the basic domestic router style, not a full > blown firewall requiring port mappings Strange, then you must have some other firmware, because I just can't get it registered at all, let alone make calls. We do have proxies for RTP ;-) -- Andreas SikkemaBBeyond Software EngineerPlaneetbaan 4 +31 (0)23 70743422132 HZ Hoofddorp ___ --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] E61
> Anyone here use the Nokia E61 ? I am looking to invest in a > wifi phone and I want to get the best. Is it good as far as > reception ? That is of most importance to me. Thanks. I've tried it in the last couple of days. The biggest issue for me ist that it HAS to be on the same side of a NAT as the server it talks to (asterisk, ser, etc). If it is on the private side of a NAT and the server is on the public side, it doesn't work. I've read something on the Nokia forums that Nokia is aware of the problem and it will be solved. My problem is that they want to solve this using STUN etc, while I would prefer they also wouldn't have the software care if it is on the inside of a NAT like most other CPE's so our platform can take care of things. -- Andreas SikkemaBBeyond Software EngineerPlaneetbaan 4 +31 (0)23 70743422132 HZ Hoofddorp ___ --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] billed calls when cellullar phone is unreachable
> When we route a call to an unreachable cellular phone we know
> it cause
> we get a particular ${HANGUPCAUSE} so we don't bill that call even if
> billsec is > 0 (the duration of the "cellular is unreachable bla bla"
> message), but the customer says their system too records the
> call as >
> 0 and their expected behaviour is to have the call recorded
> as duration
> == 0.
> (I'm supposing the customer is noticing the cellphones cause
> of the high
> traffic, but probably this happens also with other kind of "service
> messages" which aren't to be billed, have to try)
I would expect these kind of messages to be played without
"answering" the call. IIRC on ISDN this is done using a
FACILITY message with a "start audio" IE included (I'm not
that much into ISDN to be more precise).
On H.323 this is called "early media" and in SIP this would be
signalled using a 186 Session Progress message.
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RE: [Asterisk-Users] SIPCALLID, but which callid?
> Andreas Sikkema wrote: > > > > Hi, > > > > To combine two sources of CDR's I want Asterisk to save the > SIP callid for > > all calls. I know there's a variable that contains the SIP > CallID value, > > but is this the callid value of the incoming INVITE message or the > > outgoing > > message? Are they the same? (I've not yet checked a trace, > I'm sorry for > > that). I've tried to read chan_sip, but couldn't find > something in the > > time > > I had today. I've found hardly any documentation o this variable, > > apart from > > that it exists and that it contains "the" SIP CallID value. > > > > Can anyone enlighten me? > They are the same on both sides. Is this new behaviour? I've got an asterisk 1.2.5 installation that does not use the same CallID on both the incoming and outgoing side of a call through our Asterisk machine. -- Andreas ___ --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] SIPCALLID, but which callid?
Title: SIPCALLID, but which callid? Hi, To combine two sources of CDR's I want Asterisk to save the SIP callid for all calls. I know there's a variable that contains the SIP CallID value, but is this the callid value of the incoming INVITE message or the outgoing message? Are they the same? (I've not yet checked a trace, I'm sorry for that). I've tried to read chan_sip, but couldn't find something in the time I had today. I've found hardly any documentation o this variable, apart from that it exists and that it contains "the" SIP CallID value. Can anyone enlighten me? -- Andreas Sikkema ___ --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] Camp on?
> I believe what you refer to is called "Ring Back When Free" > at least thats how I know it in the UK. Ah yes, no I remember. We called it "Automatic Ring Back". So we had "normal ARB", or "ARB on next use". -- Andreas Sikkema BBned NV Software EngineerPlaneetbaan 4 +31 (0)23 70743422132 HZ Hoofddorp ___ --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] Camp on?
