Re: [Asterisk-Users] Re: Nokai E60 and E61 , working fine with Asterisk , with new access points
--- Markus Schuster [EMAIL PROTECTED] wrote: Could you please post some details (or even better: write them in some sort of Wiki) on the configuration you did on the Nokia? I'm thinking about buying a Nokia E60 but after a short web search there seem to be some problems about the correct configuration of the phone. I tried to put some details on Voip-info.org , please check the link http://www.voip-info.org/wiki/view/Nokia thanks Joseph John Send instant messages to your online friends http://uk.messenger.yahoo.com ___ --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
AW: [Asterisk-Users] Compiling mpg123 under 1.2.9.1 / Ubuntu 6.06
Good morning, Why still use mpg123? Start using format_mp3 from asterisk-addons and your * will play mp3 by itself... good point - did that, and everything's working again. Thanks! Marc Rohlfing ___ --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] delay in MeetMe
No , actually I am using Asterisk-1.2.9.1 I will try the q option though Thanks and regards, Amna On 6/14/06, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: I assume you are using 1.0.x.Add the q option to the Meetmeextension.1.0.x has a known issue where enter/exit sounds cause increasing delays.amna saleem wrote: Hi All! I am facing some delay in conferencing. Using DIAX for Voip calls ,no hardware used yet I am using MeetMe to achieve conferencingand am having a lot of delays. Can anyone tell me how to reduce the delay Regards, Amna Saleem ___ --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--Now accepting new clients in Birmingham, Atlanta, Huntsville,Chattanooga, and Montgomery. ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: delay in MeetMe
Hi! I am using Asterisk-1.2.9.1 Zaptel 1.2.6 And my system has Linux Kernal 2.4 Best Regards, Amna On 6/14/06, Tony Mountifield [EMAIL PROTECTED] wrote: In article [EMAIL PROTECTED] ,amna saleem [EMAIL PROTECTED] wrote: Hi All! I am facing some delay in conferencing. Using DIAX for Voip calls ,no hardware used yet I am using MeetMe to achieve conferencingand am having a lot of delays. Can anyone tell me how to reduce the delayWhat version of Zaptel are you using, what version of Asterisk, andwhich Linux kernel does your system have? CheersTony--Tony MountifieldWork: [EMAIL PROTECTED] - http://www.softins.co.ukPlay: [EMAIL PROTECTED] - http://tony.mountifield.org___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ztdummy
--- Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com wrote: On Jun 13, 2006, at 7:52 PM, Josu頃onti wrote: ?? Doug, If you it will not have hardware and if ztdummy will not have installed its moh will not function correctly I believe this is no longer be true with the new Native music on hold... Marty I was under the impression that you had to use ztdummy as long as you were using the 2.4 Linux kernel. It didn't matter whether it was the native MoH or not. If you are running the 2.6 Linux kernel, you can safely ignore using ztdummy. Undrhil ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Zap/QSig with ChanIsAvailable
Hey, we're running an asterisk system connected to another telco system using qsig. I'm currently trying to use ChanIsAvailable to get the current phone status out of the foreign system. ChanIsAvailable always return 0 - UNKNOWN. The Qsig protocoll itself supports the feature to query the status of a given phone. Is there any other way beside to use ChanIsAvailable? Thanks in advance! Greetings, Michael ___ --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] ztdummy
I believe this is no longer be true with the new Native music on hold... Ah. I suspected as much, when my home server wasn't running any zaptel for a bit I found that MoH was working fine, even though last time I tried without ztdummy a while ago it was severely broken. (The reason I turned off zaptel for a bit was that every time I started asterisk it segfaulted, restarted, then locked the server, even after I removed the x100 card. Turned out that disabling h.323 fixed the problem...) Thanks James ___ --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] SIP, Microsoft RTC, and Originate problem
It seems that Microsoft RTC has some problems with originated calls from Asterisk. If I execute Manager API originate application, with SIP channel as parameter, the Microsoft RTC softphone will start to ring after a couple of seconds delay, but nothing more happens after when I answer there is no second call to an extension. When I looked through the sip debug, I noticed that Microsoft RTC fails to properly respond to INVITE messages (I have attached the sip debug). Asterisk has to retransmit INVITE message for 6 times and even then the RTC still doesn't respond in a proper time. However, if I do direct call to that problematic Microsoft RTC based softphone, everything works fine, eventhough very same INVITE messages are being transmited to it from Asterisk. Does anyone have any ideas for a workaround? Regards, Alex Reliably Transmitting (no NAT) to 111.111.111.50:1: INVITE sip:111.111.111.50:1 SIP/2.0 Via: SIP/2.0/UDP 111.111.111.8:5060;branch=z9hG4bK360502ae;rport From: asterisk sip:[EMAIL PROTECTED];tag=as348de10b To: sip:111.111.111.50:1 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 13 Jun 2006 07:25:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 242 v=0 o=root 10295 10295 IN IP4 111.111.111.8 s=session c=IN IP4 111.111.111.8 t=0 0 m=audio 12742 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- [Kasterisk*CLI Retransmitting #1 (no NAT) to 111.111.111.50:1: INVITE sip:111.111.111.50:1 SIP/2.0 Via: SIP/2.0/UDP 111.111.111.8:5060;branch=z9hG4bK360502ae;rport From: asterisk sip:[EMAIL PROTECTED];tag=as348de10b To: sip:111.111.111.50:1 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 13 Jun 2006 07:25:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 242 v=0 o=root 10295 10295 IN IP4 111.111.111.8 s=session c=IN IP4 111.111.111.8 t=0 0 m=audio 12742 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Retransmitting #2 (no NAT) to 111.111.111.50:1: INVITE sip:111.111.111.50:1 SIP/2.0 Via: SIP/2.0/UDP 111.111.111.8:5060;branch=z9hG4bK360502ae;rport From: asterisk sip:[EMAIL PROTECTED];tag=as348de10b To: sip:111.111.111.50:1 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 13 Jun 2006 07:25:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 242 v=0 o=root 10295 10295 IN IP4 111.111.111.8 s=session c=IN IP4 111.111.111.8 t=0 0 m=audio 12742 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- [Kasterisk*CLI Retransmitting #3 (no NAT) to 111.111.111.50:1: INVITE sip:111.111.111.50:1 SIP/2.0 Via: SIP/2.0/UDP 111.111.111.8:5060;branch=z9hG4bK360502ae;rport From: asterisk sip:[EMAIL PROTECTED];tag=as348de10b To: sip:111.111.111.50:1 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 13 Jun 2006 07:25:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 242 v=0 o=root 10295 10295 IN IP4 111.111.111.8 s=session c=IN IP4 111.111.111.8 t=0 0 m=audio 12742 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- [Kasterisk*CLI Retransmitting #4 (no NAT) to 111.111.111.50:1: INVITE sip:111.111.111.50:1 SIP/2.0 Via: SIP/2.0/UDP 111.111.111.8:5060;branch=z9hG4bK360502ae;rport From: asterisk sip:[EMAIL PROTECTED];tag=as348de10b To: sip:111.111.111.50:1 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 13 Jun 2006 07:25:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 242 v=0 o=root 10295 10295 IN IP4 111.111.111.8 s=session c=IN IP4 111.111.111.8 t=0 0 m=audio 12742 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- [Kasterisk*CLI Retransmitting #5 (no NAT) to 111.111.111.50:1: INVITE sip:111.111.111.50:1 SIP/2.0 Via: SIP/2.0/UDP 111.111.111.8:5060;branch=z9hG4bK360502ae;rport From: asterisk sip:[EMAIL PROTECTED];tag=as348de10b
RE: [Asterisk-Users] Festival RPM?
festival.i386 1.4.2-25 Too older. And does anyone know if I can add Festival voices to Flite (slightly off-topic... sorry...)? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Colin Anderson Sent: Tuesday, June 13, 2006 5:56 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Festival RPM? um, yum install festival worked for me. -Original Message- From: Mimmus [mailto:[EMAIL PROTECTED] Sent: Tuesday, June 13, 2006 9:47 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Festival RPM? Hi, is there a RHEL4 RPM for the Festival text-to-speech system? Thanks -- Domenico Viggiani ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OPENSER / SER and Asterisk
Santosh Rao a écrit : asterisk has a extremely cool documentation. The wiki has everything a newbie like me could hope for.. with samples and everyhting./. where as we are having a very dificult time finding proper documentation or samples and stuff like thtt for SER.. may be if someone good with SER could update ther voip-info/wiki and write some basics abt the ser.cfg or somethjing .. then it would be great. You can find some very good SER tutorials on onsip.org. You need to subscribe though, but it's free. Cheers, Jean-Michel. -- Jean-Michel Hiver - http://ykoz.net/ Découvrez la Réunion des Technologies IP Telecom TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE ___ --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] DTMF when using g.729
Hello How do I get Asterisk to receive DTMF from our Snom phones, when I use G.729? Regards Jon -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.8.4/363 - Release Date: 13-06-2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] echo sidetone grandstream and tdm400p
Hi Marty, thank you for your suggestion, but... just done and nothing happens. The perfect RX gain for me is 6.0; if I change the TX nothing really happens, but if I move it to a value less than 0.0, less than that I cannot ear anything... strange but true. As I said, on internal calls everything works fine, without any echo. When I make external calls (pstn with digium TDM400P) I ear an echo just at the begin and at the end of any speech. If I say short words (sounds) then I hear a lot of echos. Isn't it a sidetone effect? Any other ideas? Thanks again, Marco On Tue, 13 Jun 2006 12:02:20 -0700, Martin Joseph wrote On Jun 13, 2006, at 9:54 AM, Marco Sajeva wrote: First, thank you for your quick and kind answer. I cannot change the TX gain on the Grandstream phones, or atleast I don't know how to... Can anybody help, please? Actually since you suggested the problem is only with your PSTN calls (ie Zap channels), I think the answer is more likely in the zapata.conf? I would experiment a bit more there with the gains... Marty Thanks in advance, Marco On Tue, 13 Jun 2006 08:54:31 -0600, Colin Anderson wrote Turn down your microphone TX gains on the phones. On my TDM400 with Vista 350's I had to crank the mic value way down. This is not specific to FXS phones, on my Snom 200's sidetone is so bad, that an appropriate setting for mic gain is '2' (out of 8) hth -Original Message- From: Marco Sajeva [mailto:[EMAIL PROTECTED] Sent: Tuesday, June 13, 2006 8:43 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] echo sidetone grandstream and tdm400p Hi all, thanks to the all of you. This list is very interesting also for a newby like me. My problem: I just setup my first full working asterisk installation with this config: 1. n.1 GXP-2000 2. n.4 Budgetone 102 3. n.1 TDM400p (3 FXS, 1 FXO) Everything seems to work fine, but the sidetone... it's really annoying! We can hear the sidetone only when we call to the outside (PSTN), it doesn't matter if we call a local, a mobile or a longdistance call. Only we hear the echo, not the called party. We do not ear any echo in internal call to each other extensions. I tryed every possible setting of the echotraining, of the rx and of the tx gain, but with no success. Any idea or help? Thank you in advance, Marco __ Dott. Ing. Marco Sajeva Visioni - we network http://www.visioni.info ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Dott. Ing. Marco Sajeva Visioni - we network http://www.visioni.info ___ --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] sound quality problem on mISDN
Have you only one BN-Card? or more? i have two cards, had compareable problems. PCM was the magic word ... from my misdn-init.conf: card=1,0x8,pcm_slave,ignore_pcm_frameclock //important! option=9,master_clock // 9 for port 9 pcm=1,1 //not sure, if this is really neaded Hi I've problem with incoming call quality to GSM gateway connected to beronet card (BN8S0), - [ GSM Gateway ] --- [ BN8S0 ] asterisk ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP, Microsoft RTC, and Originate problem
Hi, What is your setup? By MS RTC do you mean Office Communicator? If you are using MS OC, do you use SER in between (to convert SIP UDP2TCP)? Please share some more details J Cheers, Ohad From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk Sent: Wednesday, June 14, 2006 9:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] SIP, Microsoft RTC, and Originate problem It seems that Microsoft RTC has some problems with originated calls from Asterisk. If I execute Manager API originate application, with SIP channel as parameter, the Microsoft RTC softphone will start to ring after a couple of seconds delay, but nothing more happens after when I answer there is no second call to an extension. When I looked through the sip debug, I noticed that Microsoft RTC fails to properly respond to INVITE messages (I have attached the sip debug). Asterisk has to retransmit INVITE message for 6 times and even then the RTC still doesn't respond in a proper time. However, if I do direct call to that problematic Microsoft RTC based softphone, everything works fine, eventhough very same INVITE messages are being transmited to it from Asterisk. Does anyone have any ideas for a workaround? Regards, Alex ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] voip to voip bridge
Hi, Check if reinvites are enabled, and that you dont use any parameter in the dial command that forces asterisk to stay in the loop. Ohad From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Erick Baum Sent: Wednesday, June 14, 2006 5:00 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] voip to voip bridge Has anyone had any good experiences with a voip to voip bridge... where you have an incoming call on a voip line which is redirected out another voip line to a regular phone line? Whenever we do this, the connected call is kinda lagged and the quality isn't always that great. It seems to me this is just a problem with the inherent delay in the voip connections. But I was wondering if there's any special configurations that could make the situation better? Erick ___ --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] Linksys SRW224P POE Switch
I was just about to suggest the Powersense module that Cory mentioned. And no, the G models do not support 802.3af. Cory, there was some discussion about just doing the cable only works on dumb poe injectors, not the ones that only send power if requested. I was under the impression the Linksys only sent power if requested, and if that was the case only the Powersense would work. Admittedly, I have not tried the cable only approach. The discussions were all from this list several months ago. There was also someone that said the cable that converts standard POE to use for the Polycom non standard POE phones would work as well. I haven't tried this yet either. I may do some experimenting today. Guess I should locate my oldest 7960 first, in case there are sparks and a fire :-) On 6/13/06, Cory Andrews [EMAIL PROTECTED] wrote: There is an RJ45 cabling guide on the WIKI that shows how to create areverse polarity crossover cable to power Cisco legacy PoE phones, and I can attest that it works with all the applications I have tried.Belkin/Powersense also makes an inline module for Cisco CDP that isrelatively inexpensive.Cory J AndrewsVOIPSupply.com 454 Sonwil DriveBuffalo, NY 14225++voice - 716.630.1555 X22email - [EMAIL PROTECTED]AIM - B2CORY- Original Message -From: Mike Fedyk [EMAIL PROTECTED]To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Sent: Tuesday, June 13, 2006 8:54 PMSubject: Re: [Asterisk-Users] Linksys SRW224P POE Switch Tom wrote: Most of the latest generation POE switches including the Linksys SRW224P provide their power on the data pairs, not the unused pairs.So if both the data and the power are on the same pairs, how do you make a cable adapter to work with the 7960G? Maybe bridge the unused pairs with the data pairs? I haven't tried it as I don't have any old style PoE, but it seems plausible. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Lacy MooreAspendora, Inc. ___ --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] GXP-2000 Audio Quality
G729 uses 8kbps but with the IP overhead it actually uses 30kbps so for 256k upstream you should be able to handle 8 calls but this is in ideal conditions. If you were to use IAX and enable trunking then you would use 30kbps for the 1st call and 10kbps for each additional call. See http://www.voip-info.org/wiki/index.php?page=Asterisk+bandwidth+iax2 On Wed, 2006-06-14 at 04:17, Daniel Salama wrote: I have a client with about 16 GXP-2000. They complain that the audio quality is terrible after 2 or 3 simultaneous conversations. They are behind DSL 1.5Mbps down and 256Kbps up. Because they are using G711.u codec, I know they upstream bandwidth is the limiting factor and they most likely won't be able to have more than 3 simultaneous conversations, and if they're surfing the net and/or checking email, things will only get worse. So, I purchased some g729 codec licenses and forced their sip peer configuration to g729 codec. We made sample test calls and were able to make 8 simultaneous calls. On the eighth call, the audio started to sound choppy. Then we dropped the eighth call and tested with 7. We could hear just fine on the GXP-2000 but the remote end heard us a bit choppy and/or with a robot-like voice. So, we kept dropping calls until they were of acceptable quality. My question is, if they were using g729 which, in theory uses 8kbps plus overhead, they should have been just fine handling eight calls. All the computers were turned off on the network, so there shouldn't have been any other traffic but VoIP. Does anyone have any ideas? How can I improve their audio quality? I requested BellSouth to upgrade their capacity but because of where they are located, the best they can get is 900Kbps/256Kbps, so the upstream continues to be the limiting factor. I purchased a Dlink-1226G switch to allow me to control QoS on the LAN. I also upgraded their Netopia DSL router to the latest firmware which allows me to configure VLANs and DiffServ. All the computers are connected to the PC port on the phone because there is no available second wiring. Can anyone suggest how to configure the QoS settings on the phones, the Dlink and the Netopia? While there was no traffic on the wire, pinging from/to the Asterisk box gave me about 47ms latency. When we went passed the 4th call, the latency started increasing significantly and when we got to 8 calls, the latency was up in the 2000ms. Obviously, if anything I did in the QoS configuration gave VoIP a priority, then ICMP packets would have the lowest priority and I could understand that to be the reason for such result. However, I'm not sure I configured QoS properly and that's why I'm asking for help. Thanks, Daniel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How much bandwidth needed?
Hi Friends,I am implementing Asterisk PBX in our office with 180 extensions. In our office, we will make 3 calls to USA daily. We have 1 MBPS bandwidth from ISP and 100 MBPS bandwidth in our LAN. I have two doubts.1) How much bandwidth should we allocate for making VOIP calls? What can be the projected use of bandwidth to make International VOIP calls?2) Can I use Pentium IV system, 2.6 GHz processor speed with 512 MB RAM as dedicated Asterisk server?3) Now I am making calls to USA using Voipjet.com provider. How can I receive incoming calls through VoIP? How can I get my own incoming VoIP number?Looking forward for your reply.ThanksRegards,Chandra. __Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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] How much bandwidth needed?
Crazy Boy wrote: Hi Friends, I am implementing Asterisk PBX in our office with 180 extensions. In our office, we will make 3 calls to USA daily. We have 1 MBPS bandwidth from ISP and 100 MBPS bandwidth in our LAN. I have two doubts. 1) How much bandwidth should we allocate for making VOIP calls? What can be the projected use of bandwidth to make International VOIP calls? It depends on codec you are using. G711 needs 64kbps for raw voice stream (add protocol overhead). 2) Can I use Pentium IV system, 2.6 GHz processor speed with 512 MB RAM as dedicated Asterisk server? Of course. It will be enough. But if you want to transcode fe. between iLBC and Speex it will consume much more CPU power. 3) Now I am making calls to USA using Voipjet.com provider. How can I receive incoming calls through VoIP? How can I get my own incoming VoIP number? Look for DID service. Looking forward for your reply. ThanksRegards, Chandra. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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 -- Marcin Kwiatkowski System Administrator Mob: +48 663 617 664 Fix: +48 33 819 04 60 ext. 32 ___ --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] sound quality problem on mISDN
On Wed, Jun 14, 2006 at 10:04:04AM +0200, Kai Ober wrote: Have you only one BN-Card? or more? I've one BN8S0 card and one TE110P - T1 connection to Rhino channelbank. i have two cards, had compareable problems. PCM was the magic word ... from my misdn-init.conf: card=1,0x8,pcm_slave,ignore_pcm_frameclock //important! option=9,master_clock // 9 for port 9 pcm=1,1 //not sure, if this is really neaded Intresting I'm going to try this today . I thinking also about 'ulaw' option to 'card=' . My channelbank is T1 and this will eliminate transcoding from isdn to T1. thx for help. /pch -- Dyslexia bug unpatched since 1977 ... exploit has been leaked to the underground. ___ --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] How much bandwidth needed?
Crazy Boy wrote: Hi Friends, I am implementing Asterisk PBX in our office with 180 extensions. In our office, we will make 3 calls to USA daily. We have 1 MBPS bandwidth from ISP and 100 MBPS bandwidth in our LAN. I have two doubts. 1) How much bandwidth should we allocate for making VOIP calls? What can be the projected use of bandwidth to make International VOIP calls? Go have a look here... http://www.asteriskguru.com/tools/bandwidth_calculator.php 2) Can I use Pentium IV system, 2.6 GHz processor speed with 512 MB RAM as dedicated Asterisk server? 3 calls a day ? How many simultaneous calls maximum to any destination ? you can probably do 40-50 simultaneous calls with the most cpu intense codecs. (g729, iLBC, speex) and 2 to 3 times more with ulaw/alaw/gsm 3) Now I am making calls to USA using Voipjet.com provider. How can I receive incoming calls through VoIP? How can I get my own incoming VoIP number? Looking forward for your reply. ThanksRegards, Chandra. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk server
Hi,I have to build Asterisk server for about 30 user (30 concurrent calls). I decided to buy this box:-- motherboard Intel E7210 + Hence Rapids-- processor P4 3.0 GHz-- RAM 2x512 MB DDR ECC-- network interface Intel 82541 GI Is this configuration enough to handle 30 users at the same time. I am not planning to use any transcoding (everything will be alaw).CheersAndrew ___ --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] nortel meridian option 11c and asterisk te110p
Hi Koen Van Impe Thanks for the meridian config and asterisk. I will defenitly try them And let every one know. Just a few words and correct me if I am wrong There are two things 1 E1 : the 32 channels once both the equipment see each other and the ccs/hdb3 encoding/format is read the LED infront of interface goes green and this makes the lower layer work. 2 ISDN PRI: once step one is complete we can proceed to the signaling of ISDN PRI that is euro isdn or 5ess or any . I might be wrong But the problem that I face is the first step the e1 never comes up I have and the LED never goes green. I have checked the cable it work s fine with other pri which interms confirms the card also. But with the new config that u have given me I pray it works bcz it is very critical for my organization as we are tired of paying Nortel bags and bags of money and with this idea of using asterisk and interface it with the existing meridian system we see a hope of expanding with very little investment. Thanks and regards mohammad ___ --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 server
Its overkill, go get some more employees :) So yes, its just fine and there's room for expansion. Zoa Andrew Nowrot wrote: Hi, I have to build Asterisk server for about 30 user (30 concurrent calls). I decided to buy this box: -- motherboard Intel E7210 + Hence Rapids -- processor P4 3.0 GHz -- RAM 2x512 MB DDR ECC -- network interface Intel 82541 GI Is this configuration enough to handle 30 users at the same time. I am not planning to use any transcoding (everything will be alaw). Cheers Andrew ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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 server
Andrew Nowrot wrote: Hi, I have to build Asterisk server for about 30 user (30 concurrent calls). I decided to buy this box: -- motherboard Intel E7210 + Hence Rapids -- processor P4 3.0 GHz -- RAM 2x512 MB DDR ECC -- network interface Intel 82541 GI Is this configuration enough to handle 30 users at the same time. I am not planning to use any transcoding (everything will be alaw). Cheers Andrew Yes. /M -- Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP, Microsoft RTC, and Originate problem
Nope, it's just the Microsoft RTC Core 1.3 library ... more or less a single DLL J. And I'm almost sure there is no SER in between should there be one? It's pretty much a straightforward thing I have a few SIP clients defined in my sip.conf, like this: [general] context=default allowguest=yes realm=timd.si bindport=5060 bindaddr=0.0.0.0 srvlookup=yes domain=timd.si,from-sip domain=111.111.111.8,from-sip videosupport=yes disallow=all allow=alaw allow=ulaw musicclass=default rtptimeout=100 rtpholdtimeout=100 tos=0x18 canreinvite=yes [SIPClient001] username= SIPClient001 secret= mysecret type=friend host=dynamic context=from-sip disallow=all allow=alaw allow=ulaw qualify=yes [SIPClient002] username= SIPClient002 secret= mysecret type=friend host=dynamic context=from-sip disallow=all allow=alaw allow=ulaw qualify=yes And there is an MS RTC based Softphone, that I made, on the other side that registers to Asterisk, using this profile XML string: provision key=5B29C449-29EE-4fd8-9E3F-04AED077690E name=Asterisk user account=SIPClient001 uri=sip:[EMAIL PROTECTED] / sipsrv addr=111.111.111.8 protocol=udp auth=digest role=registrar session party=first type=pc2ph / /sipsrv /provision Now, doing an originate to CHANNEL=SIP/SIPClient002, and some extension, will randomly fail, for example (see OriginateFailure reponse as well): action: Originate actionid: 123 exten: 03020846051635424 channel: SIP/SIPClient002 timeout: 3 priority: 1 context: asttel async: true Event: OriginateFailure Privilege: call,all ActionID: 123 Channel: SIP/ SIPClient002 Context: asttel Exten: 03020846051635424 Reason: 1 Uniqueid: null From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, June 14, 2006 10:14 AM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] SIP, Microsoft RTC, and Originate problem Hi, What is your setup? By MS RTC do you mean Office Communicator? If you are using MS OC, do you use SER in between (to convert SIP UDP2TCP)? Please share some more details J Cheers, Ohad From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk Sent: Wednesday, June 14, 2006 9:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] SIP, Microsoft RTC, and Originate problem It seems that Microsoft RTC has some problems with originated calls from Asterisk. If I execute Manager API originate application, with SIP channel as parameter, the Microsoft RTC softphone will start to ring after a couple of seconds delay, but nothing more happens after when I answer there is no second call to an extension. When I looked through the sip debug, I noticed that Microsoft RTC fails to properly respond to INVITE messages (I have attached the sip debug). Asterisk has to retransmit INVITE message for 6 times and even then the RTC still doesn't respond in a proper time. However, if I do direct call to that problematic Microsoft RTC based softphone, everything works fine, eventhough very same INVITE messages are being transmited to it from Asterisk. Does anyone have any ideas for a workaround? Regards, Alex ___ --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] nortel meridian option 11c and asterisk te110p
Hi there sir Thanks for ur suggestion but the problem with us is that we are running the whole distributed call center in three different cities of pakistan. So we can not take risk on that behalf we just want that our expansion need to be fulfilled by expanding throght asterisk which far cheaper than existing Nortel. So thanks any ways for ur suggestion Regards zeeshan Another approach you might take would be to keep your Meridian phones but get rid of your PBX, utilising a Citel SIP Handset Gateway to interface the phones to the Asterisk server. See http://www.citel.com for more details. ___ --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] Eicon Diva Server with v3.0 drivers
Hi, I'm trying to get an Eicon Diva Server4BRI card running under Ubuntu 6.06 - by downloading the v3 driver package from Melware and compiling everything. Yet, after activating the necessary modules (divas and divadidd) and interactively configuring the card (/usr/lib/divas/Config), starting up the adapter fails! The error /usr/lib/divas/divactrl load -c 1 -Debug produces complains about the missing protocol image A: can't open protocol image: (/usr/lib/divas/te_etsi.2q0) I know from version 2.0 that you had to download these images from either isdn4linux.org or melware.de, yet none of them still have the files available. /usr/lib/divas/ does contain some *etsi* files - do they help somehow? Any hint appreciated. Marc Rohlfing ___ --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] Strange problem with MusicOnHold - works outgoing - works with extension - but not incoming!