> I believe this is called camp on. Found some examples on voip-info.org > but they assume that you do not hangup the originating phone. Anyone > have an idea how to implement this feature as described above? When I worked at Philips there were two variants: - camp on busy - camp on no answer The second one is tricky; after the destination number has been used again, the switch will dial the originator and then the destination and connect the two legs. -- Andreas Sikkema BBned NV Software EngineerPlaneetbaan 4 +31 (0)23 70743422132 HZ Hoofddorp ___ --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] Polling Asterisk for Life
> So, simply respawning asterisk, or checking to see if it's running > isn't good enough, because asterisk is indeed running. We need to > access asterisk and issue a command, and see if asterisk responds > appropriately. If not, we can assume it has died, and we can kill it > off (killall -9 asterisk) and then start it back up again (or reboot > the whole server if necessary). The _only_ way to reliably (well, in as much as that is possible) to test if your Asterisk is working, is to build a monitoring system that does more or less the same as a typical user would do. We have a system with two modems connected to ATA's and they dial each other via multiple routes so we test all of the major scenarios. We only test if calls are routed through, not if the call itself establishes (media running) to prevent major costs from such a system. I works reasonably well, it seems to detect 99% of the major problems. -- Andreas Sikkema BBned NV Software EngineerPlaneetbaan 4 +31 (0)23 70743422132 HZ Hoofddorp ___ --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] sniffing sip password/uri/host info
> Ethereal would probably be a batter analyzer. Not sure how well it > seppurts sip, though. Unlike tcpdump it won't work on-the-fly. But you > can also get tcpdump to dump raw data and analyze it off-line with > ethereal. Ethereal can also show SIP traffic on-the-fly! "update list of packets in real time" and "automatic scrolling in live capture" A "sip" display filter is needed so you only see SIP traffic, a sip capture filter might be needed for very busy networks -- Andreas Sikkema BBned NV Software EngineerPlaneetbaan 4 +31 (0)23 70743422132 HZ Hoofddorp ___ --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] Fwd: Asterisk on Solaris 10 (AMD Opteron, Sun Fire X2100)
> While it doesn't explicity say so, it seems to > very strongly imply that either a PCI card or > ztdummy are *required* for some Asterisk > functionality (namely music-on-hold and > conferencing, apparently). Is this actually not the case? I'd say support for one of these options should be available whenever Asterisk generates _any_ media by itself, including conferencing. IVR functionality and the like become much better when ztdummy or another timing source supported by Asterisk is available. -- Andreas Sikkema BBned NV Software EngineerPlaneetbaan 4 +31 (0)23 70743422132 HZ Hoofddorp ___ --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] DTMF Simultaneous Inband and RFC2833 performedby Asterisk => Duplicate tones
> I have seen the following effect in Asterisk, though: where > it converts > an inband DTMF (eg coming off a Zap channel) into an > indication, it mutes > the audio where that tone is. But sometimes it leaves a > teeny bit of the > tone behind. > > If you take such a call over say IAX to somewhere and then > back out a Zap > channel, you end up with the teeny remaining bit of the > original tone, > PLUS the regenerated tone. > > If you are very unlucky a remote DTMF receiver can hear two digits. The same thing can happen when a SIP "ATA" is configured to use rfc2833 but is also a little to lote with the filtering out of the DTMF. So sometimes it's not Asterisks fault at all ;-) And then there's some IVR's that don't notice it at all, while others are totally unusable. -- Andreas Sikkema BBned NV Software EngineerPlaneetbaan 4 +31 (0)23 70743422132 HZ Hoofddorp ___ --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] Nested MySQL Commands
> Is it possible to have nested MySQL queries in extensions.conf? > > Ie, perform a query, grab a value, and then jump to another > location in the dialplan and do another query based on that > original value. I'm having problems with the result and > fetchid's and I'm not sure if it's even possible to do this or not. Just make sure that you use different variable names for each query if the values should stay available after the next query. What we tend to do is grab the data from the database and the stuff that should stay around for a longer time is assigned to a new and appropriately named variable. So the original variable can be used again. We've got loads of queries in our extensions.conf. -- Andreas Sikkema BBned NV Software EngineerPlaneetbaan 4 +31 (0)23 70743422132 HZ Hoofddorp ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk MySQL CDR - MySQL starting too late
> Well that didn't work. When I rebooted MySQL didn't start at all The level doesn't set _when_ something starts, just _if_ something starts. Some daemons should start in single user mode, some not. Some others should only start when in GUI mode, others not, etc. This is what level controls. When something starts is usually controlled with the name of the start/stop scripts in /etc/rcx.d/ (or something like that). Files starting with the S00 prefix are started first, files with S99 are started last for that runlevel. The same for K00 and K99, but that describes the time when processes are killed. So if Asterisk is started using S80asterisk, and MySQL using S50mysqld, then it obviously isn't going to work as intended. The same also when both are started with S99, because asterisk will be started before mysqld... I usually mess around with the numbers, but that is not very reproducable, dependencies listed in the rpm file (or equivalent) usually takes care of this. When isntalling from source, you're on your own. -- Andreas Sikkema BBned NV Software EngineerPlaneetbaan 4 +31 (0)23 70743422132 HZ Hoofddorp ___ --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] "open" asterisk?