I've got a strange situation with musiconhold. It works if I dial my extension 6000: From extensions.conf: exten = 6000,1,Answer exten = 6000,2,MusicOnHold() Debug output if I call 6000: -- Executing Answer(SIP/gs1-b6ee, ) in new stack -- Executing MusicOnHold(SIP/gs1-b6ee, ) in new stack -- Started music on hold, class 'default', on SIP/gs1-b6ee -- Stopped music on hold on SIP/gs1-b6ee server*CLI If I dial out and put a call on hold the other party hears the musiconhold: Debug output when I do an outgoing call: -- Executing SetCallerID(SIP/gs1-cb7a, Anonymous 0031x) in new stack -- Executing Dial(SIP/gs1-cb7a, SIP/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] -- SIP/voipbuster-ac66 is making progress passing it to SIP/gs1-cb7a -- SIP/voipbuster-ac66 answered SIP/gs1-cb7a -- Attempting native bridge of SIP/gs1-cb7a and SIP/voipbuster-ac66 -- Started music on hold, class 'default', on SIP/voipbuster-ac66 -- Stopped music on hold on SIP/voipbuster-ac66 But If somebody rings me and I put him on hold he hears nothing: Debug output for incoming call: -- Executing SetCallerID(SIP/gw02-mci.budgetphone.nl-42ba1908, prive xx) in new stack -- Executing Dial(SIP/gw02-mci.budgetphone.nl-42ba1908, SIP/sipuraSIP/gs4) in new stack -- Called sipura -- Called gs4 -- SIP/sipura-7685 is ringing -- SIP/gs4-4a86 is ringing -- SIP/gs4-4a86 answered SIP/gw02-mci.budgetphone.nl-42ba1908 -- Attempting native bridge of SIP/gw02-mci.budgetphone.nl-42ba1908 and SIP/gs4-4a86 -- Started music on hold, class 'default', on SIP/gw02-mci.budgetphone.nl-42ba1908 According to the logs it starts the music on hold, the same way as in the other calls but it stays quiet! I've tried everything but I don't know what else to check. I've got Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k on a debian machine. In my sip.conf: [general] musicclass=default musiconhold=default (I tried it with only miscclass, only musiconhold, and without both, nothing changes) In musiconhold.conf [classes] default = quietmp3:/usr/share/asterisk/mohmp3 What can be wrong, what else can I check? Kind regards, De Boer ___ --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] Eicon Diva Server with v3.0 drivers
On Wed, 14 Jun 2006, Marc Rohlfing wrote: Hi, I'm trying to get an Eicon Diva Server4BRI card running under Ubuntu 6.06 - by downloading the v3 driver package from Melware and compiling everything. Yet, after activating the necessary modules (divas and divadidd) and interactively configuring the card (/usr/lib/divas/Config), starting up the adapter fails! The error /usr/lib/divas/divactrl load -c 1 -Debug produces complains about the missing protocol image A: can't open protocol image: (/usr/lib/divas/te_etsi.2q0) I know from version 2.0 that you had to download these images from either isdn4linux.org or melware.de, yet none of them still have the files available. /usr/lib/divas/ does contain some *etsi* files - do they help somehow? It is not necessary any more to download any firmware files (they are incompatible anyway). All needed files are part of the v3 package. It is odd, that the file te_etsi.* is searched. This is needed only if the DMLT code (te_dmlt.*) is not available. Can you please provide the list of files which are installed in /usr/lib/divas and possible logs /var/log/diva* ? Which card (board revision) do you have? Thanks, Armin ___ --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] RES: DISA Password Authenntication with Grandstream 488
Hi I can use now DISA settings like this one when I set E1 card connected directly to Asterisk. In this way every call dialed with pass 29 will be accepted. I have a billing which filters caller ID number and address calls to each account with same caller ID number previously set [frommt] exten = 1536,1,Answer exten = 1536,2,DigitTimeout(5) exten = 1536,3,ResponseTimeout(10) exten = 1536,4,Authenticate(29) exten = 1536,5,DISA(no-password|brasil) exten = 1536,6,Hangup Now I need to add a Grandstream 488 for DISA to remote landlines. So asterisk will receive phone number from the landline connected to this grandstream and also the sip account which is linked to Asterisk. But I cant decode caller phone number who dialed to the landline connected to asterisk. Is that possible with Asterisk to create a variable to collect a dialed password and then present that password which I can read it and then manipulate that pass ? Regards from Brazil Kind Regards, Diretoria Comercial - Newton Medina PABX 11.3085.1536 MSN[EMAIL PROTECTED] Rua Augusta 2.212 SL 26 Jardins 01412001 São Paulo - Brasil ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP, Microsoft RTC, and Originate problem
Hmm.. Interesting, I didnt try to implement it this way... but, if its the same libraries used for Office communicator, than it supports only SIP over TCP or TLS, since asterisk doesnt support any of those its impossible to connect them directly... If udp works, maybe the registration part is problematic, try configuring asterisk with autocreatepeer (just for testing) to see if you can dial out without being registered. Ohad From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk Sent: Wednesday, June 14, 2006 11:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] SIP, Microsoft RTC, and Originate problem Nope, it's just the Microsoft RTC Core 1.3 library ... more or less a single DLL J. And I'm almost sure there is no SER in between should there be one? It's pretty much a straightforward thing I have a few SIP clients defined in my sip.conf, like this: [general] context=default allowguest=yes realm=timd.si bindport=5060 bindaddr=0.0.0.0 srvlookup=yes domain=timd.si,from-sip domain=111.111.111.8,from-sip videosupport=yes disallow=all allow=alaw allow=ulaw musicclass=default rtptimeout=100 rtpholdtimeout=100 tos=0x18 canreinvite=yes [SIPClient001] username= SIPClient001 secret= mysecret type=friend host=dynamic context=from-sip disallow=all allow=alaw allow=ulaw qualify=yes [SIPClient002] username= SIPClient002 secret= mysecret type=friend host=dynamic context=from-sip disallow=all allow=alaw allow=ulaw qualify=yes And there is an MS RTC based Softphone, that I made, on the other side that registers to Asterisk, using this profile XML string: provision key=5B29C449-29EE-4fd8-9E3F-04AED077690E name=Asterisk user account=SIPClient001 uri=sip:[EMAIL PROTECTED] / sipsrv addr=111.111.111.8 protocol=udp auth=digest role=registrar session party=first type=pc2ph / /sipsrv /provision Now, doing an originate to CHANNEL=SIP/SIPClient002, and some extension, will randomly fail, for example (see OriginateFailure reponse as well): action: Originate actionid: 123 exten: 03020846051635424 channel: SIP/SIPClient002 timeout: 3 priority: 1 context: asttel async: true Event: OriginateFailure Privilege: call,all ActionID: 123 Channel: SIP/ SIPClient002 Context: asttel Exten: 03020846051635424 Reason: 1 Uniqueid: null From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, June 14, 2006 10:14 AM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] SIP, Microsoft RTC, and Originate problem Hi, What is your setup? By MS RTC do you mean Office Communicator? If you are using MS OC, do you use SER in between (to convert SIP UDP2TCP)? Please share some more details J Cheers, Ohad From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk Sent: Wednesday, June 14, 2006 9:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] SIP, Microsoft RTC, and Originate problem It seems that Microsoft RTC has some problems with originated calls from Asterisk. If I execute Manager API originate application, with SIP channel as parameter, the Microsoft RTC softphone will start to ring after a couple of seconds delay, but nothing more happens after when I answer there is no second call to an extension. When I looked through the sip debug, I noticed that Microsoft RTC fails to properly respond to INVITE messages (I have attached the sip debug). Asterisk has to retransmit INVITE message for 6 times and even then the RTC still doesn't respond in a proper time. However, if I do direct call to that problematic Microsoft RTC based softphone, everything works fine, eventhough very same INVITE messages are being transmited to it from Asterisk. Does anyone have any ideas for a workaround? Regards, Alex ___ --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
AW: [Asterisk-Users] Eicon Diva Server with v3.0 drivers
Hi, The error /usr/lib/divas/divactrl load -c 1 -Debug produces complains about the missing protocol image A: can't open protocol image: (/usr/lib/divas/te_etsi.2q0) It is not necessary any more to download any firmware files (they are incompatible anyway). All needed files are part of the v3 package. It is odd, that the file te_etsi.* is searched. This is needed only if the DMLT code (te_dmlt.*) is not available. Can you please provide the list of files which are installed in /usr/lib/divas and possible logs /var/log/diva* ? As usual, the second I sent my request, I tried something else and it worked (^_^) Seriously: If I run the autogenerated startup script (/usr/lib/divas/divas_cfg.rc), the card is activated just fine. capiinfo shows all 8 B-channels, so I guess I'm good to go. Maybe this should be stated more clearly in the INSTALL and README files - it's especially confusing for veterans who try to do things the 2.1 way... In any case: Thanks for the quick reply. Marc Rohlfing ___ --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 server
Hi, With 30 users and NO transcoding, that is certainly enough. Even if you use real-time configuration (that requires a SQL server) Now, if you system will be accessible both from inside (LAN) and outside (Internet), I would advice a second network card (10/100) Regards, T. Jacobson From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Nowrot Sent: mercredi 14 juin 2006 11:23 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Asterisk server Hi, I have to build Asterisk server for about 30 user (30 concurrent calls). I decided to buy this box: -- motherboard Intel E7210 + Hence Rapids -- processor P4 3.0 GHz -- RAM 2x512 MB DDR ECC -- network interface Intel 82541 GI Is this configuration enough to handle 30 users at the same time. I am not planning to use any transcoding (everything will be alaw). Cheers Andrew -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.8.4/363 - Release Date: 13/06/2006 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.8.4/363 - Release Date: 13/06/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] GXP-2000 1.1.0.13 Issues
I have had 2 GXP-2000 for a while now and been slowly following the firmware releases made by Grandstream and am now up to 1.1.0.13. This version works really well on these 2 original phones (MAC's 00:0B:82:06:xx:xx), so I went ahead and ordered another 2 phones (MAC's 00:0B:82:09:xx:xx). One of these I upgraded to 1.1.0.13 (it came with 1.1.0.5) and pressed it into use. The Speaker phone does not work at all (no sound from the Speaker) and the phone completely hangs doing a soft-reboot, other than that the phone seems to work well. Unfortunatly I do not have a copy of 1.1.0.5 so cannot downgrade the phone. Has anyone else noticed these problems, or does anyone have a copy of 1.1.0.5. -Drew- ___ --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] How to find out which line in extensions.conf?
When trying to figure out why something's not working, is there any way to have the output specify which line of extensions.conf was being executed? I mean, sure, I could pour a million NoOp()'s into it, but that's not exactly scalable, nor easy. It would be really nice if, instead, along with timestamp, it mentioned either a line number, or -- more likely -- a context/extension/priority triplet. Is there anything like that? Thanks, Ken D'Ambrosio ___ --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] AddQueueMember and Local channels
Following on from a posting yesterday from Kevin, I have the following
in the dialplan:
exten = 709,1,AddQueueMember(SomeQueue|Local/[EMAIL PROTECTED])
I am on extension 706.
From the CLI:
SomeQueue has 0 calls (max unlimited) in 'rrmemory' strategy (0s
holdtime), W:0, C:0, A:3, SL:0.0% within 60s
No Members
No Callers
I call 709, get a console message
NOTICE[30879]: app_queue.c:3122 aqm_exec: Added interface
'Local/[EMAIL PROTECTED]' to queue 'SomeQueue'
from the CLI:
SomeQueue has 0 calls (max unlimited) in 'rrmemory' strategy (0s
holdtime), W:0, C:0, A:3, SL:0.0% within 60s
Members:
Local/[EMAIL PROTECTED] (dynamic) (In use) has taken no calls yet
No Callers
Notice the (In use) on the member. When I call the queue, the call is
not passed onto the member, and there is no activity on the cli.
Eventually the call times out.
If I add SIP/706 instead of Local/[EMAIL PROTECTED] then it all works as
expected.
Any clues or help ? Many thanks !
Julian.
Kevin P. Fleming wrote:
- Julian Lyndon-Smith [EMAIL PROTECTED] wrote:
Now, I want to be able to use a device, rather than agents. So I can
use addQueueMember and add my SIP device. However, I still want to
do a couple of things before the device is called.
This is what the Local channel (chan_local) is for.
If your SIP device is called myfancyphone, then instead of adding
SIP/myfancyphone to the queue using AddQueueMember, add (instead)
Local/[EMAIL PROTECTED], and then in your dialplan:
[members]
exten = myfancyphone,1,...
exten = myfancyphone,n,...
exten = myfancyphone,n,Dial(SIP/${EXTEN})
___
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Re: [Asterisk-Users] GXP-2000 1.1.0.13 Issues
The only issue with 1.1.0.13 which affects only certain versions of the gxp-2000 is the display blanking issue on very early phones. It sounds like you have a faulty phone and should return it for a replacement. On Wed, 2006-06-14 at 11:57, [EMAIL PROTECTED] wrote: I have had 2 GXP-2000 for a while now and been slowly following the firmware releases made by Grandstream and am now up to 1.1.0.13. This version works really well on these 2 original phones (MAC's 00:0B:82:06:xx:xx), so I went ahead and ordered another 2 phones (MAC's 00:0B:82:09:xx:xx). One of these I upgraded to 1.1.0.13 (it came with 1.1.0.5) and pressed it into use. The Speaker phone does not work at all (no sound from the Speaker) and the phone completely hangs doing a soft-reboot, other than that the phone seems to work well. Unfortunatly I do not have a copy of 1.1.0.5 so cannot downgrade the phone. Has anyone else noticed these problems, or does anyone have a copy of 1.1.0.5. -Drew- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Realtime queue_members and penalties nost escalating (clue anyone?)
Howdy, have working realtime queues using queue_members looking something like; queuea|Local/[EMAIL PROTECTED]|0 queuea|Local/[EMAIL PROTECTED]|1 queuea|Local/[EMAIL PROTECTED]|10 Regardless of what strategy is used in the queues (roundrobin,rrmemory,ringall etc) it wont escalate on NOANSWER Asterisk SVN-branch-1.2-r33841 Any clues are appreciated! /Danny ___ --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 server
Thanks for all replies Now, if you system will be accessible both from inside (LAN) and outside (Internet), I would advice a second network card (10/100)Actually the machine has two interfaces - 1000 and 100 Mbit/s CheersAndrew ___ --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: AW: [Asterisk-Users] Eicon Diva Server with v3.0 drivers
On Wed, 14 Jun 2006, Marc Rohlfing wrote: Hi, The error /usr/lib/divas/divactrl load -c 1 -Debug produces complains about the missing protocol image A: can't open protocol image: (/usr/lib/divas/te_etsi.2q0) It is not necessary any more to download any firmware files (they are incompatible anyway). All needed files are part of the v3 package. It is odd, that the file te_etsi.* is searched. This is needed only if the DMLT code (te_dmlt.*) is not available. Can you please provide the list of files which are installed in /usr/lib/divas and possible logs /var/log/diva* ? As usual, the second I sent my request, I tried something else and it worked (^_^) Seriously: If I run the autogenerated startup script (/usr/lib/divas/divas_cfg.rc), the card is activated just fine. capiinfo shows all 8 B-channels, so I guess I'm good to go. Maybe this should be stated more clearly in the INSTALL and README files - it's especially confusing for veterans who try to do things the 2.1 way... Ah yes, I thought you did /usr/lib/divas/divactrl load -c 1 -Debug for debugging purposes only. Sure, divactrl load without -CfgLib cannot be used with the new v3 any more. I will try to added some notes in the README, thanks for pointing out. Armin ___ --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] GXP-2000 1.1.0.13 Issues
Thats what I thought the problem might be, so I have just now upgraded the other phone to 1.1.0.13 and its exactly the same, no speaker phone and hangs from a soft reboot. I also tried the audio loopback in the factory functions menu, this loopback's fine with the older 1.1.0.13 phones but does not with the newer ones (by older I mean MAC's 00:0B:82:06:xx:xx and newer I mean MAC's 00:0B:82:09:xx:xx). -Drew- On Wed, 14 Jun 2006, Gareth Blades wrote: The only issue with 1.1.0.13 which affects only certain versions of the gxp-2000 is the display blanking issue on very early phones. It sounds like you have a faulty phone and should return it for a replacement. On Wed, 2006-06-14 at 11:57, [EMAIL PROTECTED] wrote: I have had 2 GXP-2000 for a while now and been slowly following the firmware releases made by Grandstream and am now up to 1.1.0.13. This version works really well on these 2 original phones (MAC's 00:0B:82:06:xx:xx), so I went ahead and ordered another 2 phones (MAC's 00:0B:82:09:xx:xx). One of these I upgraded to 1.1.0.13 (it came with 1.1.0.5) and pressed it into use. The Speaker phone does not work at all (no sound from the Speaker) and the phone completely hangs doing a soft-reboot, other than that the phone seems to work well. Unfortunatly I do not have a copy of 1.1.0.5 so cannot downgrade the phone. Has anyone else noticed these problems, or does anyone have a copy of 1.1.0.5. -Drew- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sangoma driver update?
Can you help me, how to update the old sangoma driver? I downloaded the last driver from sangoma's web. kind regards, Szolke ___ --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] GXP-2000 1.1.0.13 Issues
If can help, I have 80 00:0b:82:08 :xx:xx GXP-2000 phones and they works well with 1.1.0.11 firmware. I can send you this firmware, if you mail me off-list. Bye DV -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, June 14, 2006 1:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] GXP-2000 1.1.0.13 Issues Thats what I thought the problem might be, so I have just now upgraded the other phone to 1.1.0.13 and its exactly the same, no speaker phone and hangs from a soft reboot. I also tried the audio loopback in the factory functions menu, this loopback's fine with the older 1.1.0.13 phones but does not with the newer ones (by older I mean MAC's 00:0B:82:06:xx:xx and newer I mean MAC's 00:0B:82:09:xx:xx). -Drew- On Wed, 14 Jun 2006, Gareth Blades wrote: The only issue with 1.1.0.13 which affects only certain versions of the gxp-2000 is the display blanking issue on very early phones. It sounds like you have a faulty phone and should return it for a replacement. On Wed, 2006-06-14 at 11:57, [EMAIL PROTECTED] wrote: I have had 2 GXP-2000 for a while now and been slowly following the firmware releases made by Grandstream and am now up to 1.1.0.13. This version works really well on these 2 original phones (MAC's 00:0B:82:06:xx:xx), so I went ahead and ordered another 2 phones (MAC's 00:0B:82:09:xx:xx). One of these I upgraded to 1.1.0.13 (it came with 1.1.0.5) and pressed it into use. The Speaker phone does not work at all (no sound from the Speaker) and the phone completely hangs doing a soft-reboot, other than that the phone seems to work well. Unfortunatly I do not have a copy of 1.1.0.5 so cannot downgrade the phone. Has anyone else noticed these problems, or does anyone have a copy of 1.1.0.5. -Drew- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] sound quality problem on mISDN
Piotr Chytla schrieb: On Wed, Jun 14, 2006 at 10:04:04AM +0200, Kai Ober wrote: Have you only one BN-Card? or more? I've one BN8S0 card and one TE110P - T1 connection to Rhino channelbank. i have two cards, had compareable problems. PCM was the magic word ... from my misdn-init.conf: card=1,0x8,pcm_slave,ignore_pcm_frameclock //important! option=9,master_clock // 9 for port 9 pcm=1,1 //not sure, if this is really neaded Intresting I'm going to try this today . I thinking also about 'ulaw' option to 'card=' . My channelbank is T1 and this will eliminate transcoding from isdn to T1.i hmm, my S0 cards are connected over a pcm bus ( the BN8S0 provides this, ). I don't think the pcm stuff will solve your problem, but hey, give it a try kai ___ --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] delay in MeetMe
The problem was fixed in 1.2.0 amna saleem wrote: No , actually I am using Asterisk-1.2.9.1 I will try the q option though Thanks and regards, Amna On 6/14/06, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: I assume you are using 1.0.x. Add the q option to the Meetme extension. 1.0.x has a known issue where enter/exit sounds cause increasing delays. amna saleem wrote: Hi All! I am facing some delay in conferencing. Using DIAX for Voip calls ,no hardware used yet I am using MeetMe to achieve conferencing and am having a lot of delays. Can anyone tell me how to reduce the delay Regards, Amna Saleem ___ --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 -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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] GXP-2000 1.1.0.13 Issues
Thanks for the offer, but I have just tried 1.1.0.11, it is available publicly and it has the same problems on these 2 phones. On Wed, 14 Jun 2006, Mimmus wrote: If can help, I have 80 00:0b:82:08 :xx:xx GXP-2000 phones and they works well with 1.1.0.11 firmware. I can send you this firmware, if you mail me off-list. Bye DV -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, June 14, 2006 1:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] GXP-2000 1.1.0.13 Issues Thats what I thought the problem might be, so I have just now upgraded the other phone to 1.1.0.13 and its exactly the same, no speaker phone and hangs from a soft reboot. I also tried the audio loopback in the factory functions menu, this loopback's fine with the older 1.1.0.13 phones but does not with the newer ones (by older I mean MAC's 00:0B:82:06:xx:xx and newer I mean MAC's 00:0B:82:09:xx:xx). -Drew- On Wed, 14 Jun 2006, Gareth Blades wrote: The only issue with 1.1.0.13 which affects only certain versions of the gxp-2000 is the display blanking issue on very early phones. It sounds like you have a faulty phone and should return it for a replacement. On Wed, 2006-06-14 at 11:57, [EMAIL PROTECTED] wrote: I have had 2 GXP-2000 for a while now and been slowly following the firmware releases made by Grandstream and am now up to 1.1.0.13. This version works really well on these 2 original phones (MAC's 00:0B:82:06:xx:xx), so I went ahead and ordered another 2 phones (MAC's 00:0B:82:09:xx:xx). One of these I upgraded to 1.1.0.13 (it came with 1.1.0.5) and pressed it into use. The Speaker phone does not work at all (no sound from the Speaker) and the phone completely hangs doing a soft-reboot, other than that the phone seems to work well. Unfortunatly I do not have a copy of 1.1.0.5 so cannot downgrade the phone. Has anyone else noticed these problems, or does anyone have a copy of 1.1.0.5. -Drew- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] NCS patch