> There should be other "voices" worth while... > > Give other people the chance > > The market is growing... > > Be open :) I'd _love_ a different voice for the default distribution. To my (European) ears Allison is practically incomprehensible. -- Andreas Sikkema BBned NV Software EngineerPlaneetbaan 4 +31 (0)23 70743422132 HZ Hoofddorp ___ --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] How to check how many G729 codec licenseinstalled
> On Mon, 2005-11-14 at 07:46 -0500, Sean Cook wrote: > > Easy: > > > show g729 > > > > This will show total in use and total available channels for g729 > > doesnt work for me, maybe its a version difference. > > I do have g729 loaded, and that was verified. Do you have the Digium G.729 codec installed? This one provides "show g729 " I have no idea if the IPP hack provides a similar interface. -- Andreas Sikkema BBned NV Software EngineerPlaneetbaan 4 +31 (0)23 70743422132 HZ Hoofddorp ___ --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] Don't call
[EMAIL PROTECTED] wrote: > I receive a call, but don't call... > Asterisk show this message. > Are codecs the problem? > > Sep 30 11:25:54 WARNING[4475]: chan_sip.c:1899 > create_addr: No such host: sip.uni.it,r If you pasted this directly from Asterisk, then there's an error in your configuration somewhere. Host names cannot contain , characters. -- Andreas Sikkema bbned NV Van Vollenhovenstraat 33016 BE Rotterdam t: +31 (0)10 2245544 f: +31 (0)10 413 65 45 ___ --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] Removing "-" (Dash) from Dialed Numbers
[EMAIL PROTECTED] wrote:
>> I am trying to enable dial-by-email by using LDAPget to query
>> an Active Directory server. I've got it retrieving the phone
>> number fine. Unforunately, the numbers stored in active
>> directory are either in the format: (xxx) xxx- or
>> xxx-xxx-.
> Is there
>> any way to parse characters out of the dialed phone number so
>> that I only end up with digits (remove spaces, parenthesis and
>> dashes)? From there, my outbound routes can take care of
>> where to send the call.
> This would be darned easy to do with the AGI and a perl script.
>
> IE:
>
> exten => _X.,1,agi,fixnumbers|${MyNumber}
> exten => _X.,2,Dial(ZAP/g0/1${MyNumber})
>
> Then, in a perl script called "fixnumbers" and inside the agi-bin
> directory:
>
> ## START CODE #
> #!/usr/bin/perl -w
> use strict;
> use Asterisk::AGI;
> $AGI = new Asterisk::AGI;
> my %input = $AGI->ReadParse();
>
> my $number=$ARGV[0];
> $number=~s/-//g;
> $number=~s/ //g;
> $number=~s/\(//g;
> $number=~s/\)//g;
>
> print $AGI->set_variable('MyNumber',"$number");
>
> exit;
>
> ### END CODE
Depending on how many calls per second you want to perform,
some dialplan magic might be cheaper than starting up a
perl process.
I'd write a diaplan macro for this. If the numbers are in a
fixed format (4th character is a -, 7th character is a -,
etc), then it's really simple.
Something like this:
exten => s,1,SetVar(strPart1 = ${myNumber:0:3}
exten => s,2,SetVar(strPart2 = ${myNumber:4:3}
exten => s,3,SetVar(strPart3 = ${myNumber:7:3}
exten => s,4,SetVar(myNumber = $strPart1$strPart2$strPart3
But I'm using quite an old Asterisk, so current syntax might
be a little different, but the Wiki suggests this still works.
--
Andreas Sikkema bbned NV
Van Vollenhovenstraat 33016 BE Rotterdam
t: +31 (0)10 2245544 f: +31 (0)10 413 65 45
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RE: [Asterisk-Users] Early Media with Asterisk
[EMAIL PROTECTED] wrote: > Now, I traced RTP packets and see how sip2.provider1.de sends > packets to my Asterisk but the port seems closed on my server so the > inquiring server of > provider1 will never get an answer and sends a "port unreachable". Did provider1 send the exact same SIP message types to you as provider2? It looks to me like provider1 is not sending a 183 Session Progress message. Which is usually used for this kind of functionality I think. -- Andreas Sikkema bbned NV Van Vollenhovenstraat 33016 BE Rotterdam t: +31 (0)10 2245544 f: +31 (0)10 413 65 45 ___ --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] Storing extension prefs. in MySQL
[EMAIL PROTECTED] wrote: > I would like to store these seetings in a mysql database, so > that they are more easily accessible from a user > configuration page on a webserver. Since these settings need > to be checked in the dialplan for each call to the extension, > it seems a bit to much to have to connect, query and > disconnect from mysql every time. Is there any way to keep a > persistent connection to mysql that can be queried from the > dialplan? Well, if you do this before answering, nobody is going to notice. Even querying during an answered call will have hardly any outside consequences... -- Andreas Sikkema bbned NV Van Vollenhovenstraat 33016 BE Rotterdam t: +31 (0)10 2245544 f: +31 (0)10 413 65 45 ___ --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] Huge Echo
[EMAIL PROTECTED] wrote: > In the following setup: > call coming from a pstn line -> into FXO card -> asterisk -> SIP > phone > > i get an incredible loud echo in the SIP phone (about 0,5-1s) > (everything i speak into SIP phone microphone i hear in its > speaker). The person calling from PSTN is not getting any echo. Make sure you're not playing the recorded sound from your microphone back to your loudspeakers. -- Andreas Sikkema bbned NV Van Vollenhovenstraat 33016 BE Rotterdam t: +31 (0)10 2245544 f: +31 (0)10 413 65 45 ___ --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] DTMF and "breaking through" voice prompts
Sherwood McGowan wrote: > Has anyone else had problems with users being able to press key > tones during a voice prompt? I have a few users complaining that > some systems will not recognize key presses during them. You are using Backgroudn() to play the prompts? -- Andreas Sikkema bbned NV Van Vollenhovenstraat 33016 BE Rotterdam t: +31 (0)10 2245544 f: +31 (0)10 413 65 45 ___ --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] How to use * and # as part of numberindialcommand
[EMAIL PROTECTED] wrote: > What is CFU and CFNR? Call forwarding unconditional call forward not reachable -- Andreas Sikkema bbned NV Van Vollenhovenstraat 33016 BE Rotterdam t: +31 (0)10 2245544 f: +31 (0)10 413 65 45 ___ --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] Dell 2850 anyone ...
[EMAIL PROTECTED] wrote: > I just setup a Dell 1800, not a 2850, which is working > awesome.. only had to > disable USB, which realistically no-one on a phone system > would care about > anyways. Oh, really? Only if you're running a 2.6 kernel or using a zaptel card you don't need it. -- Andreas Sikkema bbned NV Van Vollenhovenstraat 33016 BE Rotterdam t: +31 (0)10 2245544 f: +31 (0)10 413 65 45 ___ --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] DECT gateways