Hi, I have cable modems Arris with MGCP protocol. And I need PacketCable NCSpatch for Asterisk. http://asterisk.urtho.net/ doesn't work!-- Pagarbiai,Giedrius AugysSiauliu Universitetas, ISTIP telefonijos inzinieriusTel. 8 41 590408Mob. Tel. 8 678 05790el. pastas [EMAIL PROTECTED] ___ --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] Compiling mpg123 under 1.2.9.1 / Ubuntu 6.06
On Tue, Jun 13, 2006 at 01:47:27PM +0200, Koen Van Impe wrote: Why still use mpg123? Start using format_mp3 from asterisk-addons and your * will play mp3 by itself... Not to mention that an mpg123 package is availble in Debian-nonfree . http://packages.debian.org/mpg123 http://packages.ubuntu.com/mpg123 -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 [EMAIL PROTECTED] http://www.xorcom.com ___ --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] GXP-2000 and Configdownload via TFTP
Hi, i got my Grandstream GXP-2000 phone today and want to configure it with TFTP. I downloaded the firmware 1.1.0.13 and put it into my tftp-server directory. Then I downloaded the template from: http://www.grandstream.com/DOWNLOAD/Configuration_Tool/Linux_Unix/Grandstream_Configuration_File_Template_1.0.6.x.txt renamed it to cfgmac-address Did the configuration in the new file and rebooted my phone. I can see in the log file from my tftp server that all files are loaded, the phone did a firmware upgrade. But it doesn't seems that the configuration file is loaded. Is it necessary to define on any place something that the phone use the config-file via tftp? Best regards, Matthias -- Programming today is a race between software engineers striving to build bigger and better idiot-proof programs, and the universe trying to produce bigger and better idiots. So far, the universe is winning. -- Rich Cook ___ --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] SPA-941 Disable call waiting or Disable Call waiting via asterisk
I'm trying to disable call waiting for Linksys SPA-941, but unfortunately as far as I have seen, there are no parameters on the web interface regarding this feature. I just want callers to hear the busy tone when the called party is at the phone. Probably I can accomplish this by using the disable call waiting in asterisk as well, but I have not been able to find any documentation for this. I have found this http://www.voip-info.org/wiki/index.php?page=PBX+Disable+Call+Waiting about call waiting, but it's quite unusefull. Thanks Tommaso Calosi ___ --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 when using g.729
Is new to me that using G729 codec is a problem when sending DTMF. Could it be that you are a little bit confused? Usually the problems with DTMF depend on how the phone is configured and how Asterisk is configured (DTMF using SIP INFO, RFC2833 etc), check this out: http://www.voip-info.org/wiki/view/Asterisk+sip+dtmfmode Regards. On 6/14/06, Jon Schøpzinsky [EMAIL PROTECTED] wrote: Hello How do I get Asterisk to receive DTMF from our Snom phones, when I use G.729? Regards Jon -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.8.4/363 - Release Date: 13-06-2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --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] Re: g729 or another
On Fri, Jun 09, 2006 at 04:45:51PM -0400, William Piper wrote: GSM and what is the size in KB that gsm spent? bp On 6/9/06, Pablo Allietti [EMAIL PROTECTED] wrote: hi all, i saw in digium that the codec g729 is not free. exist another codec with low bandwith to use in asterisk for free? -- .- Pablo Allietti E-mail: [EMAIL PROTECTED] | LACNIC Phone : +598 2 604 | [3]http://LACNIC.NET VoIp: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by [5]Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: [6]http://lists.digium.com/mailman/listinfo/asterisk-users References 1. mailto:[EMAIL PROTECTED] 2. mailto:[EMAIL PROTECTED] 3. http://LACNIC.NET/ 4. mailto:[EMAIL PROTECTED] 5. http://Easynews.com/ 6. http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---end quoted text--- -- .- Pablo Allietti E-mail: [EMAIL PROTECTED] | LACNIC Phone : +598 2 604 | http://LACNIC.NET ___ --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] Realtime queue_members and penalties nost escalating (clue anyone?)
- Danny Froberg [EMAIL PROTECTED] wrote: Regardless of what strategy is used in the queues (roundrobin,rrmemory,ringall etc) it wont escalate on NOANSWER That is not how penalties are supposed to work. Calls are delivered to the lowest-penalty members that are considered available (i.e. not busy and not unreachable). The queue application does not turn 'noanswer' into 'unavailable'. -- Kevin P. Fleming Senior Software Engineer Digium, Inc. ___ --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] GXP-2000 and Configdownload via TFTP
Hi, I was now successful in getting syslog messages. Syslog says the following: Jun 14 15:43:57 192.168.0.117 GS_LOG: [MAC][708][FF71][0101000D] ERROR 4099 GET cfgMAC What does errorcode 4099 mean? Best regards, Matthias -- Programming today is a race between software engineers striving to build bigger and better idiot-proof programs, and the universe trying to produce bigger and better idiots. So far, the universe is winning. -- Rich Cook ___ --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] Which application to open Zap channel?
I'm sure this a very common and easy thing to do with Asterisk, but for the life of me I can't find the application that will allow me to open a Zap channel. Real world example: To be able to connect to an open Zap channel, so it would allow me to say, join in on a call that was originally answered by a PSTN phone (ie. just like you would by simply picking up another PSTN phone..!). There is ZapBarge, but allows no speaking, which is useless for this situation. Maybe I just have to use Dial in some way? Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Xorcom Rapid
Hi Sorry for the long response time. I was away for a while and am now going over the asterisk-users backlog. On Sun, Jun 11, 2006 at 06:12:50PM +0200, Olivier Saulnier wrote: Tzafrir Cohen a écrit : I'm still not hapy with that as a default. It should provide you a basis for manual editing at this stage. But I wonder what else could the script configured there differently. Are those sane defaults for BRI on France? I've modified zaptel-channels.conf file , because, nothing happen when i call from an external phone inside the company. It's my problem, i don't know how name the QuadBRI interface, and how to use it in extensions files Do you hace some samples to give me, or explain me how i can detect the name to use? I'll just note that the standard zaptel.conf and zapata.conf samples that come with the qozap source could be found at /usr/share/doc/zaptel-source/examples/qozap Note to self: it should be back in the package zaptel, not in the package zaptel-source . Will be moved there. I'll answer your other mail shortly... -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 [EMAIL PROTECTED] http://www.xorcom.com ___ --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] Xorcom Rapid
On Sun, Jun 11, 2006 at 07:07:12PM +0200, Olivier Saulnier wrote:
Tzafrir Cohen a écrit :
Jut as usual with Zaptel: Zap/NNN (e.g: Zap/1 , Zap/2) for individual
channels. And gNNN and similar work just the same.
OK, in extensions.conf, i put the contexts PSTN and INTERNAL as:
[PSTN] ; for in coming calls - defin in zapata.conf
exten = s,1,Dial(IAX2/300,20)
exten = s,2,Voicemail, u300)
[INTERNAL] ; for internal AND outgoing call - actually just outgoing calls
exten = _0.,1,Dial(ZAP/g1/${EXTEN:1})
For hardware, how can i know on which interface is connected my ISDN line??
If all of them are defined but you only get no D channel message for
some, probably only those few are disconnected.
For outgoing call, i name the channel ZAP/1 in extensions.conf file, but
i dont know if it's correct.
And i always have the message timeout, but no rule 't' in context
What's mean??
There is no extension named t in that context to handle timeouts.
Your dialplan reads:
[PSTN]
exten = 1,1,Dial (IAX2/300,20)
exten = s,2,Voicemail, u300)
So no timeout action is specified. Ignore it if you don't just want to
have the call disconnected on timeout without taking any other action.
I'm not sure if the space after Dial is legal. I figure it may be the
source to your problem. Do you get an error in the CLI when reloading?
Before reloading:
set verbose 1
to see only the relevant warnings.
I have the same message!
Do you know how i can stop messages from qozap (they fill the screen
either asterisk is down!!!)
Simply don't define in zapata.conf (or any included configs) that span.
--
Tzafrir Cohen sip:[EMAIL PROTECTED]
icq#16849755 iax:[EMAIL PROTECTED]
+972-50-7952406
[EMAIL PROTECTED] http://www.xorcom.com
___
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Re: [Asterisk-Users] GXP-2000 and Configdownload via TFTP
You need to run the java based tool from the grandstream website to convert the template to a format the phone understands. On Wed, 2006-06-14 at 14:05, Matthias Fechner wrote: Hi, i got my Grandstream GXP-2000 phone today and want to configure it with TFTP. I downloaded the firmware 1.1.0.13 and put it into my tftp-server directory. Then I downloaded the template from: http://www.grandstream.com/DOWNLOAD/Configuration_Tool/Linux_Unix/Grandstream_Configuration_File_Template_1.0.6.x.txt renamed it to cfgmac-address Did the configuration in the new file and rebooted my phone. I can see in the log file from my tftp server that all files are loaded, the phone did a firmware upgrade. But it doesn't seems that the configuration file is loaded. Is it necessary to define on any place something that the phone use the config-file via tftp? Best regards, Matthias ___ --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] GXP-2000 and Configdownload via TFTP
On Wed, 2006-06-14 at 15:46 +0200, Matthias Fechner wrote: Hi, I was now successful in getting syslog messages. Syslog says the following: Jun 14 15:43:57 192.168.0.117 GS_LOG: [MAC][708][FF71][0101000D] ERROR 4099 GET cfgMAC What does errorcode 4099 mean? I don't know but it looks like it can't download the cfg file from your tft server. I've seen this with Cisco phones and boxes booting via PXE. Make sure the cfgMAC file has the right read permissions ie with chmod 644 cfgMAC. You can probably run the tftpserver with one or more -v arguments so you may get more info. Worth a try. Regards, Patrick ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk wengophone
Hi I use Asterisk with some SIP phone (grandstrea), while with my notebook when I'm out of home/office I use X-lite and all work. Some days ago I try to install wengophone and I decided that I want replace X-lite for use wengophone as client for my Asterisk. One of the reasons is that wengophone support g729 codec. I make some test and I see that is possible to configure other sip server (es. Asterisk) but every login wengo download from his site the conf. Now I want that wengo download the conf from my http server with my conf.:) Now I work on this using patient and ethereal, is anyone make wengo and Asterisk work or make this test? -- Pasqualotto Enrico email: pasqu AT linux.it || enrico AT pasqualotto.org web: http://www.pasqualotto.org skype: epasqualotto smime.p7s Description: S/MIME Cryptographic Signature ___ --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] Which application to open Zap channel?