[EMAIL PROTECTED] wrote: > Nice looking device. > Does it support DECT repeaters? > I cannot rely on 1 basestation for my handset when I walk > around in the location. The Siemens stuff has 10 > antennas/repeaters/extenders in the building. It's a bit > overkill, but the Siemens guys tend to love doing it BIG. > I think 3 to 6 repeaters will be way enough for the cases we have > open now. Buildings do strange things with radio. What might seem like overkill could be mandated by liftcages, firestairs etc. Also if you want to add dect coverage in a busy area where lots of people with dect handsets gather (meeting rooms, canteens) you need lots of basestations. I've worked in a building where there seemed to be an overkill of basestations every hallway had 3 or 4, (every 20 meters or so) and still there were areas with insufficient coverage... -- Andreas Sikkema bbned NV Van Vollenhovenstraat 33016 BE Rotterdam t: +31 (0)10 2245544 f: +31 (0)10 413 65 45 ___ 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] g729 recording on asterisk using g729 enabledphone
[EMAIL PROTECTED] wrote: > i have installed asterisk on my system and using only g729 > enabled phones. > from what i understand, we would not be needing any g729 > licenses as all my > voicemail prompts are also in g729 and asterisk is not doing any > transcoding. when i use the voicemail function to record, the > message is not recorded (0 byte file is created) and it gives the > following errors - > > "unable to convert from g729 to slin" You can force Record to record to G.729, but I'm not sure the voicemail application has the same possibility. -- Andreas Sikkema bbned NV Van Vollenhovenstraat 33016 BE Rotterdam t: +31 (0)10 2245544 f: +31 (0)10 413 65 45 ___ 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] TFTP Secondary Ports
Chad Brown wrote: > I'm publishing tftp through my firewall to support external Cisco > 7960 sip phones. I hope the files requested by the Cisco phones don't contain username / password information. Passing that in cleartext is just so wrong ;-) -- Andreas Sikkema bbned NV Van Vollenhovenstraat 33016 BE Rotterdam t: +31 (0)10 2245544 f: +31 (0)10 413 65 45 ___ 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] Zaptel rpm spec file with udev support
Hi, Has anyone written a SPEC file for zaptel, with kernel 2.6 and udev support? I can find some spec files here and there, but from what I can see they're all kernel 2.4 / non udev... -- Andreas Sikkema bbned NV Van Vollenhovenstraat 33016 BE Rotterdam t: +31 (0)10 2245544 f: +31 (0)10 413 65 45 ___ 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] Dell Hardware
[EMAIL PROTECTED] wrote: > And it's a great shame Digium hardware has such problems on > Dell kit, since > there's so much of it about :( If you don't use digium hardware, there's of course no problems with using Dell. -- Andreas Sikkema bbned NV Van Vollenhovenstraat 33016 BE Rotterdam t: +31 (0)10 2245544 f: +31 (0)10 413 65 45 ___ 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] Re: Braodvoice - UK Non Geographic Numbers
[EMAIL PROTECTED] wrote: > http://www.serviceview.bt.com/list/current/docs/Call_Charges.boo/00165.htm > Of course these are BT retail rates but I fully expect wholesale > rates based on call prefix will be available for carriers / ITSP In some countries there's a company (companies?) providing access to a database which telcos can use to find the rates on this kind of numbers. -- Andreas Sikkema bbned NV Van Vollenhovenstraat 33016 BE Rotterdam t: +31 (0)10 2245544 f: +31 (0)10 413 65 45 ___ 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] Asterisk with Intel Blade Machine...
[EMAIL PROTECTED] wrote: >> We've had Asterisk running on a blade for some time. Blades as >> such can be used but with a couple of restrictions: >> - There's probably no room for PCI cards, so no zap hardware >> - Check the kind of USB supported on the board (UHCI vs OHCI, >> for ztdummy support, see wiki) > > If you have no zaptel hardware and must rely on software you should > use kernel 2.6's ztdummy, don't you? It is better, and also does > not rely on USB. Yes, true, but this entirely depends on how the blade is set up. We had no control over the distro installed on the blade. -- Andreas SikkemaRits tele.com Van Vollenhovenstraat 33016 BE Rotterdam t: +31 (0)10 2245544 f: +31 (0)10 413 65 45 ___ 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] Asterisk with Intel Blade Machine...
Nahid Hossain wrote: > I would like to use Intel Blade machine for running Asterisk. Is > there anyone who already use Intel Blade server for running > Asterisk? Can you please explain, how perform Asterisk with Intel > Blade machine? We've had Asterisk running on a blade for some time. Blades as such can be used but with a couple of restrictions: - There's probably no room for PCI cards, so no zap hardware - Check the kind of USB supported on the board (UHCI vs OHCI, for ztdummy support, see wiki) For some reasons we've moved away (non blade related) from the blades, but not because blades don't work. We really liked them. -- Andreas SikkemaRits tele.com Van Vollenhovenstraat 33016 BE Rotterdam t: +31 (0)10 2245544 f: +31 (0)10 413 65 45 ___ 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] SER and Asterisk question
[EMAIL PROTECTED] wrote: > Actually what happens is that from SER debug I can see the call is > looping between Asterisk and SER. but adding a number makes no > loops. Check what the origin (IP/DNS name) of the incoming SIP message is. If it's from asterisk, send it to the user, if it is not from asterisk, it must be meant to go to asterisk. Add a couple of other tests (known user, etc) to it and then I think you'll have what you're looking for. -- Andreas SikkemaRits tele.com Van Vollenhovenstraat 33016 BE Rotterdam t: +31 (0)10 2245544 f: +31 (0)10 413 65 45 ___ 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] Ztdummy usage
[EMAIL PROTECTED] wrote: > On Tue, May 31, 2005 at 12:35:32PM +0100, Gentian Bajraktari wrote: > >> Then try to 'modprobe zaptel' and then 'modprobe ztdummy' > > 'modprobe ztdummy' should load zaptel as well. I've seen this faul, when only modprobe zaptel first would help. (Debian sarge) -- Andreas SikkemaRits tele.com Van Vollenhovenstraat 33016 BE Rotterdam t: +31 (0)10 2245544 f: +31 (0)10 413 65 45 ___ 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] How to connect to IPTEL.ORG
[EMAIL PROTECTED] wrote: > Hi, how I can connect Astrisk to my iptel account??? > [iptel.org] > type=friend > host=iptel.org > fromuser=my_account_name > secret= > nat=yes We've had problems as well when the friend in sip.conf was named the same as the domain. Call it something else (iptel-out?) and maybe that solves your problem. -- Andreas SikkemaRits tele.com Van Vollenhovenstraat 33016 BE Rotterdam t: +31 (0)10 2245544 f: +31 (0)10 413 65 45 ___ 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] DEBUG output on sip extensions
[EMAIL PROTECTED] wrote: > Marty Mastera wrote: >> May 17 10:50:23 DEBUG[2030]: Didn't get a frame from channel: >> SIP/105-1ae4 May 17 10:50:23 DEBUG[2030]: Bridge stops bridging >> channels SIP/105-1ae4 and SIP/outbound-7dc3 > I am noticing these in my logs also. I looks like it is the result > of the person hanging up, but I have had a few comlaints of > dropped calls > the last few days. These messages also appear at the times of the > dropped calls. I have been watching CPU usage and it doesn't look > like my machine was really loaded or anything. I see these with every single call. I (naturally I'd say) also have reports of dropped calls, but have never been able to relate them to these messages. The messages happen much more often. -- Andreas SikkemaRits tele.com Van Vollenhovenstraat 33016 BE Rotterdam t: +31 (0)10 2245544 f: +31 (0)10 413 65 45 ___ 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] Philips - [QSIG] - Alcatel - [H323] - Asterisk -[SIP] - Users.
DT wrote: > Firstly we have to connect our Asterisk system to a Philips PBX > throught QSIG protocol (interfaces S0), but we doesn't find any > documentation about the support of QSIG and S0 interfaces by > Asterisk. > > [PSTN/ISDN] <---> Philips <-[QSIG over S0]-> Asterisk <-[SIP]-> > Final users. > > Is it possible? > does Asterisk support QSIG and S0 interfaces? As far as I know, Asterisk doesn't support QSIG. Do you _have to_ use QSIG? I'd just use a PRI interface (DTU-PH IIRC) to connect to Asterisk with a sutable PCI card in the server. -- Andreas SikkemaRits tele.com Van Vollenhovenstraat 33016 BE Rotterdam t: +31 (0)10 2245544 f: +31 (0)10 413 65 45 ___ 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] i like my colors, thanks..