This will just pick up the line exten = *01,1,Dial(ZAP/1/) _ Mobilcom http://www.mobilcom.net - Original Message - From: Carey O'Shea [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, June 14, 2006 9:48 AM Subject: [Asterisk-Users] Which application to open Zap channel? I'm sure this a very common and easy thing to do with Asterisk, but for the life of me I can't find the application that will allow me to open a Zap channel. Real world example: To be able to connect to an open Zap channel, so it would allow me to say, join in on a call that was originally answered by a PSTN phone (ie. just like you would by simply picking up another PSTN phone..!). There is ZapBarge, but allows no speaking, which is useless for this situation. Maybe I just have to use Dial in some way? Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] OPENSER / SER and Asterisk
-Original Message- From: Martin Joseph [mailto:[EMAIL PROTECTED] Sent: Tuesday, June 13, 2006 10:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] OPENSER / SER and Asterisk On Jun 13, 2006, at 8:29 PM, Douglas Garstang wrote: If you do this, and not have Asterisk in the call setup path, your going to lose the ability to do a lot of features. What about black/white lists, rate centers, pic codes, intra company extension dialling and other advanced features? Sure, you might be able to do them with SER but good luck trying to find documentation. So, your saying asterisk has better documentation? I just want to be sure I understand you ;~) Absolutely. The SER/OpenSER documentation is terrible, and if you post to the OpenSER mailing list, you get very cryptic replies. ___ --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] OPENSER / SER and Asterisk
Agreed. -Original Message- From: Santosh Rao [mailto:[EMAIL PROTECTED] Sent: Tuesday, June 13, 2006 11:19 PM To: Martin Joseph Cc: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] OPENSER / SER and Asterisk asterisk has a extremely cool documentation. The wiki has everything a newbie like me could hope for.. with samples and everyhting./. where as we are having a very dificult time finding proper documentation or samples and stuff like thtt for SER.. may be if someone good with SER could update ther voip-info/wiki and write some basics abt the ser.cfg or somethjing .. then it would be great. Regards Santosh Rao Martin Joseph wrote: On Jun 13, 2006, at 8:29 PM, Douglas Garstang wrote: If you do this, and not have Asterisk in the call setup path, your going to lose the ability to do a lot of features. What about black/white lists, rate centers, pic codes, intra company extension dialling and other advanced features? Sure, you might be able to do them with SER but good luck trying to find documentation. So, your saying asterisk has better documentation? I just want to be sure I understand you ;~) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] OPENSER / SER and Asterisk
-Original Message- From: Jean-Michel Hiver [mailto:[EMAIL PROTECTED] Sent: Wednesday, June 14, 2006 1:47 AM To: Santosh Rao; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] OPENSER / SER and Asterisk Santosh Rao a écrit : asterisk has a extremely cool documentation. The wiki has everything a newbie like me could hope for.. with samples and everyhting./. where as we are having a very dificult time finding proper documentation or samples and stuff like thtt for SER.. may be if someone good with SER could update ther voip-info/wiki and write some basics abt the ser.cfg or somethjing .. then it would be great. You can find some very good SER tutorials on onsip.org. You need to subscribe though, but it's free. I haven't read the tutorials, so I could be wrong, but I doubt they'd be very much use. They probably don't do more than give a basic overview, and I'm sure they don't touch things like avpops. ___ --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
SV: [Asterisk-Users] DTMF when using g.729
I should note that we are not running the Digium g729 implementation, but the intel one. Also, to not angry people, this ofcourse isn't used in our production environment, only for testing if we want g.729. Jon -Oprindelig meddelelse- Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af Moises Silva Sendt: 14. juni 2006 15:18 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [Asterisk-Users] DTMF when using g.729 Is new to me that using G729 codec is a problem when sending DTMF. Could it be that you are a little bit confused? Usually the problems with DTMF depend on how the phone is configured and how Asterisk is configured (DTMF using SIP INFO, RFC2833 etc), check this out: http://www.voip-info.org/wiki/view/Asterisk+sip+dtmfmode Regards. On 6/14/06, Jon Schøpzinsky [EMAIL PROTECTED] wrote: Hello How do I get Asterisk to receive DTMF from our Snom phones, when I use G.729? Regards Jon -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.8.4/363 - Release Date: 13-06-2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.8.4/363 - Release Date: 13-06-2006 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.8.4/363 - Release Date: 13-06-2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Hard drive write cache
99.999% I suspect you will see this drop as traditional PBX'es start to use commodity parts. My Mitel ICP 3300 has a Maxtor 10 gig hard drive in it (same as an Xbox!) ___ --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] GXP-2000 and Configdownload via TFTP
You need to encode txt configuration file using tool provided on GS site. DV -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthias Fechner Sent: Wednesday, June 14, 2006 3:06 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] GXP-2000 and Configdownload via TFTP Hi, i got my Grandstream GXP-2000 phone today and want to configure it with TFTP. I downloaded the firmware 1.1.0.13 and put it into my tftp-server directory. Then I downloaded the template from: http://www.grandstream.com/DOWNLOAD/Configuration_Tool/Linux_U nix/Grandstream_Configuration_File_Template_1.0.6.x.txt renamed it to cfgmac-address Did the configuration in the new file and rebooted my phone. I can see in the log file from my tftp server that all files are loaded, the phone did a firmware upgrade. But it doesn't seems that the configuration file is loaded. Is it necessary to define on any place something that the phone use the config-file via tftp? Best regards, Matthias ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk auto conference
Hi Please I want to make a schedule to make list of extensions in a conference, automatically the system call them and put them in a conference mode Can any one help me Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk auto conference
Hi Please I want to make a schedule to make list of extensions in a conference, automatically the system call them and put them in a conference mode Can any one help me Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --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] SIP call disconnected after answer
Hi, calling a partner on the other side of a SIP trunk, call gets disconnected immediately after answer. This is the content of log file: Jun 14 16:25:14 DEBUG[14380] channel.c: Didn't get a frame from channel: SIP/cerved-out-6eba Jun 14 16:25:14 DEBUG[14380] channel.c: Bridge stops bridging channels SIP/232-2e41 and SIP/cerved-out-6eba Jun 14 16:25:14 DEBUG[14380] channel.c: Hanging up channel 'SIP/cerved-out-6eba' Jun 14 16:25:14 DEBUG[14380] chan_sip.c: Hangup call SIP/cerved-out-6eba, SIP callid [EMAIL PROTECTED]) Jun 14 16:25:14 DEBUG[14380] chan_sip.c: update_call_counter(9704) - decrement call limit counter Jun 14 16:25:14 DEBUG[14380] chan_sip.c: Updating call counter for outgoing call Jun 14 16:25:14 DEBUG[14380] app_dial.c: Exiting with DIALSTATUS=ANSWER. I have Asterisk 1.2.8 but remote server has 1.2.4. Any help? -- Domenico Viggiani ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk auto conference
Hi Please I want to make a schedule to make list of extensions in a conference, automatically the system call them and put them in a conference mode Can any one help me Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --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] Which application to open Zap channel?
I swear Dial(Zap/X) was the first thing I tried and it didn't work, but now it works fine... hmmm maybe I forgot to reload my extensions or something like that. Thanks though. On Wed, 2006-06-14 at 10:03 -0400, Mailing List wrote: This will just pick up the line exten = *01,1,Dial(ZAP/1/) _ Mobilcom http://www.mobilcom.net - Original Message - From: Carey O'Shea [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, June 14, 2006 9:48 AM Subject: [Asterisk-Users] Which application to open Zap channel? I'm sure this a very common and easy thing to do with Asterisk, but for the life of me I can't find the application that will allow me to open a Zap channel. Real world example: To be able to connect to an open Zap channel, so it would allow me to say, join in on a call that was originally answered by a PSTN phone (ie. just like you would by simply picking up another PSTN phone..!). There is ZapBarge, but allows no speaking, which is useless for this situation. Maybe I just have to use Dial in some way? Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GXP-2000 and Configdownload via TFTP
Hi Gareth, Gareth Blades wrote: You need to run the java based tool from the grandstream website to convert the template to a format the phone understands. thx that was the problem. Now it works fine. Best regards, Matthias -- Programming today is a race between software engineers striving to build bigger and better idiot-proof programs, and the universe trying to produce bigger and better idiots. So far, the universe is winning. -- Rich Cook ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP, Microsoft RTC, and Originate problem
I tried your suggestion and found out that someone/something I don't know whether that is an MS RTC or Asterisk is having problems if the same Windows application is using Manager and SIP at the same time. At least for now, it has always worked, if I tried to initiate Originate command from one application, and had MS RTC in another. As soon as I put these two things in the same application, it stops working...weird. Has anyone experienced anything like that before? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, June 14, 2006 12:50 PM To: asterisk-users@lists.digium.com Cc: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] SIP, Microsoft RTC, and Originate problem Hmm.. Interesting, I didnt try to implement it this way... but, if its the same libraries used for Office communicator, than it supports only SIP over TCP or TLS, since asterisk doesnt support any of those its impossible to connect them directly... If udp works, maybe the registration part is problematic, try configuring asterisk with autocreatepeer (just for testing) to see if you can dial out without being registered. Ohad From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk Sent: Wednesday, June 14, 2006 11:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] SIP, Microsoft RTC, and Originate problem Nope, it's just the Microsoft RTC Core 1.3 library ... more or less a single DLL J. And I'm almost sure there is no SER in between should there be one? It's pretty much a straightforward thing I have a few SIP clients defined in my sip.conf, like this: [general] context=default allowguest=yes realm=timd.si bindport=5060 bindaddr=0.0.0.0 srvlookup=yes domain=timd.si,from-sip domain=111.111.111.8,from-sip videosupport=yes disallow=all allow=alaw allow=ulaw musicclass=default rtptimeout=100 rtpholdtimeout=100 tos=0x18 canreinvite=yes [SIPClient001] username= SIPClient001 secret= mysecret type=friend host=dynamic context=from-sip disallow=all allow=alaw allow=ulaw qualify=yes [SIPClient002] username= SIPClient002 secret= mysecret type=friend host=dynamic context=from-sip disallow=all allow=alaw allow=ulaw qualify=yes And there is an MS RTC based Softphone, that I made, on the other side that registers to Asterisk, using this profile XML string: provision key=5B29C449-29EE-4fd8-9E3F-04AED077690E name=Asterisk user account=SIPClient001 uri=sip:[EMAIL PROTECTED] / sipsrv addr=111.111.111.8 protocol=udp auth=digest role=registrar session party=first type=pc2ph / /sipsrv /provision Now, doing an originate to CHANNEL=SIP/SIPClient002, and some extension, will randomly fail, for example (see OriginateFailure reponse as well): action: Originate actionid: 123 exten: 03020846051635424 channel: SIP/SIPClient002 timeout: 3 priority: 1 context: asttel async: true Event: OriginateFailure Privilege: call,all ActionID: 123 Channel: SIP/ SIPClient002 Context: asttel Exten: 03020846051635424 Reason: 1 Uniqueid: null From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, June 14, 2006 10:14 AM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] SIP, Microsoft RTC, and Originate problem Hi, What is your setup? By MS RTC do you mean Office Communicator? If you are using MS OC, do you use SER in between (to convert SIP UDP2TCP)? Please share some more details J Cheers, Ohad From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk Sent: Wednesday, June 14, 2006 9:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] SIP, Microsoft RTC, and Originate problem It seems that Microsoft RTC has some problems with originated calls from Asterisk. If I execute Manager API originate application, with SIP channel as parameter, the Microsoft RTC softphone will start to ring after a couple of seconds delay, but nothing more happens after when I answer there is no second call to an extension. When I looked through the sip debug, I noticed that Microsoft RTC fails to properly respond to INVITE messages (I have attached the sip debug). Asterisk has to retransmit INVITE message for 6 times and even then the RTC still doesn't respond in a proper time. However, if I do direct call to that problematic Microsoft RTC based softphone, everything works fine, eventhough very same INVITE messages are being transmited to it from Asterisk. Does anyone have any ideas for a workaround? Regards, Alex ___ --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] No ring tone on outgoing calls
I am running on 1.2.7.1 and have an intermittent problem when making outgoing calls. Sometimes the calling party does not hear the ring tone in their handset, but the call goes through. From my extension I have only had 3 calls like this in the last couple of weeks, other people have had 3 or 4 calls in a single day and then not have a problem for a couple of days. The called phone number is not the problem because sometimes it works and sometimes not. We have both Aastra and Cisco phone sets and the problem occurs on both of them. We have SIP to PRI connections. I believe that this problem started after we upgraded from 1.0.9 but not 100 percent sure of that. Any help or suggestions that you have would be appreciated. Thank you, Tim ___ --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] SPA-941 Disable call waiting or Disable Call waiting via asterisk
It has a conceptual problem i have notified several times to Cisco-Linksys. It could not be disabled, i have the same problem with my queue extensions, and the way to resolve has been to use call-limit=1 in extensions. i hope this helps. Tommaso Calosi escribió: I'm trying to disable call waiting for Linksys SPA-941, but unfortunately as far as I have seen, there are no parameters on the web interface regarding this feature. I just want callers to hear the busy tone when the called party is at the phone. Probably I can accomplish this by using the disable call waiting in asterisk as well, but I have not been able to find any documentation for this. I have found this http://www.voip-info.org/wiki/index.php?page=PBX+Disable+Call+Waiting about call waiting, but it's quite unusefull. Thanks Tommaso Calosi ___ --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 -- Alberto Sagredo I+D Area (Asterisk // Cisco-Linksys) Peoplecall Email : [EMAIL PROTECTED] Blog: http://www.voipnovatos.es Tel./Ph. : +34 91 120 5080 Tel. Dir./Dir. Ph.: 700 757 139 / 91 120 50 39 Fax./Fax.: +34 91 661 9460 ___ --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] DUNDi Docs
Does anyone know where I can find some good DUNDi docs? The ones are dundi.org are absolutely horrible. The examples in dundi.conf are pretty much useless. I still can't figure out why Digium can't write some good documentation. It's their 'baby' after all. This really drives me nuts and pisses people off in general. I've been dicking around with DUNDi for over 6 months and still can't figure it out past the most basic application. Doug. ___ --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] OPENSER / SER and Asterisk
In fact the www.onsip.org documentation does include discussion about the avpops. It even gives an example of call forwarding using these functions. Simon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Douglas Garstang Sent: 14 June 2006 15:06 To: Asterisk Users Mailing List - Non-Commercial Discussion; Santosh Rao Subject: RE: [Asterisk-Users] OPENSER / SER and Asterisk -Original Message- From: Jean-Michel Hiver [mailto:[EMAIL PROTECTED] Sent: Wednesday, June 14, 2006 1:47 AM To: Santosh Rao; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] OPENSER / SER and Asterisk Santosh Rao a écrit : asterisk has a extremely cool documentation. The wiki has everything a newbie like me could hope for.. with samples and everyhting./. where as we are having a very dificult time finding proper documentation or samples and stuff like thtt for SER.. may be if someone good with SER could update ther voip-info/wiki and write some basics abt the ser.cfg or somethjing .. then it would be great. You can find some very good SER tutorials on onsip.org. You need to subscribe though, but it's free. I haven't read the tutorials, so I could be wrong, but I doubt they'd be very much use. They probably don't do more than give a basic overview, and I'm sure they don't touch things like avpops. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dynamic features on call waiting
Hello, I have problems using dynamic features while an other person is doing call waiting in a call. I define a dynamic application mapping in features.conf as the following: testfeature = *9,caller,Playback,tt-monkeys I also set DYNAMIC_FEATURES = testfeature. The mapping is working well. But during a third person is calling I'm hearing just the call waiting tone and none of my mapped features are working for this time. How can I change this behaviour? (I'm using Asterisk 1.2.4-BRIstuffed-0.3.0-PRE-1l) Thank you in advance, Henry ___ --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] DUNDi Docs
On Wed, 14 Jun 2006, Douglas Garstang wrote: The examples in dundi.conf are pretty much useless. I still can't figure out why Digium can't write some good documentation. It's their 'baby' after all. This really drives me nuts and pisses people off in general. I've been dicking around with DUNDi for over 6 months and still can't figure it out past the most basic application. What are you trying to do? -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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] Web UI - Best practices?
Hi, I'm stuck writing a Web GUI because nothing out there is exactly what I need. I'm not writing something as extensive as what _is_ out there, but just something that allows users to change where their calls are forwarded and other small things like that. What I wanted to know is what is recommended by those you successfully wrote their own UI : 1) Modifying the config directly in the AsteriskRealTime DB and and use Asterisk Realtime?This seems like the obvious choice, but I have a bad feeling about this method...especially with respect to future changes I would make to my UI or that the Asterisk dev team would make to their own tables / code 2) Using custom tables I make up myself, and querying that DB with the MySQL commanddirectly in the .conf files (or Realtime asterisk for that matter)(http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+MYSQL) ? Any input is appreciated...I really don't want to start on the wrong foot. Option 2 looks better (less dependent onAsterisk not changing from version to version) butI feel it'll make a mess out of the code. While Option 1 looks like it'llbe messybecause I have to adapt to the Realtime DB format in my PHP code, but at least the Asteriskcode will be clean. Really, opinons based on anecdotes would help me. Regards, Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SPA-941 Disable call waiting or Disable Call waiting via asterisk
Well, it does help, but it causes the announced transfer to fail, because if you set call-limit=1 you cannot dial out to announce the transfer... Alberto Sagredo wrote: It has a conceptual problem i have notified several times to Cisco-Linksys. It could not be disabled, i have the same problem with my queue extensions, and the way to resolve has been to use call-limit=1 in extensions. i hope this helps. Tommaso Calosi escribió: I'm trying to disable call waiting for Linksys SPA-941, but unfortunately as far as I have seen, there are no parameters on the web interface regarding this feature. I just want callers to hear the busy tone when the called party is at the phone. Probably I can accomplish this by using the disable call waiting in asterisk as well, but I have not been able to find any documentation for this. I have found this http://www.voip-info.org/wiki/index.php?page=PBX+Disable+Call+Waiting about call waiting, but it's quite unusefull. Thanks Tommaso Calosi ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] DUNDi Docs
Hi, Check this document, it helped me to build our DUNDi Network. http://leifmadsen.com/papers/dundi-intro.pdf Frédéric Marti == -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Douglas Garstang Sent: mercredi, 14. juin 2006 17:09 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] DUNDi Docs Does anyone know where I can find some good DUNDi docs? The ones are dundi.org are absolutely horrible. The examples in dundi.conf are pretty much useless. I still can't figure out why Digium can't write some good documentation. It's their 'baby' after all. This really drives me nuts and pisses people off in general. I've been dicking around with DUNDi for over 6 months and still can't figure it out past the most basic application. Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No ring tone on outgoing calls
Make sure you have /etc/asterisk/indications.conf set up. People that don't know any better might tell you to use the r option to Dial. Those people are confused. Don't do that until you have tried everything else. Tim Sharp wrote: I am running on 1.2.7.1 and have an intermittent problem when making outgoing calls. Sometimes the calling party does not hear the ring tone in their handset, but the call goes through. From my extension I have only had 3 calls like this in the last couple of weeks, other people have had 3 or 4 calls in a single day and then not have a problem for a couple of days. The called phone number is not the problem because sometimes it works and sometimes not. We have both Aastra and Cisco phone sets and the problem occurs on both of them. We have SIP to PRI connections. I believe that this problem started after we upgraded from 1.0.9 but not 100 percent sure of that. Any help or suggestions that you have would be appreciated. Thank you, Tim -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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] GXP-2000 Audio Quality
Wow! 22Kbps of overhead? Are you sure? That sounds like way too much overhead. I can't use IAX2 because the GXP-2000 are SIP phones :( Any other suggestion? Thanks, Daniel On Jun 14, 2006, at 4:37 AM, Gareth Blades wrote: G729 uses 8kbps but with the IP overhead it actually uses 30kbps so for 256k upstream you should be able to handle 8 calls but this is in ideal conditions. If you were to use IAX and enable trunking then you would use 30kbps for the 1st call and 10kbps for each additional call. See http://www.voip-info.org/wiki/index.php?page=Asterisk+bandwidth +iax2 On Wed, 2006-06-14 at 04:17, Daniel Salama wrote: I have a client with about 16 GXP-2000. They complain that the audio quality is terrible after 2 or 3 simultaneous conversations. They are behind DSL 1.5Mbps down and 256Kbps up. Because they are using G711.u codec, I know they upstream bandwidth is the limiting factor and they most likely won't be able to have more than 3 simultaneous conversations, and if they're surfing the net and/or checking email, things will only get worse. So, I purchased some g729 codec licenses and forced their sip peer configuration to g729 codec. We made sample test calls and were able to make 8 simultaneous calls. On the eighth call, the audio started to sound choppy. Then we dropped the eighth call and tested with 7. We could hear just fine on the GXP-2000 but the remote end heard us a bit choppy and/or with a robot-like voice. So, we kept dropping calls until they were of acceptable quality. My question is, if they were using g729 which, in theory uses 8kbps plus overhead, they should have been just fine handling eight calls. All the computers were turned off on the network, so there shouldn't have been any other traffic but VoIP. Does anyone have any ideas? How can I improve their audio quality? I requested BellSouth to upgrade their capacity but because of where they are located, the best they can get is 900Kbps/256Kbps, so the upstream continues to be the limiting factor. I purchased a Dlink-1226G switch to allow me to control QoS on the LAN. I also upgraded their Netopia DSL router to the latest firmware which allows me to configure VLANs and DiffServ. All the computers are connected to the PC port on the phone because there is no available second wiring. Can anyone suggest how to configure the QoS settings on the phones, the Dlink and the Netopia? While there was no traffic on the wire, pinging from/to the Asterisk box gave me about 47ms latency. When we went passed the 4th call, the latency started increasing significantly and when we got to 8 calls, the latency was up in the 2000ms. Obviously, if anything I did in the QoS configuration gave VoIP a priority, then ICMP packets would have the lowest priority and I could understand that to be the reason for such result. However, I'm not sure I configured QoS properly and that's why I'm asking for help. Thanks, Daniel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Which application to open Zap channel?
Carey O'Shea wrote: I swear Dial(Zap/X) was the first thing I tried and it didn't work, but now it works fine... hmmm maybe I forgot to reload my extensions or something like that. Don't expect Dial(Zap/X) to work. Expect Dial(Zap/X/) to work. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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] Web UI - Best practices?