[EMAIL PROTECTED] wrote: > On Thu, 2005-04-21 at 08:39 -0500, Matthew Boehm wrote: >> Using most recent CVS-HEAD and my terminal keeps changing colors. >> >> I'm using vt100 terminal emulation. How can I turn off asterisk's >> colors? Or at least turn off the black background. My normal >> terminal is white background, black font. But for some reason, >> asterisk is changing it to white font, black background. > > Add '-n' to your command line. 'asterisk -h' will print out a list > of all of the command line switches that it supports. On the (admittedly relatively old version we're using) this will only work when I'm logging in via SSH. When working from the console -n doesn't work. -- Andreas SikkemaRits tele.com Van Vollenhovenstraat 33016 BE Rotterdam t: +31 (0)10 2245544 f: +31 (0)10 413 65 45 ___ 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] Zaptel based timing for VoIP-only Asterisk
Hi, In a VoIP only environment, Asterisk has to use ztdummy to have any chance of playing back understandable audio files (without drops, hickups etc). I have been using ztdummy to some degree of success, but I also have a "Wildcard TDM400P REV E/F Board 1" in the Asterisk machine I'm using. I'm not using this card for anything at all, but I'm wondering how to set it up for timing only. What do I have to do (I have no experience at all with zap channels and the zaptel.conf file)? -- Andreas SikkemaRits tele.com Van Vollenhovenstraat 33016 BE Rotterdam t: +31 (0)10 2245544 f: +31 (0)10 413 65 45 ___ 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] ser+asterisk - security
[EMAIL PROTECTED] wrote: > it's impossible to use iptables due to the reason that audio > flows through asterisk and users won't be able to communicate w/ *... I was thinking of just the SIP port. I am assuming that asterisk protects its RTP ports from processing traffic from a third party. -- Andreas SikkemaRits tele.com Van Vollenhovenstraat 33016 BE Rotterdam t: +31 (0)10 2245544 f: +31 (0)10 413 65 45 ___ 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] ser+asterisk - security
Pavel Siderov - Hostmates wrote: > I can restrict forwarding calls from another sip > proxy to ser (using proxy_authorize) but how can I restrict access > to asterisk ... Now everyone can forward calls to my asterisk and > can place pstn calls. Use iptables on the asterisk machine to only allow SIP traffic from the machine with SER? -- Andreas SikkemaRits tele.com Van Vollenhovenstraat 33016 BE Rotterdam t: +31 (0)10 2245544 f: +31 (0)10 413 65 45 ___ 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] Zombie or soft hangup
Hi, What does this line of output mean? Bridge stops because we're zombie or need a soft hangup: I'm seeing this sometimes... I've looked in channel.c, but the code is not much more revealing than the debug line... -- Andreas SikkemaRits tele.com Van Vollenhovenstraat 33016 BE Rotterdam t: +31 (0)10 2245544 f: +31 (0)10 413 65 45 ___ 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] colinux fresh install, zaptel does not compile, size_t error
[EMAIL PROTECTED] wrote: > I followed the instructions on > http://www.asterisk.org/index.php?menu=download. > I picked the latest version using CVS. > Things went fine until I cd zaptel ; make clean ; make install. > > I then get an error when compiling zaptel.c > /usr/src/linux/include/linux/kernel.h:75: error: parse error > before "size_t" > > This happens very early on and I suspect that it is actually an > issue with the kernel include files on my machine. > > Nota: I am installing on a colinux debian. > uname -a > Linux colinux2 2.4.26-co-0.6.1 #1 Sat May 29 15:30:37 IDT 2004 i686 > GNU/Linux On http://www.ramdyne.nl/ you can find an article on how I got rid of the same problems you were having (on a Debian sarge install). Unfortunately the server is down for the next couple of hours... Here's a link to the google cache copy: http://66.102.9.104/search?q=cache:pR1IMCaiRcQJ:www.ramdyne.nl/index.php%3Fcat%3D11+%2Basterisk+%2Bramdyne+%2Bdebian&hl=nl&start=1 -- Andreas SikkemaRits tele.com Van Vollenhovenstraat 33016 BE Rotterdam t: +31 (0)10 2245544 f: +31 (0)10 413 65 45 ___ 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] chan_sip not 100% RFC3665 compliant - re-REGISTERsfail.
[EMAIL PROTECTED] wrote: > Is there anyone else with the same problem? Yes, we've seen the same problem. We have found a work around, but I'm unable to to look into it today. -- Andreas SikkemaRits tele.com Van Vollenhovenstraat 33016 BE Rotterdam t: +31 (0)10 2245544f: +31 (0)10 2245540 ___ 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] ATA that actually work with T.38
[EMAIL PROTECTED] wrote: > For T.38 passthrough between RTP channels it doesn't need to know a > great deal. There are some pitfalls, though, due to dumbness > in the T.38 > spec. > > Are you actually working on this? Yes, well, with a lot of other things, so progress is erratic. I've got to solve some other problems first, but Asterisk T.38 pass through is the next major issue. -- Andreas SikkemaRits tele.com Van Vollenhovenstraat 33016 BE Rotterdam t: +31 (0)10 2245544f: +31 (0)10 2245540 ___ 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] ATA that actually work with T.38
[EMAIL PROTECTED] wrote: >> Is it only the ATA that has to be T.38 compatible or does Asterisk >> have to work with T.38 also? Does Asterisk support T.38? > Asterisk must have T38 support in order to recognize the signaling. > No it doesn't at this time. We're working on Fax as well and if I'm not mistaken, there is a mode where Asterisk doesn't have to know very much about T.38 to make it work. -- Andreas SikkemaRits tele.com Van Vollenhovenstraat 33016 BE Rotterdam t: +31 (0)10 2245544f: +31 (0)10 2245540 ___ 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] List tips for new subscribers
[EMAIL PROTECTED] wrote: > (side note: If you havent bought their hardware and are using > Asterisk for "free" them again you should expect even less > assistance imo) Right, so I have to buy hardware I don't need? -- Andreas SikkemaRits tele.com Van Vollenhovenstraat 33016 BE Rotterdam t: +31 (0)10 2245544f: +31 (0)10 2245540 ___ 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] logger reload/restart hanging
Hi, We're running a very old version of Asterisk (CVS-HEAD-08/03/04) and we're having some problems with logging. Our logger.conf has the following: full => notice,warning,error,debug,verbose After having started Asterisk, asterisk will hang in "/usr/sbin/asterisk -rx 'logger reload'" unless some output has been sent to the file. I can't find anything on bugs.digium.com related to this problem. Am I the only one? Also no useful output will be sent to the log file, unless I run "asterisk -rdn" and exit from the console. Is this normal? How do I prevent neeeding this step? I know we shoul move to at least 1.0, but we're running this in production and we haven't felt the need to upgrade. If necessary I can backport... -- Andreas SikkemaRits tele.com Van Vollenhovenstraat 33016 BE Rotterdam t: +31 (0)10 2245544f: +31 (0)10 2245540 ___ 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] X-IMail-SPAM-Connection DNS Sudo ANI vs True ANI
[EMAIL PROTECTED] wrote: > I'm having problems with international calling via Global > Crossing. I'm told I need to send a true ani versus a sudo ani. > What is the difference and how can I configure asterisk to do this. > Global Crossing is denying calls with sudo anis. I'm wondering if they didn't mean a "pseudo" ani? Are you sending internal Asterisk ANI or the ANI Gobal Crossing is expecting? -- Andreas SikkemaRits tele.com Van Vollenhovenstraat 33016 BE Rotterdam t: +31 (0)10 2245544f: +31 (0)10 2245540 ___ 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] Sipua SPA-2000 and liong delay afterdialling number
[EMAIL PROTECTED] wrote: > The invite message is sent as a single message to asterisk > containing the whole number string, as apposed to each number > individually. Does SIP support non en-bloc dialling mode? -- Andreas SikkemaRits tele.com Van Vollenhovenstraat 33016 BE Rotterdam t: +31 (0)10 2245544f: +31 (0)10 2245540 ___ 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] Calculating required bandwidth
[EMAIL PROTECTED] wrote: > [EMAIL PROTECTED] wrote: >> I was posed this question: >> >> A T1 set up for voice carries 24 conversations on a circuit that is >> 1.544 megabits/second. Right? > > Yes and no. If the T1 is channelized, then yes. If it's a PRI > circuit, then it has only 23 channels to carry voice, as the 24th > channel is used for the D-channel (signalling channel). Only if you're in the US. We have 30 + 1 :-) -- Andreas SikkemaRits tele.com Van Vollenhovenstraat 33016 BE Rotterdam t: +31 (0)10 2245544f: +31 (0)10 2245540 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Dialing out to 2 clients simultaneously
[EMAIL PROTECTED] wrote: > You can't make config changes without having to restart SER If you have it running stateless (is there another way?), there's hardly any impact at all. Just a window of a few seconds where a call cannot be made. > Can change to-URI (add prefix or something) but when you do that ser > behaves strange, in statefull mode not recognizing the packets that > follow (call leg does not exist etc) Why are you running in statefull mode? > And some other minor things > > I just had too many issues with it... Maybe you're the issue? :-)) A combo of SER and Asterisk is pretty powerfull IMHO. -- Andreas SikkemaRits tele.com Van Vollenhovenstraat 33016 BE Rotterdam t: +31 (0)10 2245544f: +31 (0)10 2245540 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