On Wed, Jun 14, 2006 at 11:51:06AM -0400, Mike wrote: Hi, I'm stuck writing a Web GUI because nothing out there is exactly what I need. I'm not writing something as extensive as what _is_ out there, but just something that allows users to change where their calls are forwarded and other small things like that. What I wanted to know is what is recommended by those you successfully wrote their own UI : 1) Modifying the config directly in the Asterisk RealTime DB and and use Asterisk Realtime? This seems like the obvious choice, but I have a bad feeling about this method...especially with respect to future changes I would make to my UI or that the Asterisk dev team would make to their own tables / code Non-static real time mean that your PBX becomes non-functional if the DB server has a problem (or is even a bit loaded). Something quite similar to static real-time is to have the UI (re)write a small portion of the dialplan that will be included, and have it initiate an 'extentions reload' on any change. 2) Using custom tables I make up myself, and querying that DB with the MySQL command directly in the .conf files (or Realtime asterisk for that matter)(http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+MYSQL) ? This has basically the same weakness of non-static real-time: overhead and dependence at the time of the call. Any input is appreciated...I really don't want to start on the wrong foot. Option 2 looks better (less dependent on Asterisk not changing from version to version) It is basically the same. Unless you want to make something that is quite generic (if you do, you might as well stick with an existing UI that pays with complexity as the price for generity) you'll probably want to use some nice local features and use a customized dialplan. but I feel it'll make a mess out of the code. While Option 1 looks like it'll be messy because I have to adapt to the Realtime DB format in my PHP code, but at least the Asterisk code will be clean. Really, opinons based on anecdotes would help me. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 [EMAIL PROTECTED] http://www.xorcom.com ___ --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] QSIG
Hi all, I have to connect an asterisk box to a legacy pbx using QSIG signalling : where could i find more information or any sample ocnfiguration file? Has anyone never used it? Thanks in advance. Giordano Grandis ___ --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] DUNDi Docs
Yes, what is it you attempting? I use DUNDi extensively, though you are correct that the existing docs don't go very far in describing some things. Regards - Brad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Aaron Daniel Sent: Wednesday, June 14, 2006 11:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] DUNDi Docs On Wed, 14 Jun 2006, Douglas Garstang wrote: The examples in dundi.conf are pretty much useless. I still can't figure out why Digium can't write some good documentation. It's their 'baby' after all. This really drives me nuts and pisses people off in general. I've been dicking around with DUNDi for over 6 months and still can't figure it out past the most basic application. What are you trying to do? -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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 The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sangoma driver and zaptel
Hi, using Sangoma drivers: - doing 'lsmod', I see: zaptel ... wanpipe,wctdm24xxp,wctdm,wcfxo,wcte11xp,wct1xxp,wct4xxp,tor2 I'd like to avoid loading all these modules. What have I to do? - do I need to have 'zaptel' startup script under /etc/init.d ? Thanks -- Domenico Viggiani ___ --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] GXP-2000 Audio Quality
Welcome to the wonderful world of VoIP, where people are eager to move from 8kbps G.729 to 6.3kbps G.723.1, and accept a substantial drop in voice quality, and then throw over 20kbps of RTP, IP and related overhead on top of them. Isn't IP wonderful? :-) Regards, Steve Daniel Salama wrote: Wow! 22Kbps of overhead? Are you sure? That sounds like way too much overhead. I can't use IAX2 because the GXP-2000 are SIP phones :( Any other suggestion? Thanks, Daniel On Jun 14, 2006, at 4:37 AM, Gareth Blades wrote: G729 uses 8kbps but with the IP overhead it actually uses 30kbps so for 256k upstream you should be able to handle 8 calls but this is in ideal conditions. If you were to use IAX and enable trunking then you would use 30kbps for the 1st call and 10kbps for each additional call. See http://www.voip-info.org/wiki/index.php?page=Asterisk+bandwidth +iax2 On Wed, 2006-06-14 at 04:17, Daniel Salama wrote: I have a client with about 16 GXP-2000. They complain that the audio quality is terrible after 2 or 3 simultaneous conversations. They are behind DSL 1.5Mbps down and 256Kbps up. Because they are using G711.u codec, I know they upstream bandwidth is the limiting factor and they most likely won't be able to have more than 3 simultaneous conversations, and if they're surfing the net and/or checking email, things will only get worse. So, I purchased some g729 codec licenses and forced their sip peer configuration to g729 codec. We made sample test calls and were able to make 8 simultaneous calls. On the eighth call, the audio started to sound choppy. Then we dropped the eighth call and tested with 7. We could hear just fine on the GXP-2000 but the remote end heard us a bit choppy and/or with a robot-like voice. So, we kept dropping calls until they were of acceptable quality. My question is, if they were using g729 which, in theory uses 8kbps plus overhead, they should have been just fine handling eight calls. All the computers were turned off on the network, so there shouldn't have been any other traffic but VoIP. Does anyone have any ideas? How can I improve their audio quality? I requested BellSouth to upgrade their capacity but because of where they are located, the best they can get is 900Kbps/256Kbps, so the upstream continues to be the limiting factor. I purchased a Dlink-1226G switch to allow me to control QoS on the LAN. I also upgraded their Netopia DSL router to the latest firmware which allows me to configure VLANs and DiffServ. All the computers are connected to the PC port on the phone because there is no available second wiring. Can anyone suggest how to configure the QoS settings on the phones, the Dlink and the Netopia? While there was no traffic on the wire, pinging from/to the Asterisk box gave me about 47ms latency. When we went passed the 4th call, the latency started increasing significantly and when we got to 8 calls, the latency was up in the 2000ms. Obviously, if anything I did in the QoS configuration gave VoIP a priority, then ICMP packets would have the lowest priority and I could understand that to be the reason for such result. However, I'm not sure I configured QoS properly and that's why I'm asking for help. Thanks, Daniel ___ --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] GXP-2000 Audio Quality
Well, with 16 phones, it might be worth putting a 'satellite' asterisk in their office, have it handle local transfers, and act as a protocol converter, talking sip to the phones and (trunked) IAX2 to the outside world. An embedded low power system would do fine. You might even get away with an nslu2, but I'm not sure it has the RAM for 16 calls. A better alternative is to get them to upgrade the DSL to 512 uplink. Tim. On 14 Jun 2006, at 17:11, Daniel Salama wrote: Wow! 22Kbps of overhead? Are you sure? That sounds like way too much overhead. I can't use IAX2 because the GXP-2000 are SIP phones :( Any other suggestion? Thanks, Daniel On Jun 14, 2006, at 4:37 AM, Gareth Blades wrote: G729 uses 8kbps but with the IP overhead it actually uses 30kbps so for 256k upstream you should be able to handle 8 calls but this is in ideal conditions. If you were to use IAX and enable trunking then you would use 30kbps for the 1st call and 10kbps for each additional call. See http://www.voip-info.org/wiki/index.php?page=Asterisk+bandwidth +iax2 On Wed, 2006-06-14 at 04:17, Daniel Salama wrote: I have a client with about 16 GXP-2000. They complain that the audio quality is terrible after 2 or 3 simultaneous conversations. They are behind DSL 1.5Mbps down and 256Kbps up. Because they are using G711.u codec, I know they upstream bandwidth is the limiting factor and they most likely won't be able to have more than 3 simultaneous conversations, and if they're surfing the net and/or checking email, things will only get worse. So, I purchased some g729 codec licenses and forced their sip peer configuration to g729 codec. We made sample test calls and were able to make 8 simultaneous calls. On the eighth call, the audio started to sound choppy. Then we dropped the eighth call and tested with 7. We could hear just fine on the GXP-2000 but the remote end heard us a bit choppy and/or with a robot-like voice. So, we kept dropping calls until they were of acceptable quality. My question is, if they were using g729 which, in theory uses 8kbps plus overhead, they should have been just fine handling eight calls. All the computers were turned off on the network, so there shouldn't have been any other traffic but VoIP. Does anyone have any ideas? How can I improve their audio quality? I requested BellSouth to upgrade their capacity but because of where they are located, the best they can get is 900Kbps/256Kbps, so the upstream continues to be the limiting factor. I purchased a Dlink-1226G switch to allow me to control QoS on the LAN. I also upgraded their Netopia DSL router to the latest firmware which allows me to configure VLANs and DiffServ. All the computers are connected to the PC port on the phone because there is no available second wiring. Can anyone suggest how to configure the QoS settings on the phones, the Dlink and the Netopia? While there was no traffic on the wire, pinging from/to the Asterisk box gave me about 47ms latency. When we went passed the 4th call, the latency started increasing significantly and when we got to 8 calls, the latency was up in the 2000ms. Obviously, if anything I did in the QoS configuration gave VoIP a priority, then ICMP packets would have the lowest priority and I could understand that to be the reason for such result. However, I'm not sure I configured QoS properly and that's why I'm asking for help. Thanks, Daniel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Tim Panton [EMAIL PROTECTED] ___ --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] dial plan return values
Is there a method for detecting return values of applications in the dial plan? Thanks Mark Price UNETA ___ --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] transcoding problem
I am having a problem with asterisk transcoding GSM and G729 codecs, the error message is below: Jun 14 09:38:12 WARNING[18292]: channel.c:2693 ast_channel_make_compatible: No path to translate from SIP/3004-fcfb(256) to SIP/3003-c1c3(2) Jun 14 09:38:12 WARNING[18292]: app_dial.c:1586 dial_exec_full: Had to drop call because I couldn't make SIP/3004-fcfb compatible with SIP/3003-c1c3 == Spawn extension (test, 3003, 1) exited non-zero on 'SIP/3004-fcfb' sip.conf [3004] type=friend secret=x context=test callerid=test1 3004 nat=yes disallow=all allow=g729 host=dynamic canreinvite=yes dtmfmode=rfc2833 [3003] type=friend secret=x context=test callerid=test2 3003 nat=yes disallow=all allow=gsm host=dynamic canreinvite=yes dtmfmode=rfc2833 ___ --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] dial plan return values
I think each application returns it's own value in a variable defined by the application. Mark Price wrote: Is there a method for detecting return values of applications in the dial plan? Thanks Mark Price UNETA ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Realtime queue_members and penalties nost escalating (clue anyone?)
Thanks for clearing that up Kevin. Now on to figure out how to PauseQueueMember when enough NOANSWER's has been detected so he don't fubar the entire queue. Would be alot cleaner than sending callers to ever higher level queues *sigh* Kevin P. Fleming wrote: Regardless of what strategy is used in the queues (roundrobin,rrmemory,ringall etc) it wont escalate on NOANSWER That is not how penalties are supposed to work. Calls are delivered to the lowest-penalty members that are considered available (i.e. not busy and not unreachable). The queue application does not turn 'noanswer' into 'unavailable'. ___ --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] DUNDi Users
I have three Asterisk boxes.
Each has the following in dundi.conf:
180net = dundi_local,0,IAX,dundi:[EMAIL PROTECTED]/${NUMBER},nopartial
180q = dundi_q_pbx1,1,IAX,dundi:[EMAIL PROTECTED]/${NUMBER},nopartial
180q = dundi_q_pbx2,2,IAX,dundi:[EMAIL PROTECTED]/${NUMBER},nopartial
180q = dundi_q_pbx3,3,IAX,dundi:[EMAIL PROTECTED]/${NUMBER},nopartial
My iax.conf on all three Asterisk boxes has this:
[dundi]
type=user
dbsecret=dundi/secret
context=dundi_local
disallow=all
allow=ulaw
allow=g729
I can do a lookup on pbx2 to find where a number is:
hermes*CLI dundi lookup [EMAIL PROTECTED]
1. 1 IAX2/dundi:[EMAIL PROTECTED]/oe_main (EXISTS)
from 00:0e:0c:a1:92:6f, expires in 0 s
2. 1 IAX2/dundi:[EMAIL PROTECTED]/oe_main (EXISTS)
from 00:0e:0c:a1:92:4d, expires in 0 s
DUNDi lookup completed in 53 ms
However, when I dial the DUNDi path, this is what pbx1 logs on the console:
Jun 14 10:51:39 NOTICE[22424]: chan_iax2.c:7215 socket_read: Rejected connect
attempt from xxx.187.142.204, request '[EMAIL PROTECTED]' does not exist
I tried adding the contexts to [dundi] in iax.conf:
[dundi]
type=user
dbsecret=dundi/secret
context=dundi_local
context=dundi_q_pbx1
context=dundi_q_pbx2
context=dundi_q_pbx3
disallow=all
allow=ulaw
allow=g729
However, the call on pbx1 is still routed to the dundi_local context instead of
dundi_q_pbx1.
Do I have to go and modify dundi.conf, so that every dundi entry uses a
different DUNDi user, like this?
180q = dundi_q_pbx1,1,IAX,dundi1:[EMAIL PROTECTED]/${NUMBER},nopartial
180q = dundi_q_pbx2,2,IAX,dundi2:[EMAIL PROTECTED]/${NUMBER},nopartial
180q = dundi_q_pbx3,3,IAX,dundi3:[EMAIL PROTECTED]/${NUMBER},nopartial
And then add users dundi1, dundi2 and dundi3 to iax.conf?
I sure hope not. What a horrible way to have to do it.
Doug.
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Re: [Asterisk-Users] transcoding problem
Contact Digium to purchase a G729 license. Osama Kamal wrote: I am having a problem with asterisk transcoding GSM and G729 codecs, the error message is below: Jun 14 09:38:12 WARNING[18292]: channel.c:2693 ast_channel_make_compatible: No path to translate from SIP/3004-fcfb(256) to SIP/3003-c1c3(2) Jun 14 09:38:12 WARNING[18292]: app_dial.c:1586 dial_exec_full: Had to drop call because I couldn't make SIP/3004-fcfb compatible with SIP/3003-c1c3 == Spawn extension (test, 3003, 1) exited non-zero on 'SIP/3004-fcfb' sip.conf [3004] type=friend secret=x context=test callerid=test1 3004 nat=yes disallow=all allow=g729 host=dynamic canreinvite=yes dtmfmode=rfc2833 [3003] type=friend secret=x context=test callerid=test2 3003 nat=yes disallow=all allow=gsm host=dynamic canreinvite=yes dtmfmode=rfc2833 ___ --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 -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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] DUNDi Docs
-Original Message- From: Aaron Daniel [mailto:[EMAIL PROTECTED] Sent: Wednesday, June 14, 2006 9:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] DUNDi Docs On Wed, 14 Jun 2006, Douglas Garstang wrote: The examples in dundi.conf are pretty much useless. I still can't figure out why Digium can't write some good documentation. It's their 'baby' after all. This really drives me nuts and pisses people off in general. I've been dicking around with DUNDi for over 6 months and still can't figure it out past the most basic application. What are you trying to do? I am trying to implement distributed ACD queues. A user dials the main queue number 2944000. The primary Asterisk server for that user has 2944000 in it's dialplan. It does a DUNDi lookup of a number, oe_main (it has to be different to 2944000 of course), to determine what the primary asterisk box is for this number, oemain, which is really the ACD Queue. I therefore need to have a DUNDi context that maps to three dialplan contexts. The context are slightly different on each Asterisk server, so that the queue has a primary, secondary, and tertiary server. Like this...: PBX1: [pbx_pri] exten = oe_main,1,Dial(SIP/2944000,20,tr) [pbx_sec] [pbx_ter] PBX2: [pbx_pri] [pbx_sec] exten = oe_main,1,Dial(SIP/2944000,20,tr) [pbx_ter] PBX3: [pbx_pri] [pbx_sec] [pbx_ter] exten = oe_main,1,Dial(SIP/2944000,20,tr) The queue accessed by oe_main is primary on pbx, secondary on pbx2, and tertiary on pbx3. Doug. ___ --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
